aboutsummaryrefslogtreecommitdiffstats
path: root/ChangeLog
blob: 7f55a974c3fa855ab3660daa57c705578d4ba3d3 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
1129
1130
1131
1132
1133
1134
1135
1136
1137
1138
1139
1140
1141
1142
1143
1144
1145
1146
1147
1148
1149
1150
1151
1152
1153
1154
1155
1156
1157
1158
1159
1160
1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
1171
1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
1189
1190
1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
1205
1206
1207
1208
1209
1210
1211
1212
1213
1214
1215
1216
1217
1218
1219
1220
1221
1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235
1236
1237
1238
1239
1240
1241
1242
1243
1244
1245
1246
1247
1248
1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
1262
1263
1264
1265
1266
1267
1268
1269
1270
1271
1272
1273
1274
1275
1276
1277
1278
1279
1280
1281
1282
1283
1284
1285
1286
1287
1288
1289
1290
1291
1292
1293
1294
1295
1296
1297
1298
1299
1300
1301
1302
1303
1304
1305
1306
1307
1308
1309
1310
1311
1312
1313
1314
1315
1316
1317
1318
1319
1320
1321
1322
1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
1338
1339
1340
1341
1342
1343
1344
1345
1346
1347
1348
1349
1350
1351
1352
1353
1354
1355
1356
1357
1358
1359
1360
1361
1362
1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
1373
1374
1375
1376
1377
1378
1379
1380
1381
1382
1383
1384
1385
1386
1387
1388
1389
1390
1391
1392
1393
1394
1395
1396
1397
1398
1399
1400
1401
1402
1403
1404
1405
1406
1407
1408
1409
1410
1411
1412
1413
1414
1415
1416
1417
1418
1419
1420
1421
1422
1423
1424
1425
1426
1427
1428
1429
1430
1431
1432
1433
1434
1435
1436
1437
1438
1439
1440
1441
1442
1443
1444
1445
1446
1447
1448
1449
1450
1451
1452
1453
1454
1455
1456
1457
1458
1459
1460
1461
1462
1463
1464
1465
1466
1467
1468
1469
1470
1471
1472
1473
1474
1475
1476
1477
1478
1479
1480
1481
1482
1483
1484
1485
1486
1487
1488
1489
1490
1491
1492
1493
1494
1495
1496
1497
1498
1499
1500
1501
1502
1503
1504
1505
1506
1507
1508
1509
1510
1511
1512
1513
1514
1515
1516
1517
1518
1519
1520
1521
1522
1523
1524
1525
1526
1527
1528
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
1541
1542
1543
1544
1545
1546
1547
1548
1549
1550
1551
1552
1553
1554
1555
1556
1557
1558
1559
1560
1561
1562
1563
1564
1565
1566
1567
1568
1569
1570
1571
1572
1573
1574
1575
1576
1577
1578
1579
1580
1581
1582
1583
1584
1585
1586
1587
1588
1589
1590
1591
1592
1593
1594
1595
1596
1597
1598
1599
1600
1601
1602
1603
1604
1605
1606
1607
1608
1609
1610
1611
1612
1613
1614
1615
1616
1617
1618
1619
1620
1621
1622
1623
1624
1625
1626
1627
1628
1629
1630
1631
1632
1633
1634
1635
1636
1637
1638
1639
1640
1641
1642
1643
1644
1645
1646
1647
1648
1649
1650
1651
1652
1653
1654
1655
1656
1657
1658
1659
1660
1661
1662
1663
1664
1665
1666
1667
1668
1669
1670
1671
1672
1673
1674
1675
1676
1677
1678
1679
1680
1681
1682
1683
1684
1685
1686
1687
1688
1689
1690
1691
1692
1693
1694
1695
1696
1697
1698
1699
1700
1701
1702
1703
1704
1705
1706
1707
1708
1709
1710
1711
1712
1713
1714
1715
1716
1717
1718
1719
1720
1721
1722
1723
1724
1725
1726
1727
1728
1729
1730
1731
1732
1733
1734
1735
1736
1737
1738
1739
1740
1741
1742
1743
1744
1745
1746
1747
1748
1749
1750
1751
1752
1753
1754
1755
1756
1757
1758
1759
1760
1761
1762
1763
1764
1765
1766
1767
1768
1769
1770
1771
1772
1773
1774
1775
1776
1777
1778
1779
1780
1781
1782
1783
1784
1785
1786
1787
1788
1789
1790
1791
1792
1793
1794
1795
1796
1797
1798
1799
1800
1801
1802
1803
1804
1805
1806
1807
1808
1809
1810
1811
1812
1813
1814
1815
1816
1817
1818
1819
1820
1821
1822
1823
1824
1825
1826
1827
1828
1829
1830
1831
1832
1833
1834
1835
1836
1837
1838
1839
1840
1841
1842
1843
1844
1845
1846
1847
1848
1849
1850
1851
1852
1853
1854
1855
1856
1857
1858
1859
1860
1861
1862
1863
1864
1865
1866
1867
1868
1869
1870
1871
1872
1873
1874
1875
1876
1877
1878
1879
1880
1881
1882
1883
1884
1885
1886
1887
1888
1889
1890
1891
1892
1893
1894
1895
1896
1897
1898
1899
1900
1901
1902
1903
1904
1905
1906
1907
1908
1909
1910
1911
1912
1913
1914
1915
1916
1917
1918
1919
1920
1921
1922
1923
1924
1925
1926
1927
1928
1929
1930
1931
1932
1933
1934
1935
1936
1937
1938
1939
1940
1941
1942
1943
1944
1945
1946
1947
1948
1949
1950
1951
1952
1953
1954
1955
1956
1957
1958
1959
1960
1961
1962
1963
1964
1965
1966
1967
1968
1969
1970
1971
1972
1973
1974
1975
1976
1977
1978
1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
1989
1990
1991
1992
1993
1994
1995
1996
1997
1998
1999
2000
2001
2002
2003
2004
2005
2006
2007
2008
2009
2010
2011
2012
2013
2014
2015
2016
2017
2018
2019
2020
2021
2022
2023
2024
2025
2026
2027
2028
2029
2030
2031
2032
2033
2034
2035
2036
2037
2038
2039
2040
2041
2042
2043
2044
2045
2046
2047
2048
2049
2050
2051
2052
2053
2054
2055
2056
2057
2058
2059
2060
2061
2062
2063
2064
2065
2066
2067
2068
2069
2070
2071
2072
2073
2074
2075
2076
2077
2078
2079
2080
2081
2082
2083
2084
2085
2086
2087
2088
2089
2090
2091
2092
2093
2094
2095
2096
2097
2098
2099
2100
2101
2102
2103
2104
2105
2106
2107
2108
2109
2110
2111
2112
2113
2114
2115
2116
2117
2118
2119
2120
2121
2122
2123
2124
2125
2126
2127
2128
2129
2130
2131
2132
2133
2134
2135
2136
2137
2138
2139
2140
2141
2142
2143
2144
2145
2146
2147
2148
2149
2150
2151
2152
2153
2154
2155
2156
2157
2158
2159
2160
2161
2162
2163
2164
2165
2166
2167
2168
2169
2170
2171
2172
2173
2174
2175
2176
2177
2178
2179
2180
2181
2182
2183
2184
2185
2186
2187
2188
2189
2190
2191
2192
2193
2194
2195
2196
2197
2198
2199
2200
2201
2202
2203
2204
2205
2206
2207
2208
2209
2210
2211
2212
2213
2214
2215
2216
2217
2218
2219
2220
2221
2222
2223
2224
2225
2226
2227
2228
2229
2230
2231
2232
2233
2234
2235
2236
2237
2238
2239
2240
2241
2242
2243
2244
2245
2246
2247
2248
2249
2250
2251
2252
2253
2254
2255
2256
2257
2258
2259
2260
2261
2262
2263
2264
2265
2266
2267
2268
2269
2270
2271
2272
2273
2274
2275
2276
2277
2278
2279
2280
2281
2282
2283
2284
2285
2286
2287
2288
2289
2290
2291
2292
2293
2294
2295
2296
2297
2298
2299
2300
2301
2302
2303
2304
2305
2306
2307
2308
2309
2310
2311
2312
2313
2314
2315
2316
2317
2318
2319
2320
2321
2322
2323
2324
2325
2326
2327
2328
2329
2330
2331
2332
2333
2334
2335
2336
2337
2338
2339
2340
2341
2342
2343
2344
2345
2346
2347
2348
2349
2350
2351
2352
2353
2354
2355
2356
2357
2358
2359
2360
2361
2362
2363
2364
2365
2366
2367
2368
2369
2370
2371
2372
2373
2374
2375
2376
2377
2378
2379
2380
2381
2382
2383
2384
2385
2386
2387
2388
2389
2390
2391
2392
2393
2394
2395
2396
2397
2398
2399
2400
2401
2402
2403
2404
2405
2406
2407
2408
2409
2410
2411
2412
2413
2414
2415
2416
2417
2418
2419
2420
2421
2422
2423
2424
2425
2426
2427
2428
2429
2430
2431
2432
2433
2434
2435
2436
2437
2438
2439
2440
2441
2442
2443
2444
2445
2446
2447
2448
2449
2450
2451
2452
2453
2454
2455
2456
2457
2458
2459
2460
2461
2462
2463
2464
2465
2466
2467
2468
2469
2470
2471
2472
2473
2474
2475
2476
2477
2478
2479
2480
2481
2482
2483
2484
2485
2486
2487
2488
2489
2490
2491
2492
2493
2494
2495
2496
2497
2498
2499
2500
2501
2502
2503
2504
2505
2506
2507
2508
2509
2510
2511
2512
2513
2514
2515
2516
2517
2518
2519
2520
2521
2522
2523
2524
2525
2526
2527
2528
2529
2530
2531
2532
2533
2534
2535
2536
2537
2538
2539
2540
2541
2542
2543
2544
2545
2546
2547
2548
2549
2550
2551
2552
2553
2554
2555
2556
2557
2558
2559
2560
2561
2562
2563
2564
2565
2566
2567
2568
2569
2570
2571
2572
2573
2574
2575
2576
2577
2578
2579
2580
2581
2582
2583
2584
2585
2586
2587
2588
2589
2590
2591
2592
2593
2594
2595
2596
2597
2598
2599
2600
2601
2602
2603
2604
2605
2606
2607
2608
2609
2610
2611
2612
2613
2614
2615
2616
2617
2618
2619
2620
2621
2622
2623
2624
2625
2626
2627
2628
2629
2630
2631
2632
2633
2634
2635
2636
2637
2638
2639
2640
2641
2642
2643
2644
2645
2646
2647
2648
2649
2650
2651
2652
2653
2654
2655
2656
2657
2658
2659
2660
2661
2662
2663
2664
2665
2666
2667
2668
2669
2670
2671
2672
2673
2674
2675
2676
2677
2678
2679
2680
2681
2682
2683
2684
2685
2686
2687
2688
2689
2690
2691
2692
2693
2694
2695
2696
2697
2698
2699
2700
2701
2702
2703
2704
2705
2706
2707
2708
2709
2710
2711
2712
2713
2714
2715
2716
2717
2718
2719
2720
2721
2722
2723
2724
2725
2726
2727
2728
2729
2730
2731
2732
2733
2734
2735
2736
2737
2738
2739
2740
2741
2742
2743
2744
2745
2746
2747
2748
2749
2750
2751
2752
2753
2754
2755
2756
2757
2758
2759
2760
2761
2762
2763
2764
2765
2766
2767
2768
2769
2770
2771
2772
2773
2774
2775
2776
2777
2778
2779
2780
2781
2782
2783
2784
2785
2786
2787
2788
2789
2790
2791
2792
2793
2794
2795
2796
2797
2798
2799
2800
2801
2802
2803
2804
2805
2806
2807
2808
2809
2810
2811
2812
2813
2814
2815
2816
2817
2818
2819
2820
2821
2822
2823
2824
2825
2826
2827
2828
2829
2830
2831
2832
2833
2834
2835
2836
2837
2838
2839
2840
2841
2842
2843
2844
2845
2846
2847
2848
2849
2850
2851
2852
2853
2854
2855
2856
2857
2858
2859
2860
2861
2862
2863
2864
2865
2866
2867
2868
2869
2870
2871
2872
2873
2874
2875
2876
2877
2878
2879
2880
2881
2882
2883
2884
2885
2886
2887
2888
2889
2890
2891
2892
2893
2894
2895
2896
2897
2898
2899
2900
2901
2902
2903
2904
2905
2906
2907
2908
2909
2910
2911
2912
2913
2914
2915
2916
2917
2918
2919
2920
2921
2922
2923
2924
2925
2926
2927
2928
2929
2930
2931
2932
2933
2934
2935
2936
2937
2938
2939
2940
2941
2942
2943
2944
2945
2946
2947
2948
2949
2950
2951
2952
2953
2954
2955
2956
2957
2958
2959
2960
2961
2962
2963
2964
2965
2966
2967
2968
2969
2970
2971
2972
2973
2974
2975
2976
2977
2978
2979
2980
2981
2982
2983
2984
2985
2986
2987
2988
2989
2990
2991
2992
2993
2994
2995
2996
2997
2998
2999
3000
3001
3002
3003
3004
3005
3006
3007
3008
3009
3010
3011
3012
3013
3014
3015
3016
3017
3018
3019
3020
3021
3022
3023
3024
3025
3026
3027
3028
3029
3030
3031
3032
3033
3034
3035
3036
3037
3038
3039
3040
3041
3042
3043
3044
3045
3046
3047
3048
3049
3050
3051
3052
3053
3054
3055
3056
3057
3058
3059
3060
3061
3062
3063
3064
3065
3066
3067
3068
3069
3070
3071
3072
3073
3074
3075
3076
3077
3078
3079
3080
3081
3082
3083
3084
3085
3086
3087
3088
3089
3090
3091
3092
3093
3094
3095
3096
3097
3098
3099
3100
3101
3102
3103
3104
3105
3106
3107
3108
3109
3110
3111
3112
3113
3114
3115
3116
3117
3118
3119
3120
3121
3122
3123
3124
3125
3126
3127
3128
3129
3130
3131
3132
3133
3134
3135
3136
3137
3138
3139
3140
3141
3142
3143
3144
3145
3146
3147
3148
3149
3150
3151
3152
3153
3154
3155
3156
3157
3158
3159
3160
3161
3162
3163
3164
3165
3166
3167
3168
3169
3170
3171
3172
3173
3174
3175
3176
3177
3178
3179
3180
3181
3182
3183
3184
3185
3186
3187
3188
3189
3190
3191
3192
3193
3194
3195
3196
3197
3198
3199
3200
3201
3202
3203
3204
3205
3206
3207
3208
3209
3210
3211
3212
3213
3214
3215
3216
3217
3218
3219
3220
3221
3222
3223
3224
3225
3226
3227
3228
3229
3230
3231
3232
3233
3234
3235
3236
3237
3238
3239
3240
3241
3242
3243
3244
3245
3246
3247
3248
3249
3250
3251
3252
3253
3254
3255
3256
3257
3258
3259
3260
3261
3262
3263
3264
3265
3266
3267
3268
3269
3270
3271
3272
3273
3274
3275
3276
3277
3278
3279
3280
3281
3282
3283
3284
3285
3286
3287
3288
3289
3290
3291
3292
3293
3294
3295
3296
3297
3298
3299
3300
3301
3302
3303
3304
3305
3306
3307
3308
3309
3310
3311
3312
3313
3314
3315
3316
3317
3318
3319
3320
3321
3322
3323
3324
3325
3326
3327
3328
3329
3330
3331
3332
3333
3334
3335
3336
3337
3338
3339
3340
3341
3342
3343
3344
3345
3346
3347
3348
3349
3350
3351
3352
3353
3354
3355
3356
3357
3358
3359
3360
3361
3362
3363
3364
3365
3366
3367
3368
3369
3370
3371
3372
3373
3374
3375
3376
3377
3378
3379
3380
3381
3382
3383
3384
3385
3386
3387
3388
3389
3390
3391
3392
3393
3394
3395
3396
3397
3398
3399
3400
3401
3402
3403
3404
3405
3406
3407
3408
3409
3410
3411
3412
3413
3414
3415
3416
3417
3418
3419
3420
3421
3422
3423
3424
3425
3426
3427
3428
3429
3430
3431
3432
3433
3434
3435
3436
3437
3438
3439
3440
3441
3442
3443
3444
3445
3446
3447
3448
3449
3450
3451
3452
3453
3454
3455
3456
3457
3458
3459
3460
3461
3462
3463
3464
3465
3466
3467
3468
3469
3470
3471
3472
3473
3474
3475
3476
3477
3478
3479
3480
3481
3482
3483
3484
3485
3486
3487
3488
3489
3490
3491
3492
3493
3494
3495
3496
3497
3498
3499
3500
3501
3502
3503
3504
3505
3506
3507
3508
3509
3510
3511
3512
3513
3514
3515
3516
3517
3518
3519
3520
3521
3522
3523
3524
3525
3526
3527
3528
3529
3530
3531
3532
3533
3534
3535
3536
3537
3538
3539
3540
3541
3542
3543
3544
3545
3546
3547
3548
3549
3550
3551
3552
3553
3554
3555
3556
3557
3558
3559
3560
3561
3562
3563
3564
3565
3566
3567
3568
3569
3570
3571
3572
3573
3574
3575
3576
3577
3578
3579
3580
3581
3582
3583
3584
3585
3586
3587
3588
3589
3590
3591
3592
3593
3594
3595
3596
3597
3598
3599
3600
3601
3602
3603
3604
3605
3606
3607
3608
3609
3610
3611
3612
3613
3614
3615
3616
3617
3618
3619
3620
3621
3622
3623
3624
3625
3626
3627
3628
3629
3630
3631
3632
3633
3634
3635
3636
3637
3638
3639
3640
3641
3642
3643
3644
3645
3646
3647
3648
3649
3650
3651
3652
3653
3654
3655
3656
3657
3658
3659
3660
3661
3662
3663
3664
3665
3666
3667
3668
3669
3670
3671
3672
3673
3674
3675
3676
3677
3678
3679
3680
3681
3682
3683
3684
3685
3686
3687
3688
3689
3690
3691
3692
3693
3694
3695
3696
3697
3698
3699
3700
3701
3702
3703
3704
3705
3706
3707
3708
3709
3710
3711
3712
3713
3714
3715
3716
3717
3718
3719
3720
3721
3722
3723
3724
3725
3726
3727
3728
3729
3730
3731
3732
3733
3734
3735
3736
3737
3738
3739
3740
3741
3742
3743
3744
3745
3746
3747
3748
3749
3750
3751
3752
3753
3754
3755
3756
3757
3758
3759
3760
3761
3762
3763
3764
3765
3766
3767
3768
3769
3770
3771
3772
3773
3774
3775
3776
3777
3778
3779
3780
3781
3782
3783
3784
3785
3786
3787
3788
3789
3790
3791
3792
3793
3794
3795
3796
3797
3798
3799
3800
3801
3802
3803
3804
3805
3806
3807
3808
3809
3810
3811
3812
3813
3814
3815
3816
3817
3818
3819
3820
3821
3822
3823
3824
3825
3826
3827
3828
3829
3830
3831
3832
3833
3834
3835
3836
3837
3838
3839
3840
3841
3842
3843
3844
3845
3846
3847
3848
3849
3850
3851
3852
3853
3854
3855
3856
3857
3858
3859
3860
3861
3862
3863
3864
3865
3866
3867
3868
3869
3870
3871
3872
3873
3874
3875
3876
3877
3878
3879
3880
3881
3882
3883
3884
3885
3886
3887
3888
3889
3890
3891
3892
3893
3894
3895
3896
3897
3898
3899
3900
3901
3902
3903
3904
3905
3906
3907
3908
3909
3910
3911
3912
3913
3914
3915
3916
3917
3918
3919
3920
3921
3922
3923
3924
3925
3926
3927
3928
3929
3930
3931
3932
3933
3934
3935
3936
3937
3938
3939
3940
3941
3942
3943
3944
3945
3946
3947
3948
3949
3950
3951
3952
3953
3954
3955
3956
3957
3958
3959
3960
3961
3962
3963
3964
3965
3966
3967
3968
3969
3970
3971
3972
3973
3974
3975
3976
3977
3978
3979
3980
3981
3982
3983
3984
3985
3986
3987
3988
3989
3990
3991
3992
3993
3994
3995
3996
3997
3998
3999
4000
4001
4002
4003
4004
4005
4006
4007
4008
4009
4010
4011
4012
4013
4014
4015
4016
4017
4018
4019
4020
4021
4022
4023
4024
4025
4026
4027
4028
4029
4030
4031
4032
4033
4034
4035
4036
4037
4038
4039
4040
4041
4042
4043
4044
4045
4046
4047
4048
4049
4050
4051
4052
4053
4054
4055
4056
4057
4058
4059
4060
4061
4062
4063
4064
4065
4066
4067
4068
4069
4070
4071
4072
4073
4074
4075
4076
4077
4078
4079
4080
4081
4082
4083
4084
4085
4086
4087
4088
4089
4090
4091
4092
4093
4094
4095
4096
4097
4098
4099
4100
4101
4102
4103
4104
4105
4106
4107
4108
4109
4110
4111
4112
4113
4114
4115
4116
4117
4118
4119
4120
4121
4122
4123
4124
4125
4126
4127
4128
4129
4130
4131
4132
4133
4134
4135
4136
4137
4138
4139
4140
4141
4142
4143
4144
4145
4146
4147
4148
4149
4150
4151
4152
4153
4154
4155
4156
4157
4158
4159
4160
4161
4162
4163
4164
4165
4166
4167
4168
4169
4170
4171
4172
4173
4174
4175
4176
4177
4178
4179
4180
4181
4182
4183
4184
4185
4186
4187
4188
4189
4190
4191
4192
4193
4194
4195
4196
4197
4198
4199
4200
4201
4202
4203
4204
4205
4206
4207
4208
4209
4210
4211
4212
4213
4214
4215
4216
4217
4218
4219
4220
4221
4222
4223
4224
4225
4226
4227
4228
4229
4230
4231
4232
4233
4234
4235
4236
4237
4238
4239
4240
4241
4242
4243
4244
4245
4246
4247
4248
4249
4250
4251
4252
4253
4254
4255
4256
4257
4258
4259
4260
4261
4262
4263
4264
4265
4266
4267
4268
4269
4270
4271
4272
4273
4274
4275
4276
4277
4278
4279
4280
4281
4282
4283
4284
4285
4286
4287
4288
4289
4290
4291
4292
4293
4294
4295
4296
4297
4298
4299
4300
4301
4302
4303
4304
4305
4306
4307
4308
4309
4310
4311
4312
4313
4314
4315
4316
4317
4318
4319
4320
4321
4322
4323
4324
4325
4326
4327
4328
4329
4330
4331
4332
4333
4334
4335
4336
4337
4338
4339
4340
4341
4342
4343
4344
4345
4346
4347
4348
4349
4350
4351
4352
4353
4354
4355
4356
4357
4358
4359
4360
4361
4362
4363
4364
4365
4366
4367
4368
4369
4370
4371
4372
4373
4374
4375
4376
4377
4378
4379
4380
4381
4382
4383
4384
4385
4386
4387
4388
4389
4390
4391
4392
4393
4394
4395
4396
4397
4398
4399
4400
4401
4402
4403
4404
4405
4406
4407
4408
4409
4410
4411
4412
4413
4414
4415
4416
4417
4418
4419
4420
4421
4422
4423
4424
4425
4426
4427
4428
4429
4430
4431
4432
4433
4434
4435
4436
4437
4438
4439
4440
4441
4442
4443
4444
4445
4446
4447
4448
4449
4450
4451
4452
4453
4454
4455
4456
4457
4458
4459
4460
4461
4462
4463
4464
4465
4466
4467
4468
4469
4470
4471
4472
4473
4474
4475
4476
4477
4478
4479
4480
4481
4482
4483
4484
4485
4486
4487
4488
4489
4490
4491
4492
4493
4494
4495
4496
4497
4498
4499
4500
4501
4502
4503
4504
4505
4506
4507
4508
4509
4510
4511
4512
4513
4514
4515
4516
4517
4518
4519
4520
4521
4522
4523
4524
4525
4526
4527
4528
4529
4530
4531
4532
4533
4534
4535
4536
4537
4538
4539
4540
4541
4542
4543
4544
4545
4546
4547
4548
4549
4550
4551
4552
4553
4554
4555
4556
4557
4558
4559
4560
4561
4562
4563
4564
4565
4566
4567
4568
4569
4570
4571
4572
4573
4574
4575
4576
4577
4578
4579
4580
4581
4582
4583
4584
4585
4586
4587
4588
4589
4590
4591
4592
4593
4594
4595
4596
4597
4598
4599
4600
4601
4602
4603
4604
4605
4606
4607
4608
4609
4610
4611
4612
4613
4614
4615
4616
4617
4618
4619
4620
4621
4622
4623
4624
4625
4626
4627
4628
4629
4630
4631
4632
4633
4634
4635
4636
4637
4638
4639
4640
4641
4642
4643
4644
4645
4646
4647
4648
4649
4650
4651
4652
4653
4654
4655
4656
4657
4658
4659
4660
4661
4662
4663
4664
4665
4666
4667
4668
4669
4670
4671
4672
4673
4674
4675
4676
4677
4678
4679
4680
4681
4682
4683
4684
4685
4686
4687
4688
4689
4690
4691
4692
4693
4694
4695
4696
4697
4698
4699
4700
4701
4702
4703
4704
4705
4706
4707
4708
4709
4710
4711
4712
4713
4714
4715
4716
4717
4718
4719
4720
4721
4722
4723
4724
4725
4726
4727
4728
4729
4730
4731
4732
4733
4734
4735
4736
4737
4738
4739
4740
4741
4742
4743
4744
4745
4746
4747
4748
4749
4750
4751
4752
4753
4754
4755
4756
4757
4758
4759
4760
4761
4762
4763
4764
4765
4766
4767
4768
4769
4770
4771
4772
4773
4774
4775
4776
4777
4778
4779
4780
4781
4782
4783
4784
4785
4786
4787
4788
4789
4790
4791
4792
4793
4794
4795
4796
4797
4798
4799
4800
4801
4802
4803
4804
4805
4806
4807
4808
4809
4810
4811
4812
4813
4814
4815
4816
4817
4818
4819
4820
4821
4822
4823
4824
4825
4826
4827
4828
4829
4830
4831
4832
4833
4834
4835
4836
4837
4838
4839
4840
4841
4842
4843
4844
4845
4846
4847
4848
4849
4850
4851
4852
4853
4854
4855
4856
4857
4858
4859
4860
4861
4862
4863
4864
4865
4866
4867
4868
4869
4870
4871
4872
4873
4874
4875
4876
4877
4878
4879
4880
4881
4882
4883
4884
4885
4886
4887
4888
4889
4890
4891
4892
4893
4894
4895
4896
4897
4898
4899
4900
4901
4902
4903
4904
4905
4906
4907
4908
4909
4910
4911
4912
4913
4914
4915
4916
4917
4918
4919
4920
4921
4922
4923
4924
4925
4926
4927
4928
4929
4930
4931
4932
4933
4934
4935
4936
4937
4938
4939
4940
4941
4942
4943
4944
4945
4946
4947
4948
4949
4950
4951
4952
4953
4954
4955
4956
4957
4958
4959
4960
4961
4962
4963
4964
4965
4966
4967
4968
4969
4970
4971
4972
4973
4974
4975
4976
4977
4978
4979
4980
4981
4982
4983
4984
4985
4986
4987
4988
4989
4990
4991
4992
4993
4994
4995
4996
4997
4998
4999
5000
5001
5002
5003
5004
5005
5006
5007
5008
5009
5010
5011
5012
5013
5014
5015
5016
5017
5018
5019
5020
5021
5022
5023
5024
5025
5026
5027
5028
5029
5030
5031
5032
5033
5034
5035
5036
5037
5038
5039
5040
5041
5042
5043
5044
5045
5046
5047
5048
5049
5050
5051
5052
5053
5054
5055
5056
5057
5058
5059
5060
5061
5062
5063
5064
5065
5066
5067
5068
5069
5070
5071
5072
5073
5074
5075
5076
5077
5078
5079
5080
5081
5082
5083
5084
5085
5086
5087
5088
5089
5090
5091
5092
5093
5094
5095
5096
5097
5098
5099
5100
5101
5102
5103
5104
5105
5106
5107
5108
5109
5110
5111
5112
5113
5114
5115
5116
5117
5118
5119
5120
5121
5122
5123
5124
5125
5126
5127
5128
5129
5130
5131
5132
5133
5134
5135
5136
5137
5138
5139
5140
5141
5142
5143
5144
5145
5146
5147
5148
5149
5150
5151
5152
5153
5154
5155
5156
5157
5158
5159
5160
5161
5162
5163
5164
5165
5166
5167
5168
5169
5170
5171
5172
5173
5174
5175
5176
5177
5178
5179
5180
5181
5182
5183
5184
5185
5186
5187
5188
5189
5190
5191
5192
5193
5194
5195
5196
5197
5198
5199
5200
5201
5202
5203
5204
5205
5206
5207
5208
5209
5210
5211
5212
5213
5214
5215
5216
5217
5218
5219
5220
5221
5222
5223
5224
5225
5226
5227
5228
5229
5230
5231
5232
5233
5234
5235
5236
5237
5238
5239
5240
5241
5242
5243
5244
5245
5246
5247
5248
5249
5250
5251
5252
5253
5254
5255
5256
5257
5258
5259
5260
5261
5262
5263
5264
5265
5266
5267
5268
5269
5270
5271
5272
5273
5274
5275
5276
5277
5278
5279
5280
5281
5282
5283
5284
5285
5286
5287
5288
5289
5290
5291
5292
5293
5294
5295
5296
5297
5298
5299
5300
5301
5302
5303
5304
5305
5306
5307
5308
5309
5310
5311
5312
5313
5314
5315
5316
5317
5318
5319
5320
5321
5322
5323
5324
5325
5326
5327
5328
5329
5330
5331
5332
5333
5334
5335
5336
5337
5338
5339
5340
5341
5342
5343
5344
5345
5346
5347
5348
5349
5350
5351
5352
5353
5354
5355
5356
5357
5358
5359
5360
5361
5362
5363
5364
5365
5366
5367
5368
5369
5370
5371
5372
5373
5374
5375
5376
5377
5378
5379
5380
5381
5382
5383
5384
5385
5386
5387
5388
5389
5390
5391
5392
5393
5394
5395
5396
5397
5398
5399
5400
5401
5402
5403
5404
5405
5406
5407
5408
5409
5410
5411
5412
5413
5414
5415
5416
5417
5418
5419
5420
5421
5422
5423
5424
5425
5426
5427
5428
5429
5430
5431
5432
5433
5434
5435
5436
5437
5438
5439
5440
5441
5442
5443
5444
5445
5446
5447
5448
5449
5450
5451
5452
5453
5454
5455
5456
5457
5458
5459
5460
5461
5462
5463
5464
5465
5466
5467
5468
5469
5470
5471
5472
5473
5474
5475
5476
5477
5478
5479
5480
5481
5482
5483
5484
5485
5486
5487
5488
5489
5490
5491
5492
5493
5494
5495
5496
5497
5498
5499
5500
5501
5502
5503
5504
5505
5506
5507
5508
5509
5510
5511
5512
5513
5514
5515
5516
5517
5518
5519
5520
5521
5522
5523
5524
5525
5526
5527
5528
5529
5530
5531
5532
5533
5534
5535
5536
5537
5538
5539
5540
5541
5542
5543
5544
5545
5546
5547
5548
5549
5550
5551
5552
5553
5554
5555
5556
5557
5558
5559
5560
5561
5562
5563
5564
5565
5566
5567
5568
5569
5570
5571
5572
5573
5574
5575
5576
5577
5578
5579
5580
5581
5582
5583
5584
5585
5586
5587
5588
5589
5590
5591
5592
5593
5594
5595
5596
5597
5598
5599
5600
5601
5602
5603
5604
5605
5606
5607
5608
5609
5610
5611
5612
5613
5614
5615
5616
5617
5618
5619
5620
5621
5622
5623
5624
5625
5626
5627
5628
5629
5630
5631
5632
5633
5634
5635
5636
5637
5638
5639
5640
5641
5642
5643
5644
5645
5646
5647
5648
5649
5650
5651
5652
5653
5654
5655
5656
5657
5658
5659
5660
5661
5662
5663
5664
5665
5666
5667
5668
5669
5670
5671
5672
5673
5674
5675
5676
5677
5678
5679
5680
5681
5682
5683
5684
5685
5686
5687
5688
5689
5690
5691
5692
5693
5694
5695
5696
5697
5698
5699
5700
5701
5702
5703
5704
5705
5706
5707
5708
5709
5710
5711
5712
5713
5714
5715
5716
5717
5718
5719
5720
5721
5722
5723
5724
5725
5726
5727
5728
5729
5730
5731
5732
5733
5734
5735
5736
5737
5738
5739
5740
5741
5742
5743
5744
5745
5746
5747
5748
5749
5750
5751
5752
5753
5754
5755
5756
5757
5758
5759
5760
5761
5762
5763
5764
5765
5766
5767
5768
5769
5770
5771
5772
5773
5774
5775
5776
5777
5778
5779
5780
5781
5782
5783
5784
5785
5786
5787
5788
5789
5790
5791
5792
5793
5794
5795
5796
5797
5798
5799
5800
5801
5802
5803
5804
5805
5806
5807
5808
5809
5810
5811
5812
5813
5814
5815
5816
5817
5818
5819
5820
5821
5822
5823
5824
5825
5826
5827
5828
5829
5830
5831
5832
5833
5834
5835
5836
5837
5838
5839
5840
5841
5842
5843
5844
5845
5846
5847
5848
5849
5850
5851
5852
5853
5854
5855
5856
5857
5858
5859
5860
5861
5862
5863
5864
5865
5866
5867
5868
5869
5870
5871
5872
5873
5874
5875
5876
5877
5878
5879
5880
5881
5882
5883
5884
5885
5886
5887
5888
5889
5890
5891
5892
5893
5894
5895
5896
5897
5898
5899
5900
5901
5902
5903
5904
5905
5906
5907
5908
5909
5910
5911
5912
5913
5914
5915
5916
5917
5918
5919
5920
5921
5922
5923
5924
5925
5926
5927
5928
5929
5930
5931
5932
5933
5934
5935
5936
5937
5938
5939
5940
5941
5942
5943
5944
5945
5946
5947
5948
5949
5950
5951
5952
5953
5954
5955
5956
5957
5958
5959
5960
5961
5962
5963
5964
5965
5966
5967
5968
5969
5970
5971
5972
5973
5974
5975
5976
5977
5978
5979
5980
5981
5982
5983
5984
5985
5986
5987
5988
5989
5990
5991
5992
5993
5994
5995
5996
5997
5998
5999
6000
6001
6002
6003
6004
6005
6006
6007
6008
6009
6010
6011
6012
6013
6014
6015
6016
6017
6018
6019
6020
6021
6022
6023
6024
6025
6026
6027
6028
6029
6030
6031
6032
6033
6034
6035
6036
6037
6038
6039
6040
6041
6042
6043
6044
6045
6046
6047
6048
6049
6050
6051
6052
6053
6054
6055
6056
6057
6058
6059
6060
6061
6062
6063
6064
6065
6066
6067
6068
6069
6070
6071
6072
6073
6074
6075
6076
6077
6078
6079
6080
6081
6082
6083
6084
6085
6086
6087
6088
6089
6090
6091
6092
6093
6094
6095
6096
6097
6098
6099
6100
6101
6102
6103
6104
6105
6106
6107
6108
6109
6110
6111
6112
6113
6114
6115
6116
6117
6118
6119
6120
6121
6122
6123
6124
6125
6126
6127
6128
6129
6130
6131
6132
6133
6134
6135
6136
6137
6138
6139
6140
6141
6142
6143
6144
6145
6146
6147
6148
6149
6150
6151
6152
6153
6154
6155
6156
6157
6158
6159
6160
6161
6162
6163
6164
6165
6166
6167
6168
6169
6170
6171
6172
6173
6174
6175
6176
6177
6178
6179
6180
6181
6182
6183
6184
6185
6186
6187
6188
6189
6190
6191
6192
6193
6194
6195
6196
6197
6198
6199
6200
6201
6202
6203
6204
6205
6206
6207
6208
6209
6210
6211
6212
6213
6214
6215
6216
6217
6218
6219
6220
6221
6222
6223
6224
6225
6226
6227
6228
6229
6230
6231
6232
6233
6234
6235
6236
6237
6238
6239
6240
6241
6242
6243
6244
6245
6246
6247
6248
6249
6250
6251
6252
6253
6254
6255
6256
6257
6258
6259
6260
6261
6262
6263
6264
6265
6266
6267
6268
6269
6270
6271
6272
6273
6274
6275
6276
6277
6278
6279
6280
6281
6282
6283
6284
6285
6286
6287
6288
6289
6290
6291
6292
6293
6294
6295
6296
6297
6298
6299
6300
6301
6302
6303
6304
6305
6306
6307
6308
6309
6310
6311
6312
6313
6314
6315
6316
6317
6318
6319
6320
6321
6322
6323
6324
6325
6326
6327
6328
6329
6330
6331
6332
6333
6334
6335
6336
6337
6338
6339
6340
6341
6342
6343
6344
6345
6346
6347
6348
6349
6350
6351
6352
6353
6354
6355
6356
6357
6358
6359
6360
6361
6362
6363
6364
6365
6366
6367
6368
6369
6370
6371
6372
6373
6374
6375
6376
6377
6378
6379
6380
6381
6382
6383
6384
6385
6386
6387
6388
6389
6390
6391
6392
6393
6394
6395
6396
6397
6398
6399
6400
6401
6402
6403
6404
6405
6406
6407
6408
6409
6410
6411
6412
6413
6414
6415
6416
6417
6418
6419
6420
6421
6422
6423
6424
6425
6426
6427
6428
6429
6430
6431
6432
6433
6434
6435
6436
6437
6438
6439
6440
6441
6442
6443
6444
6445
6446
6447
6448
6449
6450
6451
6452
6453
6454
6455
6456
6457
6458
6459
6460
6461
6462
6463
6464
6465
6466
6467
6468
6469
6470
6471
6472
6473
6474
6475
6476
6477
6478
6479
6480
6481
6482
6483
6484
6485
6486
6487
6488
6489
6490
6491
6492
6493
6494
6495
6496
6497
6498
6499
6500
6501
6502
6503
6504
6505
6506
6507
6508
6509
6510
6511
6512
6513
6514
6515
6516
6517
6518
6519
6520
6521
6522
6523
6524
6525
6526
6527
6528
6529
6530
6531
6532
6533
6534
6535
6536
6537
6538
6539
6540
6541
6542
6543
6544
6545
6546
6547
6548
6549
6550
6551
6552
6553
6554
6555
6556
6557
6558
6559
6560
6561
6562
6563
6564
6565
6566
6567
6568
6569
6570
6571
6572
6573
6574
6575
6576
6577
6578
6579
6580
6581
6582
6583
6584
6585
6586
6587
6588
6589
6590
6591
6592
6593
6594
6595
6596
6597
6598
6599
6600
6601
6602
6603
6604
6605
6606
6607
6608
6609
6610
6611
6612
6613
6614
6615
6616
6617
6618
6619
6620
6621
6622
6623
6624
6625
6626
6627
6628
6629
6630
6631
6632
6633
6634
6635
6636
6637
6638
6639
6640
6641
6642
6643
6644
6645
6646
6647
6648
6649
6650
6651
6652
6653
6654
6655
6656
6657
6658
6659
6660
6661
6662
6663
6664
6665
6666
6667
6668
6669
6670
6671
6672
6673
6674
6675
6676
6677
6678
6679
6680
6681
6682
6683
6684
6685
6686
6687
6688
6689
6690
6691
6692
6693
6694
6695
6696
6697
6698
6699
6700
6701
6702
6703
6704
6705
6706
6707
6708
6709
6710
6711
6712
6713
6714
6715
6716
6717
6718
6719
6720
6721
6722
6723
6724
6725
6726
6727
6728
6729
6730
6731
6732
6733
6734
6735
6736
6737
6738
6739
6740
6741
6742
6743
6744
6745
6746
6747
6748
6749
6750
6751
6752
6753
6754
6755
6756
6757
6758
6759
6760
6761
6762
6763
6764
6765
6766
6767
6768
6769
6770
6771
6772
6773
6774
6775
6776
6777
6778
6779
6780
6781
6782
6783
6784
6785
6786
6787
6788
6789
6790
6791
6792
6793
6794
6795
6796
6797
6798
6799
6800
6801
6802
6803
6804
6805
6806
6807
6808
6809
6810
6811
6812
6813
6814
6815
6816
6817
6818
6819
6820
6821
6822
6823
6824
6825
6826
6827
6828
6829
6830
6831
6832
6833
6834
6835
6836
6837
6838
6839
6840
6841
6842
6843
6844
6845
6846
6847
6848
6849
6850
6851
6852
6853
6854
6855
6856
6857
6858
6859
6860
6861
6862
6863
6864
6865
6866
6867
6868
6869
6870
6871
6872
6873
6874
6875
6876
6877
6878
6879
6880
6881
6882
6883
6884
6885
6886
6887
6888
6889
6890
6891
6892
6893
6894
6895
6896
6897
6898
6899
6900
6901
6902
6903
6904
6905
6906
6907
6908
6909
6910
6911
6912
6913
6914
6915
6916
6917
6918
6919
6920
6921
6922
6923
6924
6925
6926
6927
6928
6929
6930
6931
6932
6933
6934
6935
6936
6937
6938
6939
6940
6941
6942
6943
6944
6945
6946
6947
6948
6949
6950
6951
6952
6953
6954
6955
6956
6957
6958
6959
6960
6961
6962
6963
6964
6965
6966
6967
6968
6969
6970
6971
6972
6973
6974
6975
6976
6977
6978
6979
6980
6981
6982
6983
6984
6985
6986
6987
6988
6989
6990
6991
6992
6993
6994
6995
6996
6997
6998
6999
7000
7001
7002
7003
7004
7005
7006
7007
7008
7009
7010
7011
7012
7013
7014
7015
7016
7017
7018
7019
7020
7021
7022
7023
7024
7025
7026
7027
7028
7029
7030
7031
7032
7033
7034
7035
7036
7037
7038
7039
7040
7041
7042
7043
7044
7045
7046
7047
7048
7049
7050
7051
7052
7053
7054
7055
7056
7057
7058
7059
7060
7061
7062
7063
7064
7065
7066
7067
7068
7069
7070
7071
7072
7073
7074
7075
7076
7077
7078
7079
7080
7081
7082
7083
7084
7085
7086
7087
7088
7089
7090
7091
7092
7093
7094
7095
7096
7097
7098
7099
7100
7101
7102
7103
7104
7105
7106
7107
7108
7109
7110
7111
7112
7113
7114
7115
7116
7117
7118
7119
7120
7121
7122
7123
7124
7125
7126
7127
7128
7129
7130
7131
7132
7133
7134
7135
7136
7137
7138
7139
7140
7141
7142
7143
7144
7145
7146
7147
7148
7149
7150
7151
7152
7153
7154
7155
7156
7157
7158
7159
7160
7161
7162
7163
7164
7165
7166
7167
7168
7169
7170
7171
7172
7173
7174
7175
7176
7177
7178
7179
7180
7181
7182
7183
7184
7185
7186
7187
7188
7189
7190
7191
7192
7193
7194
7195
7196
7197
7198
7199
7200
7201
7202
7203
7204
7205
7206
7207
7208
7209
7210
7211
7212
7213
7214
7215
7216
7217
7218
7219
7220
7221
7222
7223
7224
7225
7226
7227
7228
7229
7230
7231
7232
7233
7234
7235
7236
7237
7238
7239
7240
7241
7242
7243
7244
7245
7246
7247
7248
7249
7250
7251
7252
7253
7254
7255
7256
7257
7258
7259
7260
7261
7262
7263
7264
7265
7266
7267
7268
7269
7270
7271
7272
7273
7274
7275
7276
7277
7278
7279
7280
7281
7282
7283
7284
7285
7286
7287
7288
7289
7290
7291
7292
7293
7294
7295
7296
7297
7298
7299
7300
7301
7302
7303
7304
7305
7306
7307
7308
7309
7310
7311
7312
7313
7314
7315
7316
7317
7318
7319
7320
7321
7322
7323
7324
7325
7326
7327
7328
7329
7330
7331
7332
7333
7334
7335
7336
7337
7338
7339
7340
7341
7342
7343
7344
7345
7346
7347
7348
7349
7350
7351
7352
7353
7354
7355
7356
7357
7358
7359
7360
7361
7362
7363
7364
7365
7366
7367
7368
7369
7370
7371
7372
7373
7374
7375
7376
7377
7378
7379
7380
7381
7382
7383
7384
7385
7386
7387
7388
7389
7390
7391
7392
7393
7394
7395
7396
7397
7398
7399
7400
7401
7402
7403
7404
7405
7406
7407
7408
7409
7410
7411
7412
7413
7414
7415
7416
7417
7418
7419
7420
7421
7422
7423
7424
7425
7426
7427
7428
7429
7430
7431
7432
7433
7434
7435
7436
7437
7438
7439
7440
7441
7442
7443
7444
7445
7446
7447
7448
7449
7450
7451
7452
7453
7454
7455
7456
7457
7458
7459
7460
7461
7462
7463
7464
7465
7466
7467
7468
7469
7470
7471
7472
7473
7474
7475
7476
7477
7478
7479
7480
7481
7482
7483
7484
7485
7486
7487
7488
7489
7490
7491
7492
7493
7494
7495
7496
7497
7498
7499
7500
7501
7502
7503
7504
7505
7506
7507
7508
7509
7510
7511
7512
7513
7514
7515
7516
7517
7518
7519
7520
7521
7522
7523
7524
7525
7526
7527
7528
7529
7530
7531
7532
7533
7534
7535
7536
7537
7538
7539
7540
7541
7542
7543
7544
7545
7546
7547
7548
7549
7550
7551
7552
7553
7554
7555
7556
7557
7558
7559
7560
7561
7562
7563
7564
7565
7566
7567
7568
7569
7570
7571
7572
7573
7574
7575
7576
7577
7578
7579
7580
7581
7582
7583
7584
7585
7586
7587
7588
7589
7590
7591
7592
7593
7594
7595
7596
7597
7598
7599
7600
7601
7602
7603
7604
7605
7606
7607
7608
7609
7610
7611
7612
7613
7614
7615
7616
7617
7618
7619
7620
7621
7622
7623
7624
7625
7626
7627
7628
7629
7630
7631
7632
7633
7634
7635
7636
7637
7638
7639
7640
7641
7642
7643
7644
7645
7646
7647
7648
7649
7650
7651
7652
7653
7654
7655
7656
7657
7658
7659
7660
7661
7662
7663
7664
7665
7666
7667
7668
7669
7670
7671
7672
7673
7674
7675
7676
7677
7678
7679
7680
7681
7682
7683
7684
7685
7686
7687
7688
7689
7690
7691
7692
7693
7694
7695
7696
7697
7698
7699
7700
7701
7702
7703
7704
7705
7706
7707
7708
7709
7710
7711
7712
7713
7714
7715
7716
7717
7718
7719
7720
7721
7722
7723
7724
7725
7726
7727
7728
7729
7730
7731
7732
7733
7734
7735
7736
7737
7738
7739
7740
7741
7742
7743
7744
7745
7746
7747
7748
7749
7750
7751
7752
7753
7754
7755
7756
7757
7758
7759
7760
7761
7762
7763
7764
7765
7766
7767
7768
7769
7770
7771
7772
7773
7774
7775
7776
7777
7778
7779
7780
7781
7782
7783
7784
7785
7786
7787
7788
7789
7790
7791
7792
7793
7794
7795
7796
7797
7798
7799
7800
7801
7802
7803
7804
7805
7806
7807
7808
7809
7810
7811
7812
7813
7814
7815
7816
7817
7818
7819
7820
7821
7822
7823
7824
7825
7826
7827
7828
7829
7830
7831
7832
7833
7834
7835
7836
7837
7838
7839
7840
7841
7842
7843
7844
7845
7846
7847
7848
7849
7850
7851
7852
7853
7854
7855
7856
7857
7858
7859
7860
7861
7862
7863
7864
7865
7866
7867
7868
7869
7870
7871
7872
7873
7874
7875
7876
7877
7878
7879
7880
7881
7882
7883
7884
7885
7886
7887
7888
7889
7890
7891
7892
7893
7894
7895
7896
7897
7898
7899
7900
7901
7902
7903
7904
7905
7906
7907
7908
7909
7910
7911
7912
7913
7914
7915
7916
7917
7918
7919
7920
7921
7922
7923
7924
7925
7926
7927
7928
7929
7930
7931
7932
7933
7934
7935
7936
7937
7938
7939
7940
7941
7942
7943
7944
7945
7946
7947
7948
7949
7950
7951
7952
7953
7954
7955
7956
7957
7958
7959
7960
7961
7962
7963
7964
7965
7966
7967
7968
7969
7970
7971
7972
7973
7974
7975
7976
7977
7978
7979
7980
7981
7982
7983
7984
7985
7986
7987
7988
7989
7990
7991
7992
7993
7994
7995
7996
7997
7998
7999
8000
8001
8002
8003
8004
8005
8006
8007
8008
8009
8010
8011
8012
8013
8014
8015
8016
8017
8018
8019
8020
8021
8022
8023
8024
8025
8026
8027
8028
8029
8030
8031
8032
8033
8034
8035
8036
8037
8038
8039
8040
8041
8042
8043
8044
8045
8046
8047
8048
8049
8050
8051
8052
8053
8054
8055
8056
8057
8058
8059
8060
8061
8062
8063
8064
8065
8066
8067
8068
8069
8070
8071
8072
8073
8074
8075
8076
8077
8078
8079
8080
8081
8082
8083
8084
8085
8086
8087
8088
8089
8090
8091
8092
8093
8094
8095
8096
8097
8098
8099
8100
8101
8102
8103
8104
8105
8106
8107
8108
8109
8110
8111
8112
8113
8114
8115
8116
8117
8118
8119
8120
8121
8122
8123
8124
8125
8126
8127
8128
8129
8130
8131
8132
8133
8134
8135
8136
8137
8138
8139
8140
8141
8142
8143
8144
8145
8146
8147
8148
8149
8150
8151
8152
8153
8154
8155
8156
8157
8158
8159
8160
8161
8162
8163
8164
8165
8166
8167
8168
8169
8170
8171
8172
8173
8174
8175
8176
8177
8178
8179
8180
8181
8182
8183
8184
8185
8186
8187
8188
8189
8190
8191
8192
8193
8194
8195
8196
8197
8198
8199
8200
8201
8202
8203
8204
8205
8206
8207
8208
8209
8210
8211
8212
8213
8214
8215
8216
8217
8218
8219
8220
8221
8222
8223
8224
8225
8226
8227
8228
8229
8230
8231
8232
8233
8234
8235
8236
8237
8238
8239
8240
8241
8242
8243
8244
8245
8246
8247
8248
8249
8250
8251
8252
8253
8254
8255
8256
8257
8258
8259
8260
8261
8262
8263
8264
8265
8266
8267
8268
8269
8270
8271
8272
8273
8274
8275
8276
8277
8278
8279
8280
8281
8282
8283
8284
8285
8286
8287
8288
8289
8290
8291
8292
8293
8294
8295
8296
8297
8298
8299
8300
8301
8302
8303
8304
8305
8306
8307
8308
8309
8310
8311
8312
8313
8314
8315
8316
8317
8318
8319
8320
8321
8322
8323
8324
8325
8326
8327
8328
8329
8330
8331
8332
8333
8334
8335
8336
8337
8338
8339
8340
8341
8342
8343
8344
8345
8346
8347
8348
8349
8350
8351
8352
8353
8354
8355
8356
8357
8358
8359
8360
8361
8362
8363
8364
8365
8366
8367
8368
8369
8370
8371
8372
8373
8374
8375
8376
8377
8378
8379
8380
8381
8382
8383
8384
8385
8386
8387
8388
8389
8390
8391
8392
8393
8394
8395
8396
8397
8398
8399
8400
8401
8402
8403
8404
8405
8406
8407
8408
8409
8410
8411
8412
8413
8414
8415
8416
8417
8418
8419
8420
8421
8422
8423
8424
8425
8426
8427
8428
8429
8430
8431
8432
8433
8434
8435
8436
8437
8438
8439
8440
8441
8442
8443
8444
8445
8446
8447
8448
8449
8450
8451
8452
8453
8454
8455
8456
8457
8458
8459
8460
8461
8462
8463
8464
8465
8466
8467
8468
8469
8470
8471
8472
8473
8474
8475
8476
8477
8478
8479
8480
8481
8482
8483
8484
8485
8486
8487
8488
8489
8490
8491
8492
8493
8494
8495
8496
8497
8498
8499
8500
8501
8502
8503
8504
8505
8506
8507
8508
8509
8510
8511
8512
8513
8514
8515
8516
8517
8518
8519
8520
8521
8522
8523
8524
8525
8526
8527
8528
8529
8530
8531
8532
8533
8534
8535
8536
8537
8538
8539
8540
8541
8542
8543
8544
8545
8546
8547
8548
8549
8550
8551
8552
8553
8554
8555
8556
8557
8558
8559
8560
8561
8562
8563
8564
8565
8566
8567
8568
8569
8570
8571
8572
8573
8574
8575
8576
8577
8578
8579
8580
8581
8582
8583
8584
8585
8586
8587
8588
8589
8590
8591
8592
8593
8594
8595
8596
8597
8598
8599
8600
8601
8602
8603
8604
8605
8606
8607
8608
8609
8610
8611
8612
8613
8614
8615
8616
8617
8618
8619
8620
8621
8622
8623
8624
8625
8626
8627
8628
8629
8630
8631
8632
8633
8634
8635
8636
8637
8638
8639
8640
8641
8642
8643
8644
8645
8646
8647
8648
8649
8650
8651
8652
8653
8654
8655
8656
8657
8658
8659
8660
8661
8662
8663
8664
8665
8666
8667
8668
8669
8670
8671
8672
8673
8674
8675
8676
8677
8678
8679
8680
8681
8682
8683
8684
8685
8686
8687
8688
8689
8690
8691
8692
8693
8694
8695
8696
8697
8698
8699
8700
8701
8702
8703
8704
8705
8706
8707
8708
8709
8710
8711
8712
8713
8714
8715
8716
8717
8718
8719
8720
8721
8722
8723
8724
8725
8726
8727
8728
8729
8730
8731
8732
8733
8734
8735
8736
8737
8738
8739
8740
8741
8742
8743
8744
8745
8746
8747
8748
8749
8750
8751
8752
8753
8754
8755
8756
8757
8758
8759
8760
8761
8762
8763
8764
8765
8766
8767
8768
8769
8770
8771
8772
8773
8774
8775
8776
8777
8778
8779
8780
8781
8782
8783
8784
8785
8786
8787
8788
8789
8790
8791
8792
8793
8794
8795
8796
8797
8798
8799
8800
8801
8802
8803
8804
8805
8806
8807
8808
8809
8810
8811
8812
8813
8814
8815
8816
8817
8818
8819
8820
8821
8822
8823
8824
8825
8826
8827
8828
8829
8830
8831
8832
8833
8834
8835
8836
8837
8838
8839
8840
8841
8842
8843
8844
8845
8846
8847
8848
8849
8850
8851
8852
8853
8854
8855
8856
8857
8858
8859
8860
8861
8862
8863
8864
8865
8866
8867
8868
8869
8870
8871
8872
8873
8874
8875
8876
8877
8878
8879
8880
8881
8882
8883
8884
8885
8886
8887
8888
8889
8890
8891
8892
8893
8894
8895
8896
8897
8898
8899
8900
8901
8902
8903
8904
8905
8906
8907
8908
8909
8910
8911
8912
8913
8914
8915
8916
8917
8918
8919
8920
8921
8922
8923
8924
8925
8926
8927
8928
8929
8930
8931
8932
8933
8934
8935
8936
8937
8938
8939
8940
8941
8942
8943
8944
8945
8946
8947
8948
8949
8950
8951
8952
8953
8954
8955
8956
8957
8958
8959
8960
8961
8962
8963
8964
8965
8966
8967
8968
8969
8970
8971
8972
8973
8974
8975
8976
8977
8978
8979
8980
8981
8982
8983
8984
8985
8986
8987
8988
8989
8990
8991
8992
8993
8994
8995
8996
8997
8998
8999
9000
9001
9002
9003
9004
9005
9006
9007
9008
9009
9010
9011
9012
9013
9014
9015
9016
9017
9018
9019
9020
9021
9022
9023
9024
9025
9026
9027
9028
9029
9030
9031
9032
9033
9034
9035
9036
9037
9038
9039
9040
9041
9042
9043
9044
9045
9046
9047
9048
9049
9050
9051
9052
9053
9054
9055
9056
9057
9058
9059
9060
9061
9062
9063
9064
9065
9066
9067
9068
9069
9070
9071
9072
9073
9074
9075
9076
9077
9078
9079
9080
9081
9082
9083
9084
9085
9086
9087
9088
9089
9090
9091
9092
9093
9094
9095
9096
9097
9098
9099
9100
9101
9102
9103
9104
9105
9106
9107
9108
9109
9110
9111
9112
9113
9114
9115
9116
9117
9118
9119
9120
9121
9122
9123
9124
9125
9126
9127
9128
9129
9130
9131
9132
9133
9134
9135
9136
9137
9138
9139
9140
9141
9142
9143
9144
9145
9146
9147
9148
9149
9150
9151
9152
9153
9154
9155
9156
9157
9158
9159
9160
9161
9162
9163
9164
9165
9166
9167
9168
9169
9170
9171
9172
9173
9174
9175
9176
9177
9178
9179
9180
9181
9182
9183
9184
9185
9186
9187
9188
9189
9190
9191
9192
9193
9194
9195
9196
9197
9198
9199
9200
9201
9202
9203
9204
9205
9206
9207
9208
9209
9210
9211
9212
9213
9214
9215
9216
9217
9218
9219
9220
9221
9222
9223
9224
9225
9226
9227
9228
9229
9230
9231
9232
9233
9234
9235
9236
9237
9238
9239
9240
9241
9242
9243
9244
9245
9246
9247
9248
9249
9250
9251
9252
9253
9254
9255
9256
9257
9258
9259
9260
9261
9262
9263
9264
9265
9266
9267
9268
9269
9270
9271
9272
9273
9274
9275
9276
9277
9278
9279
9280
9281
9282
9283
9284
9285
9286
9287
9288
9289
9290
9291
9292
9293
9294
9295
9296
9297
9298
9299
9300
9301
9302
9303
9304
9305
9306
9307
9308
9309
9310
9311
9312
9313
9314
9315
9316
9317
9318
9319
9320
9321
9322
9323
9324
9325
9326
9327
9328
9329
9330
9331
9332
9333
9334
9335
9336
9337
9338
9339
9340
9341
9342
9343
9344
9345
9346
9347
9348
9349
9350
9351
9352
9353
9354
9355
9356
9357
9358
9359
9360
9361
9362
9363
9364
9365
9366
9367
9368
9369
9370
9371
9372
9373
9374
9375
9376
9377
9378
9379
9380
9381
9382
9383
9384
9385
9386
9387
9388
9389
9390
9391
9392
9393
9394
9395
9396
9397
9398
9399
9400
9401
9402
9403
9404
9405
9406
9407
9408
9409
9410
9411
9412
9413
9414
9415
9416
9417
9418
9419
9420
9421
9422
9423
9424
9425
9426
9427
9428
9429
9430
9431
9432
9433
9434
9435
9436
9437
9438
9439
9440
9441
9442
9443
9444
9445
9446
9447
9448
9449
9450
9451
9452
9453
9454
9455
9456
9457
9458
9459
9460
9461
9462
9463
9464
9465
9466
9467
9468
9469
9470
9471
9472
9473
9474
9475
9476
9477
9478
9479
9480
9481
9482
9483
9484
9485
9486
9487
9488
9489
9490
9491
9492
9493
9494
9495
9496
9497
9498
9499
9500
9501
9502
9503
9504
9505
9506
9507
9508
9509
9510
9511
9512
9513
9514
9515
9516
9517
9518
9519
9520
9521
9522
9523
9524
9525
9526
9527
9528
9529
9530
9531
9532
9533
9534
9535
9536
9537
9538
9539
9540
9541
9542
9543
9544
9545
9546
9547
9548
9549
9550
9551
9552
9553
9554
9555
9556
9557
9558
9559
9560
9561
9562
9563
9564
9565
9566
9567
9568
9569
9570
9571
9572
9573
9574
9575
9576
9577
9578
9579
9580
9581
9582
9583
9584
9585
9586
9587
9588
9589
9590
9591
9592
9593
9594
9595
9596
9597
9598
9599
9600
9601
9602
9603
9604
9605
9606
9607
9608
9609
9610
9611
9612
9613
9614
9615
9616
9617
9618
9619
9620
9621
9622
9623
9624
9625
9626
9627
9628
9629
9630
9631
9632
9633
9634
9635
9636
9637
9638
9639
9640
9641
9642
9643
9644
9645
9646
9647
9648
9649
9650
9651
9652
9653
9654
9655
9656
9657
9658
9659
9660
9661
9662
9663
9664
9665
9666
9667
9668
9669
9670
9671
9672
9673
9674
9675
9676
9677
9678
9679
9680
9681
9682
9683
9684
9685
9686
9687
9688
9689
9690
9691
9692
9693
9694
9695
9696
9697
9698
9699
9700
9701
9702
9703
9704
9705
9706
9707
9708
9709
9710
9711
9712
9713
9714
9715
9716
9717
9718
9719
9720
9721
9722
9723
9724
9725
9726
9727
9728
9729
9730
9731
9732
9733
9734
9735
9736
9737
9738
9739
9740
9741
9742
9743
9744
9745
9746
9747
9748
9749
9750
9751
9752
9753
9754
9755
9756
9757
9758
9759
9760
9761
9762
9763
9764
9765
9766
9767
9768
9769
9770
9771
9772
9773
9774
9775
9776
9777
9778
9779
9780
9781
9782
9783
9784
9785
9786
9787
9788
9789
9790
9791
9792
9793
9794
9795
9796
9797
9798
9799
9800
9801
9802
9803
9804
9805
9806
9807
9808
9809
9810
9811
9812
9813
9814
9815
9816
9817
9818
9819
9820
9821
9822
9823
9824
9825
9826
9827
9828
9829
9830
9831
9832
9833
9834
9835
9836
9837
9838
9839
9840
9841
9842
9843
9844
9845
9846
9847
9848
9849
9850
9851
9852
9853
9854
9855
9856
9857
9858
9859
9860
9861
9862
9863
9864
9865
9866
9867
9868
9869
9870
9871
9872
9873
9874
9875
9876
9877
9878
9879
9880
9881
9882
9883
9884
9885
9886
9887
9888
9889
9890
9891
9892
9893
9894
9895
9896
9897
9898
9899
9900
9901
9902
9903
9904
9905
9906
9907
9908
9909
9910
9911
9912
9913
9914
9915
9916
9917
9918
9919
9920
9921
9922
9923
9924
9925
9926
9927
9928
9929
9930
9931
9932
9933
9934
9935
9936
9937
9938
9939
9940
9941
9942
9943
9944
9945
9946
9947
9948
9949
9950
9951
9952
9953
9954
9955
9956
9957
9958
9959
9960
9961
9962
9963
9964
9965
9966
9967
9968
9969
9970
9971
9972
9973
9974
9975
9976
9977
9978
9979
9980
9981
9982
9983
9984
9985
9986
9987
9988
9989
9990
9991
9992
9993
9994
9995
9996
9997
9998
9999
10000
10001
10002
10003
10004
10005
10006
10007
10008
10009
10010
10011
10012
10013
10014
10015
10016
10017
10018
10019
10020
10021
10022
10023
10024
10025
10026
10027
10028
10029
10030
10031
10032
10033
10034
10035
10036
10037
10038
10039
10040
10041
10042
10043
10044
10045
10046
10047
10048
10049
10050
10051
10052
10053
10054
10055
10056
10057
10058
10059
10060
10061
10062
10063
10064
10065
10066
10067
10068
10069
10070
10071
10072
10073
10074
10075
10076
10077
10078
10079
10080
10081
10082
10083
10084
10085
10086
10087
10088
10089
10090
10091
10092
10093
10094
10095
10096
10097
10098
10099
10100
10101
10102
10103
10104
10105
10106
10107
10108
10109
10110
10111
10112
10113
10114
10115
10116
10117
10118
10119
10120
10121
10122
10123
10124
10125
10126
10127
10128
10129
10130
10131
10132
10133
10134
10135
10136
10137
10138
10139
10140
10141
10142
10143
10144
10145
10146
10147
10148
10149
10150
10151
10152
10153
10154
10155
10156
10157
10158
10159
10160
10161
10162
10163
10164
10165
10166
10167
10168
10169
10170
10171
10172
10173
10174
10175
10176
10177
10178
10179
10180
10181
10182
10183
10184
10185
10186
10187
10188
10189
10190
10191
10192
10193
10194
10195
10196
10197
10198
10199
10200
10201
10202
10203
10204
10205
10206
10207
10208
10209
10210
10211
10212
10213
10214
10215
10216
10217
10218
10219
10220
10221
10222
10223
10224
10225
10226
10227
10228
10229
10230
10231
10232
10233
10234
10235
10236
10237
10238
10239
10240
10241
10242
10243
10244
10245
10246
10247
10248
10249
10250
10251
10252
10253
10254
10255
10256
10257
10258
10259
10260
10261
10262
10263
10264
10265
10266
10267
10268
10269
10270
10271
10272
10273
10274
10275
10276
10277
10278
10279
10280
10281
10282
10283
10284
10285
10286
10287
10288
10289
10290
10291
10292
10293
10294
10295
10296
10297
10298
10299
10300
10301
10302
10303
10304
10305
10306
10307
10308
10309
10310
10311
10312
10313
10314
10315
10316
10317
10318
10319
10320
10321
10322
10323
10324
10325
10326
10327
10328
10329
10330
10331
10332
10333
10334
10335
10336
10337
10338
10339
10340
10341
10342
10343
10344
10345
10346
10347
10348
10349
10350
10351
10352
10353
10354
10355
10356
10357
10358
10359
10360
10361
10362
10363
10364
10365
10366
10367
10368
10369
10370
10371
10372
10373
10374
10375
10376
10377
10378
10379
10380
10381
10382
10383
10384
10385
10386
10387
10388
10389
10390
10391
10392
10393
10394
10395
10396
10397
10398
10399
10400
10401
10402
10403
10404
10405
10406
10407
10408
10409
10410
10411
10412
10413
10414
10415
10416
10417
10418
10419
10420
10421
10422
10423
10424
10425
10426
10427
10428
10429
10430
10431
10432
10433
10434
10435
10436
10437
10438
10439
10440
10441
10442
10443
10444
10445
10446
10447
10448
10449
10450
10451
10452
10453
10454
10455
10456
10457
10458
10459
10460
10461
10462
10463
10464
10465
10466
10467
10468
10469
10470
10471
10472
10473
10474
10475
10476
10477
10478
10479
10480
10481
10482
10483
10484
10485
10486
10487
10488
10489
10490
10491
10492
10493
10494
10495
10496
10497
10498
10499
10500
10501
10502
10503
10504
10505
10506
10507
10508
10509
10510
10511
10512
10513
10514
10515
10516
10517
10518
10519
10520
10521
10522
10523
10524
10525
10526
10527
10528
10529
10530
10531
10532
10533
10534
10535
10536
10537
10538
10539
10540
10541
10542
10543
10544
10545
10546
10547
10548
10549
10550
10551
10552
10553
10554
10555
10556
10557
10558
10559
10560
10561
10562
10563
10564
10565
10566
10567
10568
10569
10570
10571
10572
10573
10574
10575
10576
10577
10578
10579
10580
10581
10582
10583
10584
10585
10586
10587
10588
10589
10590
10591
10592
10593
10594
10595
10596
10597
10598
10599
10600
10601
10602
10603
10604
10605
10606
10607
10608
10609
10610
10611
10612
10613
10614
10615
10616
10617
10618
10619
10620
10621
10622
10623
10624
10625
10626
10627
10628
10629
10630
10631
10632
10633
10634
10635
10636
10637
10638
10639
10640
10641
10642
10643
10644
10645
10646
10647
10648
10649
10650
10651
10652
10653
10654
10655
10656
10657
10658
10659
10660
10661
10662
10663
10664
10665
10666
10667
10668
10669
10670
10671
10672
10673
10674
10675
10676
10677
10678
10679
10680
10681
10682
10683
10684
10685
10686
10687
10688
10689
10690
10691
10692
10693
10694
10695
10696
10697
10698
10699
10700
10701
10702
10703
10704
10705
10706
10707
10708
10709
10710
10711
10712
10713
10714
10715
10716
10717
10718
10719
10720
10721
10722
10723
10724
10725
10726
10727
10728
10729
10730
10731
10732
10733
10734
10735
10736
10737
10738
10739
10740
10741
10742
10743
10744
10745
10746
10747
10748
10749
10750
10751
10752
10753
10754
10755
10756
10757
10758
10759
10760
10761
10762
10763
10764
10765
10766
10767
10768
10769
10770
10771
10772
10773
10774
10775
10776
10777
10778
10779
10780
10781
10782
10783
10784
10785
10786
10787
10788
10789
10790
10791
10792
10793
10794
10795
10796
10797
10798
10799
10800
10801
10802
10803
10804
10805
10806
10807
10808
10809
10810
10811
10812
10813
10814
10815
10816
10817
10818
10819
10820
10821
10822
10823
10824
10825
10826
10827
10828
10829
10830
10831
10832
10833
10834
10835
10836
10837
10838
10839
10840
10841
10842
10843
10844
10845
10846
10847
10848
10849
10850
10851
10852
10853
10854
10855
10856
10857
10858
10859
10860
10861
10862
10863
10864
10865
10866
10867
10868
10869
10870
10871
10872
10873
10874
10875
10876
10877
10878
10879
10880
10881
10882
10883
10884
10885
10886
10887
10888
10889
10890
10891
10892
10893
10894
10895
10896
10897
10898
10899
10900
10901
10902
10903
10904
10905
10906
10907
10908
10909
10910
10911
10912
10913
10914
10915
10916
10917
10918
10919
10920
10921
10922
10923
10924
10925
10926
10927
10928
10929
10930
10931
10932
10933
10934
10935
10936
10937
10938
10939
10940
10941
10942
10943
10944
10945
10946
10947
10948
10949
10950
10951
10952
10953
10954
10955
10956
10957
10958
10959
10960
10961
10962
10963
10964
10965
10966
10967
10968
10969
10970
10971
10972
10973
10974
10975
10976
10977
10978
10979
10980
10981
10982
10983
10984
10985
10986
10987
10988
10989
10990
10991
10992
10993
10994
10995
10996
10997
10998
10999
11000
11001
11002
11003
11004
11005
11006
11007
11008
11009
11010
11011
11012
11013
11014
11015
11016
11017
11018
11019
11020
11021
11022
11023
11024
11025
11026
11027
11028
11029
11030
11031
11032
11033
11034
11035
11036
11037
11038
11039
11040
11041
11042
11043
11044
11045
11046
11047
11048
11049
11050
11051
11052
11053
11054
11055
11056
11057
11058
11059
11060
11061
11062
11063
11064
11065
11066
11067
11068
11069
11070
11071
11072
11073
11074
11075
11076
11077
11078
11079
11080
11081
11082
11083
11084
11085
11086
11087
11088
11089
11090
11091
11092
11093
11094
11095
11096
11097
11098
11099
11100
11101
11102
11103
11104
11105
11106
11107
11108
11109
11110
11111
11112
11113
11114
11115
11116
11117
11118
11119
11120
11121
11122
11123
11124
11125
11126
11127
11128
11129
11130
11131
11132
11133
11134
11135
11136
11137
11138
11139
11140
11141
11142
11143
11144
11145
11146
11147
11148
11149
11150
11151
11152
11153
11154
11155
11156
11157
11158
11159
11160
11161
11162
11163
11164
11165
11166
11167
11168
11169
11170
11171
11172
11173
11174
11175
11176
11177
11178
11179
11180
11181
11182
11183
11184
11185
11186
11187
11188
11189
11190
11191
11192
11193
11194
11195
11196
11197
11198
11199
11200
11201
11202
11203
11204
11205
11206
11207
11208
11209
11210
11211
11212
11213
11214
11215
11216
11217
11218
11219
11220
11221
11222
11223
11224
11225
11226
11227
11228
11229
11230
11231
11232
11233
11234
11235
11236
11237
11238
11239
11240
11241
11242
11243
11244
11245
11246
11247
11248
11249
11250
11251
11252
11253
11254
11255
11256
11257
11258
11259
11260
11261
11262
11263
11264
11265
11266
11267
11268
11269
11270
11271
11272
11273
11274
11275
11276
11277
11278
11279
11280
11281
11282
11283
11284
11285
11286
11287
11288
11289
11290
11291
11292
11293
11294
11295
11296
11297
11298
11299
11300
11301
11302
11303
11304
11305
11306
11307
11308
11309
11310
11311
11312
11313
11314
11315
11316
11317
11318
11319
11320
11321
11322
11323
11324
11325
11326
11327
11328
11329
11330
11331
11332
11333
11334
11335
11336
11337
11338
11339
11340
11341
11342
11343
11344
11345
11346
11347
11348
11349
11350
11351
11352
11353
11354
11355
11356
11357
11358
11359
11360
11361
11362
11363
11364
11365
11366
11367
11368
11369
11370
11371
11372
11373
11374
11375
11376
11377
11378
11379
11380
11381
11382
11383
11384
11385
11386
11387
11388
11389
11390
11391
11392
11393
11394
11395
11396
11397
11398
11399
11400
11401
11402
11403
11404
11405
11406
11407
11408
11409
11410
11411
11412
11413
11414
11415
11416
11417
11418
11419
11420
11421
11422
11423
11424
11425
11426
11427
11428
11429
11430
11431
11432
11433
11434
11435
11436
11437
11438
11439
11440
11441
11442
11443
11444
11445
11446
11447
11448
11449
11450
11451
11452
11453
11454
11455
11456
11457
11458
11459
11460
11461
11462
11463
11464
11465
11466
11467
11468
11469
11470
11471
11472
11473
11474
11475
11476
11477
11478
11479
11480
11481
11482
11483
11484
11485
11486
11487
11488
11489
11490
11491
11492
11493
11494
11495
11496
11497
11498
11499
11500
11501
11502
11503
11504
11505
11506
11507
11508
11509
11510
11511
11512
11513
11514
11515
11516
11517
11518
11519
11520
11521
11522
11523
11524
11525
11526
11527
11528
11529
11530
11531
11532
11533
11534
11535
11536
11537
11538
11539
11540
11541
11542
11543
11544
11545
11546
11547
11548
11549
11550
11551
11552
11553
11554
11555
11556
11557
11558
11559
11560
11561
11562
11563
11564
11565
11566
11567
11568
11569
11570
11571
11572
11573
11574
11575
11576
11577
11578
11579
11580
11581
11582
11583
11584
11585
11586
11587
11588
11589
11590
11591
11592
11593
11594
11595
11596
11597
11598
11599
11600
11601
11602
11603
11604
11605
11606
11607
11608
11609
11610
11611
11612
11613
11614
11615
11616
11617
11618
11619
11620
11621
11622
11623
11624
11625
11626
11627
11628
11629
11630
11631
11632
11633
11634
11635
11636
11637
11638
11639
11640
11641
11642
11643
11644
11645
11646
11647
11648
11649
11650
11651
11652
11653
11654
11655
11656
11657
11658
11659
11660
11661
11662
11663
11664
11665
11666
11667
11668
11669
11670
11671
11672
11673
11674
11675
11676
11677
11678
11679
11680
11681
11682
11683
11684
11685
11686
11687
11688
11689
11690
11691
11692
11693
11694
11695
11696
11697
11698
11699
11700
11701
11702
11703
11704
11705
11706
11707
11708
11709
11710
11711
11712
11713
11714
11715
11716
11717
11718
11719
11720
11721
11722
11723
11724
11725
11726
11727
11728
11729
11730
11731
11732
11733
11734
11735
11736
11737
11738
11739
11740
11741
11742
11743
11744
11745
11746
11747
11748
11749
11750
11751
11752
11753
11754
11755
11756
11757
11758
11759
11760
11761
11762
11763
11764
11765
11766
11767
11768
11769
11770
11771
11772
11773
11774
11775
11776
11777
11778
11779
11780
11781
11782
11783
11784
11785
11786
11787
11788
11789
11790
11791
11792
11793
11794
11795
11796
11797
11798
11799
11800
11801
11802
11803
11804
11805
11806
11807
11808
11809
11810
11811
11812
11813
11814
11815
11816
11817
11818
11819
11820
11821
11822
11823
11824
11825
11826
11827
11828
11829
11830
11831
11832
11833
11834
11835
11836
11837
11838
11839
11840
11841
11842
11843
11844
11845
11846
11847
11848
11849
11850
11851
11852
11853
11854
11855
11856
11857
11858
11859
11860
11861
11862
11863
11864
11865
11866
11867
11868
11869
11870
11871
11872
11873
11874
11875
11876
11877
11878
11879
11880
11881
11882
11883
11884
11885
11886
11887
11888
11889
11890
11891
11892
11893
11894
11895
11896
11897
11898
11899
11900
11901
11902
11903
11904
11905
11906
11907
11908
11909
11910
11911
11912
11913
11914
11915
11916
11917
11918
11919
11920
11921
11922
11923
11924
11925
11926
11927
11928
11929
11930
11931
11932
11933
11934
11935
11936
11937
11938
11939
11940
11941
11942
11943
11944
11945
11946
11947
11948
11949
11950
11951
11952
11953
11954
11955
11956
11957
11958
11959
11960
11961
11962
11963
11964
11965
11966
11967
11968
11969
11970
11971
11972
11973
11974
11975
11976
11977
11978
11979
11980
11981
11982
11983
11984
11985
11986
11987
11988
11989
11990
11991
11992
11993
11994
11995
11996
11997
11998
11999
12000
12001
12002
12003
12004
12005
12006
12007
12008
12009
12010
12011
12012
12013
12014
12015
12016
12017
12018
12019
12020
12021
12022
12023
12024
12025
12026
12027
12028
12029
12030
12031
12032
12033
12034
12035
12036
12037
12038
12039
12040
12041
12042
12043
12044
12045
12046
12047
12048
12049
12050
12051
12052
12053
12054
12055
12056
12057
12058
12059
12060
12061
12062
12063
12064
12065
12066
12067
12068
12069
12070
12071
12072
12073
12074
12075
12076
12077
12078
12079
12080
12081
12082
12083
12084
12085
12086
12087
12088
12089
12090
12091
12092
12093
12094
12095
12096
12097
12098
12099
12100
12101
12102
12103
12104
12105
12106
12107
12108
12109
12110
12111
12112
12113
12114
12115
12116
12117
12118
12119
12120
12121
12122
12123
12124
12125
12126
12127
12128
12129
12130
12131
12132
12133
12134
12135
12136
12137
12138
12139
12140
12141
12142
12143
12144
12145
12146
12147
12148
12149
12150
12151
12152
12153
12154
12155
12156
12157
12158
12159
12160
12161
12162
12163
12164
12165
12166
12167
12168
12169
12170
12171
12172
12173
12174
12175
12176
12177
12178
12179
12180
12181
12182
12183
12184
12185
12186
12187
12188
12189
12190
12191
12192
12193
12194
12195
12196
12197
12198
12199
12200
12201
12202
12203
12204
12205
12206
12207
12208
12209
12210
12211
12212
12213
12214
12215
12216
12217
12218
12219
12220
12221
12222
12223
12224
12225
12226
12227
12228
12229
12230
12231
12232
12233
12234
12235
12236
12237
12238
12239
12240
12241
12242
12243
12244
12245
12246
12247
12248
12249
12250
12251
12252
12253
12254
12255
12256
12257
12258
12259
12260
12261
12262
12263
12264
12265
12266
12267
12268
12269
12270
12271
12272
12273
12274
12275
12276
12277
12278
12279
12280
12281
12282
12283
12284
12285
12286
12287
12288
12289
12290
12291
12292
12293
12294
12295
12296
12297
12298
12299
12300
12301
12302
12303
12304
12305
12306
12307
12308
12309
12310
12311
12312
12313
12314
12315
12316
12317
12318
12319
12320
12321
12322
12323
12324
12325
12326
12327
12328
12329
12330
12331
12332
12333
12334
12335
12336
12337
12338
12339
12340
12341
12342
12343
12344
12345
12346
12347
12348
12349
12350
12351
12352
12353
12354
12355
12356
12357
12358
12359
12360
12361
12362
12363
12364
12365
12366
12367
12368
12369
12370
12371
12372
12373
12374
12375
12376
12377
12378
12379
12380
12381
12382
12383
12384
12385
12386
12387
12388
12389
12390
12391
12392
12393
12394
12395
12396
12397
12398
12399
12400
12401
12402
12403
12404
12405
12406
12407
12408
12409
12410
12411
12412
12413
12414
12415
12416
12417
12418
12419
12420
12421
12422
12423
12424
12425
12426
12427
12428
12429
12430
12431
12432
12433
12434
12435
12436
12437
12438
12439
12440
12441
12442
12443
12444
12445
12446
12447
12448
12449
12450
12451
12452
12453
12454
12455
12456
12457
12458
12459
12460
12461
12462
12463
12464
12465
12466
12467
12468
12469
12470
12471
12472
12473
12474
12475
12476
12477
12478
12479
12480
12481
12482
12483
12484
12485
12486
12487
12488
12489
12490
12491
12492
12493
12494
12495
12496
12497
12498
12499
12500
12501
12502
12503
12504
12505
12506
12507
12508
12509
12510
12511
12512
12513
12514
12515
12516
12517
12518
12519
12520
12521
12522
12523
12524
12525
12526
12527
12528
12529
12530
12531
12532
12533
12534
12535
12536
12537
12538
12539
12540
12541
12542
12543
12544
12545
12546
12547
12548
12549
12550
12551
12552
12553
12554
12555
12556
12557
12558
12559
12560
12561
12562
12563
12564
12565
12566
12567
12568
12569
12570
12571
12572
12573
12574
12575
12576
12577
12578
12579
12580
12581
12582
12583
12584
12585
12586
12587
12588
12589
12590
12591
12592
12593
12594
12595
12596
12597
12598
12599
12600
12601
12602
12603
12604
12605
12606
12607
12608
12609
12610
12611
12612
12613
12614
12615
12616
12617
12618
12619
12620
12621
12622
12623
12624
12625
12626
12627
12628
12629
12630
12631
12632
12633
12634
12635
12636
12637
12638
12639
12640
12641
12642
12643
12644
12645
12646
12647
12648
12649
12650
12651
12652
12653
12654
12655
12656
12657
12658
12659
12660
12661
12662
12663
12664
12665
12666
12667
12668
12669
12670
12671
12672
12673
12674
12675
12676
12677
12678
12679
12680
12681
12682
12683
12684
12685
12686
12687
12688
12689
12690
12691
12692
12693
12694
12695
12696
12697
12698
12699
12700
12701
12702
12703
12704
12705
12706
12707
12708
12709
12710
12711
12712
12713
12714
12715
12716
12717
12718
12719
12720
12721
12722
12723
12724
12725
12726
12727
12728
12729
12730
12731
12732
12733
12734
12735
12736
12737
12738
12739
12740
12741
12742
12743
12744
12745
12746
12747
12748
12749
12750
12751
12752
12753
12754
12755
12756
12757
12758
12759
12760
12761
12762
12763
12764
12765
12766
12767
12768
12769
12770
12771
12772
12773
12774
12775
12776
12777
12778
12779
12780
12781
12782
12783
12784
12785
12786
12787
12788
12789
12790
12791
12792
12793
12794
12795
12796
12797
12798
12799
12800
12801
12802
12803
12804
12805
12806
12807
12808
12809
12810
12811
12812
12813
12814
12815
12816
12817
12818
12819
12820
12821
12822
12823
12824
12825
12826
12827
12828
12829
12830
12831
12832
12833
12834
12835
12836
12837
12838
12839
12840
12841
12842
12843
12844
12845
12846
12847
12848
12849
12850
12851
12852
12853
12854
12855
12856
12857
12858
12859
12860
12861
12862
12863
12864
12865
12866
12867
12868
12869
12870
12871
12872
12873
12874
12875
12876
12877
12878
12879
12880
12881
12882
12883
12884
12885
12886
12887
12888
12889
12890
12891
12892
12893
12894
12895
12896
12897
12898
12899
12900
12901
12902
12903
12904
12905
12906
12907
12908
12909
12910
12911
12912
12913
12914
12915
12916
12917
12918
12919
12920
12921
12922
12923
12924
12925
12926
12927
12928
12929
12930
12931
12932
12933
12934
12935
12936
12937
12938
12939
12940
12941
12942
12943
12944
12945
12946
12947
12948
12949
12950
12951
12952
12953
12954
12955
12956
12957
12958
12959
12960
12961
12962
12963
12964
12965
12966
12967
12968
12969
12970
12971
12972
12973
12974
12975
12976
12977
12978
12979
12980
12981
12982
12983
12984
12985
12986
12987
12988
12989
12990
12991
12992
12993
12994
12995
12996
12997
12998
12999
13000
13001
13002
13003
13004
13005
13006
13007
13008
13009
13010
13011
13012
13013
13014
13015
13016
13017
13018
13019
13020
13021
13022
13023
13024
13025
13026
13027
13028
13029
13030
13031
13032
13033
13034
13035
13036
13037
13038
13039
13040
13041
13042
13043
13044
13045
13046
13047
13048
13049
13050
13051
13052
13053
13054
13055
13056
13057
13058
13059
13060
13061
13062
13063
13064
13065
13066
13067
13068
13069
13070
13071
13072
13073
13074
13075
13076
13077
13078
13079
13080
13081
13082
13083
13084
13085
13086
13087
13088
13089
13090
13091
13092
13093
13094
13095
13096
13097
13098
13099
13100
13101
13102
13103
13104
13105
13106
13107
13108
13109
13110
13111
13112
13113
13114
13115
13116
13117
13118
13119
13120
13121
13122
13123
13124
13125
13126
13127
13128
13129
13130
13131
13132
13133
13134
13135
13136
13137
13138
13139
13140
13141
13142
13143
13144
13145
13146
13147
13148
13149
13150
13151
13152
13153
13154
13155
13156
13157
13158
13159
13160
13161
13162
13163
13164
13165
13166
13167
13168
13169
13170
13171
13172
13173
13174
13175
13176
13177
13178
13179
13180
13181
13182
13183
13184
13185
13186
13187
13188
13189
13190
13191
13192
13193
13194
13195
13196
13197
13198
13199
13200
13201
13202
13203
13204
13205
13206
13207
13208
13209
13210
13211
13212
13213
13214
13215
13216
13217
13218
13219
13220
13221
13222
13223
13224
13225
13226
13227
13228
13229
13230
13231
13232
13233
13234
13235
13236
13237
13238
13239
13240
13241
13242
13243
13244
13245
13246
13247
13248
13249
13250
13251
13252
13253
13254
13255
13256
13257
13258
13259
13260
13261
13262
13263
13264
13265
13266
13267
13268
13269
13270
13271
13272
13273
13274
13275
13276
13277
13278
13279
13280
13281
13282
13283
13284
13285
13286
13287
13288
13289
13290
13291
13292
13293
13294
13295
13296
13297
13298
13299
13300
13301
13302
13303
13304
13305
13306
13307
13308
13309
13310
13311
13312
13313
13314
13315
13316
13317
13318
13319
13320
13321
13322
13323
13324
13325
13326
13327
13328
13329
13330
13331
13332
13333
13334
13335
13336
13337
13338
13339
13340
13341
13342
13343
13344
13345
13346
13347
13348
13349
13350
13351
13352
13353
13354
13355
13356
13357
13358
13359
13360
13361
13362
13363
13364
13365
13366
13367
13368
13369
13370
13371
13372
13373
13374
13375
13376
13377
13378
13379
13380
13381
13382
13383
13384
13385
13386
13387
13388
13389
13390
13391
13392
13393
13394
13395
13396
13397
13398
13399
13400
13401
13402
13403
13404
13405
13406
13407
13408
13409
13410
13411
13412
13413
13414
13415
13416
13417
13418
13419
13420
13421
13422
13423
13424
13425
13426
13427
13428
13429
13430
13431
13432
13433
13434
13435
13436
13437
13438
13439
13440
13441
13442
13443
13444
13445
13446
13447
13448
13449
13450
13451
13452
13453
13454
13455
13456
13457
13458
13459
13460
13461
13462
13463
13464
13465
13466
13467
13468
13469
13470
13471
13472
13473
13474
13475
13476
13477
13478
13479
13480
13481
13482
13483
13484
13485
13486
13487
13488
13489
13490
13491
13492
13493
13494
13495
13496
13497
13498
13499
13500
13501
13502
13503
13504
13505
13506
13507
13508
13509
13510
13511
13512
13513
13514
13515
13516
13517
13518
13519
13520
13521
13522
13523
13524
13525
13526
13527
13528
13529
13530
13531
13532
13533
13534
13535
13536
13537
13538
13539
13540
13541
13542
13543
13544
13545
13546
13547
13548
13549
13550
13551
13552
13553
13554
13555
13556
13557
13558
13559
13560
13561
13562
13563
13564
13565
13566
13567
13568
13569
13570
13571
13572
13573
13574
13575
13576
13577
13578
13579
13580
13581
13582
13583
13584
13585
13586
13587
13588
13589
13590
13591
13592
13593
13594
13595
13596
13597
13598
13599
13600
13601
13602
13603
13604
13605
13606
13607
13608
13609
13610
13611
13612
13613
13614
13615
13616
13617
13618
13619
13620
13621
13622
13623
13624
13625
13626
13627
13628
13629
13630
13631
13632
13633
13634
13635
13636
13637
13638
13639
13640
13641
13642
13643
13644
13645
13646
13647
13648
13649
13650
13651
13652
13653
13654
13655
13656
13657
13658
13659
13660
13661
13662
13663
13664
13665
13666
13667
13668
13669
13670
13671
13672
13673
13674
13675
13676
13677
13678
13679
13680
13681
13682
13683
13684
13685
13686
13687
13688
13689
13690
13691
13692
13693
13694
13695
13696
13697
13698
13699
13700
13701
13702
13703
13704
13705
13706
13707
13708
13709
13710
13711
13712
13713
13714
13715
13716
13717
13718
13719
13720
13721
13722
13723
13724
13725
13726
13727
13728
13729
13730
13731
13732
13733
13734
13735
13736
13737
13738
13739
13740
13741
13742
13743
13744
13745
13746
13747
13748
13749
13750
13751
13752
13753
13754
13755
13756
13757
13758
13759
13760
13761
13762
13763
13764
13765
13766
13767
13768
13769
13770
13771
13772
13773
13774
13775
13776
13777
13778
13779
13780
13781
13782
13783
13784
13785
13786
13787
13788
13789
13790
13791
13792
13793
13794
13795
13796
13797
13798
13799
13800
13801
13802
13803
13804
13805
13806
13807
13808
13809
13810
13811
13812
13813
13814
13815
13816
13817
13818
13819
13820
13821
13822
13823
13824
13825
13826
13827
13828
13829
13830
13831
13832
13833
13834
13835
13836
13837
13838
13839
13840
13841
13842
13843
13844
13845
13846
13847
13848
13849
13850
13851
13852
13853
13854
13855
13856
13857
13858
13859
13860
13861
13862
13863
13864
13865
13866
13867
13868
13869
13870
13871
13872
13873
13874
13875
13876
13877
13878
13879
13880
13881
13882
13883
13884
13885
13886
13887
13888
13889
13890
13891
13892
13893
13894
13895
13896
13897
13898
13899
13900
13901
13902
13903
13904
13905
13906
13907
13908
13909
13910
13911
13912
13913
13914
13915
13916
13917
13918
13919
13920
13921
13922
13923
13924
13925
13926
13927
13928
13929
13930
13931
13932
13933
13934
13935
13936
13937
13938
13939
13940
13941
13942
13943
13944
13945
13946
13947
13948
13949
13950
13951
13952
13953
13954
13955
13956
13957
13958
13959
13960
13961
13962
13963
13964
13965
13966
13967
13968
13969
13970
13971
13972
13973
13974
13975
13976
13977
13978
13979
13980
13981
13982
13983
13984
13985
13986
13987
13988
13989
13990
13991
13992
13993
13994
13995
13996
13997
13998
13999
14000
14001
14002
14003
14004
14005
14006
14007
14008
14009
14010
14011
14012
14013
14014
14015
14016
14017
14018
14019
14020
14021
14022
14023
14024
14025
14026
14027
14028
14029
14030
14031
14032
14033
14034
14035
14036
14037
14038
14039
14040
14041
14042
14043
14044
14045
14046
14047
14048
14049
14050
14051
14052
14053
14054
14055
14056
14057
14058
14059
14060
14061
14062
14063
14064
14065
14066
14067
14068
14069
14070
14071
14072
14073
14074
14075
14076
14077
14078
14079
14080
14081
14082
14083
14084
14085
14086
14087
14088
14089
14090
14091
14092
14093
14094
14095
14096
14097
14098
14099
14100
14101
14102
14103
14104
14105
14106
14107
14108
14109
14110
14111
14112
14113
14114
14115
14116
14117
14118
14119
14120
14121
14122
14123
14124
14125
14126
14127
14128
14129
14130
14131
14132
14133
14134
14135
14136
14137
14138
14139
14140
14141
14142
14143
14144
14145
14146
14147
14148
14149
14150
14151
14152
14153
14154
14155
14156
14157
14158
14159
14160
14161
14162
14163
14164
14165
14166
14167
14168
14169
14170
14171
14172
14173
14174
14175
14176
14177
14178
14179
14180
14181
14182
14183
14184
14185
14186
14187
14188
14189
14190
14191
14192
14193
14194
14195
14196
14197
14198
14199
14200
14201
14202
14203
14204
14205
14206
14207
14208
14209
14210
14211
14212
14213
14214
14215
14216
14217
14218
14219
14220
14221
14222
14223
14224
14225
14226
14227
14228
14229
14230
14231
14232
14233
14234
14235
14236
14237
14238
14239
14240
14241
14242
14243
14244
14245
14246
14247
14248
14249
14250
14251
14252
14253
14254
14255
14256
14257
14258
14259
14260
14261
14262
14263
14264
14265
14266
14267
14268
14269
14270
14271
14272
14273
14274
14275
14276
14277
14278
14279
14280
14281
14282
14283
14284
14285
14286
14287
14288
14289
14290
14291
14292
14293
14294
14295
14296
14297
14298
14299
14300
14301
14302
14303
14304
14305
14306
14307
14308
14309
14310
14311
14312
14313
14314
14315
14316
14317
14318
14319
14320
14321
14322
14323
14324
14325
14326
14327
14328
14329
14330
14331
14332
14333
14334
14335
14336
14337
14338
14339
14340
14341
14342
14343
14344
14345
14346
14347
14348
14349
14350
14351
14352
14353
14354
14355
14356
14357
14358
14359
14360
14361
14362
14363
14364
14365
14366
14367
14368
14369
14370
14371
14372
14373
14374
14375
14376
14377
14378
14379
14380
14381
14382
14383
14384
14385
14386
14387
14388
14389
14390
14391
14392
14393
14394
14395
14396
14397
14398
14399
14400
14401
14402
14403
14404
14405
14406
14407
14408
14409
14410
14411
14412
14413
14414
14415
14416
14417
14418
14419
14420
14421
14422
14423
14424
14425
14426
14427
14428
14429
14430
14431
14432
14433
14434
14435
14436
14437
14438
14439
14440
14441
14442
14443
14444
14445
14446
14447
14448
14449
14450
14451
14452
14453
14454
14455
14456
14457
14458
14459
14460
14461
14462
14463
14464
14465
14466
14467
14468
14469
14470
14471
14472
14473
14474
14475
14476
14477
14478
14479
14480
14481
14482
14483
14484
14485
14486
14487
14488
14489
14490
14491
14492
14493
14494
14495
14496
14497
14498
14499
14500
14501
14502
14503
14504
14505
14506
14507
14508
14509
14510
14511
14512
14513
14514
14515
14516
14517
14518
14519
14520
14521
14522
14523
14524
14525
14526
14527
14528
14529
14530
14531
14532
14533
14534
14535
14536
14537
14538
14539
14540
14541
14542
14543
14544
14545
14546
14547
14548
14549
14550
14551
14552
14553
14554
14555
14556
14557
14558
14559
14560
14561
14562
14563
14564
14565
14566
14567
14568
14569
14570
14571
14572
14573
14574
14575
14576
14577
14578
14579
14580
14581
14582
14583
14584
14585
14586
14587
14588
14589
14590
14591
14592
14593
14594
14595
14596
14597
14598
14599
14600
14601
14602
14603
14604
14605
14606
14607
14608
14609
14610
14611
14612
14613
14614
14615
14616
14617
14618
14619
14620
14621
14622
14623
14624
14625
14626
14627
14628
14629
14630
14631
14632
14633
14634
14635
14636
14637
14638
14639
14640
14641
14642
14643
14644
14645
14646
14647
14648
14649
14650
14651
14652
14653
14654
14655
14656
14657
14658
14659
14660
14661
14662
14663
14664
14665
14666
14667
14668
14669
14670
14671
14672
14673
14674
14675
14676
14677
14678
14679
14680
14681
14682
14683
14684
14685
14686
14687
14688
14689
14690
14691
14692
14693
14694
14695
14696
14697
14698
14699
14700
14701
14702
14703
14704
14705
14706
14707
14708
14709
14710
14711
14712
14713
14714
14715
14716
14717
14718
14719
14720
14721
14722
14723
14724
14725
14726
14727
14728
14729
14730
14731
14732
14733
14734
14735
14736
14737
14738
14739
14740
14741
14742
14743
14744
14745
14746
14747
14748
14749
14750
14751
14752
14753
14754
14755
14756
14757
14758
14759
14760
14761
14762
14763
14764
14765
14766
14767
14768
14769
14770
14771
14772
14773
14774
14775
14776
14777
14778
14779
14780
14781
14782
14783
14784
14785
14786
14787
14788
14789
14790
14791
14792
14793
14794
14795
14796
14797
14798
14799
14800
14801
14802
14803
14804
14805
14806
14807
14808
14809
14810
14811
14812
14813
14814
14815
14816
14817
14818
14819
14820
14821
14822
14823
14824
14825
14826
14827
14828
14829
14830
14831
14832
14833
14834
14835
14836
14837
14838
14839
14840
14841
14842
14843
14844
14845
14846
14847
14848
14849
14850
14851
14852
14853
14854
14855
14856
14857
14858
14859
14860
14861
14862
14863
14864
14865
14866
14867
14868
14869
14870
14871
14872
14873
14874
14875
14876
14877
14878
14879
14880
14881
14882
14883
14884
14885
14886
14887
14888
14889
14890
14891
14892
14893
14894
14895
14896
14897
14898
14899
14900
14901
14902
14903
14904
14905
14906
14907
14908
14909
14910
14911
14912
14913
14914
14915
14916
14917
14918
14919
14920
14921
14922
14923
14924
14925
14926
14927
14928
14929
14930
14931
14932
14933
14934
14935
14936
14937
14938
14939
14940
14941
14942
14943
14944
14945
14946
14947
14948
14949
14950
14951
14952
14953
14954
14955
14956
14957
14958
14959
14960
14961
14962
14963
14964
14965
14966
14967
14968
14969
14970
14971
14972
14973
14974
14975
14976
14977
14978
14979
14980
14981
14982
14983
14984
14985
14986
14987
14988
14989
14990
14991
14992
14993
14994
14995
14996
14997
14998
14999
15000
15001
15002
15003
15004
15005
15006
15007
15008
15009
15010
15011
15012
15013
15014
15015
15016
15017
15018
15019
15020
15021
15022
15023
15024
15025
15026
15027
15028
15029
15030
15031
15032
15033
15034
15035
15036
15037
15038
15039
15040
15041
15042
15043
15044
15045
15046
15047
15048
15049
15050
15051
15052
15053
15054
15055
15056
15057
15058
15059
15060
15061
15062
15063
15064
15065
15066
15067
15068
15069
15070
15071
15072
15073
15074
15075
15076
15077
15078
15079
15080
15081
15082
15083
15084
15085
15086
15087
15088
15089
15090
15091
15092
15093
15094
15095
15096
15097
15098
15099
15100
15101
15102
15103
15104
15105
15106
15107
15108
15109
15110
15111
15112
15113
15114
15115
15116
15117
15118
15119
15120
15121
15122
15123
15124
15125
15126
15127
15128
15129
15130
15131
15132
15133
15134
15135
15136
15137
15138
15139
15140
15141
15142
15143
15144
15145
15146
15147
15148
15149
15150
15151
15152
15153
15154
15155
15156
15157
15158
15159
15160
15161
15162
15163
15164
15165
15166
15167
15168
15169
15170
15171
15172
15173
15174
15175
15176
15177
15178
15179
15180
15181
15182
15183
15184
15185
15186
15187
15188
15189
15190
15191
15192
15193
15194
15195
15196
15197
15198
15199
15200
15201
15202
15203
15204
15205
15206
15207
15208
15209
15210
15211
15212
15213
15214
15215
15216
15217
15218
15219
15220
15221
15222
15223
15224
15225
15226
15227
15228
15229
15230
15231
15232
15233
15234
15235
15236
15237
15238
15239
15240
15241
15242
15243
15244
15245
15246
15247
15248
15249
15250
15251
15252
15253
15254
15255
15256
15257
15258
15259
15260
15261
15262
15263
15264
15265
15266
15267
15268
15269
15270
15271
15272
15273
15274
15275
15276
15277
15278
15279
15280
15281
15282
15283
15284
15285
15286
15287
15288
15289
15290
15291
15292
15293
15294
15295
15296
15297
15298
15299
15300
15301
15302
15303
15304
15305
15306
15307
15308
15309
15310
15311
15312
15313
15314
15315
15316
15317
15318
15319
15320
15321
15322
15323
15324
15325
15326
15327
15328
15329
15330
15331
15332
15333
15334
15335
15336
15337
15338
15339
15340
15341
15342
15343
15344
15345
15346
15347
15348
15349
15350
15351
15352
15353
15354
15355
15356
15357
15358
15359
15360
15361
15362
15363
15364
15365
15366
15367
15368
15369
15370
15371
15372
15373
15374
15375
15376
15377
15378
15379
15380
15381
15382
15383
15384
15385
15386
15387
15388
15389
15390
15391
15392
15393
15394
15395
15396
15397
15398
15399
15400
15401
15402
15403
15404
15405
15406
15407
15408
15409
15410
15411
15412
15413
15414
15415
15416
15417
15418
15419
15420
15421
15422
15423
15424
15425
15426
15427
15428
15429
15430
15431
15432
15433
15434
15435
15436
15437
15438
15439
15440
15441
15442
15443
15444
15445
15446
15447
15448
15449
15450
15451
15452
15453
15454
15455
15456
15457
15458
15459
15460
15461
15462
15463
15464
15465
15466
15467
15468
15469
15470
15471
15472
15473
15474
15475
15476
15477
15478
15479
15480
15481
15482
15483
15484
15485
15486
15487
15488
15489
15490
15491
15492
15493
15494
15495
15496
15497
15498
15499
15500
15501
15502
15503
15504
15505
15506
15507
15508
15509
15510
15511
15512
15513
15514
15515
15516
15517
15518
15519
15520
15521
15522
15523
15524
15525
15526
15527
15528
15529
15530
15531
15532
15533
15534
15535
15536
15537
15538
15539
15540
15541
15542
15543
15544
15545
15546
15547
15548
15549
15550
15551
15552
15553
15554
15555
15556
15557
15558
15559
15560
15561
15562
15563
15564
15565
15566
15567
15568
15569
15570
15571
15572
15573
15574
15575
15576
15577
15578
15579
15580
15581
15582
15583
15584
15585
15586
15587
15588
15589
15590
15591
15592
15593
15594
15595
15596
15597
15598
15599
15600
15601
15602
15603
15604
15605
15606
15607
15608
15609
15610
15611
15612
15613
15614
15615
15616
15617
15618
15619
15620
15621
15622
15623
15624
15625
15626
15627
15628
15629
15630
15631
15632
15633
15634
15635
15636
15637
15638
15639
15640
15641
15642
15643
15644
15645
15646
15647
15648
15649
15650
15651
15652
15653
15654
15655
15656
15657
15658
15659
15660
15661
15662
15663
15664
15665
15666
15667
15668
15669
15670
15671
15672
15673
15674
15675
15676
15677
15678
15679
15680
15681
15682
15683
15684
15685
15686
15687
15688
15689
15690
15691
15692
15693
15694
15695
15696
15697
15698
15699
15700
15701
15702
15703
15704
15705
15706
15707
15708
15709
15710
15711
15712
15713
15714
15715
15716
15717
15718
15719
15720
15721
15722
15723
15724
15725
15726
15727
15728
15729
15730
15731
15732
15733
15734
15735
15736
15737
15738
15739
15740
15741
15742
15743
15744
15745
15746
15747
15748
15749
15750
15751
15752
15753
15754
15755
15756
15757
15758
15759
15760
15761
15762
15763
15764
15765
15766
15767
15768
15769
15770
15771
15772
15773
15774
15775
15776
15777
15778
15779
15780
15781
15782
15783
15784
15785
15786
15787
15788
15789
15790
15791
15792
15793
15794
15795
15796
15797
15798
15799
15800
15801
15802
15803
15804
15805
15806
15807
15808
15809
15810
15811
15812
15813
15814
15815
15816
15817
15818
15819
15820
15821
15822
15823
15824
15825
15826
15827
15828
15829
15830
15831
15832
15833
15834
15835
15836
15837
15838
15839
15840
15841
15842
15843
15844
15845
15846
15847
15848
15849
15850
15851
15852
15853
15854
15855
15856
15857
15858
15859
15860
15861
15862
15863
15864
15865
15866
15867
15868
15869
15870
15871
15872
15873
15874
15875
15876
15877
15878
15879
15880
15881
15882
15883
15884
15885
15886
15887
15888
15889
15890
15891
15892
15893
15894
15895
15896
15897
15898
15899
15900
15901
15902
15903
15904
15905
15906
15907
15908
15909
15910
15911
15912
15913
15914
15915
15916
15917
15918
15919
15920
15921
15922
15923
15924
15925
15926
15927
15928
15929
15930
15931
15932
15933
15934
15935
15936
15937
15938
15939
15940
15941
15942
15943
15944
15945
15946
15947
15948
15949
15950
15951
15952
15953
15954
15955
15956
15957
15958
15959
15960
15961
15962
15963
15964
15965
15966
15967
15968
15969
15970
15971
15972
15973
15974
15975
15976
15977
15978
15979
15980
15981
15982
15983
15984
15985
15986
15987
15988
15989
15990
15991
15992
15993
15994
15995
15996
15997
15998
15999
16000
16001
16002
16003
16004
16005
16006
16007
16008
16009
16010
16011
16012
16013
16014
16015
16016
16017
16018
16019
16020
16021
16022
16023
16024
16025
16026
16027
16028
16029
16030
16031
16032
16033
16034
16035
16036
16037
16038
16039
16040
16041
16042
16043
16044
16045
16046
16047
16048
16049
16050
16051
16052
16053
16054
16055
16056
16057
16058
16059
16060
16061
16062
16063
16064
16065
16066
16067
16068
16069
16070
16071
16072
16073
16074
16075
16076
16077
16078
16079
16080
16081
16082
16083
16084
16085
16086
16087
16088
16089
16090
16091
16092
16093
16094
16095
16096
16097
16098
16099
16100
16101
16102
16103
16104
16105
16106
16107
16108
16109
16110
16111
16112
16113
16114
16115
16116
16117
16118
16119
16120
16121
16122
16123
16124
16125
16126
16127
16128
16129
16130
16131
16132
16133
16134
16135
16136
16137
16138
16139
16140
16141
16142
16143
16144
16145
16146
16147
16148
16149
16150
16151
16152
16153
16154
16155
16156
16157
16158
16159
16160
16161
16162
16163
16164
16165
16166
16167
16168
16169
16170
16171
16172
16173
16174
16175
16176
16177
16178
16179
16180
16181
16182
16183
16184
16185
16186
16187
16188
16189
16190
16191
16192
16193
16194
16195
16196
16197
16198
16199
16200
16201
16202
16203
16204
16205
16206
16207
16208
16209
16210
16211
16212
16213
16214
16215
16216
16217
16218
16219
16220
16221
16222
16223
16224
16225
16226
16227
16228
16229
16230
16231
16232
16233
16234
16235
16236
16237
16238
16239
16240
16241
16242
16243
16244
16245
16246
16247
16248
16249
16250
16251
16252
16253
16254
16255
16256
16257
16258
16259
16260
16261
16262
16263
16264
16265
16266
16267
16268
16269
16270
16271
16272
16273
16274
16275
16276
16277
16278
16279
16280
16281
16282
16283
16284
16285
16286
16287
16288
16289
16290
16291
16292
16293
16294
16295
16296
16297
16298
16299
16300
16301
16302
16303
16304
16305
16306
16307
16308
16309
16310
16311
16312
16313
16314
16315
16316
16317
16318
16319
16320
16321
16322
16323
16324
16325
16326
16327
16328
16329
16330
16331
16332
16333
16334
16335
16336
16337
16338
16339
16340
16341
16342
16343
16344
16345
16346
16347
16348
16349
16350
16351
16352
16353
16354
16355
16356
16357
16358
16359
16360
16361
16362
16363
16364
16365
16366
16367
16368
16369
16370
16371
16372
16373
16374
16375
16376
16377
16378
16379
16380
16381
16382
16383
16384
16385
16386
16387
16388
16389
16390
16391
16392
16393
16394
16395
16396
16397
16398
16399
16400
16401
16402
16403
16404
16405
16406
16407
16408
16409
16410
16411
16412
16413
16414
16415
16416
16417
16418
16419
16420
16421
16422
16423
16424
16425
16426
16427
16428
16429
16430
16431
16432
16433
16434
16435
16436
16437
16438
16439
16440
16441
16442
16443
16444
16445
16446
16447
16448
16449
16450
16451
16452
16453
16454
16455
16456
16457
16458
16459
16460
16461
16462
16463
16464
16465
16466
16467
16468
16469
16470
16471
16472
16473
16474
16475
16476
16477
16478
16479
16480
16481
16482
16483
16484
16485
16486
16487
16488
16489
16490
16491
16492
16493
16494
16495
16496
16497
16498
16499
16500
16501
16502
16503
16504
16505
16506
16507
16508
16509
16510
16511
16512
16513
16514
16515
16516
16517
16518
16519
16520
16521
16522
16523
16524
16525
16526
16527
16528
16529
16530
16531
16532
16533
16534
16535
16536
16537
16538
16539
16540
16541
16542
16543
16544
16545
16546
16547
16548
16549
16550
16551
16552
16553
16554
16555
16556
16557
16558
16559
16560
16561
16562
16563
16564
16565
16566
16567
16568
16569
16570
16571
16572
16573
16574
16575
16576
16577
16578
16579
16580
16581
16582
16583
16584
16585
16586
16587
16588
16589
16590
16591
16592
16593
16594
16595
16596
16597
16598
16599
16600
16601
16602
16603
16604
16605
16606
16607
16608
16609
16610
16611
16612
16613
16614
16615
16616
16617
16618
16619
16620
16621
16622
16623
16624
16625
16626
16627
16628
16629
16630
16631
16632
16633
16634
16635
16636
16637
16638
16639
16640
16641
16642
16643
16644
16645
16646
16647
16648
16649
16650
16651
16652
16653
16654
16655
16656
16657
16658
16659
16660
16661
16662
16663
16664
16665
16666
16667
16668
16669
16670
16671
16672
16673
16674
16675
16676
16677
16678
16679
16680
16681
16682
16683
16684
16685
16686
16687
16688
16689
16690
16691
16692
16693
16694
16695
16696
16697
16698
16699
16700
16701
16702
16703
16704
16705
16706
16707
16708
16709
16710
16711
16712
16713
16714
16715
16716
16717
16718
16719
16720
16721
16722
16723
16724
16725
16726
16727
16728
16729
16730
16731
16732
16733
16734
16735
16736
16737
16738
16739
16740
16741
16742
16743
16744
16745
16746
16747
16748
16749
16750
16751
16752
16753
16754
16755
16756
16757
16758
16759
16760
16761
16762
16763
16764
16765
16766
16767
16768
16769
16770
16771
16772
16773
16774
16775
16776
16777
16778
16779
16780
16781
16782
16783
16784
16785
16786
16787
16788
16789
16790
16791
16792
16793
16794
16795
16796
16797
16798
16799
16800
16801
16802
16803
16804
16805
16806
16807
16808
16809
16810
16811
16812
16813
16814
16815
16816
16817
16818
16819
16820
16821
16822
16823
16824
16825
16826
16827
16828
16829
16830
16831
16832
16833
16834
16835
16836
16837
16838
16839
16840
16841
16842
16843
16844
16845
16846
16847
16848
16849
16850
16851
16852
16853
16854
16855
16856
16857
16858
16859
16860
16861
16862
16863
16864
16865
16866
16867
16868
16869
16870
16871
16872
16873
16874
16875
16876
16877
16878
16879
16880
16881
16882
16883
16884
16885
16886
16887
16888
16889
16890
16891
16892
16893
16894
16895
16896
16897
16898
16899
16900
16901
16902
16903
16904
16905
16906
16907
16908
16909
16910
16911
16912
16913
16914
16915
16916
16917
16918
16919
16920
16921
16922
16923
16924
16925
16926
16927
16928
16929
16930
16931
16932
16933
16934
16935
16936
16937
16938
16939
16940
16941
16942
16943
16944
16945
16946
16947
16948
16949
16950
16951
16952
16953
16954
16955
16956
16957
16958
16959
16960
16961
16962
16963
16964
16965
16966
16967
16968
16969
16970
16971
16972
16973
16974
16975
16976
16977
16978
16979
16980
16981
16982
16983
16984
16985
16986
16987
16988
16989
16990
16991
16992
16993
16994
16995
16996
16997
16998
16999
17000
17001
17002
17003
17004
17005
17006
17007
17008
17009
17010
17011
17012
17013
17014
17015
17016
17017
17018
17019
17020
17021
17022
17023
17024
17025
17026
17027
17028
17029
17030
17031
17032
17033
17034
17035
17036
17037
17038
17039
17040
17041
17042
17043
17044
17045
17046
17047
17048
17049
17050
17051
17052
17053
17054
17055
17056
17057
17058
17059
17060
17061
17062
17063
17064
17065
17066
17067
17068
17069
17070
17071
17072
17073
17074
17075
17076
17077
17078
17079
17080
17081
17082
17083
17084
17085
17086
17087
17088
17089
17090
17091
17092
17093
17094
17095
17096
17097
17098
17099
17100
17101
17102
17103
17104
17105
17106
17107
17108
17109
17110
17111
17112
17113
17114
17115
17116
17117
17118
17119
17120
17121
17122
17123
17124
17125
17126
17127
17128
17129
17130
17131
17132
17133
17134
17135
17136
17137
17138
17139
17140
17141
17142
17143
17144
17145
17146
17147
17148
17149
17150
17151
17152
17153
17154
17155
17156
17157
17158
17159
17160
17161
17162
17163
17164
17165
17166
17167
17168
17169
17170
17171
17172
17173
17174
17175
17176
17177
17178
17179
17180
17181
17182
17183
17184
17185
17186
17187
17188
17189
17190
17191
17192
17193
17194
17195
17196
17197
17198
17199
17200
17201
17202
17203
17204
17205
17206
17207
17208
17209
17210
17211
17212
17213
17214
17215
17216
17217
17218
17219
17220
17221
17222
17223
17224
17225
17226
17227
17228
17229
17230
17231
17232
17233
17234
17235
17236
17237
17238
17239
17240
17241
17242
17243
17244
17245
17246
17247
17248
17249
17250
17251
17252
17253
17254
17255
17256
17257
17258
17259
17260
17261
17262
17263
17264
17265
17266
17267
17268
17269
17270
17271
17272
17273
17274
17275
17276
17277
17278
17279
17280
17281
17282
17283
17284
17285
17286
17287
17288
17289
17290
17291
17292
17293
17294
17295
17296
17297
17298
17299
17300
17301
17302
17303
17304
17305
17306
17307
17308
17309
17310
17311
17312
17313
17314
17315
17316
17317
17318
17319
17320
17321
17322
17323
17324
17325
17326
17327
17328
17329
17330
17331
17332
17333
17334
17335
17336
17337
17338
17339
17340
17341
17342
17343
17344
17345
17346
17347
17348
17349
17350
17351
17352
17353
17354
17355
17356
17357
17358
17359
17360
17361
17362
17363
17364
17365
17366
17367
17368
17369
17370
17371
17372
17373
17374
17375
17376
17377
17378
17379
17380
17381
17382
17383
17384
17385
17386
17387
17388
17389
17390
17391
17392
17393
17394
17395
17396
17397
17398
17399
17400
17401
17402
17403
17404
17405
17406
17407
17408
17409
17410
17411
17412
17413
17414
17415
17416
17417
17418
17419
17420
17421
17422
17423
17424
17425
17426
17427
17428
17429
17430
17431
17432
17433
17434
17435
17436
17437
17438
17439
17440
17441
17442
17443
17444
17445
17446
17447
17448
17449
17450
17451
17452
17453
17454
17455
17456
17457
17458
17459
17460
17461
17462
17463
17464
17465
17466
17467
17468
17469
17470
17471
17472
17473
17474
17475
17476
17477
17478
17479
17480
17481
17482
17483
17484
17485
17486
17487
17488
17489
17490
17491
17492
17493
17494
17495
17496
17497
17498
17499
17500
17501
17502
17503
17504
17505
17506
17507
17508
17509
17510
17511
17512
17513
17514
17515
17516
17517
17518
17519
17520
17521
17522
17523
17524
17525
17526
17527
17528
17529
17530
17531
17532
17533
17534
17535
17536
17537
17538
17539
17540
17541
17542
17543
17544
17545
17546
17547
17548
17549
17550
17551
17552
17553
17554
17555
17556
17557
17558
17559
17560
17561
17562
17563
17564
17565
17566
17567
17568
17569
17570
17571
17572
17573
17574
17575
17576
17577
17578
17579
17580
17581
17582
17583
17584
17585
17586
17587
17588
17589
17590
17591
17592
17593
17594
17595
17596
17597
17598
17599
17600
17601
17602
17603
17604
17605
17606
17607
17608
17609
17610
17611
17612
17613
17614
17615
17616
17617
17618
17619
17620
17621
17622
17623
17624
17625
17626
17627
17628
17629
17630
17631
17632
17633
17634
17635
17636
17637
17638
17639
17640
17641
17642
17643
17644
17645
17646
17647
17648
17649
17650
17651
17652
17653
17654
17655
17656
17657
17658
17659
17660
17661
17662
17663
17664
17665
17666
17667
17668
17669
17670
17671
17672
17673
17674
17675
17676
17677
17678
17679
17680
17681
17682
17683
17684
17685
17686
17687
17688
17689
17690
17691
17692
17693
17694
17695
17696
17697
17698
17699
17700
17701
17702
17703
17704
17705
17706
17707
17708
17709
17710
17711
17712
17713
17714
17715
17716
17717
17718
17719
17720
17721
17722
17723
17724
17725
17726
17727
17728
17729
17730
17731
17732
17733
17734
17735
17736
17737
17738
17739
17740
17741
17742
17743
17744
17745
17746
17747
17748
17749
17750
17751
17752
17753
17754
17755
17756
17757
17758
17759
17760
17761
17762
17763
17764
17765
17766
17767
17768
17769
17770
17771
17772
17773
17774
17775
17776
17777
17778
17779
17780
17781
17782
17783
17784
17785
17786
17787
17788
17789
17790
17791
17792
17793
17794
17795
17796
17797
17798
17799
17800
17801
17802
17803
17804
17805
17806
17807
17808
17809
17810
17811
17812
17813
17814
17815
17816
17817
17818
17819
17820
17821
17822
17823
17824
17825
17826
17827
17828
17829
17830
17831
17832
17833
17834
17835
17836
17837
17838
17839
17840
17841
17842
17843
17844
17845
17846
17847
17848
17849
17850
17851
17852
17853
17854
17855
17856
17857
17858
17859
17860
17861
17862
17863
17864
17865
17866
17867
17868
17869
17870
17871
17872
17873
17874
17875
17876
17877
17878
17879
17880
17881
17882
17883
17884
17885
17886
17887
17888
17889
17890
17891
17892
17893
17894
17895
17896
17897
17898
17899
17900
17901
17902
17903
17904
17905
17906
17907
17908
17909
17910
17911
17912
17913
17914
17915
17916
17917
17918
17919
17920
17921
17922
17923
17924
17925
17926
17927
17928
17929
17930
17931
17932
17933
17934
17935
17936
17937
17938
17939
17940
17941
17942
17943
17944
17945
17946
17947
17948
17949
17950
17951
17952
17953
17954
17955
17956
17957
17958
17959
17960
17961
17962
17963
17964
17965
17966
17967
17968
17969
17970
17971
17972
17973
17974
17975
17976
17977
17978
17979
17980
17981
17982
17983
17984
17985
17986
17987
17988
17989
17990
17991
17992
17993
17994
17995
17996
17997
17998
17999
18000
18001
18002
18003
18004
18005
18006
18007
18008
18009
18010
18011
18012
18013
18014
18015
18016
18017
18018
18019
18020
18021
18022
18023
18024
18025
18026
18027
18028
18029
18030
18031
18032
18033
18034
18035
18036
18037
18038
18039
18040
18041
18042
18043
18044
18045
18046
18047
18048
18049
18050
18051
18052
18053
18054
18055
18056
18057
18058
18059
18060
18061
18062
18063
18064
18065
18066
18067
18068
18069
18070
18071
18072
18073
18074
18075
18076
18077
18078
18079
18080
18081
18082
18083
18084
18085
18086
18087
18088
18089
18090
18091
18092
18093
18094
18095
18096
18097
18098
18099
18100
18101
18102
18103
18104
18105
18106
18107
18108
18109
18110
18111
18112
18113
18114
18115
18116
18117
18118
18119
18120
18121
18122
18123
18124
18125
18126
18127
18128
18129
18130
18131
18132
18133
18134
18135
18136
18137
18138
18139
18140
18141
18142
18143
18144
18145
18146
18147
18148
18149
18150
18151
18152
18153
18154
18155
18156
18157
18158
18159
18160
18161
18162
18163
18164
18165
18166
18167
18168
18169
18170
18171
18172
18173
18174
18175
18176
18177
18178
18179
18180
18181
18182
18183
18184
18185
18186
18187
18188
18189
18190
18191
18192
18193
18194
18195
18196
18197
18198
18199
18200
18201
18202
18203
18204
18205
18206
18207
18208
18209
18210
18211
18212
18213
18214
18215
18216
18217
18218
18219
18220
18221
18222
18223
18224
18225
18226
18227
18228
18229
18230
18231
18232
18233
18234
18235
18236
18237
18238
18239
18240
18241
18242
18243
18244
18245
18246
18247
18248
18249
18250
18251
18252
18253
18254
18255
18256
18257
18258
18259
18260
18261
18262
18263
18264
18265
18266
18267
18268
18269
18270
18271
18272
18273
18274
18275
18276
18277
18278
18279
18280
18281
18282
18283
18284
18285
18286
18287
18288
18289
18290
18291
18292
18293
18294
18295
18296
18297
18298
18299
18300
18301
18302
18303
18304
18305
18306
18307
18308
18309
18310
18311
18312
18313
18314
18315
18316
18317
18318
18319
18320
18321
18322
18323
18324
18325
18326
18327
18328
18329
18330
18331
18332
18333
18334
18335
18336
18337
18338
18339
18340
18341
18342
18343
18344
18345
18346
18347
18348
18349
18350
18351
18352
18353
18354
18355
18356
18357
18358
18359
18360
18361
18362
18363
18364
18365
18366
18367
18368
18369
18370
18371
18372
18373
18374
18375
18376
18377
18378
18379
18380
18381
18382
18383
18384
18385
18386
18387
18388
18389
18390
18391
18392
18393
18394
18395
18396
18397
18398
18399
18400
18401
18402
18403
18404
18405
18406
18407
18408
18409
18410
18411
18412
18413
18414
18415
18416
18417
18418
18419
18420
18421
18422
18423
18424
18425
18426
18427
18428
18429
18430
18431
18432
18433
18434
18435
18436
18437
18438
18439
18440
18441
18442
18443
18444
18445
18446
18447
18448
18449
18450
18451
18452
18453
18454
18455
18456
18457
18458
18459
18460
18461
18462
18463
18464
18465
18466
18467
18468
18469
18470
18471
18472
18473
18474
18475
18476
18477
18478
18479
18480
18481
18482
18483
18484
18485
18486
18487
18488
18489
18490
18491
18492
18493
18494
18495
18496
18497
18498
18499
18500
18501
18502
18503
18504
18505
18506
18507
18508
18509
18510
18511
18512
18513
18514
18515
18516
18517
18518
18519
18520
18521
18522
18523
18524
18525
18526
18527
18528
18529
18530
18531
18532
18533
18534
18535
18536
18537
18538
18539
18540
18541
18542
18543
18544
18545
18546
18547
18548
18549
18550
18551
18552
18553
18554
18555
18556
18557
18558
18559
18560
18561
18562
18563
18564
18565
18566
18567
18568
18569
18570
18571
18572
18573
18574
18575
18576
18577
18578
18579
18580
18581
18582
18583
18584
18585
18586
18587
18588
18589
18590
18591
18592
18593
18594
18595
18596
18597
18598
18599
18600
18601
18602
18603
18604
18605
18606
18607
18608
18609
18610
18611
18612
18613
18614
18615
18616
18617
18618
18619
18620
18621
18622
18623
18624
18625
18626
18627
18628
18629
18630
18631
18632
18633
18634
18635
18636
18637
18638
18639
18640
18641
18642
18643
18644
18645
18646
18647
18648
18649
18650
18651
18652
18653
18654
18655
18656
18657
18658
18659
18660
18661
18662
18663
18664
18665
18666
18667
18668
18669
18670
18671
18672
18673
18674
18675
18676
18677
18678
18679
18680
18681
18682
18683
18684
18685
18686
18687
18688
18689
18690
18691
18692
18693
18694
18695
18696
18697
18698
18699
18700
18701
18702
18703
18704
18705
18706
18707
18708
18709
18710
18711
18712
18713
18714
18715
18716
18717
18718
18719
18720
18721
18722
18723
18724
18725
18726
18727
18728
18729
18730
18731
18732
18733
18734
18735
18736
18737
18738
18739
18740
18741
18742
18743
18744
18745
18746
18747
18748
18749
18750
18751
18752
18753
18754
18755
18756
18757
18758
18759
18760
18761
18762
18763
18764
18765
18766
18767
18768
18769
18770
18771
18772
18773
18774
18775
18776
18777
18778
18779
18780
18781
18782
18783
18784
18785
18786
18787
18788
18789
18790
18791
18792
18793
18794
18795
18796
18797
18798
18799
18800
18801
18802
18803
18804
18805
18806
18807
18808
18809
18810
18811
18812
18813
18814
18815
18816
18817
18818
18819
18820
18821
18822
18823
18824
18825
18826
18827
18828
18829
18830
18831
18832
18833
18834
18835
18836
18837
18838
18839
18840
18841
18842
18843
18844
18845
18846
18847
18848
18849
18850
18851
18852
18853
18854
18855
18856
18857
18858
18859
18860
18861
18862
18863
18864
18865
18866
18867
18868
18869
18870
18871
18872
18873
18874
18875
18876
18877
18878
18879
18880
18881
18882
18883
18884
18885
18886
18887
18888
18889
18890
18891
18892
18893
18894
18895
18896
18897
18898
18899
18900
18901
18902
18903
18904
18905
18906
18907
18908
18909
18910
18911
18912
18913
18914
18915
18916
18917
18918
18919
18920
18921
18922
18923
18924
18925
18926
18927
18928
18929
18930
18931
18932
18933
18934
18935
18936
18937
18938
18939
18940
18941
18942
18943
18944
18945
18946
18947
18948
18949
18950
18951
18952
18953
18954
18955
18956
18957
18958
18959
18960
18961
18962
18963
18964
18965
18966
18967
18968
18969
18970
18971
18972
18973
18974
18975
18976
18977
18978
18979
18980
18981
18982
18983
18984
18985
18986
18987
18988
18989
18990
18991
18992
18993
18994
18995
18996
18997
18998
18999
19000
19001
19002
19003
19004
19005
19006
19007
19008
19009
19010
19011
19012
19013
19014
19015
19016
19017
19018
19019
19020
19021
19022
19023
19024
19025
19026
19027
19028
19029
19030
19031
19032
19033
19034
19035
19036
19037
19038
19039
19040
19041
19042
19043
19044
19045
19046
19047
19048
19049
19050
19051
19052
19053
19054
19055
19056
19057
19058
19059
19060
19061
19062
19063
19064
19065
19066
19067
19068
19069
19070
19071
19072
19073
19074
19075
19076
19077
19078
19079
19080
19081
19082
19083
19084
19085
19086
19087
19088
19089
19090
19091
19092
19093
19094
19095
19096
19097
19098
19099
19100
19101
19102
19103
19104
19105
19106
19107
19108
19109
19110
19111
19112
19113
19114
19115
19116
19117
19118
19119
19120
19121
19122
19123
19124
19125
19126
19127
19128
19129
19130
19131
19132
19133
19134
19135
19136
19137
19138
19139
19140
19141
19142
19143
19144
19145
19146
19147
19148
19149
19150
19151
19152
19153
19154
19155
19156
19157
19158
19159
19160
19161
19162
19163
19164
19165
19166
19167
19168
19169
19170
19171
19172
19173
19174
19175
19176
19177
19178
19179
19180
19181
19182
19183
19184
19185
19186
19187
19188
19189
19190
19191
19192
19193
19194
19195
19196
19197
19198
19199
19200
19201
19202
19203
19204
19205
19206
19207
19208
19209
19210
19211
19212
19213
19214
19215
19216
19217
19218
19219
19220
19221
19222
19223
19224
19225
19226
19227
19228
19229
19230
19231
19232
19233
19234
19235
19236
19237
19238
19239
19240
19241
19242
19243
19244
19245
19246
19247
19248
19249
19250
19251
19252
19253
19254
19255
19256
19257
19258
19259
19260
19261
19262
19263
19264
19265
19266
19267
19268
19269
19270
19271
19272
19273
19274
19275
19276
19277
19278
19279
19280
19281
19282
19283
19284
19285
19286
19287
19288
19289
19290
19291
19292
19293
19294
19295
19296
19297
19298
19299
19300
19301
19302
19303
19304
19305
19306
19307
19308
19309
19310
19311
19312
19313
19314
19315
19316
19317
19318
19319
19320
19321
19322
19323
19324
19325
19326
19327
19328
19329
19330
19331
19332
19333
19334
19335
19336
19337
19338
19339
19340
19341
19342
19343
19344
19345
19346
19347
19348
19349
19350
19351
19352
19353
19354
19355
19356
19357
19358
19359
19360
19361
19362
19363
19364
19365
19366
19367
19368
19369
19370
19371
19372
19373
19374
19375
19376
19377
19378
19379
19380
19381
19382
19383
19384
19385
19386
19387
19388
19389
19390
19391
19392
19393
19394
19395
19396
19397
19398
19399
19400
19401
19402
19403
19404
19405
19406
19407
19408
19409
19410
19411
19412
19413
19414
19415
19416
19417
19418
19419
19420
19421
19422
19423
19424
19425
19426
19427
19428
19429
19430
19431
19432
19433
19434
19435
19436
19437
19438
19439
19440
19441
19442
19443
19444
19445
19446
19447
19448
19449
19450
19451
19452
19453
19454
19455
19456
19457
19458
19459
19460
19461
19462
19463
19464
19465
19466
19467
19468
19469
19470
19471
19472
19473
19474
19475
19476
19477
19478
19479
19480
19481
19482
19483
19484
19485
19486
19487
19488
19489
19490
19491
19492
19493
19494
19495
19496
19497
19498
19499
19500
19501
19502
19503
19504
19505
19506
19507
19508
19509
19510
19511
19512
19513
19514
19515
19516
19517
19518
19519
19520
19521
19522
19523
19524
19525
19526
19527
19528
19529
19530
19531
19532
19533
19534
19535
19536
19537
19538
19539
19540
19541
19542
19543
19544
19545
19546
19547
19548
19549
19550
19551
19552
19553
19554
19555
19556
19557
19558
19559
19560
19561
19562
19563
19564
19565
19566
19567
19568
19569
19570
19571
19572
19573
19574
19575
19576
19577
19578
19579
19580
19581
19582
19583
19584
19585
19586
19587
19588
19589
19590
19591
19592
19593
19594
19595
19596
19597
19598
19599
19600
19601
19602
19603
19604
19605
19606
19607
19608
19609
19610
19611
19612
19613
19614
19615
19616
19617
19618
19619
19620
19621
19622
19623
19624
19625
19626
19627
19628
19629
19630
19631
19632
19633
19634
19635
19636
19637
19638
19639
19640
19641
19642
19643
19644
19645
19646
19647
19648
19649
19650
19651
19652
19653
19654
19655
19656
19657
19658
19659
19660
19661
19662
19663
19664
19665
19666
19667
19668
19669
19670
19671
19672
19673
19674
19675
19676
19677
19678
19679
19680
19681
19682
19683
19684
19685
19686
19687
19688
19689
19690
19691
19692
19693
19694
19695
19696
19697
19698
19699
19700
19701
19702
19703
19704
19705
19706
19707
19708
19709
19710
19711
19712
19713
19714
19715
19716
19717
19718
19719
19720
19721
19722
19723
19724
19725
19726
19727
19728
19729
19730
19731
19732
19733
19734
19735
19736
19737
19738
19739
19740
19741
19742
19743
19744
19745
19746
19747
19748
19749
19750
19751
19752
19753
19754
19755
19756
19757
19758
19759
19760
19761
19762
19763
19764
19765
19766
19767
19768
19769
19770
19771
19772
19773
19774
19775
19776
19777
19778
19779
19780
19781
19782
19783
19784
19785
19786
19787
19788
19789
19790
19791
19792
19793
19794
19795
19796
19797
19798
19799
19800
19801
19802
19803
19804
19805
19806
19807
19808
19809
19810
19811
19812
19813
19814
19815
19816
19817
19818
19819
19820
19821
19822
19823
19824
19825
19826
19827
19828
19829
19830
19831
19832
19833
19834
19835
19836
19837
19838
19839
19840
19841
19842
19843
19844
19845
19846
19847
19848
19849
19850
19851
19852
19853
19854
19855
19856
19857
19858
19859
19860
19861
19862
19863
19864
19865
19866
19867
19868
19869
19870
19871
19872
19873
19874
19875
19876
19877
19878
19879
19880
19881
19882
19883
19884
19885
19886
19887
19888
19889
19890
19891
19892
19893
19894
19895
19896
19897
19898
19899
19900
19901
19902
19903
19904
19905
19906
19907
19908
19909
19910
19911
19912
19913
19914
19915
19916
19917
19918
19919
19920
19921
19922
19923
19924
19925
19926
19927
19928
19929
19930
19931
19932
19933
19934
19935
19936
19937
19938
19939
19940
19941
19942
19943
19944
19945
19946
19947
19948
19949
19950
19951
19952
19953
19954
19955
19956
19957
19958
19959
19960
19961
19962
19963
19964
19965
19966
19967
19968
19969
19970
19971
19972
19973
19974
19975
19976
19977
19978
19979
19980
19981
19982
19983
19984
19985
19986
19987
19988
19989
19990
19991
19992
19993
19994
19995
19996
19997
19998
19999
20000
20001
20002
20003
20004
20005
20006
20007
20008
20009
20010
20011
20012
20013
20014
20015
20016
20017
20018
20019
20020
20021
20022
20023
20024
20025
20026
20027
20028
20029
20030
20031
20032
20033
20034
20035
20036
20037
20038
20039
20040
20041
20042
20043
20044
20045
20046
20047
20048
20049
20050
20051
20052
20053
20054
20055
20056
20057
20058
20059
20060
20061
20062
20063
20064
20065
20066
20067
20068
20069
20070
20071
20072
20073
20074
20075
20076
20077
20078
20079
20080
20081
20082
20083
20084
20085
20086
20087
20088
20089
20090
20091
20092
20093
20094
20095
20096
20097
20098
20099
20100
20101
20102
20103
20104
20105
20106
20107
20108
20109
20110
20111
20112
20113
20114
20115
20116
20117
20118
20119
20120
20121
20122
20123
20124
20125
20126
20127
20128
20129
20130
20131
20132
20133
20134
20135
20136
20137
20138
20139
20140
20141
20142
20143
20144
20145
20146
20147
20148
20149
20150
20151
20152
20153
20154
20155
20156
20157
20158
20159
20160
20161
20162
20163
20164
20165
20166
20167
20168
20169
20170
20171
20172
20173
20174
20175
20176
20177
20178
20179
20180
20181
20182
20183
20184
20185
20186
20187
20188
20189
20190
20191
20192
20193
20194
20195
20196
20197
20198
20199
20200
20201
20202
20203
20204
20205
20206
20207
20208
20209
20210
20211
20212
20213
20214
20215
20216
20217
20218
20219
20220
20221
20222
20223
20224
20225
20226
20227
20228
20229
20230
20231
20232
20233
20234
20235
20236
20237
20238
20239
20240
20241
20242
20243
20244
20245
20246
20247
20248
20249
20250
20251
20252
20253
20254
20255
20256
20257
20258
20259
20260
20261
20262
20263
20264
20265
20266
20267
20268
20269
20270
20271
20272
20273
20274
20275
20276
20277
20278
20279
20280
20281
20282
20283
20284
20285
20286
20287
20288
20289
20290
20291
20292
20293
20294
20295
20296
20297
20298
20299
20300
20301
20302
20303
20304
20305
20306
20307
20308
20309
20310
20311
20312
20313
20314
20315
20316
20317
20318
20319
20320
20321
20322
20323
20324
20325
20326
20327
20328
20329
20330
20331
20332
20333
20334
20335
20336
20337
20338
20339
20340
20341
20342
20343
20344
20345
20346
20347
20348
20349
20350
20351
20352
20353
20354
20355
20356
20357
20358
20359
20360
20361
20362
20363
20364
20365
20366
20367
20368
20369
20370
20371
20372
20373
20374
20375
20376
20377
20378
20379
20380
20381
20382
20383
20384
20385
20386
20387
20388
20389
20390
20391
20392
20393
20394
20395
20396
20397
20398
20399
20400
20401
20402
20403
20404
20405
20406
20407
20408
20409
20410
20411
20412
20413
20414
20415
20416
20417
20418
20419
20420
20421
20422
20423
20424
20425
20426
20427
20428
20429
20430
20431
20432
20433
20434
20435
20436
20437
20438
20439
20440
20441
20442
20443
20444
20445
20446
20447
20448
20449
20450
20451
20452
20453
20454
20455
20456
20457
20458
20459
20460
20461
20462
20463
20464
20465
20466
20467
20468
20469
20470
20471
20472
20473
20474
20475
20476
20477
20478
20479
20480
20481
20482
20483
20484
20485
20486
20487
20488
20489
20490
20491
20492
20493
20494
20495
20496
20497
20498
20499
20500
20501
20502
20503
20504
20505
20506
20507
20508
20509
20510
20511
20512
20513
20514
20515
20516
20517
20518
20519
20520
20521
20522
20523
20524
20525
20526
20527
20528
20529
20530
20531
20532
20533
20534
20535
20536
20537
20538
20539
20540
20541
20542
20543
20544
20545
20546
20547
20548
20549
20550
20551
20552
20553
20554
20555
20556
20557
20558
20559
20560
20561
20562
20563
20564
20565
20566
20567
20568
20569
20570
20571
20572
20573
20574
20575
20576
20577
20578
20579
20580
20581
20582
20583
20584
20585
20586
20587
20588
20589
20590
20591
20592
20593
20594
20595
20596
20597
20598
20599
20600
20601
20602
20603
20604
20605
20606
20607
20608
20609
20610
20611
20612
20613
20614
20615
20616
20617
20618
20619
20620
20621
20622
20623
20624
20625
20626
20627
20628
20629
20630
20631
20632
20633
20634
20635
20636
20637
20638
20639
20640
20641
20642
20643
20644
20645
20646
20647
20648
20649
20650
20651
20652
20653
20654
20655
20656
20657
20658
20659
20660
20661
20662
20663
20664
20665
20666
20667
20668
20669
20670
20671
20672
20673
20674
20675
20676
20677
20678
20679
20680
20681
20682
20683
20684
20685
20686
20687
20688
20689
20690
20691
20692
20693
20694
20695
20696
20697
20698
20699
20700
20701
20702
20703
20704
20705
20706
20707
20708
20709
20710
20711
20712
20713
20714
20715
20716
20717
20718
20719
20720
20721
20722
20723
20724
20725
20726
20727
20728
20729
20730
20731
20732
20733
20734
20735
20736
20737
20738
20739
20740
20741
20742
20743
20744
20745
20746
20747
20748
20749
20750
20751
20752
20753
20754
20755
20756
20757
20758
20759
20760
20761
20762
20763
20764
20765
20766
20767
20768
20769
20770
20771
20772
20773
20774
20775
20776
20777
20778
20779
20780
20781
20782
20783
20784
20785
20786
20787
20788
20789
20790
20791
20792
20793
20794
20795
20796
20797
20798
20799
20800
20801
20802
20803
20804
20805
20806
20807
20808
20809
20810
20811
20812
20813
20814
20815
20816
20817
20818
20819
20820
20821
20822
20823
20824
20825
20826
20827
20828
20829
20830
20831
20832
20833
20834
20835
20836
20837
20838
20839
20840
20841
20842
20843
20844
20845
20846
20847
20848
20849
20850
20851
20852
20853
20854
20855
20856
20857
20858
20859
20860
20861
20862
20863
20864
20865
20866
20867
20868
20869
20870
20871
20872
20873
20874
20875
20876
20877
20878
20879
20880
20881
20882
20883
20884
20885
20886
20887
20888
20889
20890
20891
20892
20893
20894
20895
20896
20897
20898
20899
20900
20901
20902
20903
20904
20905
20906
20907
20908
20909
20910
20911
20912
20913
20914
20915
20916
20917
20918
20919
20920
20921
20922
20923
20924
20925
20926
20927
20928
20929
20930
20931
20932
20933
20934
20935
20936
20937
20938
20939
20940
20941
20942
20943
20944
20945
20946
20947
20948
20949
20950
20951
20952
20953
20954
20955
20956
20957
20958
20959
20960
20961
20962
20963
20964
20965
20966
20967
20968
20969
20970
20971
20972
20973
20974
20975
20976
20977
20978
20979
20980
20981
20982
20983
20984
20985
20986
20987
20988
20989
20990
20991
20992
20993
20994
20995
20996
20997
20998
20999
21000
21001
21002
21003
21004
21005
21006
21007
21008
21009
21010
21011
21012
21013
21014
21015
21016
21017
21018
21019
21020
21021
21022
21023
21024
21025
21026
21027
21028
21029
21030
21031
21032
21033
21034
21035
21036
21037
21038
21039
21040
21041
21042
21043
21044
21045
21046
21047
21048
21049
21050
21051
21052
21053
21054
21055
21056
21057
21058
21059
21060
21061
21062
21063
21064
21065
21066
21067
21068
21069
21070
21071
21072
21073
21074
21075
21076
21077
21078
21079
21080
21081
21082
21083
21084
21085
21086
21087
21088
21089
21090
21091
21092
21093
21094
21095
21096
21097
21098
21099
21100
21101
21102
21103
21104
21105
21106
21107
21108
21109
21110
21111
21112
21113
21114
21115
21116
21117
21118
21119
21120
21121
21122
21123
21124
21125
21126
21127
21128
21129
21130
21131
21132
21133
21134
21135
21136
21137
21138
21139
21140
21141
21142
21143
21144
21145
21146
21147
21148
21149
21150
21151
21152
21153
21154
21155
21156
21157
21158
21159
21160
21161
21162
21163
21164
21165
21166
21167
21168
21169
21170
21171
21172
21173
21174
21175
21176
21177
21178
21179
21180
21181
21182
21183
21184
21185
21186
21187
21188
21189
21190
21191
21192
21193
21194
21195
21196
21197
21198
21199
21200
21201
21202
21203
21204
21205
21206
21207
21208
21209
21210
21211
21212
21213
21214
21215
21216
21217
21218
21219
21220
21221
21222
21223
21224
21225
21226
21227
21228
21229
21230
21231
21232
21233
21234
21235
21236
21237
21238
21239
21240
21241
21242
21243
21244
21245
21246
21247
21248
21249
21250
21251
21252
21253
21254
21255
21256
21257
21258
21259
21260
21261
21262
21263
21264
21265
21266
21267
21268
21269
21270
21271
21272
21273
21274
21275
21276
21277
21278
21279
21280
21281
21282
21283
21284
21285
21286
21287
21288
21289
21290
21291
21292
21293
21294
21295
21296
21297
21298
21299
21300
21301
21302
21303
21304
21305
21306
21307
21308
21309
21310
21311
21312
21313
21314
21315
21316
21317
21318
21319
21320
21321
21322
21323
21324
21325
21326
21327
21328
21329
21330
21331
21332
21333
21334
21335
21336
21337
21338
21339
21340
21341
21342
21343
21344
21345
21346
21347
21348
21349
21350
21351
21352
21353
21354
21355
21356
21357
21358
21359
21360
21361
21362
21363
21364
21365
21366
21367
21368
21369
21370
21371
21372
21373
21374
21375
21376
21377
21378
21379
21380
21381
21382
21383
21384
21385
21386
21387
21388
21389
21390
21391
21392
21393
21394
21395
21396
21397
21398
21399
21400
21401
21402
21403
21404
21405
21406
21407
21408
21409
21410
21411
21412
21413
21414
21415
21416
21417
21418
21419
21420
21421
21422
21423
21424
21425
21426
21427
21428
21429
21430
21431
21432
21433
21434
21435
21436
21437
21438
21439
21440
21441
21442
21443
21444
21445
21446
21447
21448
21449
21450
21451
21452
21453
21454
21455
21456
21457
21458
21459
21460
21461
21462
21463
21464
21465
21466
21467
21468
21469
21470
21471
21472
21473
21474
21475
21476
21477
21478
21479
21480
21481
21482
21483
21484
21485
21486
21487
21488
21489
21490
21491
21492
21493
21494
21495
21496
21497
21498
21499
21500
21501
21502
21503
21504
21505
21506
21507
21508
21509
21510
21511
21512
21513
21514
21515
21516
21517
21518
21519
21520
21521
21522
21523
21524
21525
21526
21527
21528
21529
21530
21531
21532
21533
21534
21535
21536
21537
21538
21539
21540
21541
21542
21543
21544
21545
21546
21547
21548
21549
21550
21551
21552
21553
21554
21555
21556
21557
21558
21559
21560
21561
21562
21563
21564
21565
21566
21567
21568
21569
21570
21571
21572
21573
21574
21575
21576
21577
21578
21579
21580
21581
21582
21583
21584
21585
21586
21587
21588
21589
21590
21591
21592
21593
21594
21595
21596
21597
21598
21599
21600
21601
21602
21603
21604
21605
21606
21607
21608
21609
21610
21611
21612
21613
21614
21615
21616
21617
21618
21619
21620
21621
21622
21623
21624
21625
21626
21627
21628
21629
21630
21631
21632
21633
21634
21635
21636
21637
21638
21639
21640
21641
21642
21643
21644
21645
21646
21647
21648
21649
21650
21651
21652
21653
21654
21655
21656
21657
21658
21659
21660
21661
21662
21663
21664
21665
21666
21667
21668
21669
21670
21671
21672
21673
21674
21675
21676
21677
21678
21679
21680
21681
21682
21683
21684
21685
21686
21687
21688
21689
21690
21691
21692
21693
21694
21695
21696
21697
21698
21699
21700
21701
21702
21703
21704
21705
21706
21707
21708
21709
21710
21711
21712
21713
21714
21715
21716
21717
21718
21719
21720
21721
21722
21723
21724
21725
21726
21727
21728
21729
21730
21731
21732
21733
21734
21735
21736
21737
21738
21739
21740
21741
21742
21743
21744
21745
21746
21747
21748
21749
21750
21751
21752
21753
21754
21755
21756
21757
21758
21759
21760
21761
21762
21763
21764
21765
21766
21767
21768
21769
21770
21771
21772
21773
21774
21775
21776
21777
21778
21779
21780
21781
21782
21783
21784
21785
21786
21787
21788
21789
21790
21791
21792
21793
21794
21795
21796
21797
21798
21799
21800
21801
21802
21803
21804
21805
21806
21807
21808
21809
21810
21811
21812
21813
21814
21815
21816
21817
21818
21819
21820
21821
21822
21823
21824
21825
21826
21827
21828
21829
21830
21831
21832
21833
21834
21835
21836
21837
21838
21839
21840
21841
21842
21843
21844
21845
21846
21847
21848
21849
21850
21851
21852
21853
21854
21855
21856
21857
21858
21859
21860
21861
21862
21863
21864
21865
21866
21867
21868
21869
21870
21871
21872
21873
21874
21875
21876
21877
21878
21879
21880
21881
21882
21883
21884
21885
21886
21887
21888
21889
21890
21891
21892
21893
21894
21895
21896
21897
21898
21899
21900
21901
21902
21903
21904
21905
21906
21907
21908
21909
21910
21911
21912
21913
21914
21915
21916
21917
21918
21919
21920
21921
21922
21923
21924
21925
21926
21927
21928
21929
21930
21931
21932
21933
21934
21935
21936
21937
21938
21939
21940
21941
21942
21943
21944
21945
21946
21947
21948
21949
21950
21951
21952
21953
21954
21955
21956
21957
21958
21959
21960
21961
21962
21963
21964
21965
21966
21967
21968
21969
21970
21971
21972
21973
21974
21975
21976
21977
21978
21979
21980
21981
21982
21983
21984
21985
21986
21987
21988
21989
21990
21991
21992
21993
21994
21995
21996
21997
21998
21999
22000
22001
22002
22003
22004
22005
22006
22007
22008
22009
22010
22011
22012
22013
22014
22015
22016
22017
22018
22019
22020
22021
22022
22023
22024
22025
22026
22027
22028
22029
22030
22031
22032
22033
22034
22035
22036
22037
22038
22039
22040
22041
22042
22043
22044
22045
22046
22047
22048
22049
22050
22051
22052
22053
22054
22055
22056
22057
22058
22059
22060
22061
22062
22063
22064
22065
22066
22067
22068
22069
22070
22071
22072
22073
22074
22075
22076
22077
22078
22079
22080
22081
22082
22083
22084
22085
22086
22087
22088
22089
22090
22091
22092
22093
22094
22095
22096
22097
22098
22099
22100
22101
22102
22103
22104
22105
22106
22107
22108
22109
22110
22111
22112
22113
22114
22115
22116
22117
22118
22119
22120
22121
22122
22123
22124
22125
22126
22127
22128
22129
22130
22131
22132
22133
22134
22135
22136
22137
22138
22139
22140
22141
22142
22143
22144
22145
22146
22147
22148
22149
22150
22151
22152
22153
22154
22155
22156
22157
22158
22159
22160
22161
22162
22163
22164
22165
22166
22167
22168
22169
22170
22171
22172
22173
22174
22175
22176
22177
22178
22179
22180
22181
22182
22183
22184
22185
22186
22187
22188
22189
22190
22191
22192
22193
22194
22195
22196
22197
22198
22199
22200
22201
22202
22203
22204
22205
22206
22207
22208
22209
22210
22211
22212
22213
22214
22215
22216
22217
22218
22219
22220
22221
22222
22223
22224
22225
22226
22227
22228
22229
22230
22231
22232
22233
22234
22235
22236
22237
22238
22239
22240
22241
22242
22243
22244
22245
22246
22247
22248
22249
22250
22251
22252
22253
22254
22255
22256
22257
22258
22259
22260
22261
22262
22263
22264
22265
22266
22267
22268
22269
22270
22271
22272
22273
22274
22275
22276
22277
22278
22279
22280
22281
22282
22283
22284
22285
22286
22287
22288
22289
22290
22291
22292
22293
22294
22295
22296
22297
22298
22299
22300
22301
22302
22303
22304
22305
22306
22307
22308
22309
22310
22311
22312
22313
22314
22315
22316
22317
22318
22319
22320
22321
22322
22323
22324
22325
22326
22327
22328
22329
22330
22331
22332
22333
22334
22335
22336
22337
22338
22339
22340
22341
22342
22343
22344
22345
22346
22347
22348
22349
22350
22351
22352
22353
22354
22355
22356
22357
22358
22359
22360
22361
22362
22363
22364
22365
22366
22367
22368
22369
22370
22371
22372
22373
22374
22375
22376
22377
22378
22379
22380
22381
22382
22383
22384
22385
22386
22387
22388
22389
22390
22391
22392
22393
22394
22395
22396
22397
22398
22399
22400
22401
22402
22403
22404
22405
22406
22407
22408
22409
22410
22411
22412
22413
22414
22415
22416
22417
22418
22419
22420
22421
22422
22423
22424
22425
22426
22427
22428
22429
22430
22431
22432
22433
22434
22435
22436
22437
22438
22439
22440
22441
22442
22443
22444
22445
22446
22447
22448
22449
22450
22451
22452
22453
22454
22455
22456
22457
22458
22459
22460
22461
22462
22463
22464
22465
22466
22467
22468
22469
22470
22471
22472
22473
22474
22475
22476
22477
22478
22479
22480
22481
22482
22483
22484
22485
22486
22487
22488
22489
22490
22491
22492
22493
22494
22495
22496
22497
22498
22499
22500
22501
22502
22503
22504
22505
22506
22507
22508
22509
22510
22511
22512
22513
22514
22515
22516
22517
22518
22519
22520
22521
22522
22523
22524
22525
22526
22527
22528
22529
22530
22531
22532
22533
22534
22535
22536
22537
22538
22539
22540
22541
22542
22543
22544
22545
22546
22547
22548
22549
22550
22551
22552
22553
22554
22555
22556
22557
22558
22559
22560
22561
22562
22563
22564
22565
22566
22567
22568
22569
22570
22571
22572
22573
22574
22575
22576
22577
22578
22579
22580
22581
22582
22583
22584
22585
22586
22587
22588
22589
22590
22591
22592
22593
22594
22595
22596
22597
22598
22599
22600
22601
22602
22603
22604
22605
22606
22607
22608
22609
22610
22611
22612
22613
22614
22615
22616
22617
22618
22619
22620
22621
22622
22623
22624
22625
22626
22627
22628
22629
22630
22631
22632
22633
22634
22635
22636
22637
22638
22639
22640
22641
22642
22643
22644
22645
22646
22647
22648
22649
22650
22651
22652
22653
22654
22655
22656
22657
22658
22659
22660
22661
22662
22663
22664
22665
22666
22667
22668
22669
22670
22671
22672
22673
22674
22675
22676
22677
22678
22679
22680
22681
22682
22683
22684
22685
22686
22687
22688
22689
22690
22691
22692
22693
22694
22695
22696
22697
22698
22699
22700
22701
22702
22703
22704
22705
22706
22707
22708
22709
22710
22711
22712
22713
22714
22715
22716
22717
22718
22719
22720
22721
22722
22723
22724
22725
22726
22727
22728
22729
22730
22731
22732
22733
22734
22735
22736
22737
22738
22739
22740
22741
22742
22743
22744
22745
22746
22747
22748
22749
22750
22751
22752
22753
22754
22755
22756
22757
22758
22759
22760
22761
22762
22763
22764
22765
22766
22767
22768
22769
22770
22771
22772
22773
22774
22775
22776
22777
22778
22779
22780
22781
22782
22783
22784
22785
22786
22787
22788
22789
22790
22791
22792
22793
22794
22795
22796
22797
22798
22799
22800
22801
22802
22803
22804
22805
22806
22807
22808
22809
22810
22811
22812
22813
22814
22815
22816
22817
22818
22819
22820
22821
22822
22823
22824
22825
22826
22827
22828
22829
22830
22831
22832
22833
22834
22835
22836
22837
22838
22839
22840
22841
22842
22843
22844
22845
22846
22847
22848
22849
22850
22851
22852
22853
22854
22855
22856
22857
22858
22859
22860
22861
22862
22863
22864
22865
22866
22867
22868
22869
22870
22871
22872
22873
22874
22875
22876
22877
22878
22879
22880
22881
22882
22883
22884
22885
22886
22887
22888
22889
22890
22891
22892
22893
22894
22895
22896
22897
22898
22899
22900
22901
22902
22903
22904
22905
22906
22907
22908
22909
22910
22911
22912
22913
22914
22915
22916
22917
22918
22919
22920
22921
22922
22923
22924
22925
22926
22927
22928
22929
22930
22931
22932
22933
22934
22935
22936
22937
22938
22939
22940
22941
22942
22943
22944
22945
22946
22947
22948
22949
22950
22951
22952
22953
22954
22955
22956
22957
22958
22959
22960
22961
22962
22963
22964
22965
22966
22967
22968
22969
22970
22971
22972
22973
22974
22975
22976
22977
22978
22979
22980
22981
22982
22983
22984
22985
22986
22987
22988
22989
22990
22991
22992
22993
22994
22995
22996
22997
22998
22999
23000
23001
23002
23003
23004
23005
23006
23007
23008
23009
23010
23011
23012
23013
23014
23015
23016
23017
23018
23019
23020
23021
23022
23023
23024
23025
23026
23027
23028
23029
23030
23031
23032
23033
23034
23035
23036
23037
23038
23039
23040
23041
23042
23043
23044
23045
23046
23047
23048
23049
23050
23051
23052
23053
23054
23055
23056
23057
23058
23059
23060
23061
23062
23063
23064
23065
23066
23067
23068
23069
23070
23071
23072
23073
23074
23075
23076
23077
23078
23079
23080
23081
23082
23083
23084
23085
23086
23087
23088
23089
23090
23091
23092
23093
23094
23095
23096
23097
23098
23099
23100
23101
23102
23103
23104
23105
23106
23107
23108
23109
23110
23111
23112
23113
23114
23115
23116
23117
23118
23119
23120
23121
23122
23123
23124
23125
23126
23127
23128
23129
23130
23131
23132
23133
23134
23135
23136
23137
23138
23139
23140
23141
23142
23143
23144
23145
23146
23147
23148
23149
23150
23151
23152
23153
23154
23155
23156
23157
23158
23159
23160
23161
23162
23163
23164
23165
23166
23167
23168
23169
23170
23171
23172
23173
23174
23175
23176
23177
23178
23179
23180
23181
23182
23183
23184
23185
23186
23187
23188
23189
23190
23191
23192
23193
23194
23195
23196
23197
23198
23199
23200
23201
23202
23203
23204
23205
23206
23207
23208
23209
23210
23211
23212
23213
23214
23215
23216
23217
23218
23219
23220
23221
23222
23223
23224
23225
23226
23227
23228
23229
23230
23231
23232
23233
23234
23235
23236
23237
23238
23239
23240
23241
23242
23243
23244
23245
23246
23247
23248
23249
23250
23251
23252
23253
23254
23255
23256
23257
23258
23259
23260
23261
23262
23263
23264
23265
23266
23267
23268
23269
23270
23271
23272
23273
23274
23275
23276
23277
23278
23279
23280
23281
23282
23283
23284
23285
23286
23287
23288
23289
23290
23291
23292
23293
23294
23295
23296
23297
23298
23299
23300
23301
23302
23303
23304
23305
23306
23307
23308
23309
23310
23311
23312
23313
23314
23315
23316
23317
23318
23319
23320
23321
23322
23323
23324
23325
23326
23327
23328
23329
23330
23331
23332
23333
23334
23335
23336
23337
23338
23339
23340
23341
23342
23343
23344
23345
23346
23347
23348
23349
23350
23351
23352
23353
23354
23355
23356
23357
23358
23359
23360
23361
23362
23363
23364
23365
23366
23367
23368
23369
23370
23371
23372
23373
23374
23375
23376
23377
23378
23379
23380
23381
23382
23383
23384
23385
23386
23387
23388
23389
23390
23391
23392
23393
23394
23395
23396
23397
23398
23399
23400
23401
23402
23403
23404
23405
23406
23407
23408
23409
23410
23411
23412
23413
23414
23415
23416
23417
23418
23419
23420
23421
23422
23423
23424
23425
23426
23427
23428
23429
23430
23431
23432
23433
23434
23435
23436
23437
23438
23439
23440
23441
23442
23443
23444
23445
23446
23447
23448
23449
23450
23451
23452
23453
23454
23455
23456
23457
23458
23459
23460
23461
23462
23463
23464
23465
23466
23467
23468
23469
23470
23471
23472
23473
23474
23475
23476
23477
23478
23479
23480
23481
23482
23483
23484
23485
23486
23487
23488
23489
23490
23491
23492
23493
23494
23495
23496
23497
23498
23499
23500
23501
23502
23503
23504
23505
23506
23507
23508
23509
23510
23511
23512
23513
23514
23515
23516
23517
23518
23519
23520
23521
23522
23523
23524
23525
23526
23527
23528
23529
23530
23531
23532
23533
23534
23535
23536
23537
23538
23539
23540
23541
23542
23543
23544
23545
23546
23547
23548
23549
23550
23551
23552
23553
23554
23555
23556
23557
23558
23559
23560
23561
23562
23563
23564
23565
23566
23567
23568
23569
23570
23571
23572
23573
23574
23575
23576
23577
23578
23579
23580
23581
23582
23583
23584
23585
23586
23587
23588
23589
23590
23591
23592
23593
23594
23595
23596
23597
23598
23599
23600
23601
23602
23603
23604
23605
23606
23607
23608
23609
23610
23611
23612
23613
23614
23615
23616
23617
23618
23619
23620
23621
23622
23623
23624
23625
23626
23627
23628
23629
23630
23631
23632
23633
23634
23635
23636
23637
23638
23639
23640
23641
23642
23643
23644
23645
23646
23647
23648
23649
23650
23651
23652
23653
23654
23655
23656
23657
23658
23659
23660
23661
23662
23663
23664
23665
23666
23667
23668
23669
23670
23671
23672
23673
23674
23675
23676
23677
23678
23679
23680
23681
23682
23683
23684
23685
23686
23687
23688
23689
23690
23691
23692
23693
23694
23695
23696
23697
23698
23699
23700
23701
23702
23703
23704
23705
23706
23707
23708
23709
23710
23711
23712
23713
23714
23715
23716
23717
23718
23719
23720
23721
23722
23723
23724
23725
23726
23727
23728
23729
23730
23731
23732
23733
23734
23735
23736
23737
23738
23739
23740
23741
23742
23743
23744
23745
23746
23747
23748
23749
23750
23751
23752
23753
23754
23755
23756
23757
23758
23759
23760
23761
23762
23763
23764
23765
23766
23767
23768
23769
23770
23771
23772
23773
23774
23775
23776
23777
23778
23779
23780
23781
23782
23783
23784
23785
23786
23787
23788
23789
23790
23791
23792
23793
23794
23795
23796
23797
23798
23799
23800
23801
23802
23803
23804
23805
23806
23807
23808
23809
23810
23811
23812
23813
23814
23815
23816
23817
23818
23819
23820
23821
23822
23823
23824
23825
23826
23827
23828
23829
23830
23831
23832
23833
23834
23835
23836
23837
23838
23839
23840
23841
23842
23843
23844
23845
23846
23847
23848
23849
23850
23851
23852
23853
23854
23855
23856
23857
23858
23859
23860
23861
23862
23863
23864
23865
23866
23867
23868
23869
23870
23871
23872
23873
23874
23875
23876
23877
23878
23879
23880
23881
23882
23883
23884
23885
23886
23887
23888
23889
23890
23891
23892
23893
23894
23895
23896
23897
23898
23899
23900
23901
23902
23903
23904
23905
23906
23907
23908
23909
23910
23911
23912
23913
23914
23915
23916
23917
23918
23919
23920
23921
23922
23923
23924
23925
23926
23927
23928
23929
23930
23931
23932
23933
23934
23935
23936
23937
23938
23939
23940
23941
23942
23943
23944
23945
23946
23947
23948
23949
23950
23951
23952
23953
23954
23955
23956
23957
23958
23959
23960
23961
23962
23963
23964
23965
23966
23967
23968
23969
23970
23971
23972
23973
23974
23975
23976
23977
23978
23979
23980
23981
23982
23983
23984
23985
23986
23987
23988
23989
23990
23991
23992
23993
23994
23995
23996
23997
23998
23999
24000
24001
24002
24003
24004
24005
24006
24007
24008
24009
24010
24011
24012
24013
24014
24015
24016
24017
24018
24019
24020
24021
24022
24023
24024
24025
24026
24027
24028
24029
24030
24031
24032
24033
24034
24035
24036
24037
24038
24039
24040
24041
24042
24043
24044
24045
24046
24047
24048
24049
24050
24051
24052
24053
24054
24055
24056
24057
24058
24059
24060
24061
24062
24063
24064
24065
24066
24067
24068
24069
24070
24071
24072
24073
24074
24075
24076
24077
24078
24079
24080
24081
24082
24083
24084
24085
24086
24087
24088
24089
24090
24091
24092
24093
24094
24095
24096
24097
24098
24099
24100
24101
24102
24103
24104
24105
24106
24107
24108
24109
24110
24111
24112
24113
24114
24115
24116
24117
24118
24119
24120
24121
24122
24123
24124
24125
24126
24127
24128
24129
24130
24131
24132
24133
24134
24135
24136
24137
24138
24139
24140
24141
24142
24143
24144
24145
24146
24147
24148
24149
24150
24151
24152
24153
24154
24155
24156
24157
24158
24159
24160
24161
24162
24163
24164
24165
24166
24167
24168
24169
24170
24171
24172
24173
24174
24175
24176
24177
24178
24179
24180
24181
24182
24183
24184
24185
24186
24187
24188
24189
24190
24191
24192
24193
24194
24195
24196
24197
24198
24199
24200
24201
24202
24203
24204
24205
24206
24207
24208
24209
24210
24211
24212
24213
24214
24215
24216
24217
24218
24219
24220
24221
24222
24223
24224
24225
24226
24227
24228
24229
24230
24231
24232
24233
24234
24235
24236
24237
24238
24239
24240
24241
24242
24243
24244
24245
24246
24247
24248
24249
24250
24251
24252
24253
24254
24255
24256
24257
24258
24259
24260
24261
24262
24263
24264
24265
24266
24267
24268
24269
24270
24271
24272
24273
24274
24275
24276
24277
24278
24279
24280
24281
24282
24283
24284
24285
24286
24287
24288
24289
24290
24291
24292
24293
24294
24295
24296
24297
24298
24299
24300
24301
24302
24303
24304
24305
24306
24307
24308
24309
24310
24311
24312
24313
24314
24315
24316
24317
24318
24319
24320
24321
24322
24323
24324
24325
24326
24327
24328
24329
24330
24331
24332
24333
24334
24335
24336
24337
24338
24339
24340
24341
24342
24343
24344
24345
24346
24347
24348
24349
24350
24351
24352
24353
24354
24355
24356
24357
24358
24359
24360
24361
24362
24363
24364
24365
24366
24367
24368
24369
24370
24371
24372
24373
24374
24375
24376
24377
24378
24379
24380
24381
24382
24383
24384
24385
24386
24387
24388
24389
24390
24391
24392
24393
24394
24395
24396
24397
24398
24399
24400
24401
24402
24403
24404
24405
24406
24407
24408
24409
24410
24411
24412
24413
24414
24415
24416
24417
24418
24419
24420
24421
24422
24423
24424
24425
24426
24427
24428
24429
24430
24431
24432
24433
24434
24435
24436
24437
24438
24439
24440
24441
24442
24443
24444
24445
24446
24447
24448
24449
24450
24451
24452
24453
24454
24455
24456
24457
24458
24459
24460
24461
24462
24463
24464
24465
24466
24467
24468
24469
24470
24471
24472
24473
24474
24475
24476
24477
24478
24479
24480
24481
24482
24483
24484
24485
24486
24487
24488
24489
24490
24491
24492
24493
24494
24495
24496
24497
24498
24499
24500
24501
24502
24503
24504
24505
24506
24507
24508
24509
24510
24511
24512
24513
24514
24515
24516
24517
24518
24519
24520
24521
24522
24523
24524
24525
24526
24527
24528
24529
24530
24531
24532
24533
24534
24535
24536
24537
24538
24539
24540
24541
24542
24543
24544
24545
24546
24547
24548
24549
24550
24551
24552
24553
24554
24555
24556
24557
24558
24559
24560
24561
24562
24563
24564
24565
24566
24567
24568
24569
24570
24571
24572
24573
24574
24575
24576
24577
24578
24579
24580
24581
24582
24583
24584
24585
24586
24587
24588
24589
24590
24591
24592
24593
24594
24595
24596
24597
24598
24599
24600
24601
24602
24603
24604
24605
24606
24607
24608
24609
24610
24611
24612
24613
24614
24615
24616
24617
24618
24619
24620
24621
24622
24623
24624
24625
24626
24627
24628
24629
24630
24631
24632
24633
24634
24635
24636
24637
24638
24639
24640
24641
24642
24643
24644
24645
24646
24647
24648
24649
24650
24651
24652
24653
24654
24655
24656
24657
24658
24659
24660
24661
24662
24663
24664
24665
24666
24667
24668
24669
24670
24671
24672
24673
24674
24675
24676
24677
24678
24679
24680
24681
24682
24683
24684
24685
24686
24687
24688
24689
24690
24691
24692
24693
24694
24695
24696
24697
24698
24699
24700
24701
24702
24703
24704
24705
24706
24707
24708
24709
24710
24711
24712
24713
24714
24715
24716
24717
24718
24719
24720
24721
24722
24723
24724
24725
24726
24727
24728
24729
24730
24731
24732
24733
24734
24735
24736
24737
24738
24739
24740
24741
24742
24743
24744
24745
24746
24747
24748
24749
24750
24751
24752
24753
24754
24755
24756
24757
24758
24759
24760
24761
24762
24763
24764
24765
24766
24767
24768
24769
24770
24771
24772
24773
24774
24775
24776
24777
24778
24779
24780
24781
24782
24783
24784
24785
24786
24787
24788
24789
24790
24791
24792
24793
24794
24795
24796
24797
24798
24799
24800
24801
24802
24803
24804
24805
24806
24807
24808
24809
24810
24811
24812
24813
24814
24815
24816
24817
24818
24819
24820
24821
24822
24823
24824
24825
24826
24827
24828
24829
24830
24831
24832
24833
24834
24835
24836
24837
24838
24839
24840
24841
24842
24843
24844
24845
24846
24847
24848
24849
24850
24851
24852
24853
24854
24855
24856
24857
24858
24859
24860
24861
24862
24863
24864
24865
24866
24867
24868
24869
24870
24871
24872
24873
24874
24875
24876
24877
24878
24879
24880
24881
24882
24883
24884
24885
24886
24887
24888
24889
24890
24891
24892
24893
24894
24895
24896
24897
24898
24899
24900
24901
24902
24903
24904
24905
24906
24907
24908
24909
24910
24911
24912
24913
24914
24915
24916
24917
24918
24919
24920
24921
24922
24923
24924
24925
24926
24927
24928
24929
24930
24931
24932
24933
24934
24935
24936
24937
24938
24939
24940
24941
24942
24943
24944
24945
24946
24947
24948
24949
24950
24951
24952
24953
24954
24955
24956
24957
24958
24959
24960
24961
24962
24963
24964
24965
24966
24967
24968
24969
24970
24971
24972
24973
24974
24975
24976
24977
24978
24979
24980
24981
24982
24983
24984
24985
24986
24987
24988
24989
24990
24991
24992
24993
24994
24995
24996
24997
24998
24999
25000
25001
25002
25003
25004
25005
25006
25007
25008
25009
25010
25011
25012
25013
25014
25015
25016
25017
25018
25019
25020
25021
25022
25023
25024
25025
25026
25027
25028
25029
25030
25031
25032
25033
25034
25035
25036
25037
25038
25039
25040
25041
25042
25043
25044
25045
25046
25047
25048
25049
25050
25051
25052
25053
25054
25055
25056
25057
25058
25059
25060
25061
25062
25063
25064
25065
25066
25067
25068
25069
25070
25071
25072
25073
25074
25075
25076
25077
25078
25079
25080
25081
25082
25083
25084
25085
25086
25087
25088
25089
25090
25091
25092
25093
25094
25095
25096
25097
25098
25099
25100
25101
25102
25103
25104
25105
25106
25107
25108
25109
25110
25111
25112
25113
25114
25115
25116
25117
25118
25119
25120
25121
25122
25123
25124
25125
25126
25127
25128
25129
25130
25131
25132
25133
25134
25135
25136
25137
25138
25139
25140
25141
25142
25143
25144
25145
25146
25147
25148
25149
25150
25151
25152
25153
25154
25155
25156
25157
25158
25159
25160
25161
25162
25163
25164
25165
25166
25167
25168
25169
25170
25171
25172
25173
25174
25175
25176
25177
25178
25179
25180
25181
25182
25183
25184
25185
25186
25187
25188
25189
25190
25191
25192
25193
25194
25195
25196
25197
25198
25199
25200
25201
25202
25203
25204
25205
25206
25207
25208
25209
25210
25211
25212
25213
25214
25215
25216
25217
25218
25219
25220
25221
25222
25223
25224
25225
25226
25227
25228
25229
25230
25231
25232
25233
25234
25235
25236
25237
25238
25239
25240
25241
25242
25243
25244
25245
25246
25247
25248
25249
25250
25251
25252
25253
25254
25255
25256
25257
25258
25259
25260
25261
25262
25263
25264
25265
25266
25267
25268
25269
25270
25271
25272
25273
25274
25275
25276
25277
25278
25279
25280
25281
25282
25283
25284
25285
25286
25287
25288
25289
25290
25291
25292
25293
25294
25295
25296
25297
25298
25299
25300
25301
25302
25303
25304
25305
25306
25307
25308
25309
25310
25311
25312
25313
25314
25315
25316
25317
25318
25319
25320
25321
25322
25323
25324
25325
25326
25327
25328
25329
25330
25331
25332
25333
25334
25335
25336
25337
25338
25339
25340
25341
25342
25343
25344
25345
25346
25347
25348
25349
25350
25351
25352
25353
25354
25355
25356
25357
25358
25359
25360
25361
25362
25363
25364
25365
25366
25367
25368
25369
25370
25371
25372
25373
25374
25375
25376
25377
25378
25379
25380
25381
25382
25383
25384
25385
25386
25387
25388
25389
25390
25391
25392
25393
25394
25395
25396
25397
25398
25399
25400
25401
25402
25403
25404
25405
25406
25407
25408
25409
25410
25411
25412
25413
25414
25415
25416
25417
25418
25419
25420
25421
25422
25423
25424
25425
25426
25427
25428
25429
25430
25431
25432
25433
25434
25435
25436
25437
25438
25439
25440
25441
25442
25443
25444
25445
25446
25447
25448
25449
25450
25451
25452
25453
25454
25455
25456
25457
25458
25459
25460
25461
25462
25463
25464
25465
25466
25467
25468
25469
25470
25471
25472
25473
25474
25475
25476
25477
25478
25479
25480
25481
25482
25483
25484
25485
25486
25487
25488
25489
25490
25491
25492
25493
25494
25495
25496
25497
25498
25499
25500
25501
25502
25503
25504
25505
25506
25507
25508
25509
25510
25511
25512
25513
25514
25515
25516
25517
25518
25519
25520
25521
25522
25523
25524
25525
25526
25527
25528
25529
25530
25531
25532
25533
25534
25535
25536
25537
25538
25539
25540
25541
25542
25543
25544
25545
25546
25547
25548
25549
25550
25551
25552
25553
25554
25555
25556
25557
25558
25559
25560
25561
25562
25563
25564
25565
25566
25567
25568
25569
25570
25571
25572
25573
25574
25575
25576
25577
25578
25579
25580
25581
25582
25583
25584
25585
25586
25587
25588
25589
25590
25591
25592
25593
25594
25595
25596
25597
25598
25599
25600
25601
25602
25603
25604
25605
25606
25607
25608
25609
25610
25611
25612
25613
25614
25615
25616
25617
25618
25619
25620
25621
25622
25623
25624
25625
25626
25627
25628
25629
25630
25631
25632
25633
25634
25635
25636
25637
25638
25639
25640
25641
25642
25643
25644
25645
25646
25647
25648
25649
25650
25651
25652
25653
25654
25655
25656
25657
25658
25659
25660
25661
25662
25663
25664
25665
25666
25667
25668
25669
25670
25671
25672
25673
25674
25675
25676
25677
25678
25679
25680
25681
25682
25683
25684
25685
25686
25687
25688
25689
25690
25691
25692
25693
25694
25695
25696
25697
25698
25699
25700
25701
25702
25703
25704
25705
25706
25707
25708
25709
25710
25711
25712
25713
25714
25715
25716
25717
25718
25719
25720
25721
25722
25723
25724
25725
25726
25727
25728
25729
25730
25731
25732
25733
25734
25735
25736
25737
25738
25739
25740
25741
25742
25743
25744
25745
25746
25747
25748
25749
25750
25751
25752
25753
25754
25755
25756
25757
25758
25759
25760
25761
25762
25763
25764
25765
25766
25767
25768
25769
25770
25771
25772
25773
25774
25775
25776
25777
25778
25779
25780
25781
25782
25783
25784
25785
25786
25787
25788
25789
25790
25791
25792
25793
25794
25795
25796
25797
25798
25799
25800
25801
25802
25803
25804
25805
25806
25807
25808
25809
25810
25811
25812
25813
25814
25815
25816
25817
25818
25819
25820
25821
25822
25823
25824
25825
25826
25827
25828
25829
25830
25831
25832
25833
25834
25835
25836
25837
25838
25839
25840
25841
25842
25843
25844
25845
25846
25847
25848
25849
25850
25851
25852
25853
25854
25855
25856
25857
25858
25859
25860
25861
25862
25863
25864
25865
25866
25867
25868
25869
25870
25871
25872
25873
25874
25875
25876
25877
25878
25879
25880
25881
25882
25883
25884
25885
25886
25887
25888
25889
25890
25891
25892
25893
25894
25895
25896
25897
25898
25899
25900
25901
25902
25903
25904
25905
25906
25907
25908
25909
25910
25911
25912
25913
25914
25915
25916
25917
25918
25919
25920
25921
25922
25923
25924
25925
25926
25927
25928
25929
25930
25931
25932
25933
25934
25935
25936
25937
25938
25939
25940
25941
25942
25943
25944
25945
25946
25947
25948
25949
25950
25951
25952
25953
25954
25955
25956
25957
25958
25959
25960
25961
25962
25963
25964
25965
25966
25967
25968
25969
25970
25971
25972
25973
25974
25975
25976
25977
25978
25979
25980
25981
25982
25983
25984
25985
25986
25987
25988
25989
25990
25991
25992
25993
25994
25995
25996
25997
25998
25999
26000
26001
26002
26003
26004
26005
26006
26007
26008
26009
26010
26011
26012
26013
26014
26015
26016
26017
26018
26019
26020
26021
26022
26023
26024
26025
26026
26027
26028
26029
26030
26031
26032
26033
26034
26035
26036
26037
26038
26039
26040
26041
26042
26043
26044
26045
26046
26047
26048
26049
26050
26051
26052
26053
26054
26055
26056
26057
26058
26059
26060
26061
26062
26063
26064
26065
26066
26067
26068
26069
26070
26071
26072
26073
26074
26075
26076
26077
26078
26079
26080
26081
26082
26083
26084
26085
26086
26087
26088
26089
26090
26091
26092
26093
26094
26095
26096
26097
26098
26099
26100
26101
26102
26103
26104
26105
26106
26107
26108
26109
26110
26111
26112
26113
26114
26115
26116
26117
26118
26119
26120
26121
26122
26123
26124
26125
26126
26127
26128
26129
26130
26131
26132
26133
26134
26135
26136
26137
26138
26139
26140
26141
26142
26143
26144
26145
26146
26147
26148
26149
26150
26151
26152
26153
26154
26155
26156
26157
26158
26159
26160
26161
26162
26163
26164
26165
26166
26167
26168
26169
26170
26171
26172
26173
26174
26175
26176
26177
26178
26179
26180
26181
26182
26183
26184
26185
26186
26187
26188
26189
26190
26191
26192
26193
26194
26195
26196
26197
26198
26199
26200
26201
26202
26203
26204
26205
26206
26207
26208
26209
26210
26211
26212
26213
26214
26215
26216
26217
26218
26219
26220
26221
26222
26223
26224
26225
26226
26227
26228
26229
26230
26231
26232
26233
26234
26235
26236
26237
26238
26239
26240
26241
26242
26243
26244
26245
26246
26247
26248
26249
26250
26251
26252
26253
26254
26255
26256
26257
26258
26259
26260
26261
26262
26263
26264
26265
26266
26267
26268
26269
26270
26271
26272
26273
26274
26275
26276
26277
26278
26279
26280
26281
26282
26283
26284
26285
26286
26287
26288
26289
26290
26291
26292
26293
26294
26295
26296
26297
26298
26299
26300
26301
26302
26303
26304
26305
26306
26307
26308
26309
26310
26311
26312
26313
26314
26315
26316
26317
26318
26319
26320
26321
26322
26323
26324
26325
26326
26327
26328
26329
26330
26331
26332
26333
26334
26335
26336
26337
26338
26339
26340
26341
26342
26343
26344
26345
26346
26347
26348
26349
26350
26351
26352
26353
26354
26355
26356
26357
26358
26359
26360
26361
26362
26363
26364
26365
26366
26367
26368
26369
26370
26371
26372
26373
26374
26375
26376
26377
26378
26379
26380
26381
26382
26383
26384
26385
26386
26387
26388
26389
26390
26391
26392
26393
26394
26395
26396
26397
26398
26399
26400
26401
26402
26403
26404
26405
26406
26407
26408
26409
26410
26411
26412
26413
26414
26415
26416
26417
26418
26419
26420
26421
26422
26423
26424
26425
26426
26427
26428
26429
26430
26431
26432
26433
26434
26435
26436
26437
26438
26439
26440
26441
26442
26443
26444
26445
26446
26447
26448
26449
26450
26451
26452
26453
26454
26455
26456
26457
26458
26459
26460
26461
26462
26463
26464
26465
26466
26467
26468
26469
26470
26471
26472
26473
26474
26475
26476
26477
26478
26479
26480
26481
26482
26483
26484
26485
26486
26487
26488
26489
26490
26491
26492
26493
26494
26495
26496
26497
26498
26499
26500
26501
26502
26503
26504
26505
26506
26507
26508
26509
26510
26511
26512
26513
26514
26515
26516
26517
26518
26519
26520
26521
26522
26523
26524
26525
26526
26527
26528
26529
26530
26531
26532
26533
26534
26535
26536
26537
26538
26539
26540
26541
26542
26543
26544
26545
26546
26547
26548
26549
26550
26551
26552
26553
26554
26555
26556
26557
26558
26559
26560
26561
26562
26563
26564
26565
26566
26567
26568
26569
26570
26571
26572
26573
26574
26575
26576
26577
26578
26579
26580
26581
26582
26583
26584
26585
26586
26587
26588
26589
26590
26591
26592
26593
26594
26595
26596
26597
26598
26599
26600
26601
26602
26603
26604
26605
26606
26607
26608
26609
26610
26611
26612
26613
26614
26615
26616
26617
26618
26619
26620
26621
26622
26623
26624
26625
26626
26627
26628
26629
26630
26631
26632
26633
26634
26635
26636
26637
26638
26639
26640
26641
26642
26643
26644
26645
26646
26647
26648
26649
26650
26651
26652
26653
26654
26655
26656
26657
26658
26659
26660
26661
26662
26663
26664
26665
26666
26667
26668
26669
26670
26671
26672
26673
26674
26675
26676
26677
26678
26679
26680
26681
26682
26683
26684
26685
26686
26687
26688
26689
26690
26691
26692
26693
26694
26695
26696
26697
26698
26699
26700
26701
26702
26703
26704
26705
26706
26707
26708
26709
26710
26711
26712
26713
26714
26715
26716
26717
26718
26719
26720
26721
26722
26723
26724
26725
26726
26727
26728
26729
26730
26731
26732
26733
26734
26735
26736
26737
26738
26739
26740
26741
26742
26743
26744
26745
26746
26747
26748
26749
26750
26751
26752
26753
26754
26755
26756
26757
26758
26759
26760
26761
26762
26763
26764
26765
26766
26767
26768
26769
26770
26771
26772
26773
26774
26775
26776
26777
26778
26779
26780
26781
26782
26783
26784
26785
26786
26787
26788
26789
26790
26791
26792
26793
26794
26795
26796
26797
26798
26799
26800
26801
26802
26803
26804
26805
26806
26807
26808
26809
26810
26811
26812
26813
26814
26815
26816
26817
26818
26819
26820
26821
26822
26823
26824
26825
26826
26827
26828
26829
26830
26831
26832
26833
26834
26835
26836
26837
26838
26839
26840
26841
26842
26843
26844
26845
26846
26847
26848
26849
26850
26851
26852
26853
26854
26855
26856
26857
26858
26859
26860
26861
26862
26863
26864
26865
26866
26867
26868
26869
26870
26871
26872
26873
26874
26875
26876
26877
26878
26879
26880
26881
26882
26883
26884
26885
26886
26887
26888
26889
26890
26891
26892
26893
26894
26895
26896
26897
26898
26899
26900
26901
26902
26903
26904
26905
26906
26907
26908
26909
26910
26911
26912
26913
26914
26915
26916
26917
26918
26919
26920
26921
26922
26923
26924
26925
26926
26927
26928
26929
26930
26931
26932
26933
26934
26935
26936
26937
26938
26939
26940
26941
26942
26943
26944
26945
26946
26947
26948
26949
26950
26951
26952
26953
26954
26955
26956
26957
26958
26959
26960
26961
26962
26963
26964
26965
26966
26967
26968
26969
26970
26971
26972
26973
26974
26975
26976
26977
26978
26979
26980
26981
26982
26983
26984
26985
26986
26987
26988
26989
26990
26991
26992
26993
26994
26995
26996
26997
26998
26999
27000
27001
27002
27003
27004
27005
27006
27007
27008
27009
27010
27011
27012
27013
27014
27015
27016
27017
27018
27019
27020
27021
27022
27023
27024
27025
27026
27027
27028
27029
27030
27031
27032
27033
27034
27035
27036
27037
27038
27039
27040
27041
27042
27043
27044
27045
27046
27047
27048
27049
27050
27051
27052
27053
27054
27055
27056
27057
27058
27059
27060
27061
27062
27063
27064
27065
27066
27067
27068
27069
27070
27071
27072
27073
27074
27075
27076
27077
27078
27079
27080
27081
27082
27083
27084
27085
27086
27087
27088
27089
27090
27091
27092
27093
27094
27095
27096
27097
27098
27099
27100
27101
27102
27103
27104
27105
27106
27107
27108
27109
27110
27111
27112
27113
27114
27115
27116
27117
27118
27119
27120
27121
27122
27123
27124
27125
27126
27127
27128
27129
27130
27131
27132
27133
27134
27135
27136
27137
27138
27139
27140
27141
27142
27143
27144
27145
27146
27147
27148
27149
27150
27151
27152
27153
27154
27155
27156
27157
27158
27159
27160
27161
27162
27163
27164
27165
27166
27167
27168
27169
27170
27171
27172
27173
27174
27175
27176
27177
27178
27179
27180
27181
27182
27183
27184
27185
27186
27187
27188
27189
27190
27191
27192
27193
27194
27195
27196
27197
27198
27199
27200
27201
27202
27203
27204
27205
27206
27207
27208
27209
27210
27211
27212
27213
27214
27215
27216
27217
27218
27219
27220
27221
27222
27223
27224
27225
27226
27227
27228
27229
27230
27231
27232
27233
27234
27235
27236
27237
27238
27239
27240
27241
27242
27243
27244
27245
27246
27247
27248
27249
27250
27251
27252
27253
27254
27255
27256
27257
27258
27259
27260
27261
27262
27263
27264
27265
27266
27267
27268
27269
27270
27271
27272
27273
27274
27275
27276
27277
27278
27279
27280
27281
27282
27283
27284
27285
27286
27287
27288
27289
27290
27291
27292
27293
27294
27295
27296
27297
27298
27299
27300
27301
27302
27303
27304
27305
27306
27307
27308
27309
27310
27311
27312
27313
27314
27315
27316
27317
27318
27319
27320
27321
27322
27323
27324
27325
27326
27327
27328
27329
27330
27331
27332
27333
27334
27335
27336
27337
27338
27339
27340
27341
27342
27343
27344
27345
27346
27347
27348
27349
27350
27351
27352
27353
27354
27355
27356
27357
27358
27359
27360
27361
27362
27363
27364
27365
27366
27367
27368
27369
27370
27371
27372
27373
27374
27375
27376
27377
27378
27379
27380
27381
27382
27383
27384
27385
27386
27387
27388
27389
27390
27391
27392
27393
27394
27395
27396
27397
27398
27399
27400
27401
27402
27403
27404
27405
27406
27407
27408
27409
27410
27411
27412
27413
27414
27415
27416
27417
27418
27419
27420
27421
27422
27423
27424
27425
27426
27427
27428
27429
27430
27431
27432
27433
27434
27435
27436
27437
27438
27439
27440
27441
27442
27443
27444
27445
27446
27447
27448
27449
27450
27451
27452
27453
27454
27455
27456
27457
27458
27459
27460
27461
27462
27463
27464
27465
27466
27467
27468
27469
27470
27471
27472
27473
27474
27475
27476
27477
27478
27479
27480
27481
27482
27483
27484
27485
27486
27487
27488
27489
27490
27491
27492
27493
27494
27495
27496
27497
27498
27499
27500
27501
27502
27503
27504
27505
27506
27507
27508
27509
27510
27511
27512
27513
27514
27515
27516
27517
27518
27519
27520
27521
27522
27523
27524
27525
27526
27527
27528
27529
27530
27531
27532
27533
27534
27535
27536
27537
27538
27539
27540
27541
27542
27543
27544
27545
27546
27547
27548
27549
27550
27551
27552
27553
27554
27555
27556
27557
27558
27559
27560
27561
27562
27563
27564
27565
27566
27567
27568
27569
27570
27571
27572
27573
27574
27575
27576
27577
27578
27579
27580
27581
27582
27583
27584
27585
27586
27587
27588
27589
27590
27591
27592
27593
27594
27595
27596
27597
27598
27599
27600
27601
27602
27603
27604
27605
27606
27607
27608
27609
27610
27611
27612
27613
27614
27615
27616
27617
27618
27619
27620
27621
27622
27623
27624
27625
27626
27627
27628
27629
27630
27631
27632
27633
27634
27635
27636
27637
27638
27639
27640
27641
27642
27643
27644
27645
27646
27647
27648
27649
27650
27651
27652
27653
27654
27655
27656
27657
27658
27659
27660
27661
27662
27663
27664
27665
27666
27667
27668
27669
27670
27671
27672
27673
27674
27675
27676
27677
27678
27679
27680
27681
27682
27683
27684
27685
27686
27687
27688
27689
27690
27691
27692
27693
27694
27695
27696
27697
27698
27699
27700
27701
27702
27703
27704
27705
27706
27707
27708
27709
27710
27711
27712
27713
27714
27715
27716
27717
27718
27719
27720
27721
27722
27723
27724
27725
27726
27727
27728
27729
27730
27731
27732
27733
27734
27735
27736
27737
27738
27739
27740
27741
27742
27743
27744
27745
27746
27747
27748
27749
27750
27751
27752
27753
27754
27755
27756
27757
27758
27759
27760
27761
27762
27763
27764
27765
27766
27767
27768
27769
27770
27771
27772
27773
27774
27775
27776
27777
27778
27779
27780
27781
27782
27783
27784
27785
27786
27787
27788
27789
27790
27791
27792
27793
27794
27795
27796
27797
27798
27799
27800
27801
27802
27803
27804
27805
27806
27807
27808
27809
27810
27811
27812
27813
27814
27815
27816
27817
27818
27819
27820
27821
27822
27823
27824
27825
27826
27827
27828
27829
27830
27831
27832
27833
27834
27835
27836
27837
27838
27839
27840
27841
27842
27843
27844
27845
27846
27847
27848
27849
27850
27851
27852
27853
27854
27855
27856
27857
27858
27859
27860
27861
27862
27863
27864
27865
27866
27867
27868
27869
27870
27871
27872
27873
27874
27875
27876
27877
27878
27879
27880
27881
27882
27883
27884
27885
27886
27887
27888
27889
27890
27891
27892
27893
27894
27895
27896
27897
27898
27899
27900
27901
27902
27903
27904
27905
27906
27907
27908
27909
27910
27911
27912
27913
27914
27915
27916
27917
27918
27919
27920
27921
27922
27923
27924
27925
27926
27927
27928
27929
27930
27931
27932
27933
27934
27935
27936
27937
27938
27939
27940
27941
27942
27943
27944
27945
27946
27947
27948
27949
27950
27951
27952
27953
27954
27955
27956
27957
27958
27959
27960
27961
27962
27963
27964
27965
27966
27967
27968
27969
27970
27971
27972
27973
27974
27975
27976
27977
27978
27979
27980
27981
27982
27983
27984
27985
27986
27987
27988
27989
27990
27991
27992
27993
27994
27995
27996
27997
27998
27999
28000
28001
28002
28003
28004
28005
28006
28007
28008
28009
28010
28011
28012
28013
28014
28015
28016
28017
28018
28019
28020
28021
28022
28023
28024
28025
28026
28027
28028
28029
28030
28031
28032
28033
28034
28035
28036
28037
28038
28039
28040
28041
28042
28043
28044
28045
28046
28047
28048
28049
28050
28051
28052
28053
28054
28055
28056
28057
28058
28059
28060
28061
28062
28063
28064
28065
28066
28067
28068
28069
28070
28071
28072
28073
28074
28075
28076
28077
28078
28079
28080
28081
28082
28083
28084
28085
28086
28087
28088
28089
28090
28091
28092
28093
28094
28095
28096
28097
28098
28099
28100
28101
28102
28103
28104
28105
28106
28107
28108
28109
28110
28111
28112
28113
28114
28115
28116
28117
28118
28119
28120
28121
28122
28123
28124
28125
28126
28127
28128
28129
28130
28131
28132
28133
28134
28135
28136
28137
28138
28139
28140
28141
28142
28143
28144
28145
28146
28147
28148
28149
28150
28151
28152
28153
28154
28155
28156
28157
28158
28159
28160
28161
28162
28163
28164
28165
28166
28167
28168
28169
28170
28171
28172
28173
28174
28175
28176
28177
28178
28179
28180
28181
28182
28183
28184
28185
28186
28187
28188
28189
28190
28191
28192
28193
28194
28195
28196
28197
28198
28199
28200
28201
28202
28203
28204
28205
28206
28207
28208
28209
28210
28211
28212
28213
28214
28215
28216
28217
28218
28219
28220
28221
28222
28223
28224
28225
28226
28227
28228
28229
28230
28231
28232
28233
28234
28235
28236
28237
28238
28239
28240
28241
28242
28243
28244
28245
28246
28247
28248
28249
28250
28251
28252
28253
28254
28255
28256
28257
28258
28259
28260
28261
28262
28263
28264
28265
28266
28267
28268
28269
28270
28271
28272
28273
28274
28275
28276
28277
28278
28279
28280
28281
28282
28283
28284
28285
28286
28287
28288
28289
28290
28291
28292
28293
28294
28295
28296
28297
28298
28299
28300
28301
28302
28303
28304
28305
28306
28307
28308
28309
28310
28311
28312
28313
28314
28315
28316
28317
28318
28319
28320
28321
28322
28323
28324
28325
28326
28327
28328
28329
28330
28331
28332
28333
28334
28335
28336
28337
28338
28339
28340
28341
28342
28343
28344
28345
28346
28347
28348
28349
28350
28351
28352
28353
28354
28355
28356
28357
28358
28359
28360
28361
28362
28363
28364
28365
28366
28367
28368
28369
28370
28371
28372
28373
28374
28375
28376
28377
28378
28379
28380
28381
28382
28383
28384
28385
28386
28387
28388
28389
28390
28391
28392
28393
28394
28395
28396
28397
28398
28399
28400
28401
28402
28403
28404
28405
28406
28407
28408
28409
28410
28411
28412
28413
28414
28415
28416
28417
28418
28419
28420
28421
28422
28423
28424
28425
28426
28427
28428
28429
28430
28431
28432
28433
28434
28435
28436
28437
28438
28439
28440
28441
28442
28443
28444
28445
28446
28447
28448
28449
28450
28451
28452
28453
28454
28455
28456
28457
28458
28459
28460
28461
28462
28463
28464
28465
28466
28467
28468
28469
28470
28471
28472
28473
28474
28475
28476
28477
28478
28479
28480
28481
28482
28483
28484
28485
28486
28487
28488
28489
28490
28491
28492
28493
28494
28495
28496
28497
28498
28499
28500
28501
28502
28503
28504
28505
28506
28507
28508
28509
28510
28511
28512
28513
28514
28515
28516
28517
28518
28519
28520
28521
28522
28523
28524
28525
28526
28527
28528
28529
28530
28531
28532
28533
28534
28535
28536
28537
28538
28539
28540
28541
28542
28543
28544
28545
28546
28547
28548
28549
28550
28551
28552
28553
28554
28555
28556
28557
28558
28559
28560
28561
28562
28563
28564
28565
28566
28567
28568
28569
28570
28571
28572
28573
28574
28575
28576
28577
28578
28579
28580
28581
28582
28583
28584
28585
28586
28587
28588
28589
28590
28591
28592
28593
28594
28595
28596
28597
28598
28599
28600
28601
28602
28603
28604
28605
28606
28607
28608
28609
28610
28611
28612
28613
28614
28615
28616
28617
28618
28619
28620
28621
28622
28623
28624
28625
28626
28627
28628
28629
28630
28631
28632
28633
28634
28635
28636
28637
28638
28639
28640
28641
28642
28643
28644
28645
28646
28647
28648
28649
28650
28651
28652
28653
28654
28655
28656
28657
28658
28659
28660
28661
28662
28663
28664
28665
28666
28667
28668
28669
28670
28671
28672
28673
28674
28675
28676
28677
28678
28679
28680
28681
28682
28683
28684
28685
28686
28687
28688
28689
28690
28691
28692
28693
28694
28695
28696
28697
28698
28699
28700
28701
28702
28703
28704
28705
28706
28707
28708
28709
28710
28711
28712
28713
28714
28715
28716
28717
28718
28719
28720
28721
28722
28723
28724
28725
28726
28727
28728
28729
28730
28731
28732
28733
28734
28735
28736
28737
28738
28739
28740
28741
28742
28743
28744
28745
28746
28747
28748
28749
28750
28751
28752
28753
28754
28755
28756
28757
28758
28759
28760
28761
28762
28763
28764
28765
28766
28767
28768
28769
28770
28771
28772
28773
28774
28775
28776
28777
28778
28779
28780
28781
28782
28783
28784
28785
28786
28787
28788
28789
28790
28791
28792
28793
28794
28795
28796
28797
28798
28799
28800
28801
28802
28803
28804
28805
28806
28807
28808
28809
28810
28811
28812
28813
28814
28815
28816
28817
28818
28819
28820
28821
28822
28823
28824
28825
28826
28827
28828
28829
28830
28831
28832
28833
28834
28835
28836
28837
28838
28839
28840
28841
28842
28843
28844
28845
28846
28847
28848
28849
28850
28851
28852
28853
28854
28855
28856
28857
28858
28859
28860
28861
28862
28863
28864
28865
28866
28867
28868
28869
28870
28871
28872
28873
28874
28875
28876
28877
28878
28879
28880
28881
28882
28883
28884
28885
28886
28887
28888
28889
28890
28891
28892
28893
28894
28895
28896
28897
28898
28899
28900
28901
28902
28903
28904
28905
28906
28907
28908
28909
28910
28911
28912
28913
28914
28915
28916
28917
28918
28919
28920
28921
28922
28923
28924
28925
28926
28927
28928
28929
28930
28931
28932
28933
28934
28935
28936
28937
28938
28939
28940
28941
28942
28943
28944
28945
28946
28947
28948
28949
28950
28951
28952
28953
28954
28955
28956
28957
28958
28959
28960
28961
28962
28963
28964
28965
28966
28967
28968
28969
28970
28971
28972
28973
28974
28975
28976
28977
28978
28979
28980
28981
28982
28983
28984
28985
28986
28987
28988
28989
28990
28991
28992
28993
28994
28995
28996
28997
28998
28999
29000
29001
29002
29003
29004
29005
29006
29007
29008
29009
29010
29011
29012
29013
29014
29015
29016
29017
29018
29019
29020
29021
29022
29023
29024
29025
29026
29027
29028
29029
29030
29031
29032
29033
29034
29035
29036
29037
29038
29039
29040
29041
29042
29043
29044
29045
29046
29047
29048
29049
29050
29051
29052
29053
29054
29055
29056
29057
29058
29059
29060
29061
29062
29063
29064
29065
29066
29067
29068
29069
29070
29071
29072
29073
29074
29075
29076
29077
29078
29079
29080
29081
29082
29083
29084
29085
29086
29087
29088
29089
29090
29091
29092
29093
29094
29095
29096
29097
29098
29099
29100
29101
29102
29103
29104
29105
29106
29107
29108
29109
29110
29111
29112
29113
29114
29115
29116
29117
29118
29119
29120
29121
29122
29123
29124
29125
29126
29127
29128
29129
29130
29131
29132
29133
29134
29135
29136
29137
29138
29139
29140
29141
29142
29143
29144
29145
29146
29147
29148
29149
29150
29151
29152
29153
29154
29155
29156
29157
29158
29159
29160
29161
29162
29163
29164
29165
29166
29167
29168
29169
29170
29171
29172
29173
29174
29175
29176
29177
29178
29179
29180
29181
29182
29183
29184
29185
29186
29187
29188
29189
29190
29191
29192
29193
29194
29195
29196
29197
29198
29199
29200
29201
29202
29203
29204
29205
29206
29207
29208
29209
29210
29211
29212
29213
29214
29215
29216
29217
29218
29219
29220
29221
29222
29223
29224
29225
29226
29227
29228
29229
29230
29231
29232
29233
29234
29235
29236
29237
29238
29239
29240
29241
29242
29243
29244
29245
29246
29247
29248
29249
29250
29251
29252
29253
29254
29255
29256
29257
29258
29259
29260
29261
29262
29263
29264
29265
29266
29267
29268
29269
29270
29271
29272
29273
29274
29275
29276
29277
29278
29279
29280
29281
29282
29283
29284
29285
29286
29287
29288
29289
29290
29291
29292
29293
29294
29295
29296
29297
29298
29299
29300
29301
29302
29303
29304
29305
29306
29307
29308
29309
29310
29311
29312
29313
29314
29315
29316
29317
29318
29319
29320
29321
29322
29323
29324
29325
29326
29327
29328
29329
29330
29331
29332
29333
29334
29335
29336
29337
29338
29339
29340
29341
29342
29343
29344
29345
29346
29347
29348
29349
29350
29351
29352
29353
29354
29355
29356
29357
29358
29359
29360
29361
29362
29363
29364
29365
29366
29367
29368
29369
29370
29371
29372
29373
29374
29375
29376
29377
29378
29379
29380
29381
29382
29383
29384
29385
29386
29387
29388
29389
29390
29391
29392
29393
29394
29395
29396
29397
29398
29399
29400
29401
29402
29403
29404
29405
29406
29407
29408
29409
29410
29411
29412
29413
29414
29415
29416
29417
29418
29419
29420
29421
29422
29423
29424
29425
29426
29427
29428
29429
29430
29431
29432
29433
29434
29435
29436
29437
29438
29439
29440
29441
29442
29443
29444
29445
29446
29447
29448
29449
29450
29451
29452
29453
29454
29455
29456
29457
29458
29459
29460
29461
29462
29463
29464
29465
29466
29467
29468
29469
29470
29471
29472
29473
29474
29475
29476
29477
29478
29479
29480
29481
29482
29483
29484
29485
29486
29487
29488
29489
29490
29491
29492
29493
29494
29495
29496
29497
29498
29499
29500
29501
29502
29503
29504
29505
29506
29507
29508
29509
29510
29511
29512
29513
29514
29515
29516
29517
29518
29519
29520
29521
29522
29523
29524
29525
29526
29527
29528
29529
29530
29531
29532
29533
29534
29535
29536
29537
29538
29539
29540
29541
29542
29543
29544
29545
29546
29547
29548
29549
29550
29551
29552
29553
29554
29555
29556
29557
29558
29559
29560
29561
29562
29563
29564
29565
29566
29567
29568
29569
29570
29571
29572
29573
29574
29575
29576
29577
29578
29579
29580
29581
29582
29583
29584
29585
29586
29587
29588
29589
29590
29591
29592
29593
29594
29595
29596
29597
29598
29599
29600
29601
29602
29603
29604
29605
29606
29607
29608
29609
29610
29611
29612
29613
29614
29615
29616
29617
29618
29619
29620
29621
29622
29623
29624
29625
29626
29627
29628
29629
29630
29631
29632
29633
29634
29635
29636
29637
29638
29639
29640
29641
29642
29643
29644
29645
29646
29647
29648
29649
29650
29651
29652
29653
29654
29655
29656
29657
29658
29659
29660
29661
29662
29663
29664
29665
29666
29667
29668
29669
29670
29671
29672
29673
29674
29675
29676
29677
29678
29679
29680
29681
29682
29683
29684
29685
29686
29687
29688
29689
29690
29691
29692
29693
29694
29695
29696
29697
29698
29699
29700
29701
29702
29703
29704
29705
29706
29707
29708
29709
29710
29711
29712
29713
29714
29715
29716
29717
29718
29719
29720
29721
29722
29723
29724
29725
29726
29727
29728
29729
29730
29731
29732
29733
29734
29735
29736
29737
29738
29739
29740
29741
29742
29743
29744
29745
29746
29747
29748
29749
29750
29751
29752
29753
29754
29755
29756
29757
29758
29759
29760
29761
29762
29763
29764
29765
29766
29767
29768
29769
29770
29771
29772
29773
29774
29775
29776
29777
29778
29779
29780
29781
29782
29783
29784
29785
29786
29787
29788
29789
29790
29791
29792
29793
29794
29795
29796
29797
29798
29799
29800
29801
29802
29803
29804
29805
29806
29807
29808
29809
29810
29811
29812
29813
29814
29815
29816
29817
29818
29819
29820
29821
29822
29823
29824
29825
29826
29827
29828
29829
29830
29831
29832
29833
29834
29835
29836
29837
29838
29839
29840
29841
29842
29843
29844
29845
29846
29847
29848
29849
29850
29851
29852
29853
29854
29855
29856
29857
29858
29859
29860
29861
29862
29863
29864
29865
29866
29867
29868
29869
29870
29871
29872
29873
29874
29875
29876
29877
29878
29879
29880
29881
29882
29883
29884
29885
29886
29887
29888
29889
29890
29891
29892
29893
29894
29895
29896
29897
29898
29899
29900
29901
29902
29903
29904
29905
29906
29907
29908
29909
29910
29911
29912
29913
29914
29915
29916
29917
29918
29919
29920
29921
29922
29923
29924
29925
29926
29927
29928
29929
29930
29931
29932
29933
29934
29935
29936
29937
29938
29939
29940
29941
29942
29943
29944
29945
29946
29947
29948
29949
29950
29951
29952
29953
29954
29955
29956
29957
29958
29959
29960
29961
29962
29963
29964
29965
29966
29967
29968
29969
29970
29971
29972
29973
29974
29975
29976
29977
29978
29979
29980
29981
29982
29983
29984
29985
29986
29987
29988
29989
29990
29991
29992
29993
29994
29995
29996
29997
29998
29999
30000
30001
30002
30003
30004
30005
30006
30007
30008
30009
30010
30011
30012
30013
30014
30015
30016
30017
30018
30019
30020
30021
30022
30023
30024
30025
30026
30027
30028
30029
30030
30031
30032
30033
30034
30035
30036
30037
30038
30039
30040
30041
30042
30043
30044
30045
30046
30047
30048
30049
30050
30051
30052
30053
30054
30055
30056
30057
30058
30059
30060
30061
30062
30063
30064
30065
30066
30067
30068
30069
30070
30071
30072
30073
30074
30075
30076
30077
30078
30079
30080
30081
30082
30083
30084
30085
30086
30087
30088
30089
30090
30091
30092
30093
30094
30095
30096
30097
30098
30099
30100
30101
30102
30103
30104
30105
30106
30107
30108
30109
30110
30111
30112
30113
30114
30115
30116
30117
30118
30119
30120
30121
30122
30123
30124
30125
30126
30127
30128
30129
30130
30131
30132
30133
30134
30135
30136
30137
30138
30139
30140
30141
30142
30143
30144
30145
30146
30147
30148
30149
30150
30151
30152
30153
30154
30155
30156
30157
30158
30159
30160
30161
30162
30163
30164
30165
30166
30167
30168
30169
30170
30171
30172
30173
30174
30175
30176
30177
30178
30179
30180
30181
30182
30183
30184
30185
30186
30187
30188
30189
30190
30191
30192
30193
30194
30195
30196
30197
30198
30199
30200
30201
30202
30203
30204
30205
30206
30207
30208
30209
30210
30211
30212
30213
30214
30215
30216
30217
30218
30219
30220
30221
30222
30223
30224
30225
30226
30227
30228
30229
30230
30231
30232
30233
30234
30235
30236
30237
30238
30239
30240
30241
30242
30243
30244
30245
30246
30247
30248
30249
30250
30251
30252
30253
30254
30255
30256
30257
30258
30259
30260
30261
30262
30263
30264
30265
30266
30267
30268
30269
30270
30271
30272
30273
30274
30275
30276
30277
30278
30279
30280
30281
30282
30283
30284
30285
30286
30287
30288
30289
30290
30291
30292
30293
30294
30295
30296
30297
30298
30299
30300
30301
30302
30303
30304
30305
30306
30307
30308
30309
30310
30311
30312
30313
30314
30315
30316
30317
30318
30319
30320
30321
30322
30323
30324
30325
30326
30327
30328
30329
30330
30331
30332
30333
30334
30335
30336
30337
30338
30339
30340
30341
30342
30343
30344
30345
30346
30347
30348
30349
30350
30351
30352
30353
30354
30355
30356
30357
30358
30359
30360
30361
30362
30363
30364
30365
30366
30367
30368
30369
30370
30371
30372
30373
30374
30375
30376
30377
30378
30379
30380
30381
30382
30383
30384
30385
30386
30387
30388
30389
30390
30391
30392
30393
30394
30395
30396
30397
30398
30399
30400
30401
30402
30403
30404
30405
30406
30407
30408
30409
30410
30411
30412
30413
30414
30415
30416
30417
30418
30419
30420
30421
30422
30423
30424
30425
30426
30427
30428
30429
30430
30431
30432
30433
30434
30435
30436
30437
30438
30439
30440
30441
30442
30443
30444
30445
30446
30447
30448
30449
30450
30451
30452
30453
30454
30455
30456
30457
30458
30459
30460
30461
30462
30463
30464
30465
30466
30467
30468
30469
30470
30471
30472
30473
30474
30475
30476
30477
30478
30479
30480
30481
30482
30483
30484
30485
30486
30487
30488
30489
30490
30491
30492
30493
30494
30495
30496
30497
30498
30499
30500
30501
30502
30503
30504
30505
30506
30507
30508
30509
30510
30511
30512
30513
30514
30515
30516
30517
30518
30519
30520
30521
30522
30523
30524
30525
30526
30527
30528
30529
30530
30531
30532
30533
30534
30535
30536
30537
30538
30539
30540
30541
30542
30543
30544
30545
30546
30547
30548
30549
30550
30551
30552
30553
30554
30555
30556
30557
30558
30559
30560
30561
30562
30563
30564
30565
30566
30567
30568
30569
30570
30571
30572
30573
30574
30575
30576
30577
30578
30579
30580
30581
30582
30583
30584
30585
30586
30587
30588
30589
30590
30591
30592
30593
30594
30595
30596
30597
30598
30599
30600
30601
30602
30603
30604
30605
30606
30607
30608
30609
30610
30611
30612
30613
30614
30615
30616
30617
30618
30619
30620
30621
30622
30623
30624
30625
30626
30627
30628
30629
30630
30631
30632
30633
30634
30635
30636
30637
30638
30639
30640
30641
30642
30643
30644
30645
30646
30647
30648
30649
30650
30651
30652
30653
30654
30655
30656
30657
30658
30659
30660
30661
30662
30663
30664
30665
30666
30667
30668
30669
30670
30671
30672
30673
30674
30675
30676
30677
30678
30679
30680
30681
30682
30683
30684
30685
30686
30687
30688
30689
30690
30691
30692
30693
30694
30695
30696
30697
30698
30699
30700
30701
30702
30703
30704
30705
30706
30707
30708
30709
30710
30711
30712
30713
30714
30715
30716
30717
30718
30719
30720
30721
30722
30723
30724
30725
30726
30727
30728
30729
30730
30731
30732
30733
30734
30735
30736
30737
30738
30739
30740
30741
30742
30743
30744
30745
30746
30747
30748
30749
30750
30751
30752
30753
30754
30755
30756
30757
30758
30759
30760
30761
30762
30763
30764
30765
30766
30767
30768
30769
30770
30771
30772
30773
30774
30775
30776
30777
30778
30779
30780
30781
30782
30783
30784
30785
30786
30787
30788
30789
30790
30791
30792
30793
30794
30795
30796
30797
30798
30799
30800
30801
30802
30803
30804
30805
30806
30807
30808
30809
30810
30811
30812
30813
30814
30815
30816
30817
30818
30819
30820
30821
30822
30823
30824
30825
30826
30827
30828
30829
30830
30831
30832
30833
30834
30835
30836
30837
30838
30839
30840
30841
30842
30843
30844
30845
30846
30847
30848
30849
30850
30851
30852
30853
30854
30855
30856
30857
30858
30859
30860
30861
30862
30863
30864
30865
30866
30867
30868
30869
30870
30871
30872
30873
30874
30875
30876
30877
30878
30879
30880
30881
30882
30883
30884
30885
30886
30887
30888
30889
30890
30891
30892
30893
30894
30895
30896
30897
30898
30899
30900
30901
30902
30903
30904
30905
30906
30907
30908
30909
30910
30911
30912
30913
30914
30915
30916
30917
30918
30919
30920
30921
30922
30923
30924
30925
30926
30927
30928
30929
30930
30931
30932
30933
30934
30935
30936
30937
30938
30939
30940
30941
30942
30943
30944
30945
30946
30947
30948
30949
30950
30951
30952
30953
30954
30955
30956
30957
30958
30959
30960
30961
30962
30963
30964
30965
30966
30967
30968
30969
30970
30971
30972
30973
30974
30975
30976
30977
30978
30979
30980
30981
30982
30983
30984
30985
30986
30987
30988
30989
30990
30991
30992
30993
30994
30995
30996
30997
30998
30999
31000
31001
31002
31003
31004
31005
31006
31007
31008
31009
31010
31011
31012
31013
31014
31015
31016
31017
31018
31019
31020
31021
31022
31023
31024
31025
31026
31027
31028
31029
31030
31031
31032
31033
31034
31035
31036
31037
31038
31039
31040
31041
31042
31043
31044
31045
31046
31047
31048
31049
31050
31051
31052
31053
31054
31055
31056
31057
31058
31059
31060
31061
31062
31063
31064
31065
31066
31067
31068
31069
31070
31071
31072
31073
31074
31075
31076
31077
31078
31079
31080
31081
31082
31083
31084
31085
31086
31087
31088
31089
31090
31091
31092
31093
31094
31095
31096
31097
31098
31099
31100
31101
31102
31103
31104
31105
31106
31107
31108
31109
31110
31111
31112
31113
31114
31115
31116
31117
31118
31119
31120
31121
31122
31123
31124
31125
31126
31127
31128
31129
31130
31131
31132
31133
31134
31135
31136
31137
31138
31139
31140
31141
31142
31143
31144
31145
31146
31147
31148
31149
31150
31151
31152
31153
31154
31155
31156
31157
31158
31159
31160
31161
31162
31163
31164
31165
31166
31167
31168
31169
31170
31171
31172
31173
31174
31175
31176
31177
31178
31179
31180
31181
31182
31183
31184
31185
31186
31187
31188
31189
31190
31191
31192
31193
31194
31195
31196
31197
31198
31199
31200
31201
31202
31203
31204
31205
31206
31207
31208
31209
31210
31211
31212
31213
31214
31215
31216
31217
31218
31219
31220
31221
31222
31223
31224
31225
31226
31227
31228
31229
31230
31231
31232
31233
31234
31235
31236
31237
31238
31239
31240
31241
31242
31243
31244
31245
31246
31247
31248
31249
31250
31251
31252
31253
31254
31255
31256
31257
31258
31259
31260
31261
31262
31263
31264
31265
31266
31267
31268
31269
31270
31271
31272
31273
31274
31275
31276
31277
31278
31279
31280
31281
31282
31283
31284
31285
31286
31287
31288
31289
31290
31291
31292
31293
31294
31295
31296
31297
31298
31299
31300
31301
31302
31303
31304
31305
31306
31307
31308
31309
31310
31311
31312
31313
31314
31315
31316
31317
31318
31319
31320
31321
31322
31323
31324
31325
31326
31327
31328
31329
31330
31331
31332
31333
31334
31335
31336
31337
31338
31339
31340
31341
31342
31343
31344
31345
31346
31347
31348
31349
31350
31351
31352
31353
31354
31355
31356
31357
31358
31359
31360
31361
31362
31363
31364
31365
31366
31367
31368
31369
31370
31371
31372
31373
31374
31375
31376
31377
31378
31379
31380
31381
31382
31383
31384
31385
31386
31387
31388
31389
31390
31391
31392
31393
31394
31395
31396
31397
31398
31399
31400
31401
31402
31403
31404
31405
31406
31407
31408
31409
31410
31411
31412
31413
31414
31415
31416
31417
31418
31419
31420
31421
31422
31423
31424
31425
31426
31427
31428
31429
31430
31431
31432
31433
31434
31435
31436
31437
31438
31439
31440
31441
31442
31443
31444
31445
31446
31447
31448
31449
31450
31451
31452
31453
31454
31455
31456
31457
31458
31459
31460
31461
31462
31463
31464
31465
31466
31467
31468
31469
31470
31471
31472
31473
31474
31475
31476
31477
31478
31479
31480
31481
31482
31483
31484
31485
31486
31487
31488
31489
31490
31491
31492
31493
31494
31495
31496
31497
31498
31499
31500
31501
31502
31503
31504
31505
31506
31507
31508
31509
31510
31511
31512
31513
31514
31515
31516
31517
31518
31519
31520
31521
31522
31523
31524
31525
31526
31527
31528
31529
31530
31531
31532
31533
31534
31535
31536
31537
31538
31539
31540
31541
31542
31543
31544
31545
31546
31547
31548
31549
31550
31551
31552
31553
31554
31555
31556
31557
31558
31559
31560
31561
31562
31563
31564
31565
31566
31567
31568
31569
31570
31571
31572
31573
31574
31575
31576
31577
31578
31579
31580
31581
31582
31583
31584
31585
31586
31587
31588
31589
31590
31591
31592
31593
31594
31595
31596
31597
31598
31599
31600
31601
31602
31603
31604
31605
31606
31607
31608
31609
31610
31611
31612
31613
31614
31615
31616
31617
31618
31619
31620
31621
31622
31623
31624
31625
31626
31627
31628
31629
31630
31631
31632
31633
31634
31635
31636
31637
31638
31639
31640
31641
31642
31643
31644
31645
31646
31647
31648
31649
31650
31651
31652
31653
31654
31655
31656
31657
31658
31659
31660
31661
31662
31663
31664
31665
31666
31667
31668
31669
31670
31671
31672
31673
31674
31675
31676
31677
31678
31679
31680
31681
31682
31683
31684
31685
31686
31687
31688
31689
31690
31691
31692
31693
31694
31695
31696
31697
31698
31699
31700
31701
31702
31703
31704
31705
31706
31707
31708
31709
31710
31711
31712
31713
31714
31715
31716
31717
31718
31719
31720
31721
31722
31723
31724
31725
31726
31727
31728
31729
31730
31731
31732
31733
31734
31735
31736
31737
31738
31739
31740
31741
31742
31743
31744
31745
31746
31747
31748
31749
31750
31751
31752
31753
31754
31755
31756
31757
31758
31759
31760
31761
31762
31763
31764
31765
31766
31767
31768
31769
31770
31771
31772
31773
31774
31775
31776
31777
31778
31779
31780
31781
31782
31783
31784
31785
31786
31787
31788
31789
31790
31791
31792
31793
31794
31795
31796
31797
31798
31799
31800
31801
31802
31803
31804
31805
31806
31807
31808
31809
31810
31811
31812
31813
31814
31815
31816
31817
31818
31819
31820
31821
31822
31823
31824
31825
31826
31827
31828
31829
31830
31831
31832
31833
31834
31835
31836
31837
31838
31839
31840
31841
31842
31843
31844
31845
31846
31847
31848
31849
31850
31851
31852
31853
31854
31855
31856
31857
31858
31859
31860
31861
31862
31863
31864
31865
31866
31867
31868
31869
31870
31871
31872
31873
31874
31875
31876
31877
31878
31879
31880
31881
31882
31883
31884
31885
31886
31887
31888
31889
31890
31891
31892
31893
31894
31895
31896
31897
31898
31899
31900
31901
31902
31903
31904
31905
31906
31907
31908
31909
31910
31911
31912
31913
31914
31915
31916
31917
31918
31919
31920
31921
31922
31923
31924
31925
31926
31927
31928
31929
31930
31931
31932
31933
31934
31935
31936
31937
31938
31939
31940
31941
31942
31943
31944
31945
31946
31947
31948
31949
31950
31951
31952
31953
31954
31955
31956
31957
31958
31959
31960
31961
31962
31963
31964
31965
31966
31967
31968
31969
31970
31971
31972
31973
31974
31975
31976
31977
31978
31979
31980
31981
31982
31983
31984
31985
31986
31987
31988
31989
31990
31991
31992
31993
31994
31995
31996
31997
31998
31999
32000
32001
32002
32003
32004
32005
32006
32007
32008
32009
32010
32011
32012
32013
32014
32015
32016
32017
32018
32019
32020
32021
32022
32023
32024
32025
32026
32027
32028
32029
32030
32031
32032
32033
32034
32035
32036
32037
32038
32039
32040
32041
32042
32043
32044
32045
32046
32047
32048
32049
32050
32051
32052
32053
32054
32055
32056
32057
32058
32059
32060
32061
32062
32063
32064
32065
32066
32067
32068
32069
32070
32071
32072
32073
32074
32075
32076
32077
32078
32079
32080
32081
32082
32083
32084
32085
32086
32087
32088
32089
32090
32091
32092
32093
32094
32095
32096
32097
32098
32099
32100
32101
32102
32103
32104
32105
32106
32107
32108
32109
32110
32111
32112
32113
32114
32115
32116
32117
32118
32119
32120
32121
32122
32123
32124
32125
32126
32127
32128
32129
32130
32131
32132
32133
32134
32135
32136
32137
32138
32139
32140
32141
32142
32143
32144
32145
32146
32147
32148
32149
32150
32151
32152
32153
32154
32155
32156
32157
32158
32159
32160
32161
32162
32163
32164
32165
32166
32167
32168
32169
32170
32171
32172
32173
32174
32175
32176
32177
32178
32179
32180
32181
32182
32183
32184
32185
32186
32187
32188
32189
32190
32191
32192
32193
32194
32195
32196
32197
32198
32199
32200
32201
32202
32203
32204
32205
32206
32207
32208
32209
32210
32211
32212
32213
32214
32215
32216
32217
32218
32219
32220
32221
32222
32223
32224
32225
32226
32227
32228
32229
32230
32231
32232
32233
32234
32235
32236
32237
32238
32239
32240
32241
32242
32243
32244
32245
32246
32247
32248
32249
32250
32251
32252
32253
32254
32255
32256
32257
32258
32259
32260
32261
32262
32263
32264
32265
32266
32267
32268
32269
32270
32271
32272
32273
32274
32275
32276
32277
32278
32279
32280
32281
32282
32283
32284
32285
32286
32287
32288
32289
32290
32291
32292
32293
32294
32295
32296
32297
32298
32299
32300
32301
32302
32303
32304
32305
32306
32307
32308
32309
32310
32311
32312
32313
32314
32315
32316
32317
32318
32319
32320
32321
32322
32323
32324
32325
32326
32327
32328
32329
32330
32331
32332
32333
32334
32335
32336
32337
32338
32339
32340
32341
32342
32343
32344
32345
32346
32347
32348
32349
32350
32351
32352
32353
32354
32355
32356
32357
32358
32359
32360
32361
32362
32363
32364
32365
32366
32367
32368
32369
32370
32371
32372
32373
32374
32375
32376
32377
32378
32379
32380
32381
32382
32383
32384
32385
32386
32387
32388
32389
32390
32391
32392
32393
32394
32395
32396
32397
32398
32399
32400
32401
32402
32403
32404
32405
32406
32407
32408
32409
32410
32411
32412
32413
32414
32415
32416
32417
32418
32419
32420
32421
32422
32423
32424
32425
32426
32427
32428
32429
32430
32431
32432
32433
32434
32435
32436
32437
32438
32439
32440
32441
32442
32443
32444
32445
32446
32447
32448
32449
32450
32451
32452
32453
32454
32455
32456
32457
32458
32459
32460
32461
32462
32463
32464
32465
32466
32467
32468
32469
32470
32471
32472
32473
32474
32475
32476
32477
32478
32479
32480
32481
32482
32483
32484
32485
32486
32487
32488
32489
32490
32491
32492
32493
32494
32495
32496
32497
32498
32499
32500
32501
32502
32503
32504
32505
32506
32507
32508
32509
32510
32511
32512
32513
32514
32515
32516
32517
32518
32519
32520
32521
32522
32523
32524
32525
32526
32527
32528
32529
32530
32531
32532
32533
32534
32535
32536
32537
32538
32539
32540
32541
32542
32543
32544
32545
32546
32547
32548
32549
32550
32551
32552
32553
32554
32555
32556
32557
32558
32559
32560
32561
32562
32563
32564
32565
32566
32567
32568
32569
32570
32571
32572
32573
32574
32575
32576
32577
32578
32579
32580
32581
32582
32583
32584
32585
32586
32587
32588
32589
32590
32591
32592
32593
32594
32595
32596
32597
32598
32599
32600
32601
32602
32603
32604
32605
32606
32607
32608
32609
32610
32611
32612
32613
32614
32615
32616
32617
32618
32619
32620
32621
32622
32623
32624
32625
32626
32627
32628
32629
32630
32631
32632
32633
32634
32635
32636
32637
32638
32639
32640
32641
32642
32643
32644
32645
32646
32647
32648
32649
32650
32651
32652
32653
32654
32655
32656
32657
32658
32659
32660
32661
32662
32663
32664
32665
32666
32667
32668
32669
32670
32671
32672
32673
32674
32675
32676
32677
32678
32679
32680
32681
32682
32683
32684
32685
32686
32687
32688
32689
32690
32691
32692
32693
32694
32695
32696
32697
32698
32699
32700
32701
32702
32703
32704
32705
32706
32707
32708
32709
32710
32711
32712
32713
32714
32715
32716
32717
32718
32719
32720
32721
32722
32723
32724
32725
32726
32727
32728
32729
32730
32731
32732
32733
32734
32735
32736
32737
32738
32739
32740
32741
32742
32743
32744
32745
32746
32747
32748
32749
32750
32751
32752
32753
32754
32755
32756
32757
32758
32759
32760
32761
32762
32763
32764
32765
32766
32767
32768
32769
32770
32771
32772
32773
32774
32775
32776
32777
32778
32779
32780
32781
32782
32783
32784
32785
32786
32787
32788
32789
32790
32791
32792
32793
32794
32795
32796
32797
32798
32799
32800
32801
32802
32803
32804
32805
32806
32807
32808
32809
32810
32811
32812
32813
32814
32815
32816
32817
32818
32819
32820
32821
32822
32823
32824
32825
32826
32827
32828
32829
32830
32831
32832
32833
32834
32835
32836
32837
32838
32839
32840
32841
32842
32843
32844
32845
32846
32847
32848
32849
32850
32851
32852
32853
32854
32855
32856
32857
32858
32859
32860
32861
32862
32863
32864
32865
32866
32867
32868
32869
32870
32871
32872
32873
32874
32875
32876
32877
32878
32879
32880
32881
32882
32883
32884
32885
32886
32887
32888
32889
32890
32891
32892
32893
32894
32895
32896
32897
32898
32899
32900
32901
32902
32903
32904
32905
32906
32907
32908
32909
32910
32911
32912
32913
32914
32915
32916
32917
32918
32919
32920
32921
32922
32923
32924
32925
32926
32927
32928
32929
32930
32931
32932
32933
32934
32935
32936
32937
32938
32939
32940
32941
32942
32943
32944
32945
32946
32947
32948
32949
32950
32951
32952
32953
32954
32955
32956
32957
32958
32959
32960
32961
32962
32963
32964
32965
32966
32967
32968
32969
32970
32971
32972
32973
32974
32975
32976
32977
32978
32979
32980
32981
32982
32983
32984
32985
32986
32987
32988
32989
32990
32991
32992
32993
32994
32995
32996
32997
32998
32999
33000
33001
33002
33003
33004
33005
33006
33007
33008
33009
33010
33011
33012
33013
33014
33015
33016
33017
33018
33019
33020
33021
33022
33023
33024
33025
33026
33027
33028
33029
33030
33031
33032
33033
33034
33035
33036
33037
33038
33039
33040
33041
33042
33043
33044
33045
33046
33047
33048
33049
33050
33051
33052
33053
33054
33055
33056
33057
33058
33059
33060
33061
33062
33063
33064
33065
33066
33067
33068
33069
33070
33071
33072
33073
33074
33075
33076
33077
33078
33079
33080
33081
33082
33083
33084
33085
33086
33087
33088
33089
33090
33091
33092
33093
33094
33095
33096
33097
33098
33099
33100
33101
33102
33103
33104
33105
33106
33107
33108
33109
33110
33111
33112
33113
33114
33115
33116
33117
33118
33119
33120
33121
33122
33123
33124
33125
33126
33127
33128
33129
33130
33131
33132
33133
33134
33135
33136
33137
33138
33139
33140
33141
33142
33143
33144
33145
33146
33147
33148
33149
33150
33151
33152
33153
33154
33155
33156
33157
33158
33159
33160
33161
33162
33163
33164
33165
33166
33167
33168
33169
33170
33171
33172
33173
33174
33175
33176
33177
33178
33179
33180
33181
33182
33183
33184
33185
33186
33187
33188
33189
33190
33191
33192
33193
33194
33195
33196
33197
33198
33199
33200
33201
33202
33203
33204
33205
33206
33207
33208
33209
33210
33211
33212
33213
33214
33215
33216
33217
33218
33219
33220
33221
33222
33223
33224
33225
33226
33227
33228
33229
33230
33231
33232
33233
33234
33235
33236
33237
33238
33239
33240
33241
33242
33243
33244
33245
33246
33247
33248
33249
33250
33251
33252
33253
33254
33255
33256
33257
33258
33259
33260
33261
33262
33263
33264
33265
33266
33267
33268
33269
33270
33271
33272
33273
33274
33275
33276
33277
33278
33279
33280
33281
33282
33283
33284
33285
33286
33287
33288
33289
33290
33291
33292
33293
33294
33295
33296
33297
33298
33299
33300
33301
33302
33303
33304
33305
33306
33307
33308
33309
33310
33311
33312
33313
33314
33315
33316
33317
33318
33319
33320
33321
33322
33323
33324
33325
33326
33327
33328
33329
33330
33331
33332
33333
33334
33335
33336
33337
33338
33339
33340
33341
33342
33343
33344
33345
33346
33347
33348
33349
33350
33351
33352
33353
33354
33355
33356
33357
33358
33359
33360
33361
33362
33363
33364
33365
33366
33367
33368
33369
33370
33371
33372
33373
33374
33375
33376
33377
33378
33379
33380
33381
33382
33383
33384
33385
33386
33387
33388
33389
33390
33391
33392
33393
33394
33395
33396
33397
33398
33399
33400
33401
33402
33403
33404
33405
33406
33407
33408
33409
33410
33411
33412
33413
33414
33415
33416
33417
33418
33419
33420
33421
33422
33423
33424
33425
33426
33427
33428
33429
33430
33431
33432
33433
33434
33435
33436
33437
33438
33439
33440
33441
33442
33443
33444
33445
33446
33447
33448
33449
33450
33451
33452
33453
33454
33455
33456
33457
33458
33459
33460
33461
33462
33463
33464
33465
33466
33467
33468
33469
33470
33471
33472
33473
33474
33475
33476
33477
33478
33479
33480
33481
33482
33483
33484
33485
33486
33487
33488
33489
33490
33491
33492
33493
33494
33495
33496
33497
33498
33499
33500
33501
33502
33503
33504
33505
33506
33507
33508
33509
33510
33511
33512
33513
33514
33515
33516
33517
33518
33519
33520
33521
33522
33523
33524
33525
33526
33527
33528
33529
33530
33531
33532
33533
33534
33535
33536
33537
33538
33539
33540
33541
33542
33543
33544
33545
33546
33547
33548
33549
33550
33551
33552
33553
33554
33555
33556
33557
33558
33559
33560
33561
33562
33563
33564
33565
33566
33567
33568
33569
33570
33571
33572
33573
33574
33575
33576
33577
33578
33579
33580
33581
33582
33583
33584
33585
33586
33587
33588
33589
33590
33591
33592
33593
33594
33595
33596
33597
33598
33599
33600
33601
33602
33603
33604
33605
33606
33607
33608
33609
33610
33611
33612
33613
33614
33615
33616
33617
33618
33619
33620
33621
33622
33623
33624
33625
33626
33627
33628
33629
33630
33631
33632
33633
33634
33635
33636
33637
33638
33639
33640
33641
33642
33643
33644
33645
33646
33647
33648
33649
33650
33651
33652
33653
33654
33655
33656
33657
33658
33659
33660
33661
33662
33663
33664
33665
33666
33667
33668
33669
33670
33671
33672
33673
33674
33675
33676
33677
33678
33679
33680
33681
33682
33683
33684
33685
33686
33687
33688
33689
33690
33691
33692
33693
33694
33695
33696
33697
33698
33699
33700
33701
33702
33703
33704
33705
33706
33707
33708
33709
33710
33711
33712
33713
33714
33715
33716
33717
33718
33719
33720
33721
33722
33723
33724
33725
33726
33727
33728
33729
33730
33731
33732
33733
33734
33735
33736
33737
33738
33739
33740
33741
33742
33743
33744
33745
33746
33747
33748
33749
33750
33751
33752
33753
33754
33755
33756
33757
33758
33759
33760
33761
33762
33763
33764
33765
33766
33767
33768
33769
33770
33771
33772
33773
33774
33775
33776
33777
33778
33779
33780
33781
33782
33783
33784
33785
33786
33787
33788
33789
33790
33791
33792
33793
33794
33795
33796
33797
33798
33799
33800
33801
33802
33803
33804
33805
33806
33807
33808
33809
33810
33811
33812
33813
33814
33815
33816
33817
33818
33819
33820
33821
33822
33823
33824
33825
33826
33827
33828
33829
33830
33831
33832
33833
33834
33835
33836
33837
33838
33839
33840
33841
33842
33843
33844
33845
33846
33847
33848
33849
33850
33851
33852
33853
33854
33855
33856
33857
33858
33859
33860
33861
33862
33863
33864
33865
33866
33867
33868
33869
33870
33871
33872
33873
33874
33875
33876
33877
33878
33879
33880
33881
33882
33883
33884
33885
33886
33887
33888
33889
33890
33891
33892
33893
33894
33895
33896
33897
33898
33899
33900
33901
33902
33903
33904
33905
33906
33907
33908
33909
33910
33911
33912
33913
33914
33915
33916
33917
33918
33919
33920
33921
33922
33923
33924
33925
33926
33927
33928
33929
33930
33931
33932
33933
33934
33935
33936
33937
33938
33939
33940
33941
33942
33943
33944
33945
33946
33947
33948
33949
33950
33951
33952
33953
33954
33955
33956
33957
33958
33959
33960
33961
33962
33963
33964
33965
33966
33967
33968
33969
33970
33971
33972
33973
33974
33975
33976
33977
33978
33979
33980
33981
33982
33983
33984
33985
33986
33987
33988
33989
33990
33991
33992
33993
33994
33995
33996
33997
33998
33999
34000
34001
34002
34003
34004
34005
34006
34007
34008
34009
34010
34011
34012
34013
34014
34015
34016
34017
34018
34019
34020
34021
34022
34023
34024
34025
34026
34027
34028
34029
34030
34031
34032
34033
34034
34035
34036
34037
34038
34039
34040
34041
34042
34043
34044
34045
34046
34047
34048
34049
34050
34051
34052
34053
34054
34055
34056
34057
34058
34059
34060
34061
34062
34063
34064
34065
34066
34067
34068
34069
34070
34071
34072
34073
34074
34075
34076
34077
34078
34079
34080
34081
34082
34083
34084
34085
34086
34087
34088
34089
34090
34091
34092
34093
34094
34095
34096
34097
34098
34099
34100
34101
34102
34103
34104
34105
34106
34107
34108
34109
34110
34111
34112
34113
34114
34115
34116
34117
34118
34119
34120
34121
34122
34123
34124
34125
34126
34127
34128
34129
34130
34131
34132
34133
34134
34135
34136
34137
34138
34139
34140
34141
34142
34143
34144
34145
34146
34147
34148
34149
34150
34151
34152
34153
34154
34155
34156
34157
34158
34159
34160
34161
34162
34163
34164
34165
34166
34167
34168
34169
34170
34171
34172
34173
34174
34175
34176
34177
34178
34179
34180
34181
34182
34183
34184
34185
34186
34187
34188
34189
34190
34191
34192
34193
34194
34195
34196
34197
34198
34199
34200
34201
34202
34203
34204
34205
34206
34207
34208
34209
34210
34211
34212
34213
34214
34215
34216
34217
34218
34219
34220
34221
34222
34223
34224
34225
34226
34227
34228
34229
34230
34231
34232
34233
34234
34235
34236
34237
34238
34239
34240
34241
34242
34243
34244
34245
34246
34247
34248
34249
34250
34251
34252
34253
34254
34255
34256
34257
34258
34259
34260
34261
34262
34263
34264
34265
34266
34267
34268
34269
34270
34271
34272
34273
34274
34275
34276
34277
34278
34279
34280
34281
34282
34283
34284
34285
34286
34287
34288
34289
34290
34291
34292
34293
34294
34295
34296
34297
34298
34299
34300
34301
34302
34303
34304
34305
34306
34307
34308
34309
34310
34311
34312
34313
34314
34315
34316
34317
34318
34319
34320
34321
34322
34323
34324
34325
34326
34327
34328
34329
34330
34331
34332
34333
34334
34335
34336
34337
34338
34339
34340
34341
34342
34343
34344
34345
34346
34347
34348
34349
34350
34351
34352
34353
34354
34355
34356
34357
34358
34359
34360
34361
34362
34363
34364
34365
34366
34367
34368
34369
34370
34371
34372
34373
34374
34375
34376
34377
34378
34379
34380
34381
34382
34383
34384
34385
34386
34387
34388
34389
34390
34391
34392
34393
34394
34395
34396
34397
34398
34399
34400
34401
34402
34403
34404
34405
34406
34407
34408
34409
34410
34411
34412
34413
34414
34415
34416
34417
34418
34419
34420
34421
34422
34423
34424
34425
34426
34427
34428
34429
34430
34431
34432
34433
34434
34435
34436
34437
34438
34439
34440
34441
34442
34443
34444
34445
34446
34447
34448
34449
34450
34451
34452
34453
34454
34455
34456
34457
34458
34459
34460
34461
34462
34463
34464
34465
34466
34467
34468
34469
34470
34471
34472
34473
34474
34475
34476
34477
34478
34479
34480
34481
34482
34483
34484
34485
34486
34487
34488
34489
34490
34491
34492
34493
34494
34495
34496
34497
34498
34499
34500
34501
34502
34503
34504
34505
34506
34507
34508
34509
34510
34511
34512
34513
34514
34515
34516
34517
34518
34519
34520
34521
34522
34523
34524
34525
34526
34527
34528
34529
34530
34531
34532
34533
34534
34535
34536
34537
34538
34539
34540
34541
34542
34543
34544
34545
34546
34547
34548
34549
34550
34551
34552
34553
34554
34555
34556
34557
34558
34559
34560
34561
34562
34563
34564
34565
34566
34567
34568
34569
34570
34571
34572
34573
34574
34575
34576
34577
34578
34579
34580
34581
34582
34583
34584
34585
34586
34587
34588
34589
34590
34591
34592
34593
34594
34595
34596
34597
34598
34599
34600
34601
34602
34603
34604
34605
34606
34607
34608
34609
34610
34611
34612
34613
34614
34615
34616
34617
34618
34619
34620
34621
34622
34623
34624
34625
34626
34627
34628
34629
34630
34631
34632
34633
34634
34635
34636
34637
34638
34639
34640
34641
34642
34643
34644
34645
34646
34647
34648
34649
34650
34651
34652
34653
34654
34655
34656
34657
34658
34659
34660
34661
34662
34663
34664
34665
34666
34667
34668
34669
34670
34671
34672
34673
34674
34675
34676
34677
34678
34679
34680
34681
34682
34683
34684
34685
34686
34687
34688
34689
34690
34691
34692
34693
34694
34695
34696
34697
34698
34699
34700
34701
34702
34703
34704
34705
34706
34707
34708
34709
34710
34711
34712
34713
34714
34715
34716
34717
34718
34719
34720
34721
34722
34723
34724
34725
34726
34727
34728
34729
34730
34731
34732
34733
34734
34735
34736
34737
34738
34739
34740
34741
34742
34743
34744
34745
34746
34747
34748
34749
34750
34751
34752
34753
34754
34755
34756
34757
34758
34759
34760
34761
34762
34763
34764
34765
34766
34767
34768
34769
34770
34771
34772
34773
34774
34775
34776
34777
34778
34779
34780
34781
34782
34783
34784
34785
34786
34787
34788
34789
34790
34791
34792
34793
34794
34795
34796
34797
34798
34799
34800
34801
34802
34803
34804
34805
34806
34807
34808
34809
34810
34811
34812
34813
34814
34815
34816
34817
34818
34819
34820
34821
34822
34823
34824
34825
34826
34827
34828
34829
34830
34831
34832
34833
34834
34835
34836
34837
34838
34839
34840
34841
34842
34843
34844
34845
34846
34847
34848
34849
34850
34851
34852
34853
34854
34855
34856
34857
34858
34859
34860
34861
34862
34863
34864
34865
34866
34867
34868
34869
34870
34871
34872
34873
34874
34875
34876
34877
34878
34879
34880
34881
34882
34883
34884
34885
34886
34887
34888
34889
34890
34891
34892
34893
34894
34895
34896
34897
34898
34899
34900
34901
34902
34903
34904
34905
34906
34907
34908
34909
34910
34911
34912
34913
34914
34915
34916
34917
34918
34919
34920
34921
34922
34923
34924
34925
34926
34927
34928
34929
34930
34931
34932
34933
34934
34935
34936
34937
34938
34939
34940
34941
34942
34943
34944
34945
34946
34947
34948
34949
34950
34951
34952
34953
34954
34955
34956
34957
34958
34959
34960
34961
34962
34963
34964
34965
34966
34967
34968
34969
34970
34971
34972
34973
34974
34975
34976
34977
34978
34979
34980
34981
34982
34983
34984
34985
34986
34987
34988
34989
34990
34991
34992
34993
34994
34995
34996
34997
34998
34999
35000
35001
35002
35003
35004
35005
35006
35007
35008
35009
35010
35011
35012
35013
35014
35015
35016
35017
35018
35019
35020
35021
35022
35023
35024
35025
35026
35027
35028
35029
35030
35031
35032
35033
35034
35035
35036
35037
35038
35039
35040
35041
35042
35043
35044
35045
35046
35047
35048
35049
35050
35051
35052
35053
35054
35055
35056
35057
35058
35059
35060
35061
35062
35063
35064
35065
35066
35067
35068
35069
35070
35071
35072
35073
35074
35075
35076
35077
35078
35079
35080
35081
35082
35083
35084
35085
35086
35087
35088
35089
35090
35091
35092
35093
35094
35095
35096
35097
35098
35099
35100
35101
35102
35103
35104
35105
35106
35107
35108
35109
35110
35111
35112
35113
35114
35115
35116
35117
35118
35119
35120
35121
35122
35123
35124
35125
35126
35127
35128
35129
35130
35131
35132
35133
35134
35135
35136
35137
35138
35139
35140
35141
35142
35143
35144
35145
35146
35147
35148
35149
35150
35151
35152
35153
35154
35155
35156
35157
35158
35159
35160
35161
35162
35163
35164
35165
35166
35167
35168
35169
35170
35171
35172
35173
35174
35175
35176
35177
35178
35179
35180
35181
35182
35183
35184
35185
35186
35187
35188
35189
35190
35191
35192
35193
35194
35195
35196
35197
35198
35199
35200
35201
35202
35203
35204
35205
35206
35207
35208
35209
35210
35211
35212
35213
35214
35215
35216
35217
35218
35219
35220
35221
35222
35223
35224
35225
35226
35227
35228
35229
35230
35231
35232
35233
35234
35235
35236
35237
35238
35239
35240
35241
35242
35243
35244
35245
35246
35247
35248
35249
35250
35251
35252
35253
35254
35255
35256
35257
35258
35259
35260
35261
35262
35263
35264
35265
35266
35267
35268
35269
35270
35271
35272
35273
35274
35275
35276
35277
35278
35279
35280
35281
35282
35283
35284
35285
35286
35287
35288
35289
35290
35291
35292
35293
35294
35295
35296
35297
35298
35299
35300
35301
35302
35303
35304
35305
35306
35307
35308
35309
35310
35311
35312
35313
35314
35315
35316
35317
35318
35319
35320
35321
35322
35323
35324
35325
35326
35327
35328
35329
35330
35331
35332
35333
35334
35335
35336
35337
35338
35339
35340
35341
35342
35343
35344
35345
35346
35347
35348
35349
35350
35351
35352
35353
35354
35355
35356
35357
35358
35359
35360
35361
35362
35363
35364
35365
35366
35367
35368
35369
35370
35371
35372
35373
35374
35375
35376
35377
35378
35379
35380
35381
35382
35383
35384
35385
35386
35387
35388
35389
35390
35391
35392
35393
35394
35395
35396
35397
35398
35399
35400
35401
35402
35403
35404
35405
35406
35407
35408
35409
35410
35411
35412
35413
35414
35415
35416
35417
35418
35419
35420
35421
35422
35423
35424
35425
35426
35427
35428
35429
35430
35431
35432
35433
35434
35435
35436
35437
35438
35439
35440
35441
35442
35443
35444
35445
35446
35447
35448
35449
35450
35451
35452
35453
35454
35455
35456
35457
35458
35459
35460
35461
35462
35463
35464
35465
35466
35467
35468
35469
35470
35471
35472
35473
35474
35475
35476
35477
35478
35479
35480
35481
35482
35483
35484
35485
35486
35487
35488
35489
35490
35491
35492
35493
35494
35495
35496
35497
35498
35499
35500
35501
35502
35503
35504
35505
35506
35507
35508
35509
35510
35511
35512
35513
35514
35515
35516
35517
35518
35519
35520
35521
35522
35523
35524
35525
35526
35527
35528
35529
35530
35531
35532
35533
35534
35535
35536
35537
35538
35539
35540
35541
35542
35543
35544
35545
35546
35547
35548
35549
35550
35551
35552
35553
35554
35555
35556
35557
35558
35559
35560
35561
35562
35563
35564
35565
35566
35567
35568
35569
35570
35571
35572
35573
35574
35575
35576
35577
35578
35579
35580
35581
35582
35583
35584
35585
35586
35587
35588
35589
35590
35591
35592
35593
35594
35595
35596
35597
35598
35599
35600
35601
35602
35603
35604
35605
35606
35607
35608
35609
35610
35611
35612
35613
35614
35615
35616
35617
35618
35619
35620
35621
35622
35623
35624
35625
35626
35627
35628
35629
35630
35631
35632
35633
35634
35635
35636
35637
35638
35639
35640
35641
35642
35643
35644
35645
35646
35647
35648
35649
35650
35651
35652
35653
35654
35655
35656
35657
35658
35659
35660
35661
35662
35663
35664
35665
35666
35667
35668
35669
35670
35671
35672
35673
35674
35675
35676
35677
35678
35679
35680
35681
35682
35683
35684
35685
35686
35687
35688
35689
35690
35691
35692
35693
35694
35695
35696
35697
35698
35699
35700
35701
35702
35703
35704
35705
35706
35707
35708
35709
35710
35711
35712
35713
35714
35715
35716
35717
35718
35719
35720
35721
35722
35723
35724
35725
35726
35727
35728
35729
35730
35731
35732
35733
35734
35735
35736
35737
35738
35739
35740
35741
35742
35743
35744
35745
35746
35747
35748
35749
35750
35751
35752
35753
35754
35755
35756
35757
35758
35759
35760
35761
35762
35763
35764
35765
35766
35767
35768
35769
35770
35771
35772
35773
35774
35775
35776
35777
35778
35779
35780
35781
35782
35783
35784
35785
35786
35787
35788
35789
35790
35791
35792
35793
35794
35795
35796
35797
35798
35799
35800
35801
35802
35803
35804
35805
35806
35807
35808
35809
35810
35811
35812
35813
35814
35815
35816
35817
35818
35819
35820
35821
35822
35823
35824
35825
35826
35827
35828
35829
35830
35831
35832
35833
35834
35835
35836
35837
35838
35839
35840
35841
35842
35843
35844
35845
35846
35847
35848
35849
35850
35851
35852
35853
35854
35855
35856
35857
35858
35859
35860
35861
35862
35863
35864
35865
35866
35867
35868
35869
35870
35871
35872
35873
35874
35875
35876
35877
35878
35879
35880
35881
35882
35883
35884
35885
35886
35887
35888
35889
35890
35891
35892
35893
35894
35895
35896
35897
35898
35899
35900
35901
35902
35903
35904
35905
35906
35907
35908
35909
35910
35911
35912
35913
35914
35915
35916
35917
35918
35919
35920
35921
35922
35923
35924
35925
35926
35927
35928
35929
35930
35931
35932
35933
35934
35935
35936
35937
35938
35939
35940
35941
35942
35943
35944
35945
35946
35947
35948
35949
35950
35951
35952
35953
35954
35955
35956
35957
35958
35959
35960
35961
35962
35963
35964
35965
35966
35967
35968
35969
35970
35971
35972
35973
35974
35975
35976
35977
35978
35979
35980
35981
35982
35983
35984
35985
35986
35987
35988
35989
35990
35991
35992
35993
35994
35995
35996
35997
35998
35999
36000
36001
36002
36003
36004
36005
36006
36007
36008
36009
36010
36011
36012
36013
36014
36015
36016
36017
36018
36019
36020
36021
36022
36023
36024
36025
36026
36027
36028
36029
36030
36031
36032
36033
36034
36035
36036
36037
36038
36039
36040
36041
36042
36043
36044
36045
36046
36047
36048
36049
36050
36051
36052
36053
36054
36055
36056
36057
36058
36059
36060
36061
36062
36063
36064
36065
36066
36067
36068
36069
36070
36071
36072
36073
36074
36075
36076
36077
36078
36079
36080
36081
36082
36083
36084
36085
36086
36087
36088
36089
36090
36091
36092
36093
36094
36095
36096
36097
36098
36099
36100
36101
36102
36103
36104
36105
36106
36107
36108
36109
36110
36111
36112
36113
36114
36115
36116
36117
36118
36119
36120
36121
36122
36123
36124
36125
36126
36127
36128
36129
36130
36131
36132
36133
36134
36135
36136
36137
36138
36139
36140
36141
36142
36143
36144
36145
36146
36147
36148
36149
36150
36151
36152
36153
36154
36155
36156
36157
36158
36159
36160
36161
36162
36163
36164
36165
36166
36167
36168
36169
36170
36171
36172
36173
36174
36175
36176
36177
36178
36179
36180
36181
36182
36183
36184
36185
36186
36187
36188
36189
36190
36191
36192
36193
36194
36195
36196
36197
36198
36199
36200
36201
36202
36203
36204
36205
36206
36207
36208
36209
36210
36211
36212
36213
36214
36215
36216
36217
36218
36219
36220
36221
36222
36223
36224
36225
36226
36227
36228
36229
36230
36231
36232
36233
36234
36235
36236
36237
36238
36239
36240
36241
36242
36243
36244
36245
36246
36247
36248
36249
36250
36251
36252
36253
36254
36255
36256
36257
36258
36259
36260
36261
36262
36263
36264
36265
36266
36267
36268
36269
36270
36271
36272
36273
36274
36275
36276
36277
36278
36279
36280
36281
36282
36283
36284
36285
36286
36287
36288
36289
36290
36291
36292
36293
36294
36295
36296
36297
36298
36299
36300
36301
36302
36303
36304
36305
36306
36307
36308
36309
36310
36311
36312
36313
36314
36315
36316
36317
36318
36319
36320
36321
36322
36323
36324
36325
36326
36327
36328
36329
36330
36331
36332
36333
36334
36335
36336
36337
36338
36339
36340
36341
36342
36343
36344
36345
36346
36347
36348
36349
36350
36351
36352
36353
36354
36355
36356
36357
36358
36359
36360
36361
36362
36363
36364
36365
36366
36367
36368
36369
36370
36371
36372
36373
36374
36375
36376
36377
36378
36379
36380
36381
36382
36383
36384
36385
36386
36387
36388
36389
36390
36391
36392
36393
36394
36395
36396
36397
36398
36399
36400
36401
36402
36403
36404
36405
36406
36407
36408
36409
36410
36411
36412
36413
36414
36415
36416
36417
36418
36419
36420
36421
36422
36423
36424
36425
36426
36427
36428
36429
36430
36431
36432
36433
36434
36435
36436
36437
36438
36439
36440
36441
36442
36443
36444
36445
36446
36447
36448
36449
36450
36451
36452
36453
36454
36455
36456
36457
36458
36459
36460
36461
36462
36463
36464
36465
36466
36467
36468
36469
36470
36471
36472
36473
36474
36475
36476
36477
36478
36479
36480
36481
36482
36483
36484
36485
36486
36487
36488
36489
36490
36491
36492
36493
36494
36495
36496
36497
36498
36499
36500
36501
36502
36503
36504
36505
36506
36507
36508
36509
36510
36511
36512
36513
36514
36515
36516
36517
36518
36519
36520
36521
36522
36523
36524
36525
36526
36527
36528
36529
36530
36531
36532
36533
36534
36535
36536
36537
36538
36539
36540
36541
36542
36543
36544
36545
36546
36547
36548
36549
36550
36551
36552
36553
36554
36555
36556
36557
36558
36559
36560
36561
36562
36563
36564
36565
36566
36567
36568
36569
36570
36571
36572
36573
36574
36575
36576
36577
36578
36579
36580
36581
36582
36583
36584
36585
36586
36587
36588
36589
36590
36591
36592
36593
36594
36595
36596
36597
36598
36599
36600
36601
36602
36603
36604
36605
36606
36607
36608
36609
36610
36611
36612
36613
36614
36615
36616
36617
36618
36619
36620
36621
36622
36623
36624
36625
36626
36627
36628
36629
36630
36631
36632
36633
36634
36635
36636
36637
36638
36639
36640
36641
36642
36643
36644
36645
36646
36647
36648
36649
36650
36651
36652
36653
36654
36655
36656
36657
36658
36659
36660
36661
36662
36663
36664
36665
36666
36667
36668
36669
36670
36671
36672
36673
36674
36675
36676
36677
36678
36679
36680
36681
36682
36683
36684
36685
36686
36687
36688
36689
36690
36691
36692
36693
36694
36695
36696
36697
36698
36699
36700
36701
36702
36703
36704
36705
36706
36707
36708
36709
36710
36711
36712
36713
36714
36715
36716
36717
36718
36719
36720
36721
36722
36723
36724
36725
36726
36727
36728
36729
36730
36731
36732
36733
36734
36735
36736
36737
36738
36739
36740
36741
36742
36743
36744
36745
36746
36747
36748
36749
36750
36751
36752
36753
36754
36755
36756
36757
36758
36759
36760
36761
36762
36763
36764
36765
36766
36767
36768
36769
36770
36771
36772
36773
36774
36775
36776
36777
36778
36779
36780
36781
36782
36783
36784
36785
36786
36787
36788
36789
36790
36791
36792
36793
36794
36795
36796
36797
36798
36799
36800
36801
36802
36803
36804
36805
36806
36807
36808
36809
36810
36811
36812
36813
36814
36815
36816
36817
36818
36819
36820
36821
36822
36823
36824
36825
36826
36827
36828
36829
36830
36831
36832
36833
36834
36835
36836
36837
36838
36839
36840
36841
36842
36843
36844
36845
36846
36847
36848
36849
36850
36851
36852
36853
36854
36855
36856
36857
36858
36859
36860
36861
36862
36863
36864
36865
36866
36867
36868
36869
36870
36871
36872
36873
36874
36875
36876
36877
36878
36879
36880
36881
36882
36883
36884
36885
36886
36887
36888
36889
36890
36891
36892
36893
36894
36895
36896
36897
36898
36899
36900
36901
36902
36903
36904
36905
36906
36907
36908
36909
36910
36911
36912
36913
36914
36915
36916
36917
36918
36919
36920
36921
36922
36923
36924
36925
36926
36927
36928
36929
36930
36931
36932
36933
36934
36935
36936
36937
36938
36939
36940
36941
36942
36943
36944
36945
36946
36947
36948
36949
36950
36951
36952
36953
36954
36955
36956
36957
36958
36959
36960
36961
36962
36963
36964
36965
36966
36967
36968
36969
36970
36971
36972
36973
36974
36975
36976
36977
36978
36979
36980
36981
36982
36983
36984
36985
36986
36987
36988
36989
36990
36991
36992
36993
36994
36995
36996
36997
36998
36999
37000
37001
37002
37003
37004
37005
37006
37007
37008
37009
37010
37011
37012
37013
37014
37015
37016
37017
37018
37019
37020
37021
37022
37023
37024
37025
37026
37027
37028
37029
37030
37031
37032
37033
37034
37035
37036
37037
37038
37039
37040
37041
37042
37043
37044
37045
37046
37047
37048
37049
37050
37051
37052
37053
37054
37055
37056
37057
37058
37059
37060
37061
37062
37063
37064
37065
37066
37067
37068
37069
37070
37071
37072
37073
37074
37075
37076
37077
37078
37079
37080
37081
37082
37083
37084
37085
37086
37087
37088
37089
37090
37091
37092
37093
37094
37095
37096
37097
37098
37099
37100
37101
37102
37103
37104
37105
37106
37107
37108
37109
37110
37111
37112
37113
37114
37115
37116
37117
37118
37119
37120
37121
37122
37123
37124
37125
37126
37127
37128
37129
37130
37131
37132
37133
37134
37135
37136
37137
37138
37139
37140
37141
37142
37143
37144
37145
37146
37147
37148
37149
37150
37151
37152
37153
37154
37155
37156
37157
37158
37159
37160
37161
37162
37163
37164
37165
37166
37167
37168
37169
37170
37171
37172
37173
37174
37175
37176
37177
37178
37179
37180
37181
37182
37183
37184
37185
37186
37187
37188
37189
37190
37191
37192
37193
37194
37195
37196
37197
37198
37199
37200
37201
37202
37203
37204
37205
37206
37207
37208
37209
37210
37211
37212
37213
37214
37215
37216
37217
37218
37219
37220
37221
37222
37223
37224
37225
37226
37227
37228
37229
37230
37231
37232
37233
37234
37235
37236
37237
37238
37239
37240
37241
37242
37243
37244
37245
37246
37247
37248
37249
37250
37251
37252
37253
37254
37255
37256
37257
37258
37259
37260
37261
37262
37263
37264
37265
37266
37267
37268
37269
37270
37271
37272
37273
37274
37275
37276
37277
37278
37279
37280
37281
37282
37283
37284
37285
37286
37287
37288
37289
37290
37291
37292
37293
37294
37295
37296
37297
37298
37299
37300
37301
37302
37303
37304
37305
37306
37307
37308
37309
37310
37311
37312
37313
37314
37315
37316
37317
37318
37319
37320
37321
37322
37323
37324
37325
37326
37327
37328
37329
37330
37331
37332
37333
37334
37335
37336
37337
37338
37339
37340
37341
37342
37343
37344
37345
37346
37347
37348
37349
37350
37351
37352
37353
37354
37355
37356
37357
37358
37359
37360
37361
37362
37363
37364
37365
37366
37367
37368
37369
37370
37371
37372
37373
37374
37375
37376
37377
37378
37379
37380
37381
37382
37383
37384
37385
37386
37387
37388
37389
37390
37391
37392
37393
37394
37395
37396
37397
37398
37399
37400
37401
37402
37403
37404
37405
37406
37407
37408
37409
37410
37411
37412
37413
37414
37415
37416
37417
37418
37419
37420
37421
37422
37423
37424
37425
37426
37427
37428
37429
37430
37431
37432
37433
37434
37435
37436
37437
37438
37439
37440
37441
37442
37443
37444
37445
37446
37447
37448
37449
37450
37451
37452
37453
37454
37455
37456
37457
37458
37459
37460
37461
37462
37463
37464
37465
37466
37467
37468
37469
37470
37471
37472
37473
37474
37475
37476
37477
37478
37479
37480
37481
37482
37483
37484
37485
37486
37487
37488
37489
37490
37491
37492
37493
37494
37495
37496
37497
37498
37499
37500
37501
37502
37503
37504
37505
37506
37507
37508
37509
37510
37511
37512
37513
37514
37515
37516
37517
37518
37519
37520
37521
37522
37523
37524
37525
37526
37527
37528
37529
37530
37531
37532
37533
37534
37535
37536
37537
37538
37539
37540
37541
37542
37543
37544
37545
37546
37547
37548
37549
37550
37551
37552
37553
37554
37555
37556
37557
37558
37559
37560
37561
37562
37563
37564
37565
37566
37567
37568
37569
37570
37571
37572
37573
37574
37575
37576
37577
37578
37579
37580
37581
37582
37583
37584
37585
37586
37587
37588
37589
37590
37591
37592
37593
37594
37595
37596
37597
37598
37599
37600
37601
37602
37603
37604
37605
37606
37607
37608
37609
37610
37611
37612
37613
37614
37615
37616
37617
37618
37619
37620
37621
37622
37623
37624
37625
37626
37627
37628
37629
37630
37631
37632
37633
37634
37635
37636
37637
37638
37639
37640
37641
37642
37643
37644
37645
37646
37647
37648
37649
37650
37651
37652
37653
37654
37655
37656
37657
37658
37659
37660
37661
37662
37663
37664
37665
37666
37667
37668
37669
37670
37671
37672
37673
37674
37675
37676
37677
37678
37679
37680
37681
37682
37683
37684
37685
37686
37687
37688
37689
37690
37691
37692
37693
37694
37695
37696
37697
37698
37699
37700
37701
37702
37703
37704
37705
37706
37707
37708
37709
37710
37711
37712
37713
37714
37715
37716
37717
37718
37719
37720
37721
37722
37723
37724
37725
37726
37727
37728
37729
37730
37731
37732
37733
37734
37735
37736
37737
37738
37739
37740
37741
37742
37743
37744
37745
37746
37747
37748
37749
37750
37751
37752
37753
37754
37755
37756
37757
37758
37759
37760
37761
37762
37763
37764
37765
37766
37767
37768
37769
37770
37771
37772
37773
37774
37775
37776
37777
37778
37779
37780
37781
37782
37783
37784
37785
37786
37787
37788
37789
37790
37791
37792
37793
37794
37795
37796
37797
37798
37799
37800
37801
37802
37803
37804
37805
37806
37807
37808
37809
37810
37811
37812
37813
37814
37815
37816
37817
37818
37819
37820
37821
37822
37823
37824
37825
37826
37827
37828
37829
37830
37831
37832
37833
37834
37835
37836
37837
37838
37839
37840
37841
37842
37843
37844
37845
37846
37847
37848
37849
37850
37851
37852
37853
37854
37855
37856
37857
37858
37859
37860
37861
37862
37863
37864
37865
37866
37867
37868
37869
37870
37871
37872
37873
37874
37875
37876
37877
37878
37879
37880
37881
37882
37883
37884
37885
37886
37887
37888
37889
37890
37891
37892
37893
37894
37895
37896
37897
37898
37899
37900
37901
37902
37903
37904
37905
37906
37907
37908
37909
37910
37911
37912
37913
37914
37915
37916
37917
37918
37919
37920
37921
37922
37923
37924
37925
37926
37927
37928
37929
37930
37931
37932
37933
37934
37935
37936
37937
37938
37939
37940
37941
37942
37943
37944
37945
37946
37947
37948
37949
37950
37951
37952
37953
37954
37955
37956
37957
37958
37959
37960
37961
37962
37963
37964
37965
37966
37967
37968
37969
37970
37971
37972
37973
37974
37975
37976
37977
37978
37979
37980
37981
37982
37983
37984
37985
37986
37987
37988
37989
37990
37991
37992
37993
37994
37995
37996
37997
37998
37999
38000
38001
38002
38003
38004
38005
38006
38007
38008
38009
38010
38011
38012
38013
38014
38015
38016
38017
38018
38019
38020
38021
38022
38023
38024
38025
38026
38027
38028
38029
38030
38031
38032
38033
38034
38035
38036
38037
38038
38039
38040
38041
38042
38043
38044
38045
38046
38047
38048
38049
38050
38051
38052
38053
38054
38055
38056
38057
38058
38059
38060
38061
38062
38063
38064
38065
38066
38067
38068
38069
38070
38071
38072
38073
38074
38075
38076
38077
38078
38079
38080
38081
38082
38083
38084
38085
38086
38087
38088
38089
38090
38091
38092
38093
38094
38095
38096
38097
38098
38099
38100
38101
38102
38103
38104
38105
38106
38107
38108
38109
38110
38111
38112
38113
38114
38115
38116
38117
38118
38119
38120
38121
38122
38123
38124
38125
38126
38127
38128
38129
38130
38131
38132
38133
38134
38135
38136
38137
38138
38139
38140
38141
38142
38143
38144
38145
38146
38147
38148
38149
38150
38151
38152
38153
38154
38155
38156
38157
38158
38159
38160
38161
38162
38163
38164
38165
38166
38167
38168
38169
38170
38171
38172
38173
38174
38175
38176
38177
38178
38179
38180
38181
38182
38183
38184
38185
38186
38187
38188
38189
38190
38191
38192
38193
38194
38195
38196
38197
38198
38199
38200
38201
38202
38203
38204
38205
38206
38207
38208
38209
38210
38211
38212
38213
38214
38215
38216
38217
38218
38219
38220
38221
38222
38223
38224
38225
38226
38227
38228
38229
38230
38231
38232
38233
38234
38235
38236
38237
38238
38239
38240
38241
38242
38243
38244
38245
38246
38247
38248
38249
38250
38251
38252
38253
38254
38255
38256
38257
38258
38259
38260
38261
38262
38263
38264
38265
38266
38267
38268
38269
38270
38271
38272
38273
38274
38275
38276
38277
38278
38279
38280
38281
38282
38283
38284
38285
38286
38287
38288
38289
38290
38291
38292
38293
38294
38295
38296
38297
38298
38299
38300
38301
38302
38303
38304
38305
38306
38307
38308
38309
38310
38311
38312
38313
38314
38315
38316
38317
38318
38319
38320
38321
38322
38323
38324
38325
38326
38327
38328
38329
38330
38331
38332
38333
38334
38335
38336
38337
38338
38339
38340
38341
38342
38343
38344
38345
38346
38347
38348
38349
38350
38351
38352
38353
38354
38355
38356
38357
38358
38359
38360
38361
38362
38363
38364
38365
38366
38367
38368
38369
38370
38371
38372
38373
38374
38375
38376
38377
38378
38379
38380
38381
38382
38383
38384
38385
38386
38387
38388
38389
38390
38391
38392
38393
38394
38395
38396
38397
38398
38399
38400
38401
38402
38403
38404
38405
38406
38407
38408
38409
38410
38411
38412
38413
38414
38415
38416
38417
38418
38419
38420
38421
38422
38423
38424
38425
38426
38427
38428
38429
38430
38431
38432
38433
38434
38435
38436
38437
38438
38439
38440
38441
38442
38443
38444
38445
38446
38447
38448
38449
38450
38451
38452
38453
38454
38455
38456
38457
38458
38459
38460
38461
38462
38463
38464
38465
38466
38467
38468
38469
38470
38471
38472
38473
38474
38475
38476
38477
38478
38479
38480
38481
38482
38483
38484
38485
38486
38487
38488
38489
38490
38491
38492
38493
38494
38495
38496
38497
38498
38499
38500
38501
38502
38503
38504
38505
38506
38507
38508
38509
38510
38511
38512
38513
38514
38515
38516
38517
38518
38519
38520
38521
38522
38523
38524
38525
38526
38527
38528
38529
38530
38531
38532
38533
38534
38535
38536
38537
38538
38539
38540
38541
38542
38543
38544
38545
38546
38547
38548
38549
38550
38551
38552
38553
38554
38555
38556
38557
38558
38559
38560
38561
38562
38563
38564
38565
38566
38567
38568
38569
38570
38571
38572
38573
38574
38575
38576
38577
38578
38579
38580
38581
38582
38583
38584
38585
38586
38587
38588
38589
38590
38591
38592
38593
38594
38595
38596
38597
38598
38599
38600
38601
38602
38603
38604
38605
38606
38607
38608
38609
38610
38611
38612
38613
38614
38615
38616
38617
38618
38619
38620
38621
38622
38623
38624
38625
38626
38627
38628
38629
38630
38631
38632
38633
38634
38635
38636
38637
38638
38639
38640
38641
38642
38643
38644
38645
38646
38647
38648
38649
38650
38651
38652
38653
38654
38655
38656
38657
38658
38659
38660
38661
38662
38663
38664
38665
38666
38667
38668
38669
38670
38671
38672
38673
38674
38675
38676
38677
38678
38679
38680
38681
38682
38683
38684
38685
38686
38687
38688
38689
38690
38691
38692
38693
38694
38695
38696
38697
38698
38699
38700
38701
38702
38703
38704
38705
38706
38707
38708
38709
38710
38711
38712
38713
38714
38715
38716
38717
38718
38719
38720
38721
38722
38723
38724
38725
38726
38727
38728
38729
38730
38731
38732
38733
38734
38735
38736
38737
38738
38739
38740
38741
38742
38743
38744
38745
38746
38747
38748
38749
38750
38751
38752
38753
38754
38755
38756
38757
38758
38759
38760
38761
38762
38763
38764
38765
38766
38767
38768
38769
38770
38771
38772
38773
38774
38775
38776
38777
38778
38779
38780
38781
38782
38783
38784
38785
38786
38787
38788
38789
38790
38791
38792
38793
38794
38795
38796
38797
38798
38799
38800
38801
38802
38803
38804
38805
38806
38807
38808
38809
38810
38811
38812
38813
38814
38815
38816
38817
38818
38819
38820
38821
38822
38823
38824
38825
38826
38827
38828
38829
38830
38831
38832
38833
38834
38835
38836
38837
38838
38839
38840
38841
38842
38843
38844
38845
38846
38847
38848
38849
38850
38851
38852
38853
38854
38855
38856
38857
38858
38859
38860
38861
38862
38863
38864
38865
38866
38867
38868
38869
38870
38871
38872
38873
38874
38875
38876
38877
38878
38879
38880
38881
38882
38883
38884
38885
38886
38887
38888
38889
38890
38891
38892
38893
38894
38895
38896
38897
38898
38899
38900
38901
38902
38903
38904
38905
38906
38907
38908
38909
38910
38911
38912
38913
38914
38915
38916
38917
38918
38919
38920
38921
38922
38923
38924
38925
38926
38927
38928
38929
38930
38931
38932
38933
38934
38935
38936
38937
38938
38939
38940
38941
38942
38943
38944
38945
38946
38947
38948
38949
38950
38951
38952
38953
38954
38955
38956
38957
38958
38959
38960
38961
38962
38963
38964
38965
38966
38967
38968
38969
38970
38971
38972
38973
38974
38975
38976
38977
38978
38979
38980
38981
38982
38983
38984
38985
38986
38987
38988
38989
38990
38991
38992
38993
38994
38995
38996
38997
38998
38999
39000
39001
39002
39003
39004
39005
39006
39007
39008
39009
39010
39011
39012
39013
39014
39015
39016
39017
39018
39019
39020
39021
39022
39023
39024
39025
39026
39027
39028
39029
39030
39031
39032
39033
39034
39035
39036
39037
39038
39039
39040
39041
39042
39043
39044
39045
39046
39047
39048
39049
39050
39051
39052
39053
39054
39055
39056
39057
39058
39059
39060
39061
39062
39063
39064
39065
39066
39067
39068
39069
39070
39071
39072
39073
39074
39075
39076
39077
39078
39079
39080
39081
39082
39083
39084
39085
39086
39087
39088
39089
39090
39091
39092
39093
39094
39095
39096
39097
39098
39099
39100
39101
39102
39103
39104
39105
39106
39107
39108
39109
39110
39111
39112
39113
39114
39115
39116
39117
39118
39119
39120
39121
39122
39123
39124
39125
39126
39127
39128
39129
39130
39131
39132
39133
39134
39135
39136
39137
39138
39139
39140
39141
39142
39143
39144
39145
39146
39147
39148
39149
39150
39151
39152
39153
39154
39155
39156
39157
39158
39159
39160
39161
39162
39163
39164
39165
39166
39167
39168
39169
39170
39171
39172
39173
39174
39175
39176
39177
39178
39179
39180
39181
39182
39183
39184
39185
39186
39187
39188
39189
39190
39191
39192
39193
39194
39195
39196
39197
39198
39199
39200
39201
39202
39203
39204
39205
39206
39207
39208
39209
39210
39211
39212
39213
39214
39215
39216
39217
39218
39219
39220
39221
39222
39223
39224
39225
39226
39227
39228
39229
39230
39231
39232
39233
39234
39235
39236
39237
39238
39239
39240
39241
39242
39243
39244
39245
39246
39247
39248
39249
39250
39251
39252
39253
39254
39255
39256
39257
39258
39259
39260
39261
39262
39263
39264
39265
39266
39267
39268
39269
39270
39271
39272
39273
39274
39275
39276
39277
39278
39279
39280
39281
39282
39283
39284
39285
39286
39287
39288
39289
39290
39291
39292
39293
39294
39295
39296
39297
39298
39299
39300
39301
39302
39303
39304
39305
39306
39307
39308
39309
39310
39311
39312
39313
39314
39315
39316
39317
39318
39319
39320
39321
39322
39323
39324
39325
39326
39327
39328
39329
39330
39331
39332
39333
39334
39335
39336
39337
39338
39339
39340
39341
39342
39343
39344
39345
39346
39347
39348
39349
39350
39351
39352
39353
39354
39355
39356
39357
39358
39359
39360
39361
39362
39363
39364
39365
39366
39367
39368
39369
39370
39371
39372
39373
39374
39375
39376
39377
39378
39379
39380
39381
39382
39383
39384
39385
39386
39387
39388
39389
39390
39391
39392
39393
39394
39395
39396
39397
39398
39399
39400
39401
39402
39403
39404
39405
39406
39407
39408
39409
39410
39411
39412
39413
39414
39415
39416
39417
39418
39419
39420
39421
39422
39423
39424
39425
39426
39427
39428
39429
39430
39431
39432
39433
39434
39435
39436
39437
39438
39439
39440
39441
39442
39443
39444
39445
39446
39447
39448
39449
39450
39451
39452
39453
39454
39455
39456
39457
39458
39459
39460
39461
39462
39463
39464
39465
39466
39467
39468
39469
39470
39471
39472
39473
39474
39475
39476
39477
39478
39479
39480
39481
39482
39483
39484
39485
39486
39487
39488
39489
39490
39491
39492
39493
39494
39495
39496
39497
39498
39499
39500
39501
39502
39503
39504
39505
39506
39507
39508
39509
39510
39511
39512
39513
39514
39515
39516
39517
39518
39519
39520
39521
39522
39523
39524
39525
39526
39527
39528
39529
39530
39531
39532
39533
39534
39535
39536
39537
39538
39539
39540
39541
39542
39543
39544
39545
39546
39547
39548
39549
39550
39551
39552
39553
39554
39555
39556
39557
39558
39559
39560
39561
39562
39563
39564
39565
39566
39567
39568
39569
39570
39571
39572
39573
39574
39575
39576
39577
39578
39579
39580
39581
39582
39583
39584
39585
39586
39587
39588
39589
39590
39591
39592
39593
39594
39595
39596
39597
39598
39599
39600
39601
39602
39603
39604
39605
39606
39607
39608
39609
39610
39611
39612
39613
39614
39615
39616
39617
39618
39619
39620
39621
39622
39623
39624
39625
39626
39627
39628
39629
39630
39631
39632
39633
39634
39635
39636
39637
39638
39639
39640
39641
39642
39643
39644
39645
39646
39647
39648
39649
39650
39651
39652
39653
39654
39655
39656
39657
39658
39659
39660
39661
39662
39663
39664
39665
39666
39667
39668
39669
39670
39671
39672
39673
39674
39675
39676
39677
39678
39679
39680
39681
39682
39683
39684
39685
39686
39687
39688
39689
39690
39691
39692
39693
39694
39695
39696
39697
39698
39699
39700
39701
39702
39703
39704
39705
39706
39707
39708
39709
39710
39711
39712
39713
39714
39715
39716
39717
39718
39719
39720
39721
39722
39723
39724
39725
39726
39727
39728
39729
39730
39731
39732
39733
39734
39735
39736
39737
39738
39739
39740
39741
39742
39743
39744
39745
39746
39747
39748
39749
39750
39751
39752
39753
39754
39755
39756
39757
39758
39759
39760
39761
39762
39763
39764
39765
39766
39767
39768
39769
39770
39771
39772
39773
39774
39775
39776
39777
39778
39779
39780
39781
39782
39783
39784
39785
39786
39787
39788
39789
39790
39791
39792
39793
39794
39795
39796
39797
39798
39799
39800
39801
39802
39803
39804
39805
39806
39807
39808
39809
39810
39811
39812
39813
39814
39815
39816
39817
39818
39819
39820
39821
39822
39823
39824
39825
39826
39827
39828
39829
39830
39831
39832
39833
39834
39835
39836
39837
39838
39839
39840
39841
39842
39843
39844
39845
39846
39847
39848
39849
39850
39851
39852
39853
39854
39855
39856
39857
39858
39859
39860
39861
39862
39863
39864
39865
39866
39867
39868
39869
39870
39871
39872
39873
39874
39875
39876
39877
39878
39879
39880
39881
39882
39883
39884
39885
39886
39887
39888
39889
39890
39891
39892
39893
39894
39895
39896
39897
39898
39899
39900
39901
39902
39903
39904
39905
39906
39907
39908
39909
39910
39911
39912
39913
39914
39915
39916
39917
39918
39919
39920
39921
39922
39923
39924
39925
39926
39927
39928
39929
39930
39931
39932
39933
39934
39935
39936
39937
39938
39939
39940
39941
39942
39943
39944
39945
39946
39947
39948
39949
39950
39951
39952
39953
39954
39955
39956
39957
39958
39959
39960
39961
39962
39963
39964
39965
39966
39967
39968
39969
39970
39971
39972
39973
39974
39975
39976
39977
39978
39979
39980
39981
39982
39983
39984
39985
39986
39987
39988
39989
39990
39991
39992
39993
39994
39995
39996
39997
39998
39999
40000
40001
40002
40003
40004
40005
40006
40007
40008
40009
40010
40011
40012
40013
40014
40015
40016
40017
40018
40019
40020
40021
40022
40023
40024
40025
40026
40027
40028
40029
40030
40031
40032
40033
40034
40035
40036
40037
40038
40039
40040
40041
40042
40043
40044
40045
40046
40047
40048
40049
40050
40051
40052
40053
40054
40055
40056
40057
40058
40059
40060
40061
40062
40063
40064
40065
40066
40067
40068
40069
40070
40071
40072
40073
40074
40075
40076
40077
40078
40079
40080
40081
40082
40083
40084
40085
40086
40087
40088
40089
40090
40091
40092
40093
40094
40095
40096
40097
40098
40099
40100
40101
40102
40103
40104
40105
40106
40107
40108
40109
40110
40111
40112
40113
40114
40115
40116
40117
40118
40119
40120
40121
40122
40123
40124
40125
40126
40127
40128
40129
40130
40131
40132
40133
40134
40135
40136
40137
40138
40139
40140
40141
40142
40143
40144
40145
40146
40147
40148
40149
40150
40151
40152
40153
40154
40155
40156
40157
40158
40159
40160
40161
40162
40163
40164
40165
40166
40167
40168
40169
40170
40171
40172
40173
40174
40175
40176
40177
40178
40179
40180
40181
40182
40183
40184
40185
40186
40187
40188
40189
40190
40191
40192
40193
40194
40195
40196
40197
40198
40199
40200
40201
40202
40203
40204
40205
40206
40207
40208
40209
40210
40211
40212
40213
40214
40215
40216
40217
40218
40219
40220
40221
40222
40223
40224
40225
40226
40227
40228
40229
40230
40231
40232
40233
40234
40235
40236
40237
40238
40239
40240
40241
40242
40243
40244
40245
40246
40247
40248
40249
40250
40251
40252
40253
40254
40255
40256
40257
40258
40259
40260
40261
40262
40263
40264
40265
40266
40267
40268
40269
40270
40271
40272
40273
40274
40275
40276
40277
40278
40279
40280
40281
40282
40283
40284
40285
40286
40287
40288
40289
40290
40291
40292
40293
40294
40295
40296
40297
40298
40299
40300
40301
40302
40303
40304
40305
40306
40307
40308
40309
40310
40311
40312
40313
40314
40315
40316
40317
40318
40319
40320
40321
40322
40323
40324
40325
40326
40327
40328
40329
40330
40331
40332
40333
40334
40335
40336
40337
40338
40339
40340
40341
40342
40343
40344
40345
40346
40347
40348
40349
40350
40351
40352
40353
40354
40355
40356
40357
40358
40359
40360
40361
40362
40363
40364
40365
40366
40367
40368
40369
40370
40371
40372
40373
40374
40375
40376
40377
40378
40379
40380
40381
40382
40383
40384
40385
40386
40387
40388
40389
40390
40391
40392
40393
40394
40395
40396
40397
40398
40399
40400
40401
40402
40403
40404
40405
40406
40407
40408
40409
40410
40411
40412
40413
40414
40415
40416
40417
40418
40419
40420
40421
40422
40423
40424
40425
40426
40427
40428
40429
40430
40431
40432
40433
40434
40435
40436
40437
40438
40439
40440
40441
40442
40443
40444
40445
40446
40447
40448
40449
40450
40451
40452
40453
40454
40455
40456
40457
40458
40459
40460
40461
40462
40463
40464
40465
40466
40467
40468
40469
40470
40471
40472
40473
40474
40475
40476
40477
40478
40479
40480
40481
40482
40483
40484
40485
40486
40487
40488
40489
40490
40491
40492
40493
40494
40495
40496
40497
40498
40499
40500
40501
40502
40503
40504
40505
40506
40507
40508
40509
40510
40511
40512
40513
40514
40515
40516
40517
40518
40519
40520
40521
40522
40523
40524
40525
40526
40527
40528
40529
40530
40531
40532
40533
40534
40535
40536
40537
40538
40539
40540
40541
40542
40543
40544
40545
40546
40547
40548
40549
40550
40551
40552
40553
40554
40555
40556
40557
40558
40559
40560
40561
40562
40563
40564
40565
40566
40567
40568
40569
40570
40571
40572
40573
40574
40575
40576
40577
40578
40579
40580
40581
40582
40583
40584
40585
40586
40587
40588
40589
40590
40591
40592
40593
40594
40595
40596
40597
40598
40599
40600
40601
40602
40603
40604
40605
40606
40607
40608
40609
40610
40611
40612
40613
40614
40615
40616
40617
40618
40619
40620
40621
40622
40623
40624
40625
40626
40627
40628
40629
40630
40631
40632
40633
40634
40635
40636
40637
40638
40639
40640
40641
40642
40643
40644
40645
40646
40647
40648
40649
40650
40651
40652
40653
40654
40655
40656
40657
40658
40659
40660
40661
40662
40663
40664
40665
40666
40667
40668
40669
40670
40671
40672
40673
40674
40675
40676
40677
40678
40679
40680
40681
40682
40683
40684
40685
40686
40687
40688
40689
40690
40691
40692
40693
40694
40695
40696
40697
40698
40699
40700
40701
40702
40703
40704
40705
40706
40707
40708
40709
40710
40711
40712
40713
40714
40715
40716
40717
40718
40719
40720
40721
40722
40723
40724
40725
40726
40727
40728
40729
40730
40731
40732
40733
40734
40735
40736
40737
40738
40739
40740
40741
40742
40743
40744
40745
40746
40747
40748
40749
40750
40751
40752
40753
40754
40755
40756
40757
40758
40759
40760
40761
40762
40763
40764
40765
40766
40767
40768
40769
40770
40771
40772
40773
40774
40775
40776
40777
40778
40779
40780
40781
40782
40783
40784
40785
40786
40787
40788
40789
40790
40791
40792
40793
40794
40795
40796
40797
40798
40799
40800
40801
40802
40803
40804
40805
40806
40807
40808
40809
40810
40811
40812
40813
40814
40815
40816
40817
40818
40819
40820
40821
40822
40823
40824
40825
40826
40827
40828
40829
40830
40831
40832
40833
40834
40835
40836
40837
40838
40839
40840
40841
40842
40843
40844
40845
40846
40847
40848
40849
40850
40851
40852
40853
40854
40855
40856
40857
40858
40859
40860
40861
40862
40863
40864
40865
40866
40867
40868
40869
40870
40871
40872
40873
40874
40875
40876
40877
40878
40879
40880
40881
40882
40883
40884
40885
40886
40887
40888
40889
40890
40891
40892
40893
40894
40895
40896
40897
40898
40899
40900
40901
40902
40903
40904
40905
40906
40907
40908
40909
40910
40911
40912
40913
40914
40915
40916
40917
40918
40919
40920
40921
40922
40923
40924
40925
40926
40927
40928
40929
40930
40931
40932
40933
40934
40935
40936
40937
40938
40939
40940
40941
40942
40943
40944
40945
40946
40947
40948
40949
40950
40951
40952
40953
40954
40955
40956
40957
40958
40959
40960
40961
40962
40963
40964
40965
40966
40967
40968
40969
40970
40971
40972
40973
40974
40975
40976
40977
40978
40979
40980
40981
40982
40983
40984
40985
40986
40987
40988
40989
40990
40991
40992
40993
40994
40995
40996
40997
40998
40999
41000
41001
41002
41003
41004
41005
41006
41007
41008
41009
41010
41011
41012
41013
41014
41015
41016
41017
41018
41019
41020
41021
41022
41023
41024
41025
41026
41027
41028
41029
41030
41031
41032
41033
41034
41035
41036
41037
41038
41039
41040
41041
41042
41043
41044
41045
41046
41047
41048
41049
41050
41051
41052
41053
41054
41055
41056
41057
41058
41059
41060
41061
41062
41063
41064
41065
41066
41067
41068
41069
41070
41071
41072
41073
41074
41075
41076
41077
41078
41079
41080
41081
41082
41083
41084
41085
41086
41087
41088
41089
41090
41091
41092
41093
41094
41095
41096
41097
41098
41099
41100
41101
41102
41103
41104
41105
41106
41107
41108
41109
41110
41111
41112
41113
41114
41115
41116
41117
41118
41119
41120
41121
41122
41123
41124
41125
41126
41127
41128
41129
41130
41131
41132
41133
41134
41135
41136
41137
41138
41139
41140
41141
41142
41143
41144
41145
41146
41147
41148
41149
41150
41151
41152
41153
41154
41155
41156
41157
41158
41159
41160
41161
41162
41163
41164
41165
41166
41167
41168
41169
41170
41171
41172
41173
41174
41175
41176
41177
41178
41179
41180
41181
41182
41183
41184
41185
41186
41187
41188
41189
41190
41191
41192
41193
41194
41195
41196
41197
41198
41199
41200
41201
41202
41203
41204
41205
41206
41207
41208
41209
41210
41211
41212
41213
41214
41215
41216
41217
41218
41219
41220
41221
41222
41223
41224
41225
41226
41227
41228
41229
41230
41231
41232
41233
41234
41235
41236
41237
41238
41239
41240
41241
41242
41243
41244
41245
41246
41247
41248
41249
41250
41251
41252
41253
41254
41255
41256
41257
41258
41259
41260
41261
41262
41263
41264
41265
41266
41267
41268
41269
41270
41271
41272
41273
41274
41275
41276
41277
41278
41279
41280
41281
41282
41283
41284
41285
41286
41287
41288
41289
41290
41291
41292
41293
41294
41295
41296
41297
41298
41299
41300
41301
41302
41303
41304
41305
41306
41307
41308
41309
41310
41311
41312
41313
41314
41315
41316
41317
41318
41319
41320
41321
41322
41323
41324
41325
41326
41327
41328
41329
41330
41331
41332
41333
41334
41335
41336
41337
41338
41339
41340
41341
41342
41343
41344
41345
41346
41347
41348
41349
41350
41351
41352
41353
41354
41355
41356
41357
41358
41359
41360
41361
41362
41363
41364
41365
41366
41367
41368
41369
41370
41371
41372
41373
41374
41375
41376
41377
41378
41379
41380
41381
41382
41383
41384
41385
41386
41387
41388
41389
41390
41391
41392
41393
41394
41395
41396
41397
41398
41399
41400
41401
41402
41403
41404
41405
41406
41407
41408
41409
41410
41411
41412
41413
41414
41415
41416
41417
41418
41419
41420
41421
41422
41423
41424
41425
41426
41427
41428
41429
41430
41431
41432
41433
41434
41435
41436
41437
41438
41439
41440
41441
41442
41443
41444
41445
41446
41447
41448
41449
41450
41451
41452
41453
41454
41455
41456
41457
41458
41459
41460
41461
41462
41463
41464
41465
41466
41467
41468
41469
41470
41471
41472
41473
41474
41475
41476
41477
41478
41479
41480
41481
41482
41483
41484
41485
41486
41487
41488
41489
41490
41491
41492
41493
41494
41495
41496
41497
41498
41499
41500
41501
41502
41503
41504
41505
41506
41507
41508
41509
41510
41511
41512
41513
41514
41515
41516
41517
41518
41519
41520
41521
41522
41523
41524
41525
41526
41527
41528
41529
41530
41531
41532
41533
41534
41535
41536
41537
41538
41539
41540
41541
41542
41543
41544
41545
41546
41547
41548
41549
41550
41551
41552
41553
41554
41555
41556
41557
41558
41559
41560
41561
41562
41563
41564
41565
41566
41567
41568
41569
41570
41571
41572
41573
41574
41575
41576
41577
41578
41579
41580
41581
41582
41583
41584
41585
41586
41587
41588
41589
41590
41591
41592
41593
41594
41595
41596
41597
41598
41599
41600
41601
41602
41603
41604
41605
41606
41607
41608
41609
41610
41611
41612
41613
41614
41615
41616
41617
41618
41619
41620
41621
41622
41623
41624
41625
41626
41627
41628
41629
41630
41631
41632
41633
41634
41635
41636
41637
41638
41639
41640
41641
41642
41643
41644
41645
41646
41647
41648
41649
41650
41651
41652
41653
41654
41655
41656
41657
41658
41659
41660
41661
41662
41663
41664
41665
41666
41667
41668
41669
41670
41671
41672
41673
41674
41675
41676
41677
41678
41679
41680
41681
41682
41683
41684
41685
41686
41687
41688
41689
41690
41691
41692
41693
41694
41695
41696
41697
41698
41699
41700
41701
41702
41703
41704
41705
41706
41707
41708
41709
41710
41711
41712
41713
41714
41715
41716
41717
41718
41719
41720
41721
41722
41723
41724
41725
41726
41727
41728
41729
41730
41731
41732
41733
41734
41735
41736
41737
41738
41739
41740
41741
41742
41743
41744
41745
41746
41747
41748
41749
41750
41751
41752
41753
41754
41755
41756
41757
41758
41759
41760
41761
41762
41763
41764
41765
41766
41767
41768
41769
41770
41771
41772
41773
41774
41775
41776
41777
41778
41779
41780
41781
41782
41783
41784
41785
41786
41787
41788
41789
41790
41791
41792
41793
41794
41795
41796
41797
41798
41799
41800
41801
41802
41803
41804
41805
41806
41807
41808
41809
41810
41811
41812
41813
41814
41815
41816
41817
41818
41819
41820
41821
41822
41823
41824
41825
41826
41827
41828
41829
41830
41831
41832
41833
41834
41835
41836
41837
41838
41839
41840
41841
41842
41843
41844
41845
41846
41847
41848
41849
41850
41851
41852
41853
41854
41855
41856
41857
41858
41859
41860
41861
41862
41863
41864
41865
41866
41867
41868
41869
41870
41871
41872
41873
41874
41875
41876
41877
41878
41879
41880
41881
41882
41883
41884
41885
41886
41887
41888
41889
41890
41891
41892
41893
41894
41895
41896
41897
41898
41899
41900
41901
41902
41903
41904
41905
41906
41907
41908
41909
41910
41911
41912
41913
41914
41915
41916
41917
41918
41919
41920
41921
41922
41923
41924
41925
41926
41927
41928
41929
41930
41931
41932
41933
41934
41935
41936
41937
41938
41939
41940
41941
41942
41943
41944
41945
41946
41947
41948
41949
41950
41951
41952
41953
41954
41955
41956
41957
41958
41959
41960
41961
41962
41963
41964
41965
41966
41967
41968
41969
41970
41971
41972
41973
41974
41975
41976
41977
41978
41979
41980
41981
41982
41983
41984
41985
41986
41987
41988
41989
41990
41991
41992
41993
41994
41995
41996
41997
41998
41999
42000
42001
42002
42003
42004
42005
42006
42007
42008
42009
42010
42011
42012
42013
42014
42015
42016
42017
42018
42019
42020
42021
42022
42023
42024
42025
42026
42027
42028
42029
42030
42031
42032
42033
42034
42035
42036
42037
42038
42039
42040
42041
42042
42043
42044
42045
42046
42047
42048
42049
42050
42051
42052
42053
42054
42055
42056
42057
42058
42059
42060
42061
42062
42063
42064
42065
42066
42067
42068
42069
42070
42071
42072
42073
42074
42075
42076
42077
42078
42079
42080
42081
42082
42083
42084
42085
42086
42087
42088
42089
42090
42091
42092
42093
42094
42095
42096
42097
42098
42099
42100
42101
42102
42103
42104
42105
42106
42107
42108
42109
42110
42111
42112
42113
42114
42115
42116
42117
42118
42119
42120
42121
42122
42123
42124
42125
42126
42127
42128
42129
42130
42131
42132
42133
42134
42135
42136
42137
42138
42139
42140
42141
42142
42143
42144
42145
42146
42147
42148
42149
42150
42151
42152
42153
42154
42155
42156
42157
42158
42159
42160
42161
42162
42163
42164
42165
42166
42167
42168
42169
42170
42171
42172
42173
42174
42175
42176
42177
42178
42179
42180
42181
42182
42183
42184
42185
42186
42187
42188
42189
42190
42191
42192
42193
42194
42195
42196
42197
42198
42199
42200
42201
42202
42203
42204
42205
42206
42207
42208
42209
42210
42211
42212
42213
42214
42215
42216
42217
42218
42219
42220
42221
42222
42223
42224
42225
42226
42227
42228
42229
42230
42231
42232
42233
42234
42235
42236
42237
42238
42239
42240
42241
42242
42243
42244
42245
42246
42247
42248
42249
42250
42251
42252
42253
42254
42255
42256
42257
42258
42259
42260
42261
42262
42263
42264
42265
42266
42267
42268
42269
42270
42271
42272
42273
42274
42275
42276
42277
42278
42279
42280
42281
42282
42283
42284
42285
42286
42287
42288
42289
42290
42291
42292
42293
42294
42295
42296
42297
42298
42299
42300
42301
42302
42303
42304
42305
42306
42307
42308
42309
42310
42311
42312
42313
42314
42315
42316
42317
42318
42319
42320
42321
42322
42323
42324
42325
42326
42327
42328
42329
42330
42331
42332
42333
42334
42335
42336
42337
42338
42339
42340
42341
42342
42343
42344
42345
42346
42347
42348
42349
42350
42351
42352
42353
42354
42355
42356
42357
42358
42359
42360
42361
42362
42363
42364
42365
42366
42367
42368
42369
42370
42371
42372
42373
42374
42375
42376
42377
42378
42379
42380
42381
42382
42383
42384
42385
42386
42387
42388
42389
42390
42391
42392
42393
42394
42395
42396
42397
42398
42399
42400
42401
42402
42403
42404
42405
42406
42407
42408
42409
42410
42411
42412
42413
42414
42415
42416
42417
42418
42419
42420
42421
42422
42423
42424
42425
42426
42427
42428
42429
42430
42431
42432
42433
42434
42435
42436
42437
42438
42439
42440
42441
42442
42443
42444
42445
42446
42447
42448
42449
42450
42451
42452
42453
42454
42455
42456
42457
42458
42459
42460
42461
42462
42463
42464
42465
42466
42467
42468
42469
42470
42471
42472
42473
42474
42475
42476
42477
42478
42479
42480
42481
42482
42483
42484
42485
42486
42487
42488
42489
42490
42491
42492
42493
42494
42495
42496
42497
42498
42499
42500
42501
42502
42503
42504
42505
42506
42507
42508
42509
42510
42511
42512
42513
42514
42515
42516
42517
42518
42519
42520
42521
42522
42523
42524
42525
42526
42527
42528
42529
42530
42531
42532
42533
42534
42535
42536
42537
42538
42539
42540
42541
42542
42543
42544
42545
42546
42547
42548
42549
42550
42551
42552
42553
42554
42555
42556
42557
42558
42559
42560
42561
42562
42563
42564
42565
42566
42567
42568
42569
42570
42571
42572
42573
42574
42575
42576
42577
42578
42579
42580
42581
42582
42583
42584
42585
42586
42587
42588
42589
42590
42591
42592
42593
42594
42595
42596
42597
42598
42599
42600
42601
42602
42603
42604
42605
42606
42607
42608
42609
42610
42611
42612
42613
42614
42615
42616
42617
42618
42619
42620
42621
42622
42623
42624
42625
42626
42627
42628
42629
42630
42631
42632
42633
42634
42635
42636
42637
42638
42639
42640
42641
42642
42643
42644
42645
42646
42647
42648
42649
42650
42651
42652
42653
42654
42655
42656
42657
42658
42659
42660
42661
42662
42663
42664
42665
42666
42667
42668
42669
42670
42671
42672
42673
42674
42675
42676
42677
42678
42679
42680
42681
42682
42683
42684
42685
42686
42687
42688
42689
42690
42691
42692
42693
42694
42695
42696
42697
42698
42699
42700
42701
42702
42703
42704
42705
42706
42707
42708
42709
42710
42711
42712
42713
42714
42715
42716
42717
42718
42719
42720
42721
42722
42723
42724
42725
42726
42727
42728
42729
42730
42731
42732
42733
42734
42735
42736
42737
42738
42739
42740
42741
42742
42743
42744
42745
42746
42747
42748
42749
42750
42751
42752
42753
42754
42755
42756
42757
42758
42759
42760
42761
42762
42763
42764
42765
42766
42767
42768
42769
42770
42771
42772
42773
42774
42775
42776
42777
42778
42779
42780
42781
42782
42783
42784
42785
42786
42787
42788
42789
42790
42791
42792
42793
42794
42795
42796
42797
42798
42799
42800
42801
42802
42803
42804
42805
42806
42807
42808
42809
42810
42811
42812
42813
42814
42815
42816
42817
42818
42819
42820
42821
42822
42823
42824
42825
42826
42827
42828
42829
42830
42831
42832
42833
42834
42835
42836
42837
42838
42839
42840
42841
42842
42843
42844
42845
42846
42847
42848
42849
42850
42851
42852
42853
42854
42855
42856
42857
42858
42859
42860
42861
42862
42863
42864
42865
42866
42867
42868
42869
42870
42871
42872
42873
42874
42875
42876
42877
42878
42879
42880
42881
42882
42883
42884
42885
42886
42887
42888
42889
42890
42891
42892
42893
42894
42895
42896
42897
42898
42899
42900
42901
42902
42903
42904
42905
42906
42907
42908
42909
42910
42911
42912
42913
42914
42915
42916
42917
42918
42919
42920
42921
42922
42923
42924
42925
42926
42927
42928
42929
42930
42931
42932
42933
42934
42935
42936
42937
42938
42939
42940
42941
42942
42943
42944
42945
42946
42947
42948
42949
42950
42951
42952
42953
42954
42955
42956
42957
42958
42959
42960
42961
42962
42963
42964
42965
42966
42967
42968
42969
42970
42971
42972
42973
42974
42975
42976
42977
42978
42979
42980
42981
42982
42983
42984
42985
42986
42987
42988
42989
42990
42991
42992
42993
42994
42995
42996
42997
42998
42999
43000
43001
43002
43003
43004
43005
43006
43007
43008
43009
43010
43011
43012
43013
43014
43015
43016
43017
43018
43019
43020
43021
43022
43023
43024
43025
43026
43027
43028
43029
43030
43031
43032
43033
43034
43035
43036
43037
43038
43039
43040
43041
43042
43043
43044
43045
43046
43047
43048
43049
43050
43051
43052
43053
43054
43055
43056
43057
43058
43059
43060
43061
43062
43063
43064
43065
43066
43067
43068
43069
43070
43071
43072
43073
43074
43075
43076
43077
43078
43079
43080
43081
43082
43083
43084
43085
43086
43087
43088
43089
43090
43091
43092
43093
43094
43095
43096
43097
43098
43099
43100
43101
43102
43103
43104
43105
43106
43107
43108
43109
43110
43111
43112
43113
43114
43115
43116
43117
43118
43119
43120
43121
43122
43123
43124
43125
43126
43127
43128
43129
43130
43131
43132
43133
43134
43135
43136
43137
43138
43139
43140
43141
43142
43143
43144
43145
43146
43147
43148
43149
43150
43151
43152
43153
43154
43155
43156
43157
43158
43159
43160
43161
43162
43163
43164
43165
43166
43167
43168
43169
43170
43171
43172
43173
43174
43175
43176
43177
43178
43179
43180
43181
43182
43183
43184
43185
43186
43187
43188
43189
43190
43191
43192
43193
43194
43195
43196
43197
43198
43199
43200
43201
43202
43203
43204
43205
43206
43207
43208
43209
43210
43211
43212
43213
43214
43215
43216
43217
43218
43219
43220
43221
43222
43223
43224
43225
43226
43227
43228
43229
43230
43231
43232
43233
43234
43235
43236
43237
43238
43239
43240
43241
43242
43243
43244
43245
43246
43247
43248
43249
43250
43251
43252
43253
43254
43255
43256
43257
43258
43259
43260
43261
43262
43263
43264
43265
43266
43267
43268
43269
43270
43271
43272
43273
43274
43275
43276
43277
43278
43279
43280
43281
43282
43283
43284
43285
43286
43287
43288
43289
43290
43291
43292
43293
43294
43295
43296
43297
43298
43299
43300
43301
43302
43303
43304
43305
43306
43307
43308
43309
43310
43311
43312
43313
43314
43315
43316
43317
43318
43319
43320
43321
43322
43323
43324
43325
43326
43327
43328
43329
43330
43331
43332
43333
43334
43335
43336
43337
43338
43339
43340
43341
43342
43343
43344
43345
43346
43347
43348
43349
43350
43351
43352
43353
43354
43355
43356
43357
43358
43359
43360
43361
43362
43363
43364
43365
43366
43367
43368
43369
43370
43371
43372
43373
43374
43375
43376
43377
43378
43379
43380
43381
43382
43383
43384
43385
43386
43387
43388
43389
43390
43391
43392
43393
43394
43395
43396
43397
43398
43399
43400
43401
43402
43403
43404
43405
43406
43407
43408
43409
43410
43411
43412
43413
43414
43415
43416
43417
43418
43419
43420
43421
43422
43423
43424
43425
43426
43427
43428
43429
43430
43431
43432
43433
43434
43435
43436
43437
43438
43439
43440
43441
43442
43443
43444
43445
43446
43447
43448
43449
43450
43451
43452
43453
43454
43455
43456
43457
43458
43459
43460
43461
43462
43463
43464
43465
43466
43467
43468
43469
43470
43471
43472
43473
43474
43475
43476
43477
43478
43479
43480
43481
43482
43483
43484
43485
43486
43487
43488
43489
43490
43491
43492
43493
43494
43495
43496
43497
43498
43499
43500
43501
43502
43503
43504
43505
43506
43507
43508
43509
43510
43511
43512
43513
43514
43515
43516
43517
43518
43519
43520
43521
43522
43523
43524
43525
43526
43527
43528
43529
43530
43531
43532
43533
43534
43535
43536
43537
43538
43539
43540
43541
43542
43543
43544
43545
43546
43547
43548
43549
43550
43551
43552
43553
43554
43555
43556
43557
43558
43559
43560
43561
43562
43563
43564
43565
43566
43567
43568
43569
43570
43571
43572
43573
43574
43575
43576
43577
43578
43579
43580
43581
43582
43583
43584
43585
43586
43587
43588
43589
43590
43591
43592
43593
43594
43595
43596
43597
43598
43599
43600
43601
43602
43603
43604
43605
43606
43607
43608
43609
43610
43611
43612
43613
43614
43615
43616
43617
43618
43619
43620
43621
43622
43623
43624
43625
43626
43627
43628
43629
43630
43631
43632
43633
43634
43635
43636
43637
43638
43639
43640
43641
43642
43643
43644
43645
43646
43647
43648
43649
43650
43651
43652
43653
43654
43655
43656
43657
43658
43659
43660
43661
43662
43663
43664
43665
43666
43667
43668
43669
43670
43671
43672
43673
43674
43675
43676
43677
43678
43679
43680
43681
43682
43683
43684
43685
43686
43687
43688
43689
43690
43691
43692
43693
43694
43695
43696
43697
43698
43699
43700
43701
43702
43703
43704
43705
43706
43707
43708
43709
43710
43711
43712
43713
43714
43715
43716
43717
43718
43719
43720
43721
43722
43723
43724
43725
43726
43727
43728
43729
43730
43731
43732
43733
43734
43735
43736
43737
43738
43739
43740
43741
43742
43743
43744
43745
43746
43747
43748
43749
43750
43751
43752
43753
43754
43755
43756
43757
43758
43759
43760
43761
43762
43763
43764
43765
43766
43767
43768
43769
43770
43771
43772
43773
43774
43775
43776
43777
43778
43779
43780
43781
43782
43783
43784
43785
43786
43787
43788
43789
43790
43791
43792
43793
43794
43795
43796
43797
43798
43799
43800
43801
43802
43803
43804
43805
43806
43807
43808
43809
43810
43811
43812
43813
43814
43815
43816
43817
43818
43819
43820
43821
43822
43823
43824
43825
43826
43827
43828
43829
43830
43831
43832
43833
43834
43835
43836
43837
43838
43839
43840
43841
43842
43843
43844
43845
43846
43847
43848
43849
43850
43851
43852
43853
43854
43855
43856
43857
43858
43859
43860
43861
43862
43863
43864
43865
43866
43867
43868
43869
43870
43871
43872
43873
43874
43875
43876
43877
43878
43879
43880
43881
43882
43883
43884
43885
43886
43887
43888
43889
43890
43891
43892
43893
43894
43895
43896
43897
43898
43899
43900
43901
43902
43903
43904
43905
43906
43907
43908
43909
43910
43911
43912
43913
43914
43915
43916
43917
43918
43919
43920
43921
43922
43923
43924
43925
43926
43927
43928
43929
43930
43931
43932
43933
43934
43935
43936
43937
43938
43939
43940
43941
43942
43943
43944
43945
43946
43947
43948
43949
43950
43951
43952
43953
43954
43955
43956
43957
43958
43959
43960
43961
43962
43963
43964
43965
43966
43967
43968
43969
43970
43971
43972
43973
43974
43975
43976
43977
43978
43979
43980
43981
43982
43983
43984
43985
43986
43987
43988
43989
43990
43991
43992
43993
43994
43995
43996
43997
43998
43999
44000
44001
44002
44003
44004
44005
44006
44007
44008
44009
44010
44011
44012
44013
44014
44015
44016
44017
44018
44019
44020
44021
44022
44023
44024
44025
44026
44027
44028
44029
44030
44031
44032
44033
44034
44035
44036
44037
44038
44039
44040
44041
44042
44043
44044
44045
44046
44047
44048
44049
44050
44051
44052
44053
44054
44055
44056
44057
44058
44059
44060
44061
44062
44063
44064
44065
44066
44067
44068
44069
44070
44071
44072
44073
44074
44075
44076
44077
44078
44079
44080
44081
44082
44083
44084
44085
44086
44087
44088
44089
44090
44091
44092
44093
44094
44095
44096
44097
44098
44099
44100
44101
44102
44103
44104
44105
44106
44107
44108
44109
44110
44111
44112
44113
44114
44115
44116
44117
44118
44119
44120
44121
44122
44123
44124
44125
44126
44127
44128
44129
44130
44131
44132
44133
44134
44135
44136
44137
44138
44139
44140
44141
44142
44143
44144
44145
44146
44147
44148
44149
44150
44151
44152
44153
44154
44155
44156
44157
44158
44159
44160
44161
44162
44163
44164
44165
44166
44167
44168
44169
44170
44171
44172
44173
44174
44175
44176
44177
44178
44179
44180
44181
44182
44183
44184
44185
44186
44187
44188
44189
44190
44191
44192
44193
44194
44195
44196
44197
44198
44199
44200
44201
44202
44203
44204
44205
44206
44207
44208
44209
44210
44211
44212
44213
44214
44215
44216
44217
44218
44219
44220
44221
44222
44223
44224
44225
44226
44227
44228
44229
44230
44231
44232
44233
44234
44235
44236
44237
44238
44239
44240
44241
44242
44243
44244
44245
44246
44247
44248
44249
44250
44251
44252
44253
44254
44255
44256
44257
44258
44259
44260
44261
44262
44263
44264
44265
44266
44267
44268
44269
44270
44271
44272
44273
44274
44275
44276
44277
44278
44279
44280
44281
44282
44283
44284
44285
44286
44287
44288
44289
44290
44291
44292
44293
44294
44295
44296
44297
44298
44299
44300
44301
44302
44303
44304
44305
44306
44307
44308
44309
44310
44311
44312
44313
44314
44315
44316
44317
44318
44319
44320
44321
44322
44323
44324
44325
44326
44327
44328
44329
44330
44331
44332
44333
44334
44335
44336
44337
44338
44339
44340
44341
44342
44343
44344
44345
44346
44347
44348
44349
44350
44351
44352
44353
44354
44355
44356
44357
44358
44359
44360
44361
44362
44363
44364
44365
44366
44367
44368
44369
44370
44371
44372
44373
44374
44375
44376
44377
44378
44379
44380
44381
44382
44383
44384
44385
44386
44387
44388
44389
44390
44391
44392
44393
44394
44395
44396
44397
44398
44399
44400
44401
44402
44403
44404
44405
44406
44407
44408
44409
44410
44411
44412
44413
44414
44415
44416
44417
44418
44419
44420
44421
44422
44423
44424
44425
44426
44427
44428
44429
44430
44431
44432
44433
44434
44435
44436
44437
44438
44439
44440
44441
44442
44443
44444
44445
44446
44447
44448
44449
44450
44451
44452
44453
44454
44455
44456
44457
44458
44459
44460
44461
44462
44463
44464
44465
44466
44467
44468
44469
44470
44471
44472
44473
44474
44475
44476
44477
44478
44479
44480
44481
44482
44483
44484
44485
44486
44487
44488
44489
44490
44491
44492
44493
44494
44495
44496
44497
44498
44499
44500
44501
44502
44503
44504
44505
44506
44507
44508
44509
44510
44511
44512
44513
44514
44515
44516
44517
44518
44519
44520
44521
44522
44523
44524
44525
44526
44527
44528
44529
44530
44531
44532
44533
44534
44535
44536
44537
44538
44539
44540
44541
44542
44543
44544
44545
44546
44547
44548
44549
44550
44551
44552
44553
44554
44555
44556
44557
44558
44559
44560
44561
44562
44563
44564
44565
44566
44567
44568
44569
44570
44571
44572
44573
44574
44575
44576
44577
44578
44579
44580
44581
44582
44583
44584
44585
44586
44587
44588
44589
44590
44591
44592
44593
44594
44595
44596
44597
44598
44599
44600
44601
44602
44603
44604
44605
44606
44607
44608
44609
44610
44611
44612
44613
44614
44615
44616
44617
44618
44619
44620
44621
44622
44623
44624
44625
44626
44627
44628
44629
44630
44631
44632
44633
44634
44635
44636
44637
44638
44639
44640
44641
44642
44643
44644
44645
44646
44647
44648
44649
44650
44651
44652
44653
44654
44655
44656
44657
44658
44659
44660
44661
44662
44663
44664
44665
44666
44667
44668
44669
44670
44671
44672
44673
44674
44675
44676
44677
44678
44679
44680
44681
44682
44683
44684
44685
44686
44687
44688
44689
44690
44691
44692
44693
44694
44695
44696
44697
44698
44699
44700
44701
44702
44703
44704
44705
44706
44707
44708
44709
44710
44711
44712
44713
44714
44715
44716
44717
44718
44719
44720
44721
44722
44723
44724
44725
44726
44727
44728
44729
44730
44731
44732
44733
44734
44735
44736
44737
44738
44739
44740
44741
44742
44743
44744
44745
44746
44747
44748
44749
44750
44751
44752
44753
44754
44755
44756
44757
44758
44759
44760
44761
44762
44763
44764
44765
44766
44767
44768
44769
44770
44771
44772
44773
44774
44775
44776
44777
44778
44779
44780
44781
44782
44783
44784
44785
44786
44787
44788
44789
44790
44791
44792
44793
44794
44795
44796
44797
44798
44799
44800
44801
44802
44803
44804
44805
44806
44807
44808
44809
44810
44811
44812
44813
44814
44815
44816
44817
44818
44819
44820
44821
44822
44823
44824
44825
44826
44827
44828
44829
44830
44831
44832
44833
44834
44835
44836
44837
44838
44839
44840
44841
44842
44843
44844
44845
44846
44847
44848
44849
44850
44851
44852
44853
44854
44855
44856
44857
44858
44859
44860
44861
44862
44863
44864
44865
44866
44867
44868
44869
44870
44871
44872
44873
44874
44875
44876
44877
44878
44879
44880
44881
44882
44883
44884
44885
44886
44887
44888
44889
44890
44891
44892
44893
44894
44895
44896
44897
44898
44899
44900
44901
44902
44903
44904
44905
44906
44907
44908
44909
44910
44911
44912
44913
44914
44915
44916
44917
44918
44919
44920
44921
44922
44923
44924
44925
44926
44927
44928
44929
44930
44931
44932
44933
44934
44935
44936
44937
44938
44939
44940
44941
44942
44943
44944
44945
44946
44947
44948
44949
44950
44951
44952
44953
44954
44955
44956
44957
44958
44959
44960
44961
44962
44963
44964
44965
44966
44967
44968
44969
44970
44971
44972
44973
44974
44975
44976
44977
44978
44979
44980
44981
44982
44983
44984
44985
44986
44987
44988
44989
44990
44991
44992
44993
44994
44995
44996
44997
44998
44999
45000
45001
45002
45003
45004
45005
45006
45007
45008
45009
45010
45011
45012
45013
45014
45015
45016
45017
45018
45019
45020
45021
45022
45023
45024
45025
45026
45027
45028
45029
45030
45031
45032
45033
45034
45035
45036
45037
45038
45039
45040
45041
45042
45043
45044
45045
45046
45047
45048
45049
45050
45051
45052
45053
45054
45055
45056
45057
45058
45059
45060
45061
45062
45063
45064
45065
45066
45067
45068
45069
45070
45071
45072
45073
45074
45075
45076
45077
45078
45079
45080
45081
45082
45083
45084
45085
45086
45087
45088
45089
45090
45091
45092
45093
45094
45095
45096
45097
45098
45099
45100
45101
45102
45103
45104
45105
45106
45107
45108
45109
45110
45111
45112
45113
45114
45115
45116
45117
45118
45119
45120
45121
45122
45123
45124
45125
45126
45127
45128
45129
45130
45131
45132
45133
45134
45135
45136
45137
45138
45139
45140
45141
45142
45143
45144
45145
45146
45147
45148
45149
45150
45151
45152
45153
45154
45155
45156
45157
45158
45159
45160
45161
45162
45163
45164
45165
45166
45167
45168
45169
45170
45171
45172
45173
45174
45175
45176
45177
45178
45179
45180
45181
45182
45183
45184
45185
45186
45187
45188
45189
45190
45191
45192
45193
45194
45195
45196
45197
45198
45199
45200
45201
45202
45203
45204
45205
45206
45207
45208
45209
45210
45211
45212
45213
45214
45215
45216
45217
45218
45219
45220
45221
45222
45223
45224
45225
45226
45227
45228
45229
45230
45231
45232
45233
45234
45235
45236
45237
45238
45239
45240
45241
45242
45243
45244
45245
45246
45247
45248
45249
45250
45251
45252
45253
45254
45255
45256
45257
45258
45259
45260
45261
45262
45263
45264
45265
45266
45267
45268
45269
45270
45271
45272
45273
45274
45275
45276
45277
45278
45279
45280
45281
45282
45283
45284
45285
45286
45287
45288
45289
45290
45291
45292
45293
45294
45295
45296
45297
45298
45299
45300
45301
45302
45303
45304
45305
45306
45307
45308
45309
45310
45311
45312
45313
45314
45315
45316
45317
45318
45319
45320
45321
45322
45323
45324
45325
45326
45327
45328
45329
45330
45331
45332
45333
45334
45335
45336
45337
45338
45339
45340
45341
45342
45343
45344
45345
45346
45347
45348
45349
45350
45351
45352
45353
45354
45355
45356
45357
45358
45359
45360
45361
45362
45363
45364
45365
45366
45367
45368
45369
45370
45371
45372
45373
45374
45375
45376
45377
45378
45379
45380
45381
45382
45383
45384
45385
45386
45387
45388
45389
45390
45391
45392
45393
45394
45395
45396
45397
45398
45399
45400
45401
45402
45403
45404
45405
45406
45407
45408
45409
45410
45411
45412
45413
45414
45415
45416
45417
45418
45419
45420
45421
45422
45423
45424
45425
45426
45427
45428
45429
45430
45431
45432
45433
45434
45435
45436
45437
45438
45439
45440
45441
45442
45443
45444
45445
45446
45447
45448
45449
45450
45451
45452
45453
45454
45455
45456
45457
45458
45459
45460
45461
45462
45463
45464
45465
45466
45467
45468
45469
45470
45471
45472
45473
45474
45475
45476
45477
45478
45479
45480
45481
45482
45483
45484
45485
45486
45487
45488
45489
45490
45491
45492
45493
45494
45495
45496
45497
45498
45499
45500
45501
45502
45503
45504
45505
45506
45507
45508
45509
45510
45511
45512
45513
45514
45515
45516
45517
45518
45519
45520
45521
45522
45523
45524
45525
45526
45527
45528
45529
45530
45531
45532
45533
45534
45535
45536
45537
45538
45539
45540
45541
45542
45543
45544
45545
45546
45547
45548
45549
45550
45551
45552
45553
45554
45555
45556
45557
45558
45559
45560
45561
45562
45563
45564
45565
45566
45567
45568
45569
45570
45571
45572
45573
45574
45575
45576
45577
45578
45579
45580
45581
45582
45583
45584
45585
45586
45587
45588
45589
45590
45591
45592
45593
45594
45595
45596
45597
45598
45599
45600
45601
45602
45603
45604
45605
45606
45607
45608
45609
45610
45611
45612
45613
45614
45615
45616
45617
45618
45619
45620
45621
45622
45623
45624
45625
45626
45627
45628
45629
45630
45631
45632
45633
45634
45635
45636
45637
45638
45639
45640
45641
45642
45643
45644
45645
45646
45647
45648
45649
45650
45651
45652
45653
45654
45655
45656
45657
45658
45659
45660
45661
45662
45663
45664
45665
45666
45667
45668
45669
45670
45671
45672
45673
45674
45675
45676
45677
45678
45679
45680
45681
45682
45683
45684
45685
45686
45687
45688
45689
45690
45691
45692
45693
45694
45695
45696
45697
45698
45699
45700
45701
45702
45703
45704
45705
45706
45707
45708
45709
45710
45711
45712
45713
45714
45715
45716
45717
45718
45719
45720
45721
45722
45723
45724
45725
45726
45727
45728
45729
45730
45731
45732
45733
45734
45735
45736
45737
45738
45739
45740
45741
45742
45743
45744
45745
45746
45747
45748
45749
45750
45751
45752
45753
45754
45755
45756
45757
45758
45759
45760
45761
45762
45763
45764
45765
45766
45767
45768
45769
45770
45771
45772
45773
45774
45775
45776
45777
45778
45779
45780
45781
45782
45783
45784
45785
45786
45787
45788
45789
45790
45791
45792
45793
45794
45795
45796
45797
45798
45799
45800
45801
45802
45803
45804
45805
45806
45807
45808
45809
45810
45811
45812
45813
45814
45815
45816
45817
45818
45819
45820
45821
45822
45823
45824
45825
45826
45827
45828
45829
45830
45831
45832
45833
45834
45835
45836
45837
45838
45839
45840
45841
45842
45843
45844
45845
45846
45847
45848
45849
45850
45851
45852
45853
45854
45855
45856
45857
45858
45859
45860
45861
45862
45863
45864
45865
45866
45867
45868
45869
45870
45871
45872
45873
45874
45875
45876
45877
45878
45879
45880
45881
45882
45883
45884
45885
45886
45887
45888
45889
45890
45891
45892
45893
45894
45895
45896
45897
45898
45899
45900
45901
45902
45903
45904
45905
45906
45907
45908
45909
45910
45911
45912
45913
45914
45915
45916
45917
45918
45919
45920
45921
45922
45923
45924
45925
45926
45927
45928
45929
45930
45931
45932
45933
45934
45935
45936
45937
45938
45939
45940
45941
45942
45943
45944
45945
45946
45947
45948
45949
45950
45951
45952
45953
45954
45955
45956
45957
45958
45959
45960
45961
45962
45963
45964
45965
45966
45967
45968
45969
45970
45971
45972
45973
45974
45975
45976
45977
45978
45979
45980
45981
45982
45983
45984
45985
45986
45987
45988
45989
45990
45991
45992
45993
45994
45995
45996
45997
45998
45999
46000
46001
46002
46003
46004
46005
46006
46007
46008
46009
46010
46011
46012
46013
46014
46015
46016
46017
46018
46019
46020
46021
46022
46023
46024
46025
46026
46027
46028
46029
46030
46031
46032
46033
46034
46035
46036
46037
46038
46039
46040
46041
46042
46043
46044
46045
46046
46047
46048
46049
46050
46051
46052
46053
46054
46055
46056
46057
46058
46059
46060
46061
46062
46063
46064
46065
46066
46067
46068
46069
46070
46071
46072
46073
46074
46075
46076
46077
46078
46079
46080
46081
46082
46083
46084
46085
46086
46087
46088
46089
46090
46091
46092
46093
46094
46095
46096
46097
46098
46099
46100
46101
46102
46103
46104
46105
46106
46107
46108
46109
46110
46111
46112
46113
46114
46115
46116
46117
46118
46119
46120
46121
46122
46123
46124
46125
46126
46127
46128
46129
46130
46131
46132
46133
46134
46135
46136
46137
46138
46139
46140
46141
46142
46143
46144
46145
46146
46147
46148
46149
46150
46151
46152
46153
46154
46155
46156
46157
46158
46159
46160
46161
46162
46163
46164
46165
46166
46167
46168
46169
46170
46171
46172
46173
46174
46175
46176
46177
46178
46179
46180
46181
46182
46183
46184
46185
46186
46187
46188
46189
46190
46191
46192
46193
46194
46195
46196
46197
46198
46199
46200
46201
46202
46203
46204
46205
46206
46207
46208
46209
46210
46211
46212
46213
46214
46215
46216
46217
46218
46219
46220
46221
46222
46223
46224
46225
46226
46227
46228
46229
46230
46231
46232
46233
46234
46235
46236
46237
46238
46239
46240
46241
46242
46243
46244
46245
46246
46247
46248
46249
46250
46251
46252
46253
46254
46255
46256
46257
46258
46259
46260
46261
46262
46263
46264
46265
46266
46267
46268
46269
46270
46271
46272
46273
46274
46275
46276
46277
46278
46279
46280
46281
46282
46283
46284
46285
46286
46287
46288
46289
46290
46291
46292
46293
46294
46295
46296
46297
46298
46299
46300
46301
46302
46303
46304
46305
46306
46307
46308
46309
46310
46311
46312
46313
46314
46315
46316
46317
46318
46319
46320
46321
46322
46323
46324
46325
46326
46327
46328
46329
46330
46331
46332
46333
46334
46335
46336
46337
46338
46339
46340
46341
46342
46343
46344
46345
46346
46347
46348
46349
46350
46351
46352
46353
46354
46355
46356
46357
46358
46359
46360
46361
46362
46363
46364
46365
46366
46367
46368
46369
46370
46371
46372
46373
46374
46375
46376
46377
46378
46379
46380
46381
46382
46383
46384
46385
46386
46387
46388
46389
46390
46391
46392
46393
46394
46395
46396
46397
46398
46399
46400
46401
46402
46403
46404
46405
46406
46407
46408
46409
46410
46411
46412
46413
46414
46415
46416
46417
46418
46419
46420
46421
46422
46423
46424
46425
46426
46427
46428
46429
46430
46431
46432
46433
46434
46435
46436
46437
46438
46439
46440
46441
46442
46443
46444
46445
46446
46447
46448
46449
46450
46451
46452
46453
46454
46455
46456
46457
46458
46459
46460
46461
46462
46463
46464
46465
46466
46467
46468
46469
46470
46471
46472
46473
46474
46475
46476
46477
46478
46479
46480
46481
46482
46483
46484
46485
46486
46487
46488
46489
46490
46491
46492
46493
46494
46495
46496
46497
46498
46499
46500
46501
46502
46503
46504
46505
46506
46507
46508
46509
46510
46511
46512
46513
46514
46515
46516
46517
46518
46519
46520
46521
46522
46523
46524
46525
46526
46527
46528
46529
46530
46531
46532
46533
46534
46535
46536
46537
46538
46539
46540
46541
46542
46543
46544
46545
46546
46547
46548
46549
46550
46551
46552
46553
46554
46555
46556
46557
46558
46559
46560
46561
46562
46563
46564
46565
46566
46567
46568
46569
46570
46571
46572
46573
46574
46575
46576
46577
46578
46579
46580
46581
46582
46583
46584
46585
46586
46587
46588
46589
46590
46591
46592
46593
46594
46595
46596
46597
46598
46599
46600
46601
46602
46603
46604
46605
46606
46607
46608
46609
46610
46611
46612
46613
46614
46615
46616
46617
46618
46619
46620
46621
46622
46623
46624
46625
46626
46627
46628
46629
46630
46631
46632
46633
46634
46635
46636
46637
46638
46639
46640
46641
46642
46643
46644
46645
46646
46647
46648
46649
46650
46651
46652
46653
46654
46655
46656
46657
46658
46659
46660
46661
46662
46663
46664
46665
46666
46667
46668
46669
46670
46671
46672
46673
46674
46675
46676
46677
46678
46679
46680
46681
46682
46683
46684
46685
46686
46687
46688
46689
46690
46691
46692
46693
46694
46695
46696
46697
46698
46699
46700
46701
46702
46703
46704
46705
46706
46707
46708
46709
46710
46711
46712
46713
46714
46715
46716
46717
46718
46719
46720
46721
46722
46723
46724
46725
46726
46727
46728
46729
46730
46731
46732
46733
46734
46735
46736
46737
46738
46739
46740
46741
46742
46743
46744
46745
46746
46747
46748
46749
46750
46751
46752
46753
46754
46755
46756
46757
46758
46759
46760
46761
46762
46763
46764
46765
46766
46767
46768
46769
46770
46771
46772
46773
46774
46775
46776
46777
46778
46779
46780
46781
46782
46783
46784
46785
46786
46787
46788
46789
46790
46791
46792
46793
46794
46795
46796
46797
46798
46799
46800
46801
46802
46803
46804
46805
46806
46807
46808
46809
46810
46811
46812
46813
46814
46815
46816
46817
46818
46819
46820
46821
46822
46823
46824
46825
46826
46827
46828
46829
46830
46831
46832
46833
46834
46835
46836
46837
46838
46839
46840
46841
46842
46843
46844
46845
46846
46847
46848
46849
46850
46851
46852
46853
46854
46855
46856
46857
46858
46859
46860
46861
46862
46863
46864
46865
46866
46867
46868
46869
46870
46871
46872
46873
46874
46875
46876
46877
46878
46879
46880
46881
46882
46883
46884
46885
46886
46887
46888
46889
46890
46891
46892
46893
46894
46895
46896
46897
46898
46899
46900
46901
46902
46903
46904
46905
46906
46907
46908
46909
46910
46911
46912
46913
46914
46915
46916
46917
46918
46919
46920
46921
46922
46923
46924
46925
46926
46927
46928
46929
46930
46931
46932
46933
46934
46935
46936
46937
46938
46939
46940
46941
46942
46943
46944
46945
46946
46947
46948
46949
46950
46951
46952
46953
46954
46955
46956
46957
46958
46959
46960
46961
46962
46963
46964
46965
46966
46967
46968
46969
46970
46971
46972
46973
46974
46975
46976
46977
46978
46979
46980
46981
46982
46983
46984
46985
46986
46987
46988
46989
46990
46991
46992
46993
46994
46995
46996
46997
46998
46999
47000
47001
47002
47003
47004
47005
47006
47007
47008
47009
47010
47011
47012
47013
47014
47015
47016
47017
47018
47019
47020
47021
47022
47023
47024
47025
47026
47027
47028
47029
47030
47031
47032
47033
47034
47035
47036
47037
47038
47039
47040
47041
47042
47043
47044
47045
47046
47047
47048
47049
47050
47051
47052
47053
47054
47055
47056
47057
47058
47059
47060
47061
47062
47063
47064
47065
47066
47067
47068
47069
47070
47071
47072
47073
47074
47075
47076
47077
47078
47079
47080
47081
47082
47083
47084
47085
47086
47087
47088
47089
47090
47091
47092
47093
47094
47095
47096
47097
47098
47099
47100
47101
47102
47103
47104
47105
47106
47107
47108
47109
47110
47111
47112
47113
47114
47115
47116
47117
47118
47119
47120
47121
47122
47123
47124
47125
47126
47127
47128
47129
47130
47131
47132
47133
47134
47135
47136
47137
47138
47139
47140
47141
47142
47143
47144
47145
47146
47147
47148
47149
47150
47151
47152
47153
47154
47155
47156
47157
47158
47159
47160
47161
47162
47163
47164
47165
47166
47167
47168
47169
47170
47171
47172
47173
47174
47175
47176
47177
47178
47179
47180
47181
47182
47183
47184
47185
47186
47187
47188
47189
47190
47191
47192
47193
47194
47195
47196
47197
47198
47199
47200
47201
47202
47203
47204
47205
47206
47207
47208
47209
47210
47211
47212
47213
47214
47215
47216
47217
47218
47219
47220
47221
47222
47223
47224
47225
47226
47227
47228
47229
47230
47231
47232
47233
47234
47235
47236
47237
47238
47239
47240
47241
47242
47243
47244
47245
47246
47247
47248
47249
47250
47251
47252
47253
47254
47255
47256
47257
47258
47259
47260
47261
47262
47263
47264
47265
47266
47267
47268
47269
47270
47271
47272
47273
47274
47275
47276
47277
47278
47279
47280
47281
47282
47283
47284
47285
47286
47287
47288
47289
47290
47291
47292
47293
47294
47295
47296
47297
47298
47299
47300
47301
47302
47303
47304
47305
47306
47307
47308
47309
47310
47311
47312
47313
47314
47315
47316
47317
47318
47319
47320
47321
47322
47323
47324
47325
47326
47327
47328
47329
47330
47331
47332
47333
47334
47335
47336
47337
47338
47339
47340
47341
47342
47343
47344
47345
47346
47347
47348
47349
47350
47351
47352
47353
47354
47355
47356
47357
47358
47359
47360
47361
47362
47363
47364
47365
47366
47367
47368
47369
47370
47371
47372
47373
47374
47375
47376
47377
47378
47379
47380
47381
47382
47383
47384
47385
47386
47387
47388
47389
47390
47391
47392
47393
47394
47395
47396
47397
47398
47399
47400
47401
47402
47403
47404
47405
47406
47407
47408
47409
47410
47411
47412
47413
47414
47415
47416
47417
47418
47419
47420
47421
47422
47423
47424
47425
47426
47427
47428
47429
47430
47431
47432
47433
47434
47435
47436
47437
47438
47439
47440
47441
47442
47443
47444
47445
47446
47447
47448
47449
47450
47451
47452
47453
47454
47455
47456
47457
47458
47459
47460
47461
47462
47463
47464
47465
47466
47467
47468
47469
47470
47471
47472
47473
47474
47475
47476
47477
47478
47479
47480
47481
47482
47483
47484
47485
47486
47487
47488
47489
47490
47491
47492
47493
47494
47495
47496
47497
47498
47499
47500
47501
47502
47503
47504
47505
47506
47507
47508
47509
47510
47511
47512
47513
47514
47515
47516
47517
47518
47519
47520
47521
47522
47523
47524
47525
47526
47527
47528
47529
47530
47531
47532
47533
47534
47535
47536
47537
47538
47539
47540
47541
47542
47543
47544
47545
47546
47547
47548
47549
47550
47551
47552
47553
47554
47555
47556
47557
47558
47559
47560
47561
47562
47563
47564
47565
47566
47567
47568
47569
47570
47571
47572
47573
47574
47575
47576
47577
47578
47579
47580
47581
47582
47583
47584
47585
47586
47587
47588
47589
47590
47591
47592
47593
47594
47595
47596
47597
47598
47599
47600
47601
47602
47603
47604
47605
47606
47607
47608
47609
47610
47611
47612
47613
47614
47615
47616
47617
47618
47619
47620
47621
47622
47623
47624
47625
47626
47627
47628
47629
47630
47631
47632
47633
47634
47635
47636
47637
47638
47639
47640
47641
47642
47643
47644
47645
47646
47647
47648
47649
47650
47651
47652
47653
47654
47655
47656
47657
47658
47659
47660
47661
47662
47663
47664
47665
47666
47667
47668
47669
47670
47671
47672
47673
47674
47675
47676
47677
47678
47679
47680
47681
47682
47683
47684
47685
47686
47687
47688
47689
47690
47691
47692
47693
47694
47695
47696
47697
47698
47699
47700
47701
47702
47703
47704
47705
47706
47707
47708
47709
47710
47711
47712
47713
47714
47715
47716
47717
47718
47719
47720
47721
47722
47723
47724
47725
47726
47727
47728
47729
47730
47731
47732
47733
47734
47735
47736
47737
47738
47739
47740
47741
47742
47743
47744
47745
47746
47747
47748
47749
47750
47751
47752
47753
47754
47755
47756
47757
47758
47759
47760
47761
47762
47763
47764
47765
47766
47767
47768
47769
47770
47771
47772
47773
47774
47775
47776
47777
47778
47779
47780
47781
47782
47783
47784
47785
47786
47787
47788
47789
47790
47791
47792
47793
47794
47795
47796
47797
47798
47799
47800
47801
47802
47803
47804
47805
47806
47807
47808
47809
47810
47811
47812
47813
47814
47815
47816
47817
47818
47819
47820
47821
47822
47823
47824
47825
47826
47827
47828
47829
47830
47831
47832
47833
47834
47835
47836
47837
47838
47839
47840
47841
47842
47843
47844
47845
47846
47847
47848
47849
47850
47851
47852
47853
47854
47855
47856
47857
47858
47859
47860
47861
47862
47863
47864
47865
47866
47867
47868
47869
47870
47871
47872
47873
47874
47875
47876
47877
47878
47879
47880
47881
47882
47883
47884
47885
47886
47887
47888
47889
47890
47891
47892
47893
47894
47895
47896
47897
47898
47899
47900
47901
47902
47903
47904
47905
47906
47907
47908
47909
47910
47911
47912
47913
47914
47915
47916
47917
47918
47919
47920
47921
47922
47923
47924
47925
47926
47927
47928
47929
47930
47931
47932
47933
47934
47935
47936
47937
47938
47939
47940
47941
47942
47943
47944
47945
47946
47947
47948
47949
47950
47951
47952
47953
47954
47955
47956
47957
47958
47959
47960
47961
47962
47963
47964
47965
47966
47967
47968
47969
47970
47971
47972
47973
47974
47975
47976
47977
47978
47979
47980
47981
47982
47983
47984
47985
47986
47987
47988
47989
47990
47991
47992
47993
47994
47995
47996
47997
47998
47999
48000
48001
48002
48003
48004
48005
48006
48007
48008
48009
48010
48011
48012
48013
48014
48015
48016
48017
48018
48019
48020
48021
48022
48023
48024
48025
48026
48027
48028
48029
48030
48031
48032
48033
48034
48035
48036
48037
48038
48039
48040
48041
48042
48043
48044
48045
48046
48047
48048
48049
48050
48051
48052
48053
48054
48055
48056
48057
48058
48059
48060
48061
48062
48063
48064
48065
48066
48067
48068
48069
48070
48071
48072
48073
48074
48075
48076
48077
48078
48079
48080
48081
48082
48083
48084
48085
48086
48087
48088
48089
48090
48091
48092
48093
48094
48095
48096
48097
48098
48099
48100
48101
48102
48103
48104
48105
48106
48107
48108
48109
48110
48111
48112
48113
48114
48115
48116
48117
48118
48119
48120
48121
48122
48123
48124
48125
48126
48127
48128
48129
48130
48131
48132
48133
48134
48135
48136
48137
48138
48139
48140
48141
48142
48143
48144
48145
48146
48147
48148
48149
48150
48151
48152
48153
48154
48155
48156
48157
48158
48159
48160
48161
48162
48163
48164
48165
48166
48167
48168
48169
48170
48171
48172
48173
48174
48175
48176
48177
48178
48179
48180
48181
48182
48183
48184
48185
48186
48187
48188
48189
48190
48191
48192
48193
48194
48195
48196
48197
48198
48199
48200
48201
48202
48203
48204
48205
48206
48207
48208
48209
48210
48211
48212
48213
48214
48215
48216
48217
48218
48219
48220
48221
48222
48223
48224
48225
48226
48227
48228
48229
48230
48231
48232
48233
48234
48235
48236
48237
48238
48239
48240
48241
48242
48243
48244
48245
48246
48247
48248
48249
48250
48251
48252
48253
48254
48255
48256
48257
48258
48259
48260
48261
48262
48263
48264
48265
48266
48267
48268
48269
48270
48271
48272
48273
48274
48275
48276
48277
48278
48279
48280
48281
48282
48283
48284
48285
48286
48287
48288
48289
48290
48291
48292
48293
48294
48295
48296
48297
48298
48299
48300
48301
48302
48303
48304
48305
48306
48307
48308
48309
48310
48311
48312
48313
48314
48315
48316
48317
48318
48319
48320
48321
48322
48323
48324
48325
48326
48327
48328
48329
48330
48331
48332
48333
48334
48335
48336
48337
48338
48339
48340
48341
48342
48343
48344
48345
48346
48347
48348
48349
48350
48351
48352
48353
48354
48355
48356
48357
48358
48359
48360
48361
48362
48363
48364
48365
48366
48367
48368
48369
48370
48371
48372
48373
48374
48375
48376
48377
48378
48379
48380
48381
48382
48383
48384
48385
48386
48387
48388
48389
48390
48391
48392
48393
48394
48395
48396
48397
48398
48399
48400
48401
48402
48403
48404
48405
48406
48407
48408
48409
48410
48411
48412
48413
48414
48415
48416
48417
48418
48419
48420
48421
48422
48423
48424
48425
48426
48427
48428
48429
48430
48431
48432
48433
48434
48435
48436
48437
48438
48439
48440
48441
48442
48443
48444
48445
48446
48447
48448
48449
48450
48451
48452
48453
48454
48455
48456
48457
48458
48459
48460
48461
48462
48463
48464
48465
48466
48467
48468
48469
48470
48471
48472
48473
48474
48475
48476
48477
48478
48479
48480
48481
48482
48483
48484
48485
48486
48487
48488
48489
48490
48491
48492
48493
48494
48495
48496
48497
48498
48499
48500
48501
48502
48503
48504
48505
48506
48507
48508
48509
48510
48511
48512
48513
48514
48515
48516
48517
48518
48519
48520
48521
48522
48523
48524
48525
48526
48527
48528
48529
48530
48531
48532
48533
48534
48535
48536
48537
48538
48539
48540
48541
48542
48543
48544
48545
48546
48547
48548
48549
48550
48551
48552
48553
48554
48555
48556
48557
48558
48559
48560
48561
48562
48563
48564
48565
48566
48567
48568
48569
48570
48571
48572
48573
48574
48575
48576
48577
48578
48579
48580
48581
48582
48583
48584
48585
48586
48587
48588
48589
48590
48591
48592
48593
48594
48595
48596
48597
48598
48599
48600
48601
48602
48603
48604
48605
48606
48607
48608
48609
48610
48611
48612
48613
48614
48615
48616
48617
48618
48619
48620
48621
48622
48623
48624
48625
48626
48627
48628
48629
48630
48631
48632
48633
48634
48635
48636
48637
48638
48639
48640
48641
48642
48643
48644
48645
48646
48647
48648
48649
48650
48651
48652
48653
48654
48655
48656
48657
48658
48659
48660
48661
48662
48663
48664
48665
48666
48667
48668
48669
48670
48671
48672
48673
48674
48675
48676
48677
48678
48679
48680
48681
48682
48683
48684
48685
48686
48687
48688
48689
48690
48691
48692
48693
48694
48695
48696
48697
48698
48699
48700
48701
48702
48703
48704
48705
48706
48707
48708
48709
48710
48711
48712
48713
48714
48715
48716
48717
48718
48719
48720
48721
48722
48723
48724
48725
48726
48727
48728
48729
48730
48731
48732
48733
48734
48735
48736
48737
48738
48739
48740
48741
48742
48743
48744
48745
48746
48747
48748
48749
48750
48751
48752
48753
48754
48755
48756
48757
48758
48759
48760
48761
48762
48763
48764
48765
48766
48767
48768
48769
48770
48771
48772
48773
48774
48775
48776
48777
48778
48779
48780
48781
48782
48783
48784
48785
48786
48787
48788
48789
48790
48791
48792
48793
48794
48795
48796
48797
48798
48799
48800
48801
48802
48803
48804
48805
48806
48807
48808
48809
48810
48811
48812
48813
48814
48815
48816
48817
48818
48819
48820
48821
48822
48823
48824
48825
48826
48827
48828
48829
48830
48831
48832
48833
48834
48835
48836
48837
48838
48839
48840
48841
48842
48843
48844
48845
48846
48847
48848
48849
48850
48851
48852
48853
48854
48855
48856
48857
48858
48859
48860
48861
48862
48863
48864
48865
48866
48867
48868
48869
48870
48871
48872
48873
48874
48875
48876
48877
48878
48879
48880
48881
48882
48883
48884
48885
48886
48887
48888
48889
48890
48891
48892
48893
48894
48895
48896
48897
48898
48899
48900
48901
48902
48903
48904
48905
48906
48907
48908
48909
48910
48911
48912
48913
48914
48915
48916
48917
48918
48919
48920
48921
48922
48923
48924
48925
48926
48927
48928
48929
48930
48931
48932
48933
48934
48935
48936
48937
48938
48939
48940
48941
48942
48943
48944
48945
48946
48947
48948
48949
48950
48951
48952
48953
48954
48955
48956
48957
48958
48959
48960
48961
48962
48963
48964
48965
48966
48967
48968
48969
48970
48971
48972
48973
48974
48975
48976
48977
48978
48979
48980
48981
48982
48983
48984
48985
48986
48987
48988
48989
48990
48991
48992
48993
48994
48995
48996
48997
48998
48999
49000
49001
49002
49003
49004
49005
49006
49007
49008
49009
49010
49011
49012
49013
49014
49015
49016
49017
49018
49019
49020
49021
49022
49023
49024
49025
49026
49027
49028
49029
49030
49031
49032
49033
49034
49035
49036
49037
49038
49039
49040
49041
49042
49043
49044
49045
49046
49047
49048
49049
49050
49051
49052
49053
49054
49055
49056
49057
49058
49059
49060
49061
49062
49063
49064
49065
49066
49067
49068
49069
49070
49071
49072
49073
49074
49075
49076
49077
49078
49079
49080
49081
49082
49083
49084
49085
49086
49087
49088
49089
49090
49091
49092
49093
49094
49095
49096
49097
49098
49099
49100
49101
49102
49103
49104
49105
49106
49107
49108
49109
49110
49111
49112
49113
49114
49115
49116
49117
49118
49119
49120
49121
49122
49123
49124
49125
49126
49127
49128
49129
49130
49131
49132
49133
49134
49135
49136
49137
49138
49139
49140
49141
49142
49143
49144
49145
49146
49147
49148
49149
49150
49151
49152
49153
49154
49155
49156
49157
49158
49159
49160
49161
49162
49163
49164
49165
49166
49167
49168
49169
49170
49171
49172
49173
49174
49175
49176
49177
49178
49179
49180
49181
49182
49183
49184
49185
49186
49187
49188
49189
49190
49191
49192
49193
49194
49195
49196
49197
49198
49199
49200
49201
49202
49203
49204
49205
49206
49207
49208
49209
49210
49211
49212
49213
49214
49215
49216
49217
49218
49219
49220
49221
49222
49223
49224
49225
49226
49227
49228
49229
49230
49231
49232
49233
49234
49235
49236
49237
49238
49239
49240
49241
49242
49243
49244
49245
49246
49247
49248
49249
49250
49251
49252
49253
49254
49255
49256
49257
49258
49259
49260
49261
49262
49263
49264
49265
49266
49267
49268
49269
49270
49271
49272
49273
49274
49275
49276
49277
49278
49279
49280
49281
49282
49283
49284
49285
49286
49287
49288
49289
49290
49291
49292
49293
49294
49295
49296
49297
49298
49299
49300
49301
49302
49303
49304
49305
49306
49307
49308
49309
49310
49311
49312
49313
49314
49315
49316
49317
49318
49319
49320
49321
49322
49323
49324
49325
49326
49327
49328
49329
49330
49331
49332
49333
49334
49335
49336
49337
49338
49339
49340
49341
49342
49343
49344
49345
49346
49347
49348
49349
49350
49351
49352
49353
49354
49355
49356
49357
49358
49359
49360
49361
49362
49363
49364
49365
49366
49367
49368
49369
49370
49371
49372
49373
49374
49375
49376
49377
49378
49379
49380
49381
49382
49383
49384
49385
49386
49387
49388
49389
49390
49391
49392
49393
49394
49395
49396
49397
49398
49399
49400
49401
49402
49403
49404
49405
49406
49407
49408
49409
49410
49411
49412
49413
49414
49415
49416
49417
49418
49419
49420
49421
49422
49423
49424
49425
49426
49427
49428
2009-02-13  Leif Madsen <lmadsen@digium.com>

	* Released 1.6.0.6-rc1 

2009-02-13 16:43 +0000 [r175550]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_record.c: Merged revisions 175549 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
	  file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
	  an option to keep the recorded file upon hangup. (closes issue
	  #14341) Reported by: fnordian ........

2009-02-12 21:41 +0000 [r175369]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
	  russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
	  Remove useless string copy, and make sscanf safe again ........

2009-02-12 21:27 +0000 [r175347]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, /: Merged revisions 175334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
	  | 16 lines Merged revisions 175311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
	  | 9 lines Fix crashes when receiving certain T.38 packets. Also,
	  increase the maximum size of T.38 packets and warn users when
	  they try to set the limits above those maximums. (closes issue
	  #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: schern ........
	  ................

2009-02-12 20:59 +0000 [r175299-175301]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Fix mistake in merging conflict from 175299.

	* /, main/features.c: Merged revisions 175298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
	  | 15 lines Merged revisions 175294 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
	  | 9 lines Fix ParkedCall event information for From field in the
	  case of a blind transfer If the parker information can not be
	  obtained from the peer, try and see if the BLINDTRANSFER channel
	  variable has been set. Previously, a blind transfer to the
	  ParkAndAnnounce app would return nothing for the From. Closes
	  AST-189 ........ ................

2009-02-12 20:46 +0000 [r175256-175296]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 |
	  russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines
	  Avoid using ast_strdupa() in a loop. ........

	* build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009)
	  | 4 lines Don't enable something by default that has a dependency
	  on something _not_ enabled by default. menuselect was not happy
	  with this. ........

2009-02-12 18:00 +0000 [r175189]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 175188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009)
	  | 12 lines Merged revisions 175187 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
	  | 6 lines Fix crash in event of failed attempt to transfer to
	  parking The peer may not necessarily exist, such as in the case
	  of a transfer to ParkAndAnnounce. In this case don't try to play
	  a sound to it. ........ ................

2009-02-12 17:03 +0000 [r175126]  Russell Bryant <russell@digium.com>

	* main/rtp.c, /: Merged revisions 175125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009)
	  | 35 lines Merged revisions 175124 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
	  | 27 lines Don't send DTMF for infinite time if we do not receive
	  an END event. I thought that this was going to end up being a
	  pretty gnarly fix, but it turns out that there was actually
	  already a configuration option in rtp.conf, dtmftimeout, that was
	  intended to handle this situation. However, in between Asterisk
	  1.2 and Asterisk 1.4, the code that processed the option got
	  lost. So, this commit brings it back to life. The default timeout
	  is 3 seconds. However, it is worth noting that having this be
	  configurable at all is not really the recommended behavior in RFC
	  2833. From Section 3.5 of RFC 2833: Limiting the time period of
	  extending the tone is necessary to avoid that a tone "gets
	  stuck". Regardless of the algorithm used, the tone SHOULD NOT be
	  extended by more than three packet interarrival times. A slight
	  extension of tone durations and shortening of pauses is generally
	  harmless. Three seconds will pretty much _always_ be far more
	  than three packet interarrival times. However, that behavior is
	  not required, so I'm going to leave it with our legacy behavior
	  for now. Code from svn/asterisk/team/russell/issue_14460 (closes
	  issue #14460) Reported by: moliveras ........ ................

2009-02-12 16:33 +0000 [r175122]  Mark Michelson <mmichelson@digium.com>

	* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
	  175121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 |
	  mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11
	  lines Make lock information for ao2_trylock be more useful and
	  gnarly Core show locks information involving an ao2_trylock did
	  not show the function that called ao2_trylock, but would instead
	  show ao2_trylock as the source of the lock. This is not useful
	  when trying to debug locking issues. One bizarre note is that
	  this logic is already in 1.4 but somehow did not get merged to
	  trunk or the 1.6.X branches. ........

2009-02-12 14:27 +0000 [r175059-175090]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009)
	  | 6 lines Issue a warning message if our candidate's IP is the
	  loopback address. (closes issue #13985) Reported by: jcovert
	  Tested by: phsultan ........

	* /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100
	  (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009)
	  | 12 lines Set the initiator attribute to lowercase in our
	  replies when receiving calls. This attribute contains a JID that
	  identifies the initiator of the GoogleTalk voice session. The
	  GoogleTalk client discards Asterisk's replies if the initiator
	  attribute contains uppercase characters. (closes issue #13984)
	  Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by
	  jcovert (license 551) Tested by: jcovert ........
	  ................

2009-02-11 23:04 +0000 [r174765-174949]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 174948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb
	  2009) | 35 lines Fix odd "thank you" sound playing behavior in
	  app_queue.c If someone has configured the queue to play an
	  position or holdtime announcement, then it is odd and potentially
	  unexpected to hear a "Thank you for your patience" sound when no
	  position or holdtime was actually announced. This fixes the
	  announcement so that the "thanks" sound is only played in the
	  case that a position or holdtime was actually announced. There is
	  a way that the "thank you" sound can be played without a position
	  or holdtime, and that is to set announce-frequency to a value but
	  keep announce-position and announce-holdtime both turned off.
	  (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
	  uploaded by putnopvut (license 60) Tested by: caspy
	  ................

	* apps/app_dial.c, main/channel.c, main/pbx.c, /,
	  apps/app_dictate.c, apps/app_waitforsilence.c,
	  include/asterisk/channel.h: Merged revisions 174945 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb
	  2009) | 29 lines Fix 'd' option for app_dial and add new option
	  to Answer application The 'd' option would not work for channel
	  types which use RTP to transport DTMF digits. The only way to
	  allow for this to work was to answer the channel if we saw that
	  this option was enabled. I realized that this may cause issues
	  with CDRs, specifically with giving false dispositions and answer
	  times. I therefore modified ast_answer to take another parameter
	  which would tell if the CDR should be marked answered. I also
	  extended this to the Answer application so that the channel may
	  be answered but not CDRified if desired. I also modified
	  app_dictate and app_waitforsilence to only answer the channel if
	  it is not already up, to help not allow for faulty CDR answer
	  times. All of these changes are going into Asterisk trunk. For
	  1.6.0 and 1.6.1, however, all the changes except for the change
	  to the Answer application will go in since we do not introduce
	  new features into stable branches (closes issue #14164) Reported
	  by: DennisD Patches: 14164.patch uploaded by putnopvut (license
	  60) Tested by: putnopvut Review:
	  http://reviewboard.digium.com/r/145 ........

	* apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 |
	  mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11
	  lines Fix potential for stack overflows in app_chanspy.c When
	  using the 'g' or 'e' options, the stack allocations that were
	  used could cause a stack overflow if a spyer stayed on the line
	  long enough without actually successfully spying on anyone. The
	  problem has been corrected by using static buffers and copying
	  the contents of the appropriate strings into them instead of
	  using functions like alloca or ast_strdupa ........

	* main/manager.c, /: Merged revisions 174764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 |
	  mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21
	  lines Fix an fd leak that would occur in HTTP AMI sessions The
	  explanation behind this fix is a bit complicated, and I've
	  already typed it up in the code as a huge comment inside of
	  manager.c, so I'll give the abridged version here. We needed a
	  way to separate action-specific data from session-specific data.
	  Unfortunately, the only way to maintain API compatibility and to
	  not have to change every single manager action was to rename the
	  current mansession structure and wrap it inside a new mansession
	  structure which actually contains action- specific data. (closes
	  issue #14364) Reported by: awk Patches: 14364_better.patch
	  uploaded by putnopvut (license 60) Tested by: putnopvut Review:
	  http://reviewboard.digium.com/r/148/ ........

2009-02-10 20:16 +0000 [r174711]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 |
	  file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
	  Only decrease inringing count if above zero. (issue #13238)
	  Reported by: kowalma ........

2009-02-10 18:19 +0000 [r174596]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb
	  2009) | 25 lines Merged revisions 174583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
	  2009) | 18 lines Improve behavior of jitterbuffer when
	  maxjitterbuffer is set. This change improves the way the
	  jitterbuffer handles maxjitterbuffer and dramatically reduces the
	  number of frames dropped when maxjitterbuffer is exceeded. In the
	  previous jitterbuffer, when maxjitterbuffer was exceeded, all new
	  frames were dropped until the jitterbuffer is empty. This change
	  modifies the code to only drop frames until maxjitterbuffer is no
	  longer exceeded. Also, previously when maxjitterbuffer was
	  exceeded, dropped frames were not tracked causing stats for
	  dropped frames to be incorrect, this change also addresses that
	  problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
	  by mnicholson (license 96) Tested by: mnicholson Review:
	  http://reviewboard.digium.com/r/144/ ........ ................

2009-02-10 15:39 +0000 [r174544]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 |
	  file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
	  Make the logic for inuse and inringing manipluation match that of
	  1.4. The old broken logic would reset the values back to 0 during
	  certain scenarios causing the wrong state to be reported. (closes
	  issue #14399) Reported by: caspy (issue #13238) Reported by:
	  kowalma ........

2009-02-10 05:06 +0000 [r174439]  Steve Murphy <murf@digium.com>

	* apps/app_rpt.c: For some strange reason, I didn't think 1.6.0
	  needed this fix. I was wrong. Here it is.

2009-02-09 17:28 +0000 [r174322-174328]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 |
	  mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3
	  lines Fix something I messed up in the merge I just did ........

	* /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb
	  2009) | 20 lines Merged revisions 174282 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
	  2009) | 12 lines Don't do an SRV lookup if a port is specified
	  RFC 3263 says to do A record lookups on a hostname if a port has
	  been specified, so that's what we're going to do. See section
	  4.2. (closes issue #14419) Reported by: klaus3000 Patches:
	  patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
	  (license 65) ........ ................

2009-02-09 14:50 +0000 [r174220]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon,
	  09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4
	  lines Don't overwrite our pointer to the music class when music
	  on hold stops. We will use this if it starts again to see if we
	  can resume the music where it left off. (closes issue #14407)
	  Reported by: mostyn ........ ................

2009-02-07 16:17 +0000 [r174151]  Russell Bryant <russell@digium.com>

	* /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009)
	  | 10 lines Merged revisions 174148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
	  | 2 lines Fix a race condition that could cause a crash. ........
	  ................

2009-02-06 23:59 +0000 [r174085]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009)
	  | 13 lines Merged revisions 174082 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
	  | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
	  sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
	  didn't actually upload a properly-formed patch, instead a
	  modified chan_sip.c file was uploaded. I created a patch to
	  determine the changes, then modified the suggested changes to
	  create a proper fix. The summary above is a complete description
	  of the changes. (closes issue #13547) Reported by: tecnoxarxa
	  Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
	  Tested by: tecnoxarxa ........ ................
	  ------------------------------------------------------------------------

2009-02-06 19:29 +0000 [r173986-174042]  Joshua Colp <jcolp@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4
	  lines Don't subscribe to a mailbox on pseudo channels. It is
	  futile. This solves an issue where duplicated pseudo channels
	  would cause a crash because the first one would unsubscribe and
	  the next one would also try to unsubscribe the same subscription.
	  (closes issue #14322) Reported by: amessina ........

	* /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) |
	  15 lines Merged revisions 173967-173968 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
	  lines Some clients do not put the call-id for replaces at the
	  beginning, so support it being anywhere in the string. (closes
	  issue #14350) Reported by: fhackenberger ........ r173968 | file
	  | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
	  debug message I put in by accident. ........ ................

2009-02-06 16:33 +0000 [r173963]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
	  2009) | 14 lines Merged revisions 173917 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
	  2009) | 7 lines Limit the addition of the Contact header in SIP
	  responses according to various SIP RFCs. (closes issue #13602)
	  Reported by: hjourdain Tested by: mnicholson ........
	  ................

2009-02-05 23:51 +0000 [r173774-173777]  Mark Michelson <mmichelson@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 173776 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu,
	  05 Feb 2009) | 14 lines Update extensions.conf.sample to be
	  correct. In trunk, the only necessary change pointed out was that
	  the call to ChanIsAvail uses an option that has been removed. For
	  the 1.6.1 branch, however, it appears that the sample file is
	  badly in need of updating since there are |'s used all over the
	  place there. My tentative plan is just to copy trunk's sample
	  config file to those branches since the info there is most
	  up-to-date and should be correct for use in 1.6.1 Thanks to macli
	  in #asterisk-dev for bringing this up ........

	* apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb
	  2009) | 7 lines Properly set "seen" and "unseen" flags when
	  moving messages from the new to the old folder when using IMAP
	  for voicemail storage (closes issue #13905) Reported by: jaroth
	  Patches: foldermove_v2.patch uploaded by jaroth (license 50)
	  ........

2009-02-05 21:04 +0000 [r173698]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600
	  (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009)
	  | 12 lines Add new configuration option to make shared IMAP
	  mailboxes function as expected. The new option is "imapvmshareid"
	  which is an ID to tag multiple mailboxes using the same IMAP
	  storage location to function as one mailbox. This allows all
	  messages to be retrieved for any user in the group. The patch
	  alters the 'X-Asterisk-VM-Extension' header that is responsible
	  for matching voicemails for a given user. (closes issue #13673)
	  Reported by: howardwilkinson ........ ................

2009-02-05 20:34 +0000 [r173590-173694]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 173693 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb
	  2009) | 20 lines Merged revisions 173692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
	  2009) | 12 lines Fix situations where queue members could be
	  autopaused unexpectedly Specifically, this patch prevents us from
	  autopausing members when we receive a busy or congestion frame
	  from them. (closes issue #14376) Reported by: fiddur Patches:
	  14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
	  ........ ................

	* apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600
	  (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb
	  2009) | 3 lines Add some missing cleanup to app_mixmonitor
	  ........ ................

	* apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600
	  (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb
	  2009) | 25 lines Fix a problem where a channel pointer becomes
	  invalid due to masquerading or hanging up. app_mixmonitor runs
	  its own thread to monitor the channel's activity and write the
	  mixed audio to a file. Since this thread runs independently of
	  the channel, it is possible that the mixmonitor thread's channel
	  pointer will point to freed memory when the channel either is
	  masqueraded or hangs up (technically, both cases are hangups, but
	  we need to handle the cases slightly differently). The solution
	  for this is to employ a datastore, which has the nice benefit of
	  allowing us to hook into channel masquerades and hangups and
	  update our pointer as necessary. If this looks familiar, this
	  same technique is employed in app_chanspy. app_chanspy is a bit
	  more involved since it does a lot more operations on the channel
	  that is being spied upon. app_mixmonitor does have an extra touch
	  that app_chanspy doesn't have, though. Since there is a thread
	  race between the channel's thread and the mixmonitor thread on a
	  hangup, we em- ploy a condition-and-boolean combination to ensure
	  that the channel thread finishes with our structure before the
	  mixmonitor thread attempts to free it. No crashes! (closes issue
	  #14374) Reported by: aragon Patches: 14374.patch uploaded by
	  putnopvut (license 60) Tested by: aragon, putnopvut ........
	  ................

2009-02-05 16:23 +0000 [r173554]  Jeff Peeler <jpeeler@digium.com>

	* build_tools/menuselect-deps.in: fix WORKING_FORK detection

2009-02-05 00:11 +0000 [r173548]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/menuselect-deps.in: regenerate with bootstrap.sh

2009-02-04 23:44 +0000 [r173546-173547]  Jeff Peeler <jpeeler@digium.com>

	* /: I messed up and accidentally reverted the trunk-merged prop
	  before committing 173546. Added it manually.

	* main/features.c: Merged revisions 173500 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009)
	  | 23 lines Merged revisions 173211 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
	  | 17 lines Parking attempts made to one end of a bridge no longer
	  will hang up due to a parking failure. Parking attempts made
	  using either one-touch, or doing either a blind or assisted
	  transfer to the parking extension now keep up the bridge instead
	  of hanging up the attempted parked party. Normal causes for the
	  parking attempt to fail includes the specific specified extension
	  (via PARKINGEXTEN) not being available or if all the parking
	  spaces are currently in use. To avoid having to reverse a
	  masquerade park_space_reserve was made to provide foresight if a
	  parking attempt will succeed and if so reserve the parking space.
	  (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
	  http://reviewboard.digium.com/r/133/ ........ ................

2009-02-04 22:23 +0000 [r173534]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 173507 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 |
	  mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7
	  lines Fix some areas where the incorrect interface was passed to
	  ast_device_state I swear it feels like I already did this once...
	  (closes issue #14359) Reported by: francesco_r ........

2009-02-04 18:55 +0000 [r173460]  Tilghman Lesher <tlesher@digium.com>

	* main/tcptls.c, /: Merged revisions 173458 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 |
	  tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
	  When using a socket as a FILE *, the stdio functions will
	  sometimes try to do an fseek() on the stream, which is an invalid
	  operation for a socket. Turning off buffering explicitly lets the
	  stdio functions know they cannot do this, thus avoiding a
	  potential error. (closes issue #14400) Reported by: fnordian
	  Patches: tcptls.patch uploaded by fnordian (license 110) ........

2009-02-04 17:46 +0000 [r173355-173398]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb
	  2009) | 11 lines Merged revisions 173396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
	  2009) | 3 lines Revert my previous change because it was stupid
	  ........ ................

	* apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb
	  2009) | 11 lines Merged revisions 173392 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
	  2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
	  matter, but it's needed. ........ ................

	* /, main/file.c: Merged revisions 173354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 |
	  mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30
	  lines Fix a problem where file playback would cause fds to remain
	  open forever The problem came from the fact that a frame read
	  from a format interpreter was not freed. Adding a call to
	  ast_frfree fixed this. The explanation for why this caused the
	  problem is a bit complex, but here goes: There was a problem in
	  all versions of Asterisk where the embedded frame of a filestream
	  structure was referenced after the filestream was freed. This was
	  fixed by adding reference counting to the filestream structure.
	  The refcount would increase every time that a filestream's frame
	  pointer was pointing to an actual frame of data. When the frame
	  was freed, the refcount would decrease. Once the refcount reached
	  0, the filestream was freed, and as part of the operation, the
	  open files were closed as well. Thus it becomes more clear why a
	  missing ast_frfree would cause a reference leak and cause the
	  files to not be closed. You may ask then if there was a frame
	  leak before this patch. The answer to that is actually no! The
	  filestream code was "smart" enough to know that since the frame
	  we received came from a format interpreter, the frame had no
	  malloced data and thus didn't need to be freed. Now, however,
	  there is cleanup that needs to be done when we finish with the
	  frame, so we do need to call ast_frfree on the frame to be sure
	  that the refcount for the filestream is decremented
	  appropriately. (closes issue #14384) Reported by: fiddur Patches:
	  14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
	  putnopvut ........

2009-02-04 00:45 +0000 [r173312]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 173311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 |
	  tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10
	  lines Ensure that commas placed in the middle of extension
	  character classes do not interfere with correct parsing of the
	  extension. Also, if an unterminated character class DOES make its
	  way into the pbx core (through some other method), ensure that it
	  does not crash Asterisk. (closes issue #14362) Reported by:
	  Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: Corydon76 ........

2009-02-03 23:41 +0000 [r173250]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
	  of calls. Fixes issue with IAX2 transfers not taking place. As it
	  was, a call that was being transfered would never be handed off
	  correctly to the call ends because of how call numbers were
	  stored in a hash table. The hash table, "iax_peercallno_pvt",
	  storing all the current call numbers did not take into account
	  the complications associated with transferring a call, so a
	  separate hash table was required. This second hash table
	  "iax_transfercallno_pvt" handles calls being transfered, once the
	  call transfer is complete the call is removed from the transfer
	  hash table and added to the peer hash table resuming normal
	  operations. Addition functions were created to handle storing,
	  removing, and comparing items in the iax_transfercallno_pvt
	  table. (issue #13468) Review:
	  http://reviewboard.digium.com/r/140/

2009-02-03 00:26 +0000 [r173111]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 173104 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600
	  (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
	  | 5 lines Add warning to standard config, that globals may be
	  overridden by other dialplan configuration files. (closes issue
	  #14388) Reported by: macli ........ ................

2009-02-02 23:59 +0000 [r173068]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: Merged revisions 173067 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009)
	  | 9 lines Merged revisions 173066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
	  | 2 lines Fix a feature inheritance bug I added after code review
	  ........ ................

2009-02-02 18:15 +0000 [r172896]  Leif Madsen <lmadsen@digium.com>

	* /, configs/res_ldap.conf.sample: Merged revisions 172894 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02
	  Feb 2009) | 7 lines Update the res_ldap.conf file with a better
	  working example. (closes issue #13861) Reported by: scramatte
	  Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage
	  (license 10) Tested by: jcovert ........

2009-02-01 02:45 +0000 [r172707-172742]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009)
	  | 4 lines Blank argument crashes Asterisk (closes issue #14377)
	  Reported by: amorsen ........

	* /, funcs/func_strings.c: Merged revisions 172706 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009)
	  | 7 lines Don't increment the loop, now that incrementing is
	  taken care of by the decoder function. (closes issue #14363)
	  Reported by: andrew53 Patches: func_strings_filter.patch uploaded
	  by andrew53 (license 519) ........

2009-01-31 00:06 +0000 [r172635-172637]  Terry Wilson <twilson@digium.com>

	* configs/features.conf.sample, /: Merged revisions 172581 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30
	  Jan 2009) | 2 lines Remove incorret line from sample config
	  ........

	* configs/features.conf.sample, apps/app_dial.c,
	  main/global_datastores.c, /, main/features.c,
	  include/asterisk/global_datastores.h, CHANGES: Merged revisions
	  172580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009)
	  | 44 lines Merged revisions 172517 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
	  | 37 lines Fix feature inheritance with builtin features When
	  using builtin features like parking and transfers, the
	  AST_FEATURE_* flags would not be set correctly for all instances
	  when either performing a builtin attended transfer, or parking a
	  call and getting the timeout callback. Also, there was no way on
	  a per-call basis to specify what features someone should have on
	  picking up a parked call (since that doesn't involve the Dial()
	  command). There was a global option for setting whether or not
	  all users who pickup a parked call should have
	  AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
	  PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
	  variable which can be set either in the dialplan or with setvar
	  in channels that support it. This variable can be set to any
	  combination of 't', 'k', 'w', and 'h' (case insensitive matching
	  of the equivalent dial options), to set what features should be
	  activated on this channel. The patch moves the setting of the
	  features datastores into the bridging code instead of app_dial to
	  help facilitate this. 2) adds global options parkedcallparking,
	  parkedcallhangup, and parkedcallrecording to be similar to the
	  parkedcalltransfers option for globally setting features. 3) has
	  builtin_atxfer call builtin_parkcall if being transfered to the
	  parking extension since tracking everything through multiple
	  masquerades, etc. is difficult and error-prone 4) attempts to fix
	  all cases of return calls from parking and completed builtin
	  transfers not having the correct permissions (closes issue
	  #14274) Reported by: aragon Patches:
	  fix_feature_inheritence.diff.txt uploaded by otherwiseguy
	  (license 396) Tested by: aragon, otherwiseguy Review
	  http://reviewboard.digium.com/r/138/ ........ ................

2009-01-30 22:23 +0000 [r172604]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/channel.h: Merged revisions 172598 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri,
	  30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h
	  ........

2009-01-29 23:47 +0000 [r172503]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, apps/app_nbscat.c, /, autoconf/ast_func_fork.m4,
	  apps/app_festival.c, build_tools/menuselect-deps.in, configure,
	  apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c,
	  apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c:
	  Merged revisions 172441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009)
	  | 16 lines Merged revisions 172438 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
	  | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
	  startup. Otherwise, if Asterisk runs as a non-root user and the
	  administrator does a 'restart now', Asterisk loses the ability to
	  set QOS on packets. (closes issue #14004) Reported by: nemo
	  Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
	  (license 14) Tested by: Corydon76 ........ ................

2009-01-29 21:35 +0000 [r172434]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
	  revisions 172400 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 |
	  rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12
	  lines channels/chan_dahdi.c * Added doxygen comments to the major
	  dahdi structures. * Fixed PRI and SS7 using an incorrect string
	  value if the extension delimiter is not present in the Dial()
	  function. * Fixed SS7 not checking if the dialed extension is at
	  least as long as the stripmsd option. * Fixed PRI not handling
	  unknown TON/NPI prefix letters correctly. * Fixed some
	  uninitialized string variables on FXS ports.
	  configs/chan_dahdi.conf.sample * Updated some documentation.
	  ........

2009-01-29 16:49 +0000 [r172316]  Tilghman Lesher <tlesher@digium.com>

	* configs/func_odbc.conf.sample, /: Merged revisions 172315 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29
	  Jan 2009) | 2 lines Better document mode=multirow, based upon a
	  conversation with Jared. ........

2009-01-29 13:51 +0000 [r172273]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29
	  Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a
	  couple of fields. closes issue #14339) Reported by: fiddur
	  Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678)
	  ........

2009-01-29 09:56 +0000 [r172217]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24
	  lines Merged revisions 172169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
	  lines Make sure that we always add the hangupcause headers. In
	  some cases, the owner was disconnected before we checked for the
	  cause. This patch implements a temporary storage in the pvt and
	  use that instead. The code is based on ideas from code from
	  Adomjan in issue #13385 (Add support for Reason: header) Thanks
	  to Klaus Darillion for testing! (closes issue #14294) related to
	  issue #13385 Reported by: klaus3000 and adomjan Patches:
	  bug14294b.diff uploaded by oej (license 306) Based on
	  20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
	  (license 487) Tested by: oej, klaus3000 ........ ................

2009-01-28 20:41 +0000 [r172065]  Steve Murphy <murf@digium.com>

	* apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
	  main/features.c, include/asterisk/channel.h: Merged revisions
	  172063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) |
	  52 lines Merged revisions 172030 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
	  46 lines This patch fixes h-exten running misbehavior in
	  manager-redirected situations. What it does: 1. A new Flag value
	  is defined in include/asterisk/channel.h,
	  AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
	  bridge hangup exten code not to run the h-exten there (nor
	  publish the bridge cdr there). It will done at the pbx-loop level
	  instead. 2. In the manager Redirect code, I set this flag on the
	  channel if the channel has a non-null pbx pointer. I did the same
	  for the second (chan2) channel, which gets run if name2 is set...
	  and the first succeeds. 3. I restored the ending of the cdr for
	  the pbx loop h-exten running code. Don't know why it was removed
	  in the first place. 4. The first attempt at the fix for this bug
	  was to place code directly in the async_goto routine, which was
	  called from a large number of places, and could affect a large
	  number of cases, so I tested that fix against a fair number of
	  transfer scenarios, both with and without the patch. In the
	  process, I saw that putting the fix in async_goto seemed not to
	  affect any of the blind or attended scenarios, but still, I was
	  was highly concerned that some other scenarios I had not tested
	  might be negatively impacted, so I refined the patch to its
	  current scope, and jmls tested both. In the process, tho, I saw
	  that blind xfers in one situation, when the one-touch blind-xfer
	  feature is used by the peer, we got strange h-exten behavior. So,
	  I inserted code to swap CDRs and to set the HANGUP_DONT field, to
	  get uniform behavior. 5. I added code to the bridge to obey the
	  HANGUP_DONT flag, skipping both publishing the bridge CDR, and
	  running the h-exten; they will be done at the pbx-loop (higher)
	  level instead. 6. I removed all the debug logs from the patch
	  before committing. 7. I moved the AUTOLOOP set/reset in the
	  h-exten code in res_features so it's only done if the h-exten is
	  going to be run. A very minor performance improvement, but
	  technically correct. (closes issue #14241) Reported by: jmls
	  Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
	  uploaded by murf (license 17) Tested by: murf, jmls ........
	  ................

2009-01-28 17:28 +0000 [r171965]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600
	  (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
	  Jan 2009) | 2 lines Clarify log message (suggested by manxpower
	  on #asterisk-dev) ........ ................

2009-01-28 13:18 +0000 [r171846]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Merged revisions 171838 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons,
	  28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2
	  lines Add a better explanation of the difference between the
	  device namespace and the dialplan for newbies. ........
	  ................

2009-01-27 22:00 +0000 [r171619-171692]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 171691 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600
	  (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
	  2009) | 39 lines Fix devicestate problems for "always-on" agent
	  channels A revision to chan_agent attempted to "inherit" the
	  device state of the underlying channel in order to report the
	  device state of an agent channel more accurately. The problem
	  with the logic here is that it makes no sense to use this for
	  always-on agents. If the agent is logged in, then to the
	  underlying channel, the agent will always appear to be "in use,"
	  no matter if the agent is on a call or not. The reason is that to
	  the underlying channel, the channel is currently in use on a call
	  to the AgentLogin application. The most common cause that I found
	  for this issue to occur was for a SIP channel to be the
	  underlying channel type for an Agent channel. If the SIP phone
	  re-registers, then the registration will cause the device state
	  core to query the device state of the SIP channel. Since the SIP
	  channel is in use, the Agent channel would also inherit this
	  status. Once the agent channel was set to "in use" there was no
	  way that the device state could change on that channel unless the
	  agent logged out. The solution for this problem is a bit
	  different in 1.4 than it is in the other branches. In 1.4, there
	  will be a one-line fix to make sure that only callback agents
	  will inherit device state from their underlying channel type. For
	  the other branches of Asterisk, since callback support has been
	  removed, there is also no need for device state inheritance in
	  chan_agent, so I will simply be removing it from the code. In
	  addition, the 1.4 source is getting a new comment to help the
	  next person who edits chan_agent.c. I'm adding a comment that a
	  agent_pvt's loginchan field may be used to determine if the agent
	  is a callback agent or not. (closes issue #14173) Reported by:
	  nathan Patches: 14173.patch uploaded by putnopvut (license 60)
	  Tested by: nathan, aramirez ........ ................

	* /, main/slinfactory.c: Merged revisions 171622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan
	  2009) | 26 lines Merged revisions 171621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
	  2009) | 18 lines Prevent a crash from occurring when a jitter
	  buffer interpolated frame is removed from a slinfactory
	  slinfactory used the "samples" field of an ast_frame in order to
	  determine the amount of data contained within the frame. In
	  certain cases, such as jitter buffer interpolated frames, the
	  frame would have a non-zero value for "samples" but have NULL
	  "data" This caused a problem when a memcpy call in
	  ast_slinfactory_read would attempt to access invalid memory. The
	  solution in use here is to never feed frames into the slinfactory
	  if they have NULL "data" (closes issue #13116) Reported by:
	  aragon Patches: 13116.diff uploaded by putnopvut (license 60)
	  ........ ................

	* /, apps/app_queue.c: Merged revisions 171618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 |
	  mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24
	  lines Fix queue crashes that would occur after the calling
	  channel was masqueraded. The data passed to the
	  end_bridge_callback was assumed to be data which was still
	  stack'd. The problem was that with some call features, attended
	  transfers in particular, a new bridge thread is started once the
	  feature completes, meaning that when the end_bridge_callback is
	  called, the end_bridge_callback_data was invalid. To fix this
	  problem, there are two measures taken 1. Instead of pointing to
	  stacked data, we now used heap-allocated data for passing to the
	  end_bridge_callback in app_queue 2. Since bridges can end
	  multiple times on a single logical call, we wait until the final
	  bridge is broken to actually set any queue variables. This is
	  accomplished through reference-counting and the use of an
	  end_bridge_callback_data_fixup function in app_queue.c (closes
	  issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
	  by putnopvut (license 60) Tested by: ccesario ........

2009-01-27 16:15 +0000 [r171594-171595]  Matthew Fredrickson <creslin@digium.com>

	* main/ast_expr2.c, main/ast_expr2.h: Revert some changes that
	  shouldn't have made it in

	* main/ast_expr2.c, channels/chan_dahdi.c, main/ast_expr2.h: Make
	  sure we do not go into alarm on PTMP links with non persistent
	  D-channels

2009-01-27 15:13 +0000 [r171529]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23
	  lines Solving the same issue, but a bit different in trunk...
	  Merged revisions 171527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
	  lines Use the same branch tag in CANCEL as in INVITE Originally
	  putnopvut implemented some changes in revision 142079 that
	  according to the bug report seemed to have worked then, but
	  somehow fails now. I guess code, as humans, get old and forget
	  stuff. Anyway, this bug caused CANCEL not to work with picky
	  systems. Thanks Fredrik for pointing out where the bug in the SIP
	  messaging was. (closes issue #14346) Reported by: oej Patches:
	  bug14346.diff uploaded by oej (license 306) Tested by: oej
	  ........ ................

2009-01-26 14:02 +0000 [r171327]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) |
	  17 lines Merged revisions 171264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
	  lines Don't retransmit 401 on REGISTER requests when
	  alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
	  Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000 ........
	  ................

2009-01-26 00:03 +0000 [r171189]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009)
	  | 13 lines Merged revisions 171187 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
	  | 6 lines Correctly track the hookstate (closes issue #13686)
	  Reported by: itiliti Patches: 20081013__bug13686.diff.txt
	  uploaded by Corydon76 (license 14) ........ ................

2009-01-25 13:38 +0000 [r170981]  Sean Bright <sean.bright@gmail.com>

	* /, apps/app_page.c: Merged revisions 170980 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan
	  2009) | 16 lines Merged revisions 170979 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
	  2009) | 9 lines Resolve a logic error that was causing Page() to
	  crash when more than one channel was specified. (closes issue
	  #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
	  uploaded by seanbright (license 71) Tested by: kc0bvu ........
	  ................

2009-01-25 02:50 +0000 [r170944]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009)
	  | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second
	  part of this macro is written as 0[a] instead of a[0], it will
	  force a failure if the macro is used on a C++ object that
	  overloads the [] operator. ........

2009-01-24 13:56 +0000 [r170838]  Tilghman Lesher <tlesher@digium.com>

	* configs/res_odbc.conf.sample, /: Merged revisions 170837 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600
	  (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24
	  Jan 2009) | 2 lines Remove superfluous implementation note
	  (closes issue #14319) ........ ................

2009-01-23 23:52 +0000 [r170830]  Richard Mudgett <rmudgett@digium.com>

	* /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 |
	  rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
	  Fix asterisk.pdf generation if branch name has an underscore in
	  it. ........

2009-01-23 22:59 +0000 [r170791]  Russell Bryant <russell@digium.com>

	* /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 |
	  russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines
	  Don't blow up if a branch name has an underscore in it ........

2009-01-23 20:56 +0000 [r170685-170721]  Mark Michelson <mmichelson@digium.com>

	* configs/res_odbc.conf.sample, /: Merged revisions 170720 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600
	  (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
	  2009) | 8 lines Add notes to the idlecheck explanation in
	  res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
	  Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
	  klaus3000 (license 65) ........ ................

	* contrib/i18n.testsuite.conf, /: Merged revisions 170677 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600
	  (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
	  2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
	  deprecated syntax * Convert Wait,1 to Wait(1) * Convert
	  SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
	  priorities beyond the first Also added test for Chinese numbers,
	  too. (closes issue #14320) Reported by: dant Patches:
	  i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
	  670) ........ ................

2009-01-23 20:19 +0000 [r170659]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 170652 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) |
	  11 lines Merged revisions 170648 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
	  lines When a channel is answered make sure any indications
	  currently playing stop. Usually the phone would do this but if
	  the channel was already answered then they are being generated by
	  Asterisk and we darn well need to stop them. (closes issue
	  #14249) Reported by: RadicAlish ........ ................

2009-01-23  Tilghman Lesher <tlesher@digium.com>

	* Released 1.6.0.5

	* channels/chan_iax2.c: Regression fixes for security fix AST-2009-001

2009-01-06  Tilghman Lesher <tlesher@digium.com>

	* Released 1.6.0.3

	* channels/chan_iax2.c: Security fix AST-2009-001

2008-12-03  Tilghman Lesher <tlesher@digium.com>

	* Released 1.6.0.3-rc1

2008-12-03 14:13 +0000 [r160482]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008)
	  | 14 lines Merged revisions 160480 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
	  | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I
	  guess that having only ip-phones in mind is not a good approach.
	  Since it is possible to have a sip proxy connected to asterisk we
	  could receive a 407 (unauthorized) or 483 (too many hops) as
	  response and dialog ending would not be a good behavior." So
	  modified. ........ ................

2008-12-03 00:53 +0000 [r160427]  Sean Bright <sean.bright@gmail.com>

	* Makefile: Fix some 'make menuselect' breakage introduced by
	  recent merges.

2008-12-02 23:22 +0000 [r160386-160393]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, /: Merged revisions 156388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008)
	  | 12 lines Merged revisions 156386 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
	  | 5 lines When using call limits under 1 second, infinite call
	  lengths are allowed, instead. (closes issue #13851) Reported by:
	  ruddy ........ ................

	* /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008)
	  | 11 lines Merged revisions 156289 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
	  | 3 lines For whatever reason, gcc only warned me about the
	  possible use of an uninitialized variable when compiling 1.6.1.
	  ........ ................

	* /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008)
	  | 16 lines Merged revisions 156178 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
	  | 8 lines (closes issue #13173) Reported by: pep This change adds
	  an announce_thread responsible for playing announcements to an
	  existing conference. This allows all announcing to be immediately
	  stopped if necessary but more importantly allows other threads
	  that need to play something to not block. There are multiple
	  benefits to this, but the actual bug is for solving the scenario
	  for a channel to be unusable after hang up for the entire
	  duration of the parting announcement. The parting announcement
	  can be extremely long depending on what the user recorded upon
	  joining the conference. Reviewed by Russell on Review Board:
	  http://reviewboard.digium.com/r/25/ ........ ................

	* main/astobj2.c, main/asterisk.c, apps/app_while.c,
	  apps/app_dial.c, main/pbx.c, channels/chan_misdn.c,
	  main/manager.c, /, apps/app_meetme.c, channels/chan_sip.c,
	  channels/chan_skinny.c, include/asterisk/astobj2.h,
	  channels/chan_agent.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Merged revisions
	  152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500
	  (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008)
	  | 3 lines Cannot join detached threads. See
	  http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
	  (Closes issue #13400) ........ ................ r153122 |
	  tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10
	  lines Merged revisions 153114 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008)
	  | 3 lines Turn off qualify on uncached realtime peers. (Closes
	  issue #13383) ........ ................ r154264 | tilghman |
	  2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded
	  merge of revisions 154263 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
	  | 3 lines Make the monitor thread non-detached, so it can be
	  joined (suggested by Russell on -dev list). ........
	  ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600
	  (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008)
	  | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state
	  when it receives the indication AST_CONTROL_RINGING. ........
	  ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600
	  (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008)
	  | 9 lines On busy systems, it's possible for the values checked
	  within a single line of code to change, unless the structure is
	  locked to ensure a consistent state. (closes issue #13717)
	  Reported by: kowalma Patches: 20081102__bug13717.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: kowalma ........
	  ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600
	  (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
	  | 7 lines Clarify error message. (closes issue #13809) Reported
	  by: denke Patches: 20081104__bug13809.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: denke ........ ................
	  r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov
	  2008) | 22 lines Merged revisions 155861 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov
	  2008) | 14 lines Channel drivers assume that when their indicate
	  callback is invoked, that the channel on which the callback was
	  called is locked. This patch corrects an instance in chan_agent
	  where a channel's indicate callback is called directly without
	  first locking the channel. This was leading to some observed
	  locking issues in chan_local, but considering that all channel
	  drivers operate under the same expectations, the generic fix in
	  chan_agent is the right way to go. AST-126 ........
	  ................ r156166 | russell | 2008-11-12 11:38:20 -0600
	  (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
	  | 7 lines Move the sanity check that makes sure "always fork" is
	  not set along with the console option to be after the code that
	  reads options from asterisk.conf. This resolves a situation where
	  Asterisk can start taking up 100% when misconfigured. (Thanks to
	  Bryce Porter (x86 on IRC) for letting me log in to his system to
	  figure out what was causing the 100% CPU problem.) ........
	  ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600
	  (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
	  | 6 lines If the SLA thread is not started, then reload causes a
	  memory leak. (closes issue #13889) Reported by: eliel Patches:
	  app_meetme.c.patch uploaded by eliel (license 64) ........
	  ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600
	  (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
	  | 7 lines Provide more space for all the data which can appear in
	  an originating channel name. (closes issue #13398) Reported by:
	  bamby Patches: manager.c.diff uploaded by bamby (license 430)
	  ........ ................ r156756 | tilghman | 2008-11-13
	  18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions
	  156755 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
	  | 6 lines ast_waitfordigit() requires that the channel be up, for
	  no good logical reason. This prevents While/EndWhile from working
	  within the "h" extension. Reported by: jgalarneau (for ABE C.2)
	  Fixed by: me ........ ................ r158066 | mmichelson |
	  2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged
	  revisions 158053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
	  2008) | 12 lines Make sure to set the hangup cause on the calling
	  channel in the case that ast_call() fails. For incoming SIP
	  channels, this was causing us to send a 603 instead of a 486 when
	  the call-limit was reached on the destination channel. (closes
	  issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded
	  by putnopvut (license 60) Tested by: blitzrage ........
	  ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600
	  (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
	  2008) | 16 lines We don't handle 4XX responses to BYE well.
	  According to section 15 of RFC 3261, we should terminate a dialog
	  if we receive a 481 or 408 in response to our BYE. Since I am
	  aware of at least one phone manufacturer who may sometimes send a
	  404 as well, I am being liberal and saying that any 4XX response
	  to a BYE should result in a terminated dialog. (closes issue
	  #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
	  putnopvut (license 60) Closes AST-129 ........ ................
	  r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008)
	  | 10 lines Merged revisions 158539 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
	  | 2 lines When compiling with DEBUG_THREADS, report the real
	  file/func/line for ao2_lock/ao2_unlock ........ ................
	  r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008)
	  | 12 lines Merged revisions 158600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
	  | 5 lines The passed extension may not be the same in the list as
	  the current entry, because we strip spaces when copying the
	  extension into the structure. Therefore, use the copied item to
	  place the item into the list. (found by lmadsen on -dev, fixed by
	  me) ........ ................ r159276 | tilghman | 2008-11-25
	  15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions
	  159269 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008)
	  | 7 lines Don't try to send a response on a NULL pvt. (closes
	  issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch
	  uploaded by eliel (license 64) Tested by: barthpbx ........
	  ................

	* configs/features.conf.sample, apps/app_voicemail.c,
	  apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, /,
	  channels/chan_sip.c, apps/app_queue.c: Merged revisions
	  152216,152287,152369,152467,152569,152605 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008)
	  | 13 lines Merged revisions 152215 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008)
	  | 6 lines Inherit ALL elements of CallerID across a local
	  channel. (closes issue #13368) Reported by: Peter Schlaile
	  Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
	  (license 14) ........ ................ r152287 | jpeeler |
	  2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged
	  revisions 152286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008)
	  | 2 lines Buffer policy setting for half is not needed. ........
	  ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500
	  (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008)
	  | 8 lines Reset all DIAL variables back to blank, in case Dial is
	  called multiple times per call (which could otherwise lead to
	  inconsistent status reports). (closes issue #13216) Reported by:
	  ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76
	  (license 14) Tested by: ruddy ........ ................ r152467 |
	  tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10
	  lines Merged revisions 152463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008)
	  | 3 lines Quoting in the wrong direction (Fixes AST-107) ........
	  ................ r152569 | russell | 2008-10-29 00:34:26 -0500
	  (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008)
	  | 7 lines Fix an incorrect usage of sizeof() (closes issue
	  #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch
	  uploaded by andrew53 (license 519) ........ ................
	  r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) |
	  22 lines Merged revisions 152538 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) |
	  14 lines A little documentation cross-ref between features and
	  dial and queue... I wasted some time (stupidly) trying to get the
	  one-touch parking stuff working, because it didn't occur to me
	  that I had to also have the corresponding options in the dial
	  command! Duh! (In all this time, I never set this up before!) So,
	  to keep some poor fool from suffering the same fate, I made the
	  features.conf.sample file mention the corresponding opts in
	  dial/queue; and the docs for dial/app specifically mention the
	  corresponding decls in the feature.conf file. I hope this doesn't
	  spoil some vast, eternal plan... ........ ................

	* apps/app_speech_utils.c, apps/app_voicemail.c, Makefile,
	  channels/chan_dahdi.c, /, channels/chan_sip.c,
	  include/asterisk/audiohook.h, apps/app_waitforsilence.c,
	  main/features.c, main/audiohook.c, apps/app_queue.c: Merged
	  revisions
	  147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed,
	  08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2
	  lines If we receive DTMF make sure that the state of the speech
	  structure goes back to being not ready. (issue #LUMENVOX-8)
	  ........ ................ r147689 | kpfleming | 2008-10-08
	  17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions
	  147681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct
	  2008) | 3 lines when parsing a text configuration option, ensure
	  that the buffer on the stack is actually large enough to hold the
	  legal values of that option, and also ensure that sscanf() knows
	  to stop parsing if it would overrun the buffer (without these
	  changes, specifying "buffers=...,immediate" would overflow the
	  buffer on the stack, and could not have worked as expected)
	  ........ ................ r148000 | tilghman | 2008-10-09
	  14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions
	  147997 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008)
	  | 4 lines When blank, callerid name and number should display
	  "unknown caller" in voicemail emails. (Closes issue #13643)
	  ........ ................ r148112 | mmichelson | 2008-10-09
	  18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions
	  146026 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) |
	  18 lines (closes issue #13579) Reported by: dwagner (closes issue
	  #13584) Reported by: dwagner Tested by: murf, putnopvut The
	  thought occurred to me that the res= from the extension spawn was
	  ending up being returned from the bridge. "Thou shalt not poison
	  the return value". Made the change and it appears to allow blind
	  xfers to work as normal. If I'm wrong, reopen the bugs. But it
	  looks good to me! Many thanks to putnopvut for helping me
	  reproduce this! ........ ................ r148268 | tilghman |
	  2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged
	  revisions 148257 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008)
	  | 7 lines User not notified of temporary greeting, if ODBC
	  storage is in use. (closes issue #13659) Reported by: moliveras
	  Patches: 20081009__bug13659.diff.txt uploaded by Corydon76
	  (license 14) Tested by: moliveras ........ ................
	  r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008)
	  | 11 lines Merged revisions 148916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008)
	  | 4 lines Ensure that mail headers are 7-bit clean, even when
	  UTF-8 characters are used in headers like 'Subject' and 'To'.
	  Closes AST-107. ........ ................ r148988 | tilghman |
	  2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged
	  revisions 148987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008)
	  | 2 lines Some compilers warn, some don't. Fixing. ........
	  ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500
	  (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008)
	  | 6 lines Check correct values in the return of ast_waitfor();
	  also, get rid of a possible memory leak. (closes issue #13658)
	  Reported by: explidous Patch by: me ........ ................
	  r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct
	  2008) | 15 lines Merged revisions 149130 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct
	  2008) | 7 lines Don't allow reserved characters to be used in
	  register lines in sip.conf. (closes issue #13570) Reported by:
	  putnopvut ........ ................ r149201 | mmichelson |
	  2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged
	  revisions 149200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct
	  2008) | 12 lines Update the queue with the correct number of
	  calls and whether the call was completed within the service level
	  when a transfer takes place. This way, we do not "break" the
	  leastrecent and fewestcalls strategies by not logging a call
	  until after the transferred call has ended. (closes issue #13395)
	  Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
	  by Marquis (license 32) ........ ................ r149205 |
	  mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20
	  lines Merged revisions 149204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct
	  2008) | 12 lines Add a tolerance period for sync-triggered
	  audiohooks so that if packetization of audio is close (but not
	  equal) we don't end up flushing the audiohooks over small
	  inconsistencies in synchronization. Related to issue #13005, and
	  solves the issue for most people who were experiencing the
	  problem. However, a small number of people are still experiencing
	  the problem on long calls, so I am not closing the issue yet
	  ........ ................ r149208 | mmichelson | 2008-10-14
	  18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions
	  149207 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct
	  2008) | 9 lines Call register_peer_exten even in the case that
	  the peer's IP/port does not change. (closes issue #13309)
	  Reported by: dimas Patches: v2-13309.patch uploaded by dimas
	  (license 88) ........ ................

	* channels/misdn/isdn_lib.c, Makefile, channels/chan_dahdi.c,
	  channels/chan_misdn.c, main/manager.c, /: Merged revisions
	  115313,121770,123272,139624,140205,144257 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008)
	  | 10 lines Merged revisions 115312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008)
	  | 2 lines Reverse order, such that user configs override default
	  selections ........ ................ r121770 | crichter |
	  2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines Merged
	  revisions 121751 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008)
	  | 1 line fixed issue with previous commit, the find_free_channel
	  test for channels which where inuse was broken. ........
	  ................ r123272 | russell | 2008-06-17 10:52:13 -0500
	  (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008)
	  | 4 lines Fix a memory leak in astobj2 that was pointed out by
	  seanbright. When a container got destroyed, the underlying bucket
	  list entry for each object that was in the container at that time
	  did not get free'd. ........ ................ r139624 | jpeeler |
	  2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines Merged
	  revisions 139621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008)
	  | 5 lines (closes issue #13359) Reported by: Laureano Patches:
	  originate_channel_check.patch uploaded by Laureano (license 265)
	  ........ ................ r140205 | jpeeler | 2008-08-26 13:48:55
	  -0500 (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008)
	  | 9 lines (closes issue #12071) Reported by: tzafrir Patches:
	  dahdi_close.diff uploaded by tzafrir (license 46) Tested by:
	  tzafrir, jpeeler This patch fixes closing open file descriptors
	  in the case of an error. ........ ................ r144257 |
	  crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
	  Merged revisions 144238 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008)
	  | 1 line improved helptext of misdn_set_opt. ........
	  ................

2008-12-02 18:05 +0000 [r160326-160337]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008)
	  | 1 line remove duplicate comment that I accidentally merged
	  ........

	* channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008)
	  | 7 lines (closes issue #13786) Reported by: tzafrir Readding
	  DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which
	  fixes not being able to make outgoing calls on some FXO adapters:
	  http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
	  ........

2008-12-02 18:01 +0000 [r160228-160322]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008)
	  | 17 lines Merged revisions 160297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
	  | 10 lines When the text does not match exactly (e.g. RTP/SAVP),
	  then the %n conversion fails, and the resulting integer is
	  garbage. Thus, we must initialize the integer and check it
	  afterwards for success. (closes issue #14000) Reported by: folke
	  Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke
	  (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by
	  folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
	  uploaded by folke (license 626) ........ ................

	* include/asterisk/stringfields.h, apps/app_voicemail.c,
	  main/cli.c, main/pbx.c, main/frame.c, /,
	  channels/chan_features.c: Merged revisions 160208 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600
	  (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
	  | 3 lines Ensure that Asterisk builds with --enable-dev-mode,
	  even on the latest gcc and glibc. ........ ................

2008-12-01 23:41 +0000 [r160173]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged
	  revisions 160170-160172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec
	  2008) | 1 line Pay attention to the return value of system(),
	  even if we basically ignore it. ................ r160171 |
	  seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1
	  line Silence a build warning. (chan_phone.c:810: warning: value
	  computed is not used) ................ r160172 | seanbright |
	  2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged
	  revisions 159976 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
	  | 3 lines Get rid of the useless format string and argument in
	  the Bogus/ manager channelname. Noted by kpfleming and name
	  Bogus/manager suggested by eliel ........ ................

2008-12-01  Tilghman Lesher <tlesher@digium.com>

	* Released 1.6.0.2

2008-12-01 21:45 +0000 [r160100]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 160097 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008)
	  | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or
	  bad things happen. ........

2008-12-01 21:07 +0000 [r160096]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk.h, /: Merged revisions 154919 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 |
	  seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2
	  lines Fix a problem found while building res_snmp. ........

2008-12-01 17:39 +0000 [r160005]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 160004 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r160004 | russell | 2008-12-01 11:34:31 -0600
	  (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008)
	  | 6 lines Apply some logic used in iax2_indicate() to
	  iax2_setoption(), as well, since they both have the potential to
	  send control frames in the middle of call setup. We have to wait
	  until we have received a message back from the remote end before
	  we try to send any more frames. Otherwise, the remote end will
	  consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
	  ........ ................

2008-12-01 16:04 +0000 [r159974]  Michiel van Baak <michiel@vanbaak.info>

	* main/manager.c, /: Merged revisions 159898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008)
	  | 11 lines Merged revisions 159897 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
	  | 4 lines make manager compile on OpenBSD. The last (10th)
	  argument to ast_channel_alloc here should be a pointer and NULL
	  is not really a pointer. ........ ................

2008-12-01 14:56 +0000 [r159915]  Russell Bryant <russell@digium.com>

	* .cleancount, /: Merged revisions 159911 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008)
	  | 10 lines Merged revisions 159900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
	  | 2 lines Force a "make clean" to avoid a bizarre build issue ...
	  ........ ................

2008-11-29 18:37 +0000 [r159855]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, Makefile,
	  include/asterisk/logger.h, include/asterisk/res_odbc.h,
	  main/srv.c, channels/chan_misdn.c,
	  include/asterisk/linkedlists.h, main/event.c,
	  include/asterisk/strings.h, utils/extconf.c, makeopts.in,
	  include/asterisk/stringfields.h, utils/check_expr.c,
	  channels/chan_vpb.cc, /, main/utils.c, res/res_config_sqlite.c,
	  utils/frame.c, channels/misdn_config.c, include/asterisk/astmm.h,
	  include/asterisk/compat.h, configure, channels/misdn/ie.c,
	  include/asterisk/module.h, main/features.c, main/dns.c,
	  funcs/Makefile, include/asterisk/devicestate.h,
	  include/asterisk/utils.h, channels/chan_sip.c, main/Makefile,
	  include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac,
	  channels/chan_agent.c, utils/astman.c, include/asterisk/cli.h,
	  include/asterisk/channel.h, include/jitterbuf.h,
	  include/asterisk/manager.h: Merged revisions 159818 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov
	  2008) | 18 lines incorporates r159808 from branches/1.4:
	  ------------------------------------------------------------------------
	  r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
	  2008) | 7 lines update dev-mode compiler flags to match the ones
	  used by default on Ubuntu Intrepid, so all developers will see
	  the same warnings and errors since this branch already had some
	  printf format attributes, enable checking for them and tag
	  functions that didn't have them format attributes in a consistent
	  way
	  ------------------------------------------------------------------------
	  in addition: move some format attributes from main/utils.c to the
	  header files they belong in, and fix up references to the
	  relevant functions based on new compiler warnings ........

2008-11-26 19:58 +0000 [r159558]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /: Merged revisions 159554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 |
	  mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19
	  lines Add some necessary hangup commands in the case that
	  forwarding a call fails 1) Hang up the original destination if
	  the local channel cannot be requested. 2) Hang up the local
	  channel (in addition to the original destination) if ast_call
	  fails when calling the newly created local channel. This prevents
	  channels from sticking around forever in the case of a botched
	  call forward (e.g. to an extension which does not exist). (closes
	  issue #13764) Reported by: davidw Patches: 13764_v2.patch
	  uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
	  ........

2008-11-26 19:18 +0000 [r159536]  Kevin P. Fleming <kpfleming@digium.com>

	* agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
	  Makefile.rules: Merged revisions 159534 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov
	  2008) | 11 lines Merged revisions 159476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
	  2008) | 7 lines simplify (and slightly bug-fix) the recent
	  developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
	  removes dependency files for .i files that are created in
	  COMPILE_DOUBLE mode ........ ................

2008-11-26 18:40 +0000 [r159478]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, /: Merged revisions 159475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 |
	  tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines
	  If the config file does not exist, then the first use crashes
	  Asterisk. (closes issue #13848) Reported by: klaus3000 Patches:
	  udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage
	  ........

2008-11-26 15:01 +0000 [r159439]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Merged revisions 159437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 |
	  mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10
	  lines Don't allow for configuration options to overwrite options
	  set via channel variables on a reload. (closes issue #13921)
	  Reported by: davidw Patches: 13921.patch uploaded by putnopvut
	  (license 60) Tested by: davidw ........

2008-11-25 23:09 +0000 [r159374]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159360 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue,
	  25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
	  15 lines (closes issue #12694) Reported by: yraber Patches:
	  12694.2nd.diff uploaded by murf (license 17) Tested by: murf,
	  laurav Thanks to file (Joshua Colp) for his IAX fix. the change
	  to cdr.c allows no-answer to percolate up into CDR's, and feels
	  like the right place to locate this fix; if BUSY is done here,
	  no-answer should be, too. ........ ................

2008-11-25 22:28 +0000 [r159314]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: I don't care what anyone says, this change is
	  going into 1.6.0. Otherwise, the simple act of logging an agent
	  in spams the CLI with warning messages about failed reads of the
	  alertpipe.

2008-11-25 21:43 +0000 [r159248]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 159247 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600
	  (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
	  (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008)
	  | 7 lines Regression fix for last security fix. Set the iseqno
	  correctly. (closes issue #13918) Reported by: ffloimair Patches:
	  20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: ffloimair ........ ................ ................

2008-11-25 16:21 +0000 [r159024-159094]  Terry Wilson <twilson@digium.com>

	* /, apps/app_festival.c: Merged revisions 159093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 |
	  twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines
	  Add missing variable declaration for PPC code ........

	* channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008)
	  | 2 lines Make chan_usbradio compile under dev mode ........

2008-11-21 22:40 +0000 [r158545]  Steve Murphy <murf@digium.com>

	* /, main/features.c: Merged revisions 158484 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) |
	  19 lines Merged revisions 158483 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
	  11 lines (closes issue #13871) Reported by: mdu113 This one is
	  totally my fault. The code doesn't even create a bridge CDR if
	  the channel CDR has POST_DISABLED. I didn't check for that at the
	  end of the bridge. Fixed with a few small insertions. Tested.
	  Looks good. No cdr generated, no crash, no unnecc. data objects
	  created either. ........ ................

2008-11-21 22:13 +0000 [r158542]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
	  158540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008)
	  | 10 lines Merged revisions 158539 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
	  | 2 lines When compiling with DEBUG_THREADS, report the real
	  file/func/line for ao2_lock/ao2_unlock ........ ................

2008-11-21 20:44 +0000 [r158451]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt,
	  UPGRADE-1.6.txt, CHANGES: Merged revisions 158449 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov
	  2008) | 3 lines as suggested by jtodd, document the purposes of
	  the CHANGES and UPGRADE files ........

2008-11-21 17:12 +0000 [r158376]  Terry Wilson <twilson@digium.com>

	* cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 |
	  twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines
	  Reloading the config and having no changes still initialized some
	  settings to 0. Initialize settings after doing all of the cfg
	  checks. (closes issue #13942) Reported by: davidw Patches:
	  cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by:
	  davidw ........

2008-11-21 01:23 +0000 [r158231-158267]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 158265-158266 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu,
	  20 Nov 2008) | 4 lines Use some magic constants to get the right
	  size for this sscanf statement. Thanks Richard! ........ r158266
	  | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3
	  lines Use a more expressive constant for a 64-bit scanned int
	  ........

	* /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 |
	  mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6
	  lines Fix the build for 32-bit systems. %lu is only 32-bits on
	  32-bit systems, so we need to use %llu instead. Of course %llu is
	  128-bits on 64-bit systems, so we have to cast to unsigned long
	  long. No harm, but it's sure annoying. ........

	* /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 |
	  mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20
	  lines Change the remote user agent session version variable from
	  an int to a uint64_t. This prevents potential comparison problems
	  from happening if the version string exceeds INT_MAX. This was an
	  apparent problem for one user who could not properly place a call
	  on hold since the version in the SDP of the re-INVITE to place
	  the call on hold greatly exceeded INT_MAX. This also aligns with
	  RFC 2327 better since it recommends using an NTP timestamp for
	  the version (which is a 64-bit number). (closes issue #13531)
	  Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut
	  (license 60) Tested by: sgofferj ........

2008-11-20 19:42 +0000 [r158190]  Sean Bright <sean.bright@gmail.com>

	* res/ael/pval.c, /: Merged revisions 158188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 |
	  seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10
	  lines Fix one case where the application argument was not
	  converted from a pipe to a comma. This was causing problems with
	  switch statements with empty expressions. (closes issue #13901)
	  Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by
	  seanbright (license 71) Tested by: seanbright Reviewed by: murf
	  ........

2008-11-20 00:12 +0000 [r157738-157976]  Kevin P. Fleming <kpfleming@digium.com>

	* main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
	  channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael,
	  channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
	  codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
	  channels/misdn, main/db1-ast/mpool, Makefile.rules, res/snmp,
	  pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600
	  (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
	  2008) | 7 lines the gcc optimizer frequently finds broken code
	  (use of uninitalized variables, unreachable code, etc.), which is
	  good. however, developers usually compile with the optimizer
	  turned off, because if they need to debug the resulting code,
	  optimized code makes that process very difficult. this means that
	  we get code changes committed that weren't adequately checked
	  over for these sorts of problems. with this build system change,
	  if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is
	  turned on, when a source file is compiled it will actually be
	  preprocessed (into a .i or .ii file), then compiled once with
	  optimization (with the result sent to /dev/null) and again
	  without optimization (but only if the first compile succeeded, of
	  course). while making these changes, i did some cleanup work in
	  Makefile.rules to move commonly-used combinations of flag
	  variables into their own variables, to make the file easier to
	  read and maintain ........ ................

	* /, res/res_agi.c: Merged revisions 157743 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 |
	  kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line
	  correct small bug introduced during API conversion ........

	* apps/app_stack.c, include/asterisk/agi.h, /, channels/chan_sip.c,
	  res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged
	  revisions 157706 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 |
	  kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5
	  lines make some corrections to the ast_agi_register_multiple(),
	  ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to
	  be consistent with API guidelines also, move UPGRADE.txt to
	  UPGRADE-1.6.txt and make the new UPGRADE.txt contain information
	  about upgrading between Asterisk 1.6 releases ........

2008-11-19 00:33 +0000 [r157601]  Sean Bright <sean.bright@gmail.com>

	* Makefile, /, build_tools/make_version, configure, configure.ac,
	  build_tools/make_buildopts_h, makeopts.in: Merged revisions
	  157600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 |
	  seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10
	  lines Fix a few build problems on Solaris (and check for an md5
	  utility in configure instead of the icky loop I was doing
	  before). (closes issue #13842) Reported by: snuffy Patches:
	  bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff
	  uploaded by seanbright (license 71) Tested by: snuffy ........

2008-11-18 22:59 +0000 [r157307-157541]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Merged revisions 157512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov
	  2008) | 21 lines Merged revisions 157503 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
	  2008) | 13 lines Add some missing invite state changes necessary
	  in the sip_write function. Not setting the invite state correctly
	  on the call was resulting in the Record application leaving empty
	  files. I also have updated the doxygen comment next to the
	  declaration of the INV_EARLY_MEDIA constant to reflect that we
	  also use this state when we *send* a 18X response to an INVITE.
	  (closes issue #13878) Reported by: nahuelgreco Patches:
	  sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
	  (license 162) Tested by: putnopvut ........ ................

	* channels/chan_sip.c: Once again, Russell to the rescue. Use the
	  builtin astobj1 lock of the sip_peer and sip_user instead of
	  adding a new one

	* /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 |
	  mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6
	  lines Based on Russell's advice on the asterisk-dev list, I have
	  changed from using a global lock in update_call_counter to using
	  the locks within the sip_pvt and sip_peer structures instead.
	  ........

	* /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 |
	  mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13
	  lines * Add a lock to be used in the update_call_counter
	  function. * Revert logic to mirror 1.4's in the sense that it
	  will not allow the call counter to dip below 0. These two
	  measures prevent potential races that could cause a SIP peer to
	  appear to be busy forever. (closes issue #13668) Reported by: mjc
	  Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic
	  (license 586) ........

	* apps/app_dial.c, channels/chan_local.c, /, main/features.c,
	  include/asterisk/channel.h, apps/app_followme.c: Merged revisions
	  157306 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov
	  2008) | 20 lines Merged revisions 157305 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
	  2008) | 12 lines Fix a crash in the end_bridge_callback of
	  app_dial and app_followme which would occur at the end of an
	  attended transfer. The error occurred because we initially stored
	  a pointer to an ast_channel which then was hung up due to a
	  masquerade. This commit adds a "fixup" callback to the
	  bridge_config structure to allow for end_bridge_callback_data to
	  be changed in the case that a new channel pointer is needed for
	  the end_bridge_callback. ........ ................

2008-11-18 18:10 +0000 [r157303]  Steve Murphy <murf@digium.com>

	* main/config.c, /: Merged revisions 157302 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 |
	  murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines
	  (closes issue #13420) Reported by: alex70 Patches:
	  13420.13539.patch uploaded by murf (license 17) Tested by: murf,
	  awk This fixes two problems: a spurious linefeed insertion
	  probably left over from pre-precomment times. Only generated when
	  category had no previous comments. The other problem: Insertions
	  could get the line-numbering out of whack and generate negative
	  line numbers, causing chunks of line numbers to be emitted, on
	  the scale of the number of lines up to that point in the file. In
	  such cases, abort the looping, and all is well. ........

2008-11-15 19:47 +0000 [r157107-157165]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
	  revisions 157164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov
	  2008) | 13 lines Merged revisions 157162-157163 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
	  2008) | 1 line dist-clean should remove dependency information
	  files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
	  +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory
	  dist-clean is run, run clean in that directory first, and when
	  running top-level dist-clean, do not run subdirectory clean
	  operations twice ........ ................

	* /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15
	  Nov 2008) | 13 lines major update to doxygen configuration file:
	  1) update to doxygen 1.5.x style file, as used in trunk 2) tell
	  doxygen where are header files are, so include-file processing
	  can be done 3) make all macros that are used to define
	  variables/functions be expanded, so that doxygen will properly
	  document the resulting variable/function 4) make all macros that
	  are used to provide the contents of a variable (structure) be
	  expanded, so that doxygen will be able to document the resulting
	  fields 5) suppress compiler attributes (__attribute__(xxx)) from
	  being seen by doxygen, so it will properly match up function
	  definition and usage (for an example of th effect of this, look
	  at the doxygen docs for ast_log() from before and afte this
	  commit) ........

2008-11-14 17:03 +0000 [r156912]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 156911 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 |
	  tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines
	  Ping is missing the standard double-newline after the event.
	  (closes issue #13903) Reported by: kebl0155 ........

2008-11-14 16:55 +0000 [r156818-156889]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/strings.h, apps/app_queue.c: This is the 1.6.0
	  version of revision 156883 of trunk. This is different in that it
	  preserves the case-sensitiveness of processing queues from
	  configuration. closes issue #13703

	* apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600
	  (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov
	  2008) | 10 lines If the prompt to reenter a voicemail password
	  timed out, it resulted in the password not being saved, even if
	  the input matched what you gave when first prompted to enter a
	  new password. This is because the return value of ast_readstring
	  was checked, but not checked properly. This bug was discovered by
	  Jared Smith during an Asterisk training course. Thanks for
	  reporting it! ........ ................

2008-11-13 19:26 +0000 [r156652-156653]  Brandon Kruse <bkruse@digium.com>

	* main/manager.c: Update to Coding Guidelines

	* main/manager.c, /: Merged revisions 156017 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 |
	  pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines
	  Patch by Ryan Brindley -- Make sure that manager refuses any
	  duplicate 'new category' requests in updateconfig (closes issue
	  #13539) ........

2008-11-12 19:56 +0000 [r156319]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 156299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) |
	  26 lines Merged revisions 156297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
	  18 lines It turns out that the 0x0XX00 codes being returned for
	  N, X, and Z are off by one, as per conversation with jsmith on
	  #asterisk-dev; he was teaching a class and disconcerted that this
	  published rule was not being followed, with patterns _NXX,
	  _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
	  have been. This change, tested on these 3 patterns now picks the
	  proper one. However, this change may surprise users who set up
	  dialplans based on previous behavior, which has been there for
	  what, 2 and half years or so now. ........ ................

2008-11-12 18:57 +0000 [r156251]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 156243 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600
	  (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008)
	  | 11 lines Revert revision 132506, since it occasionally caused
	  IAX2 HANGUP packets not to be sent, and instead, schedule a task
	  to destroy the iax2 pvt structure 10 seconds later. This allows
	  the IAX2 HANGUP packet to be queued, transmitted, and ACKed
	  before the pvt is destroyed. (closes issue #13645) Reported by:
	  dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: vazir Reviewed:
	  http://reviewboard.digium.com/r/51/ ........ ................

2008-11-12 17:47 +0000 [r156170]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /: Merged revisions 156169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov
	  2008) | 15 lines Merged revisions 156167 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
	  2008) | 7 lines When doing some tests, I was having a crash at
	  the end of every call if an attended transfer occurred during the
	  call. I traced the cause to the CDR on one of the channels being
	  NULL. murf suggested a check in the end bridge callback to be
	  sure the CDR is non-NULL before proceeding, so that's what I'm
	  adding. ........ ................

2008-11-11 21:28 +0000 [r156012]  Russell Bryant <russell@digium.com>

	* apps/app_directory.c: Don't blow up if we get NULL when trying to
	  parse out the full name field (fixed for Jared in the training
	  room)

2008-11-11 20:04 +0000 [r156007]  Michiel van Baak <michiel@vanbaak.info>

	* /: remove prop that shouldn't be here

2008-11-11 19:49 +0000 [r155815-156004]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_realtime.c: Merged revisions 155862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 |
	  tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines
	  Make documentation of update method match documentation and
	  update update2 method to match. Reported by: atis, via -dev
	  mailing list. Fixed by: me ........

	* doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 |
	  tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
	  I got tired of saying this in every single bugnote referring to
	  this file. ........

2008-11-09 01:34 +0000 [r155555]  Sean Bright <sean.bright@gmail.com>

	* apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
	  apps/app_followme.c, apps/app_queue.c: Merged revisions 155554
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500
	  (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
	  2008) | 6 lines Use static functions here instead of nested ones.
	  This requires a small change to the ast_bridge_config struct as
	  well. To understand the reason for this change, see the following
	  post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
	  ........ ................

2008-11-07 23:42 +0000 [r155361-155468]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 |
	  mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12
	  lines Set the invite state to INV_CANCELLED in a place that makes
	  more sense. Where it was set before, it was impossible to
	  actually delay sending a CANCEL if we had not yet received a
	  provisional response to an INVITE. (closes issue #13626) Reported
	  by: atis Patches: 13626.patch uploaded by putnopvut (license 60)
	  Tested by: atis ........

	* /, configs/voicemail.conf.sample: Merged revisions 155360 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri,
	  07 Nov 2008) | 8 lines Remove one more instance of the sample
	  configuration lying about what's possible. The tz cannot be set
	  in a context like this. It can only be set in the general section
	  or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing
	  this out ........

2008-11-06 22:50 +0000 [r155123]  Kevin P. Fleming <kpfleming@digium.com>

	* /, res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: Merged
	  revisions 155121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 |
	  kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3
	  lines don't blindly assume that Darwin and Cygwin need
	  GLOB_ABORTED defined; only define it if it is not already defined
	  ........

2008-11-06 19:47 +0000 [r155013]  Mark Michelson <mmichelson@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 155012 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600
	  (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
	  2008) | 8 lines The documentation listed the ability to set
	  'maxmsg' per context. The truth is that you can only set this in
	  the general section or per mailbox. Thus I am updating the sample
	  config file to be more accurate. Thanks to sasargen on IRC for
	  bringing up this issue. ........ ................

2008-11-03 22:30 +0000 [r154062-154081]  Tilghman Lesher <tlesher@digium.com>

	* /: Recorded merge of revisions 154072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008)
	  | 12 lines Merged revisions 154066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008)
	  | 5 lines Attempting to expunge a mailbox when the mailstream is
	  NULL will crash Asterisk. (Closes issue #13829) Reported by:
	  jaroth Patch by: me (modified jaroth's patch) ........
	  ................

	* main/rtp.c, /: Merged revisions 154060 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
	  | 3 lines Remove the potential for a division by zero error.
	  (Closes issue #13810) ........

2008-11-03 00:53 +0000 [r153743-153746]  Kevin P. Fleming <kpfleming@digium.com>

	* /: record revisions that were manually merged

	* apps/app_stack.c, include/asterisk/agi.h, configure,
	  include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
	  configure.ac, include/asterisk/compiler.h: Merge revision 153709
	  from trunk
	  ------------------------------------------------------------------------
	  r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov
	  2008) | 3 lines instead of trying to forcibly load res_agi when
	  app_stack is loaded (even if the administrator didn't want it
	  loaded), use GCC weak symbols to determine whether it was loaded
	  already or not; if it was loaded, then use it.
	  ------------------------------------------------------------------------

	* channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c,
	  channels/chan_h323.c, main/file.c, apps/app_sms.c,
	  pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c,
	  apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c,
	  main/asterisk.c, apps/app_voicemail.c, utils/muted.c,
	  apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c,
	  res/res_musiconhold.c, formats/format_wav_gsm.c,
	  res/res_jabber.c, channels/chan_iax2.c, utils/frame.c,
	  utils/stereorize.c, main/channel.c, channels/chan_dahdi.c,
	  main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c,
	  main/ast_expr2f.c, res/res_agi.c, main/logger.c, main/http.c,
	  formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c,
	  channels/chan_sip.c, formats/format_wav.c, apps/app_festival.c,
	  main/db1-ast/hash/hash_page.c, res/ael/ael.y, res/res_crypto.c,
	  agi/eagi-test.c, utils/astman.c, pbx/pbx_lua.c,
	  formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c:
	  port gcc 4.3.x warning fixes from trunk to this branch

2008-10-31 21:49 +0000 [r153265]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
	  apps/app_followme.c, apps/app_queue.c: Merged revisions 153181
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31
	  Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h'
	  exten into the bridging code, so variables that were set after
	  ast_bridge_call was called would not show up in the 'h' exten.
	  Added a callback function to handle setting variables, etc. from
	  w/in the bridging code. Calls back into a nested function within
	  the function calling ast_bridge_call (closes issue #13793)
	  Reported by: greenfieldtech ........

2008-10-30 21:00 +0000 [r152994]  Sean Bright <sean.bright@gmail.com>

	* /, bootstrap.sh: Merged revisions 152993 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct
	  2008) | 10 lines Merged revisions 152992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct
	  2008) | 2 lines The -I argument to aclocal needs a space before
	  the include directory name. ........ ................

2008-10-30 16:54 +0000 [r152813]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cdr.c, /: Merged revisions 152812 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct
	  2008) | 9 lines Merged revisions 152811 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct
	  2008) | 3 lines instead of comparing the string pointer to 0,
	  let's compare the value that was actually parsed out of the
	  string (found by sparse) ........ ................

2008-10-30 04:28 +0000 [r152772]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 152765 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29
	  Oct 2008) | 5 lines Set up an example stdexten that preserves the
	  original context and extension in the CDR. (Related to issue
	  #13799) Reported by: davidw ........

2008-10-29 20:54 +0000 [r152647]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_directory.c: Merged revisions 152646 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct
	  2008) | 9 lines If there was no named defined in a voicemail.conf
	  mailbox entry, then app_directory would crash when attempting to
	  read that entry from the file. We now check for the NULL or empty
	  string properly so that there will be no crash. (closes issue
	  #13804) Reported by: bluecrow76 ........

2008-10-29 20:13 +0000 [r152644]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c: Small modification to putnopvut's patch to fix
	  this issue. Thanks for all the help, putnopvut! (closes issue
	  #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch
	  uploaded by otherwiseguy (license 396) Tested by: otherwiseguy

2008-10-28 21:39 +0000 [r152443]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c, /: Merged revisions 152442 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008)
	  | 7 lines Only re-add the io port if it was closed, otherwise
	  reload causes a memory leak. (closes issue #13785) Reported by:
	  eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64)
	  ........

2008-10-27 16:33 +0000 [r152157]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c, /: Merged revisions 152134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 |
	  tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines
	  Oops, only delete the ARG variables once upon release. The
	  following section would have removed them again (removing
	  variables from 2 stack frames, instead of just one). ........

2008-10-26 20:26 +0000 [r152062]  Sean Bright <sean.bright@gmail.com>

	* /, funcs/func_strings.c: Merged revisions 152060 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r152060 | seanbright | 2008-10-26 16:25:08 -0400
	  (Sun, 26 Oct 2008) | 15 lines Merged revisions 152059 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct
	  2008) | 7 lines Since passing \0 as the second argument to strchr
	  is valid (and will match the trailing \0 of a string) we need to
	  check that first, otherwise we end up with incorrect results. Fix
	  suggested by reporter. (closes issue #13787) Reported by:
	  meitinger ........ ................

2008-10-23 16:12 +0000 [r151765]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix some memory leaks. These issues are
	  1.6.0 specific. - Freeing the peer got accidentally removed from
	  the peer's destructor. It is still needed for astobj, but not for
	  astobj2. - Fix some places that called find_user or find_peer,
	  but did not release the reference that was returned. (closes
	  issue #13331) Reported by: sergee Patches:
	  chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138)
	  Tested by: sergee

2008-10-20 05:03 +0000 [r151244]  Kevin P. Fleming <kpfleming@digium.com>

	* autoconf (added), autoconf/ast_check_pwlib.m4,
	  autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
	  autoconf/ast_gcc_attribute.m4, bootstrap.sh,
	  autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
	  autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4,
	  autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4,
	  autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4,
	  /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
	  configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4:
	  Merged revisions 151242-151243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct
	  2008) | 9 lines Merged revisions 151240 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct
	  2008) | 3 lines break up acinclude.m4 into individual files,
	  which will make it easier to maintain, easier to add new macros
	  (less patching) and will ease maintenance of these macros across
	  Asterisk branches ........ ................ r151243 | kpfleming |
	  2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines Merged
	  revisions 151241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct
	  2008) | 2 lines rename this macro to properly reflect what it
	  does ........ ................

2008-10-18 02:35 +0000 [r150854]  BJ Weschke <bweschke@btwtech.com>

	* main/manager.c, /: Merged revisions 150817 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r150817 |
	  bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines
	  Using the GetVar handler in AMI is potentially dangerous
	  (insta-crash [tm]) when you use a dialplan function that requires
	  a channel and then you don't provide one or provide an invalid
	  one in the Channel: parameter. We'll handle this situation
	  exactly the same way it was handled in pbx.c back on r61766.
	  We'll create a bogus channel for the function call and destroy it
	  when we're done. If we have trouble allocating the bogus channel
	  then we're not going to try executing the function call at all
	  and run the risk of crashing. (closes issue #13715) reported by:
	  makoto patch by: bweschke ........

2008-10-17 00:19 +0000 [r150308-150313]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Instead of merging commit 150307 to 1.6.0, I
	  had meant to block it in 1.6.1...time to go home :)

	* /, channels/chan_sip.c: Merged revisions 150307 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 |
	  mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14
	  lines After a long discussion on #asterisk-bugs, it seems kind of
	  odd that a channel would be named after the port on which it came
	  in on. For endpoints that always include ":5060" as part of the
	  From: header, it will mean that you have a ton of channels with
	  names like "SIP/5060-3ea38a8b." I am boldly moving forward with
	  this change in trunk, but I'm not touching other branches with
	  this one since this definitely would qualify as a behavior
	  change. If there is a problem with this commit, and I haven't
	  seen the obvious reason why you'd want to name the channel after
	  the port from which the call originated, then please feel free to
	  revert this ........

2008-10-16 16:10 +0000 [r150126]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 150125 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r150125 | rmudgett | 2008-10-16 11:04:45 -0500
	  (Thu, 16 Oct 2008) | 9 lines Merged revisions 150124 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16
	  Oct 2008) | 1 line Fix memory leak found by customer ........
	  ................

2008-10-15 20:18 +0000 [r149757]  BJ Weschke <bweschke@btwtech.com>

	* configs/agents.conf.sample, /: Merged revisions 149756 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400
	  (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008)
	  | 4 lines An update to the documentation/example of
	  agents.conf.sample with the correct parameter for this feature as
	  defined in chan_agent.c (closes issue #13709) ........
	  ................

2008-10-15 11:29 +0000 [r149495]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 149487 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct
	  2008) | 9 lines Merged revisions 149452 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct
	  2008) | 3 lines fix some problems when parsing SIP messages that
	  have the maximum number of headers or body lines that we support
	  ........ ................

2008-10-14 17:39 +0000 [r148914]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c, /: Merged revisions 148913 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r148913 | mmichelson | 2008-10-14 12:38:06 -0500
	  (Tue, 14 Oct 2008) | 17 lines Merged revisions 148912 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct
	  2008) | 9 lines Deadlock prevention in chan_local. (closes issue
	  #13676) Reported by: tacvbo Patches: 13676.patch uploaded by
	  putnopvut (license 60) Tested by: tacvbo ........
	  ................

2008-10-14 10:34 +0000 [r148613-148739]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 148738 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct
	  2008) | 9 lines Merged revisions 148736 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct
	  2008) | 3 lines on Ubuntu (at least), recent versions of ld in
	  binutils delete all debugging symbols when -x is supplied; since
	  the reasons why -x is being passed are lost in the mists of time,
	  remove it so debugging will work properly ........
	  ................

	* /, main/translate.c: Merged revisions 148612 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct
	  2008) | 9 lines Merged revisions 148611 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct
	  2008) | 3 lines it would be nice if this message printing code
	  had actually been tested before it was committed... ........
	  ................

2008-10-10 21:18 +0000 [r148374]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 148373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 |
	  mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8
	  lines Make sure that the inUse and inRinging fields for a sip
	  peer cannot go below zero. This is a regression from 1.4 and so
	  it will be applied to 1.6.0 as well. (closes issue #13668)
	  Reported by: mjc ........

2008-10-10 01:25 +0000 [r148201-148204]  Sean Bright <sean.bright@gmail.com>

	* res/res_config_sqlite.c, apps/app_voicemail.c,
	  include/asterisk.h, /, main/tdd.c, main/cryptostub.c: Merged
	  revisions 148200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 |
	  seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12
	  lines Don't include logger.h in asterisk.h by default as it is
	  causing problems building app_voicemail. Instead, include it
	  where it is needed. This turned out to be a relatively minor
	  issue because other headers include logger.h as well. Need to
	  test -addons before merging this back to 1.6.0. (closes issue
	  #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff
	  uploaded by seanbright (license 71) Tested by: mmichelson
	  ........

	* apps/app_rpt.c: Somehow we got conflict markers checked in! Might
	  need a 1.6.0.1 sooner than we'd like.

2008-10-09 23:31 +0000 [r148147]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 148144 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct
	  2008) | 10 lines Read the callerid in the correct order and make
	  sure to read the Urgent flag value from the IMAP headers. (closes
	  issue #13652) Reported by: jaroth Patches: imapheaders.patch
	  uploaded by jaroth (license 50) ........

2008-10-09 23:26 +0000 [r148124]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/res_ldap.conf.sample: Merged revisions 148120 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09
	  Oct 2008) | 6 lines Fix example schema (closes issue #12860)
	  Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn
	  (license 503) ........

2008-10-09 17:51 +0000 [r147900]  Michiel van Baak <michiel@vanbaak.info>

	* include/asterisk/endian.h, /: Merged revisions 147899 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09
	  Oct 2008) | 5 lines only include this for OpenBSD. At least
	  FreeBSD is borked when including it (closes issue #13649)
	  Reported by: ys ........

2008-10-09 17:47 +0000 [r147897]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 147896 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09
	  Oct 2008) | 4 lines Remove "second form" of extensions, as it no
	  longer applies. Also, cleanup the grammar, formatting, and
	  introduce several clarifications to the text. (Closes issue
	  #13654) ........

2008-10-09 14:56 +0000 [r147809]  Steve Murphy <murf@digium.com>

	* main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c,
	  main/cli.c, configure, channels/console_gui.c, utils/extconf.c,
	  main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, /,
	  include/asterisk/autoconfig.h.in, main/translate.c,
	  channels/vcodecs.c, configure.ac, channels/console_video.c,
	  channels/chan_iax2.c: Merged revisions 147807 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 |
	  murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
	  (closes issue #13557) Reported by: nickpeirson Patches:
	  pbx.c.patch uploaded by nickpeirson (license 579)
	  replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
	  Tested by: nickpeirson, murf 1. replaced all refs to bzero and
	  bcopy to memset and memmove instead. 2. added a note to the
	  CODING-GUIDELINES 3. add two macros to asterisk.h to prevent
	  bzero, bcopy from creeping back into the source 4. removed bzero
	  from configure, configure.ac, autoconfig.h.in ........

2008-10-08 12:16 +0000 [r147458]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample: Remove the sample configuration
	  for configuration sections in chan_dahdi.conf. This code was not
	  merged into 1.6.0. Reported by: angler (closes AST-119)

2008-10-08  Russell Bryant <russell@digium.com>

	* Asterisk 1.6.0.1 released.

	* configs/chan_dahdi.conf.sample: Remove mention of configuration
	  sections for defining channels in chan_dahdi.conf.  This code
	  is in 1.6.1, and was not merged into 1.6.0.

2008-10-01  Russell Bryant <russell@digium.com>

	* Asterisk 1.6.0 released.

2008-09-09  Russell Bryant <russell@digium.com>

	* Asterisk 1.6.0-rc6 released.

2008-09-09 15:44 +0000 [r142065]  Russell Bryant <russell@digium.com>

	* /, main/features.c: Merged revisions 142064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008)
	  | 13 lines Merged revisions 142063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008)
	  | 5 lines Ensure that the stored CDR reference is still valid
	  after the bridge before poking at it. Also, keep the channel
	  locked while messing with this CDR. (fixes crashes reported in
	  issue #13409) ........ ................

2008-09-09 12:34 +0000 [r141996-141999]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 |
	  mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8
	  lines Fix a memory leak in chan_oss (closes issue #13311)
	  Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel
	  (license 64) ........

2008-09-09 01:49 +0000 [r141950]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 141949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 |
	  russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines
	  Modify ast_answer() to not hold the channel lock while calling
	  ast_safe_sleep() or when calling ast_waitfor(). These are
	  inappropriate times to hold the channel lock. This is what has
	  caused "could not get the channel lock" messages from chan_sip
	  and has likely caused a negative impact on performance results of
	  SIP in Asterisk 1.6. Thanks to file for pointing out this section
	  of code. (closes issue #13287) (closes issue #13115) ........

2008-09-08 21:07 +0000 [r141808]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 141807 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008)
	  | 15 lines Merged revisions 141806 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
	  | 7 lines When doing an async goto, detect if the channel is
	  already in the middle of a masquerade. This can happen when
	  chan_local is trying to optimize itself out. If this happens,
	  fail the async goto instead of bursting into flames. (closes
	  issue #13435) Reported by: geoff2010 ........ ................

2008-09-08  Russell Bryant <russell@digium.com>

	* Asterisk 1.6.0-rc5 released.

2008-09-08 20:19 +0000 [r141746]  Jason Parker <jparker@digium.com>

	* Makefile, /, redhat (removed): Merged revisions 141745 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r141745 | qwell | 2008-09-08 15:18:17 -0500
	  (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) |
	  8 lines Remove RPM package targets from Makefile (and all
	  associated parts). This has never worked in 1.4, and we decided
	  that it makes no sense to be done here. There are many distros
	  out there that already have "proper" spec files that can be
	  (re)used. Closes issue #13113 Closes issue #10950 Closes issue
	  #10952 ........ ................

2008-09-08 17:14 +0000 [r141683]  Sean Bright <sean.bright@gmail.com>

	* /, build_tools/make_buildopts_h: Merged revisions 141682 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon,
	  08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on
	  various platforms doesn't choke on the special characters (like
	  ^). (closes issue #13417) Reported by: dougm Patches:
	  13417.make_buildopts_h.patch uploaded by seanbright (license 71)
	  Tested by: dougm ........

2008-09-06 20:21 +0000 [r141567]  Steve Murphy <murf@digium.com>

	* /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9
	  lines Merged revisions 141565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
	  line This fix comes from Joshua Colp The Brilliant, who, given
	  the trace, came up with a solution. This will most likely will
	  close 13235 and 13409. I'll wait till Monday to verify, and then
	  close these bugs. ........ ................

2008-09-06 15:40 +0000 [r141505-141508]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 141504 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008)
	  | 12 lines Merged revisions 141503 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008)
	  | 4 lines Reverting behavior change (AGI should not exit non-zero
	  on SUCCESS) (closes issue #13434) Reported by: francesco_r
	  ........ ................

2008-09-05 22:06 +0000 [r141368-141426]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 141367 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500
	  (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep
	  2008) | 7 lines Agent's should not try to call a channel's
	  indicate callback if the channel has been hung up. It will likely
	  crash otherwise ABE-1159 ........ ................

2008-09-05 14:24 +0000 [r141116-141158]  Steve Murphy <murf@digium.com>

	* main/channel.c, /: Merged revisions 141157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9
	  lines Merged revisions 141156 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
	  line A small change to prevent double-posting of CDR's; thanks to
	  Daniel Ferrer for bringing it to our attention ........
	  ................

	* pbx/ael/ael-test/ref.ael-vtest25 (added), /,
	  pbx/ael/ael-test/ael-vtest25/extensions.ael,
	  pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
	  pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
	  revisions 141115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) |
	  78 lines Merged revisions 141094 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) |
	  70 lines (closes issue #13357) Reported by: pj Tested by: murf
	  (closes issue #13416) Reported by: yarns Tested by: murf If you
	  find this message overly verbose, relax, it's probably not meant
	  for you. This message is meant for probably only two people in
	  the whole world: me, or the poor schnook that has to maintain
	  this code because I'm either dead or unavailable at the moment.
	  This fix solves two reports, both having to do with embedding a
	  function call in a ${} construct. It was tricky because the
	  funccall syntax has parenthesis () in it. And up till now, the
	  'word' token in the flex stuff didn't allow that, because it
	  would tend to steal the LP and RP tokens. To be truthful, the
	  "word" token was the trickiest, most unstable thing in the whole
	  lexer. I was lucky it made this long without complaints. I had to
	  choose every character in the pattern with extreme care, and I
	  knew that someday I'd have to revisit it. Well, the day has come.
	  So, my brilliant idea (and I'm being modest), was to use the
	  surrounding ${} construct to make a state machine and capture
	  everything in it, no matter what it contains. But, I have to now
	  treat the word token like I did with comments, in that I turn the
	  whole thing into a state-machine sort of spec, with new contexts
	  "curlystate", "wordstate", and "brackstate". Wait a minute,
	  "brackstate"? Yes, well, it didn't take very many regression
	  tests to point out if I do this for ${} constructs, I also have
	  to do it with the $[] constructs, too. I had to create a separate
	  pcbstack2 and pcbstack3 because these constructs can occur inside
	  macro argument lists, and when we have two state machines
	  operating on the same structures we'd get problems otherwise. I
	  guess I could have stopped at pcbstack2 and had the brackstate
	  stuff share it, but it doesn't hurt to be safe. So, the pcbpush
	  and pcbpop routines also now have versions for "2" and "3". I had
	  to add the {KEYWORD} construct to the initial pattern for "word",
	  because previously word would match stuff like "default7",
	  because it was a longer match than the keyword "default". But,
	  not any more, because the word pattern only matches only one or
	  two characters now, and it will always lose. So, I made it the
	  winner again by making an optional match on any of the keywords
	  before it's normal pattern. I added another regression test to
	  make sure we don't lose this in future edits, and had to fix just
	  one regression, where it no longer reports a 'cascaded' error,
	  which I guess is a plus. I've given some thought as to whether to
	  apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
	  decided to put it in 1.4 because one of the bug reports was
	  against 1.4; and it is unexpected that AEL cannot handle this
	  situation. It actually reduced the amount of useless "cascade"
	  error messages that appeared in the regressions (by one line,
	  ehhem). There is a possible side-effect in that it does now do
	  more careful checking of what's in those ${} constructs, as far
	  as matching parens, and brackets are concerned. Some users may
	  find a an insidious problem and correct it this way. This should
	  be exceedingly rare, I hope. ........ ................

2008-09-04 18:35 +0000 [r141086]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c, res/res_agi.c: Merged revisions 141039 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500
	  (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008)
	  | 7 lines (closes issue #11979) Fixes multiple parking problems:
	  Crash when executing a park on an extension dialed by AGI due to
	  not returning the proper return code. Crash when using a builtin
	  feature that was a subset of a enabled dynamic feature. Crash due
	  to always hanging up the peer despite the fact that the peer was
	  supposed to be parked. ........ ................

2008-09-03 20:18 +0000 [r140976]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 140975 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 |
	  mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4
	  lines Fix some locking order issues in app_queue. This was
	  brought up by atis on IRC a while ago. ........

2008-09-03  Russell Bryant <russell@digium.com>

	* Asterisk 1.6.0-rc4 released.

2008-09-03 14:17 +0000 [r140825-140827]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 140749 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) |
	  11 lines Merged revisions 140747 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1
	  line I am turning the warnings generated in ast_cdr_free and
	  post_cdr into verbose level 2 messages. Really, they matter
	  little to end users. You either get the CDR's you wanted, or you
	  don't, and it is a bug. For trunk, I am going one step further.
	  These messages were pretty worthless even for debug, so I'm
	  completely removing them. ........ ................

	* main/channel.c, /: Merged revisions 140692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) |
	  13 lines Merged revisions 140690 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1
	  line After reconsidering, with respect to 13409, ast_cdr_detach
	  should be OK, better in fact, than ast_cdr_free, which generates
	  lots of useless warnings that will undoubtably generate
	  complaints. Hmmm. It doesn't hush the useless warnings, but it
	  does allow control of posting via the detach and post routines,
	  for those possible situations, where you'd want to post
	  single-channel cdrs. ........ ................

	* main/channel.c, main/pbx.c, /: Merged revisions 140691 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue,
	  02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) |
	  14 lines (closes issue #13409) Reported by: tomaso Patches:
	  asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license
	  564) I basically spent the day, verifying that this patch solves
	  the problem, and doesn't hurt in non-problem cases. Why valgrind
	  did not plainly reveal this leak absolutely mystifies and stuns
	  me. Many, many thanks to tomaso for finding and providing the
	  fix. ........ ................

2008-09-03 13:27 +0000 [r140818]  Russell Bryant <russell@digium.com>

	* main/poll.c, /: Merged revisions 140817 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008)
	  | 12 lines Merged revisions 140816 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008)
	  | 4 lines Don't freak out if the poll emulation receives NULL for
	  the pollfds array (closes issue #13307) Reported by: jcovert
	  ........ ................

2008-09-02 18:17 +0000 [r140607]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400
	  (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep
	  2008) | 8 lines Make sure to use the correct length of the
	  mohinterpret and mohsuggest buffers when copying configuration
	  values. (closes issue #13336) Reported by:
	  decryptus_proformatique Patches:
	  chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
	  by decryptus (license 555) ........ ................

2008-09-02 15:12 +0000 [r140564-140567]  Russell Bryant <russell@digium.com>

	* apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
	  140566 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 |
	  russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines
	  Update instructions for getting libresample ........

2008-08-27 20:15 +0000 [r140302-140304]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Revert commit 140302. Should not be merging
	  changes like that into a release-candidate branch

	* channels/chan_sip.c: Merged revisions 140301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug
	  2008) | 19 lines Merged revisions 140299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
	  2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when
	  in pedantic mode. The problem was that the wrong tags would be
	  compared depending on the direction of the call. (closes issue
	  #13353) Reported by: flefoll Patches:
	  chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
	  (license 244) ........ ................

2008-08-26 18:12 +0000 [r140170]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 140169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 |
	  russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines
	  Fix building menuselect-tree with PRINT_DIR set. We _must_ use
	  the --quiet flag here, or else some arbitrary text will end up in
	  the resulting menuselect-tree file and things will explode.
	  ........

2008-08-25 21:33 +0000 [r139918]  Sean Bright <sean.bright@gmail.com>

	* build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
	  revisions 139915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug
	  2008) | 17 lines Merged revisions 139909 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
	  2008) | 9 lines Some versions of awk (nawk, for example) don't
	  like empty regular expressions so be slightly more verbose.
	  (closes issue #13374) Reported by: dougm Patches: 13374.diff
	  uploaded by seanbright (license 71) Tested by: dougm ........
	  ................

2008-08-25 21:05 +0000 [r139872]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008)
	  | 10 lines Merged revisions 139869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
	  | 2 lines Make SIPADDHEADER() propagate indefinitely ........
	  ................

2008-08-25 16:00 +0000 [r139774]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /, main/features.c: Merged revisions 139770 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon,
	  25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
	  lines This patch reverts the changes made via 139347, and 139635,
	  as users are seeing adverse difference. I will un-close 13251.
	  Back to the drawing board/ concept/ beginning/ whatever! ........
	  ................

2008-08-24 16:30 +0000 [r139705-139708]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 |
	  tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines
	  Memory leak ........

2008-08-22 22:35 +0000 [r139628-139671]  Steve Murphy <murf@digium.com>

	* /, main/features.c: Merged revisions 139662 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) |
	  14 lines Merged revisions 139635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
	  lines I found some problems with the code I committed earlier,
	  when I merged them into trunk, so I'm coming back to clean up.
	  And, in the process, I found an error in the code I added to
	  trunk and 1.6.x, that I'll fix using this patch also. ........
	  ................

	* apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
	  139627 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) |
	  59 lines Merged revisions 139347 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
	  47 lines (closes issue #13251) Reported by: sergee Tested by:
	  murf THis is a bold move for a static release fix, but I wouldn't
	  have made it if I didn't feel confident (at least a *bit*
	  confident) that it wouldn't mess everyone up. The reasoning goes
	  something like this: 1. We simply cannot do anything with CDR's
	  at the current point (in pbx.c, after the __ast_pbx_run loop).
	  It's way too late to have any affect on the CDRs. The CDR is
	  already posted and gone, and the remnants have been cleared. 2. I
	  was very much afraid that moving the running of the 'h' extension
	  down into the bridge code (where it would be now practical to do
	  it), would result in a lot more calls to the 'h' exten, so I
	  implemented it as another exten under another name, but found, to
	  my pleasant surprise, that there was a 1:1 correspondence to the
	  running of the 'h' exten in the pbx_run loop, and the new spot at
	  the end of the bridge. So, I ifdef'd out the current 'h' loop,
	  and moved it into the bridge code. The only difference I can see
	  is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this
	  is still an important decision point, I can replicate it if there
	  are complaints. To be perfectly honest, the KEEPALIVE situation
	  is not totally clear to me, and how it relates to a post-bridge
	  situation is less clear. I suspect the users will point out
	  everything in total clarity if this steps on anyone's toes! 3. I
	  temporarily swap the bridge_cdr into the channel before running
	  the 'h' exten, which makes it possible for users to edit the cdr
	  before it goes out the door. And, of course, with the
	  endbeforehexten config var set, the users can also get at the
	  billsec/duration vals. After the h exten finishes, the cdr is
	  swapped back and processing continues as normal. Please, all who
	  deal with CDR's, please test this version of Asterisk, and file
	  bug reports as appropriate! ........ I also made a little fix to
	  the app_dial's 'e' option, that is related to my updates.
	  ................

2008-08-22 20:21 +0000 [r139458-139564]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/threadstorage.h, /: Merged revisions 139554 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500
	  (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
	  2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
	  selected (closes issue #13298) Reported by: snuffy Patches:
	  bug13298_20080822.diff uploaded by snuffy (license 35) ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500
	  (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug
	  2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500
	  (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug
	  2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from
	  incorrect locking order between iax2_pvt and ast_channel
	  structures. AST-13 ........ ................

2008-08-21 23:46 +0000 [r139400]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500
	  (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008)
	  | 3 lines Fixes loop that could possibly never exit in the event
	  of a channel never being able to be opened or specify after a
	  restart. (closes issue #11017) ........ ................

2008-08-21 10:02 +0000 [r139282]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008)
	  | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel!
	  (closes issue #13310) Reported by: eliel Patches:
	  chan_gtalk.c.patch uploaded by eliel (license 64) ........

2008-08-20  Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.6.0-rc3 released.

2008-08-20 22:17 +0000 [r139216]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008)
	  | 19 lines Merged revisions 139213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
	  | 11 lines Fix a crash in the ChanSpy application. The issue here
	  is that if you call ChanSpy and specify a spy group, and sit in
	  the application long enough looping through the channel list, you
	  will eventually run out of stack space and the application with
	  exit with a seg fault. The backtrace was always inside of a
	  harmless snprintf() call, so it was tricky to track down.
	  However, it turned out that the call to snprintf() was just the
	  biggest stack consumer in this code path, so it would always be
	  the first one to hit the boundary. (closes issue #13338) Reported
	  by: ruddy ........ ................

2008-08-20 20:12 +0000 [r139155]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Fix bug where the samples were not accurate
	  when in G723 mode, which would cause the timestamp field of the
	  RTP header to be invalid.

2008-08-20 17:30 +0000 [r139104]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 139083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) |
	  20 lines Merged revisions 139074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
	  12 lines (closes issue #13263) Reported by: brainy Tested by:
	  murf The specialized reset routine is tromping on the flags field
	  of the CDR. I made a change to not reset the DISABLED bit. This
	  should get rid of this problem. ........ ................

2008-08-20 15:39 +0000 [r138889-139017]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug
	  2008) | 14 lines Merged revisions 139015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
	  2008) | 6 lines sip_read should properly handle a NULL return
	  from sip_rtp_read. (closes issue #13257) Reported by: travishein
	  ........ ................

	* apps/app_chanspy.c: Manually add revision 138887 from trunk to
	  the 1.6.0 branch. I had misunderstood the policy for when to
	  merge to 1.6.0 since it moved to rc status.

2008-08-19 16:38 +0000 [r138846-138847]  Steve Murphy <murf@digium.com>

	* utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y,
	  res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug
	  2008) | 1 line Oops. put a decl in a generated file. My bad, but
	  fixed now. ........

	* main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y,
	  res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 |
	  murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines
	  These changes are in regards to bug 13249, where users are being
	  surprised by the changes made to the Set app in trunk/1.6.x, as
	  they come from the 1.4 world. They are only bitten if they write
	  their AEL dialplan in the 1.4 world, and then carry it over to a
	  trunk/1.6.x installation where a "make samples" was executed, or
	  where they hand-edited the asterisk.conf file and added the
	  [compat] category with app_set = 1.6 (or higher). (this commit
	  does not totally solve 13249, at least not yet) The change
	  involves issueing a single warning while the AEL file is loading,
	  if: 1. app_set is present in the config file, and set to 1.6 or
	  higher. 2. there are double quotes in an assignment statement (eg
	  x = "hi there";) 3. the warning was not already issued. The
	  standalone app, aelparse, does not (yet) issue this warning. I'd
	  have to have it read in the asterisk.conf file, and that's a bit
	  of hassle. I'll add it if users request it, tho. ........

2008-08-19 00:15 +0000 [r138776-138781]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 138778-138780 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon,
	  18 Aug 2008) | 1 line While we're at it, make this machine
	  parseable too. ........ r138779 | seanbright | 2008-08-18
	  20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we
	  don't need anymore. ........ r138780 | seanbright | 2008-08-18
	  20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now,
	  too (woops) ........

	* /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 |
	  seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3
	  lines Change event header to RegistrationTime to be more
	  consistent (and avoid breaking existing frameworks). Pointed out
	  by Laureano on #asterisk-dev. ........

2008-08-18 20:23 +0000 [r138688-138695]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 138687 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug
	  2008) | 18 lines Merged revisions 138685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
	  2008) | 10 lines Change the inequalities used in app_queue with
	  regards to timeouts from being strict to non-strict for more
	  accuracy. (closes issue #13239) Reported by: atis Patches:
	  app_queue_timeouts_v2.patch uploaded by atis (license 242)
	  ........ ................

2008-08-18 15:54 +0000 [r138632]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 138631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 |
	  qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line
	  Remove option that isn't valid here. ........

2008-08-18 02:14 +0000 [r138519]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008)
	  | 1 line add missing define for SS7 in dahdi_restart ........

2008-08-17 14:14 +0000 [r138443-138483]  Sean Bright <sean.bright@gmail.com>

	* /, main/features.c: Merged revisions 138482 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 |
	  seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6
	  lines Move Uniqueid to the end of the event for those that rely
	  on the position of the name/value pairs, pointed out by
	  snuffy-home on #asterisk-commits. For those of you who rely on
	  the position of name/value pairs in manager events... stop...
	  that is why associative arrays were invented. ........

	* /, main/features.c: Merged revisions 138479 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 |
	  seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7
	  lines Add Uniqueid header to ParkedCall manager event. (closes
	  issue #13323) Reported by: srt Patches:
	  13323_unique_id_for_parkedcalls_event.diff uploaded by srt
	  (license 378) ........

	* main/rtp.c, /: Merged revisions 138476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 |
	  seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7
	  lines Add missing colons to RTCPReceived and RTCPSent manager
	  events. (closes issue #13319) Reported by: srt Patches:
	  13319_rtcp_manager_event_headers.diff uploaded by srt (license
	  378) ........

	* /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug
	  2008) | 7 lines Fix the output of the JitterBufStats manager
	  event. (closes issue #13324) Reported by: srt Patches:
	  13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
	  (license 378) ........

	* configs/cdr_tds.conf.sample, /: Merged revisions 138442 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat,
	  16 Aug 2008) | 4 lines Since it's introduction in revision 3497,
	  cdr_tds has *never* read the port configuration option from
	  cdr_tds.conf. So go ahead and remove it from the sample config.
	  ........

2008-08-16 13:07 +0000 [r138410-138413]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008)
	  | 2 lines Fix compilation warnings (found with dev-mode) ........

2008-08-16 01:14 +0000 [r138333-138362]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500
	  (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
	  Aug 2008) | 1 line fixes use count to properly decrement if an
	  active dahdi channel is destroyed allowing module to be unloaded
	  ........ ................

	* channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500
	  (Fri, 15 Aug 2008) | 20 lines Merged revisions
	  138119,138151,138238 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
	  | 4 lines Fixes the dahdi restart functionality. Dahdi restart
	  allows one to restart all DAHDI channels, even if they are
	  currently in use. This is different from unloading and then
	  loading the module since unloading requires the use count to be
	  zero. Reloading the module is different in that the signalling is
	  not changed from what it was originally configured. Also, this
	  fixes not closing all the file descriptors for D-channels upon
	  module unload (which would prevent loading the module
	  afterwards). (closes issue #11017) ........ r138151 | jpeeler |
	  2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
	  static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
	  r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
	  | 1 line initialize condition variable ss_thread_complete using
	  ast_cond_init ........ ................

2008-08-15 23:03 +0000 [r138207-138262]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  138260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008)
	  | 16 lines Merged revisions 138258 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
	  | 8 lines More fixes for realtime peers. (closes issue #12921)
	  Reported by: Nuitari Patches: 20080804__bug12921.diff.txt
	  uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: Corydon76 ........
	  ................

	* configs/extensions.conf.sample, main/pbx.c, /: Merged revisions
	  138206 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 |
	  tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines
	  Remove deprecated syntax from sample config file (closes issue
	  #13314) Reported by: kue ........

2008-08-15 20:20 +0000 [r138156-138157]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
	  dfd to match 1.4 (left over from DAHDI transition)

2008-08-15 15:12 +0000 [r138029]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: Merged revisions 138028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008)
	  | 17 lines Merged revisions 138027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
	  | 9 lines Ensure that when a hangup occurs in autoservice, that a
	  hangup frame gets properly deferred to be read from the channel
	  owner when it gets taken out of autoservice. (closes issue
	  #12874) Reported by: dimas Patches: v1-12874.patch uploaded by
	  dimas (license 88) ........ ................

2008-08-15 15:04 +0000 [r138025]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 138024 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500
	  (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008)
	  | 8 lines Additional check for more string specifiers than
	  arguments. (closes issue #13299) Reported by: adomjan Patches:
	  20080813__bug13299.diff.txt uploaded by Corydon76 (license 14)
	  func_strings.c-sprintf.patch uploaded by adomjan (license 487)
	  Tested by: adomjan ........ ................

2008-08-14 22:43 +0000 [r137988]  Russell Bryant <russell@digium.com>

	* /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 |
	  russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines
	  Fix a bashism that causes an error when trying to build the pdf
	  on ubuntu ........

2008-08-14 18:48 +0000 [r137934]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug
	  2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes
	  issue #13304) Reported by: eliel Patches: sqlite.patch uploaded
	  by eliel (license 64) (Slightly modified by me) ........

2008-08-14 17:01 +0000 [r137849-137852]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500
	  (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008)
	  | 9 lines When creating the secondary subchannel name, it is
	  necessary to compare to the existing channel name without the
	  "Zap/" or "DAHDI/" prefix, since our test string is also without
	  that prefix. (closes issue #13027) Reported by: dferrer Patches:
	  chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
	  (Slightly modified by me, to compensate for both names) ........
	  ................

2008-08-14  Jason Parker <jparker@digium.com>

	* Asterisk 1.6.0-rc2 released.

2008-08-14 15:37 +0000 [r137814]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 |
	  qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines
	  Make sure we set the socket port, so we don't try to use <ip
	  address>:0. (closes issue #13255) Reported by: falves11 Patches:
	  13255-socketport.diff uploaded by qwell (license 4) Tested by:
	  falves11 ........

2008-08-14 15:20 +0000 [r137783]  Russell Bryant <russell@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r137732 | russell | 2008-08-14 09:15:50 -0500
	  (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008)
	  | 4 lines Comments in this config file were aligned only if your
	  tab size was set to 8. So, convert tabs to spaces so that things
	  should be aligned regardless of what tab size you use in your
	  editor. ........ ................

2008-08-14 15:05 +0000 [r137781]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 |
	  seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8
	  lines If we detect that we are no longer connected, try to
	  reconnect a few times before giving up. This relies on the
	  timeout settings in the freetds.conf file and, unfortunately, on
	  a recent version of FreeTDS (0.82 or newer). I either need to
	  change the current execs to be non-blocking (which I do not want
	  to do) or we have to force people to run with the latest and
	  greatest of FreeTDS. I'm on the fence... ........

2008-08-14 02:04 +0000 [r137681]  Kevin P. Fleming <kpfleming@digium.com>

	* /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug
	  2008) | 9 lines Merged revisions 137679 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
	  2008) | 1 line forgot one module name that changed ........
	  ................

2008-08-13  Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.6.0-rc1 released.

2008-08-13 23:00 +0000 [r137631-137641]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug
	  2008) | 1 line make this script actually work ........

	* /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions
	  137627 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug
	  2008) | 9 lines Merged revisions 137530 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug
	  2008) | 1 line add document describing what users will need to be
	  aware of when upgrading to this version and using DAHDI ........
	  ................

2008-08-13 21:09 +0000 [r137497-137533]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 |
	  qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines
	  Correctly end locally ended calls. (closes issue #12170) Reported
	  by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff
	  uploaded by bbryant (license 36) Tested by: bbryant, pabelanger
	  ........

	* /, apps/app_fax.c: Merged revisions 137496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 |
	  qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines
	  Add FAXMODE variable with what fax transport was used. (closes
	  issue #13252) Patches: v1-13252.patch uploaded by dimas (license
	  88) ........

2008-08-13 14:47 +0000 [r137350-137407]  Sean Bright <sean.bright@gmail.com>

	* /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400
	  (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed,
	  13 Aug 2008) | 1 line Update docs to reflect the change to
	  cdr_tds ........ ................

	* cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 |
	  seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1
	  line Use the ast_vasprintf macro instead of vasprintf directly.
	  ........

2008-08-12 19:48 +0000 [r137300-137302]  Russell Bryant <russell@digium.com>

	* doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008)
	  | 2 lines Grammar hax from Qwell ........

	* doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008)
	  | 3 lines Note that developer documentation belongs in doxygen,
	  and not integrated with the user manual stuff in doc/tex/.
	  ........

2008-08-11 16:15 +0000 [r137240]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 137239 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 |
	  russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines
	  Make PRINT_DIR work as advertised. ........

2008-08-11 14:31 +0000 [r137217]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon,
	  11 Aug 2008) | 7 lines Log the userfield CDR variable like the
	  other CDR backends, assuming the column is actually there. If
	  it's not, we still log everything else as before. (closes issue
	  #13281) Reported by: falves11 ........

2008-08-11 00:27 +0000 [r137160]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, /: Merged revisions 137150 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008)
	  | 13 lines Merged revisions 137138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008)
	  | 5 lines Deallocate database connection handle on disconnect, as
	  we allocate another one on connect. (closes issue #13271)
	  Reported by: dveiga ........ ................

2008-08-09 15:27 +0000 [r136948]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
	  revisions 136947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008)
	  | 18 lines Merged revisions 136946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500
	  (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
	  | 2 lines Regression fixes for Solaris ........ ................
	  ................

2008-08-09 01:16 +0000 [r136860]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 136859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 |
	  tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines
	  Update documentation as to the behavior of AGI in 1.6.0 and
	  higher. Also, add an OOB message that answers the question of, if
	  AGI no longer shuts down the connection on hangup, how will
	  FastAGI know when to stop processing the call? ........

2008-08-08 15:33 +0000 [r136785]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug
	  2008) | 3 lines Fix compilation for ODBC voicemail ........

2008-08-08 06:45 +0000 [r136778]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /,
	  pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h,
	  utils/ael_main.c: Merged revisions 136746 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) |
	  40 lines Merged revisions 136726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) |
	  32 lines (closes issue #13236) Reported by: korihor Wow, this one
	  was a challenge! I regrouped and ran a new strategy for setting
	  the ~~MACRO~~ value; I set it once per extension, up near the
	  top. It is only set if there is a switch in the extension. So, I
	  had to put in a chunk of code to detect a switch in the pval
	  tree. I moved the code to insert the set of ~~exten~~ up to the
	  beginning of the gen_prios routine, instead of down in the switch
	  code. I learned that I have to push the detection of the switches
	  down into the code, so everywhere I create a new exten in
	  gen_prios, I make sure to pass onto it the values of the
	  mother_exten first, and the exten next. I had to add a couple
	  fields to the exten struct to accomplish this, in the
	  ael_structs.h file. The checked field makes it so we don't repeat
	  the switch search if it's been done. I also updated the
	  regressions. ........ ................

2008-08-08 02:36 +0000 [r136753]  Tilghman Lesher <tlesher@digium.com>

	* /: Merged revisions 136751 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 |
	  tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines
	  Removing bad properties ........

2008-08-07 23:42 +0000 [r136719-136724]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a
	  bunch of functions over one level during a merge.

	* apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug
	  2008) | 3 lines Remove one last batch of debug messages ........

	* apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug
	  2008) | 18 lines Merging the imap_consistency_trunk branch to
	  trunk. For an explanation of what "imap_consistency" is, please
	  see svn revision 134223 to the 1.4 branch. Coincidentally, this
	  also fixes a recent bug report regarding the inability to save
	  messages to the new folder when using IMAP storage since they
	  will would be flagged as "seen" and not be recognized as new
	  messages. (closes issue #13234) Reported by: jaroth ........

2008-08-07 20:41 +0000 [r136672-136674]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Removing code that was commented out.

	* codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder
	  interface in the DAHDI. (Issue: DAHDI-42)

2008-08-07 20:26 +0000 [r136632-136663]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: Merged revisions 136660 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 |
	  mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4
	  lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears
	  once for every bridged call ........

	* main/pbx.c, /: Merged revisions 136635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 |
	  mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5
	  lines Don't allow Answer() to accept a negative argument.
	  Negative argument means an infinite delay and we don't want that.
	  ........

	* main/channel.c, /: Merged revisions 136633 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 |
	  mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7
	  lines Fix a calculation error I had made in the poll. The poll
	  would reset to 500 ms every time a non-voice frame was received.
	  The total time we poll should be 500 ms, so now we save the
	  amount of time left after the poll returned and use that as our
	  argument for the next call to poll ........

	* main/channel.c, /: Merged revisions 136631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 |
	  mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13
	  lines Scrap the 500 ms delay when Asterisk auto-answers a
	  channel. Instead, poll the channel until receiving a voice frame.
	  The cap on this poll is 500 ms. The optional delay is still
	  allowable in the Answer() application, but the delay has been
	  moved back to its original position, after the call to the
	  channel's answer callback. The poll for the voice frame will not
	  happen if a delay is specified when calling Answer(). (closes
	  issue #12708) Reported by: kactus ........

2008-08-07 19:19 +0000 [r136598]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn_config.c, channels/chan_misdn.c, /,
	  configs/misdn.conf.sample: Merged revisions 136594 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500
	  (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008)
	  | 5 lines * The allowed_bearers setting in misdn.conf misspelled
	  one of its options: digital_restricted. * Fixed some other
	  spelling errors and typos. ........ ................

2008-08-07 17:44 +0000 [r136506-136543]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/doxyref.h, /: Merged revisions 136542 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500
	  (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  ........ ................

2008-08-07 16:57 +0000 [r136490]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: Merged revisions 136489 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008)
	  | 15 lines Merged revisions 136488 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008)
	  | 7 lines Update persistent state on all exit conditions. (closes
	  issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt
	  uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon
	  ........ ................

2008-08-06 20:16 +0000 [r136113-136192]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500
	  (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008)
	  | 4 lines -C option takes a filename, not a directory path.
	  (closes issue #13007) Reported by: klaus3000 ........
	  ................

	* /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008)
	  | 7 lines Persist DIALGROUP() values in astdb (closes issue
	  #13138) Reported by: Corydon76 Patches:
	  20080725__bug13138.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: pj ........

2008-08-06 16:00 +0000 [r136064]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500
	  (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug
	  2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame
	  type, there are places where ast_rtp_new_source may be called
	  where the tech_pvt of a channel may not yet have an rtp structure
	  allocated. This caused a crash in chan_skinny, which was fixed
	  earlier, but now the same crash has been reported against
	  chan_h323 as well. It seems that the best solution is to modify
	  ast_rtp_new_source to not attempt to set the marker bit if the
	  rtp structure passed in is NULL. This change to
	  ast_rtp_new_source also allows the removal of what is now a
	  redundant pointer check from chan_skinny. (closes issue #13247)
	  Reported by: pj ........ ................

2008-08-06 13:59 +0000 [r136006]  Olle Johansson <oej@edvina.net>

	* /, res/res_jabber.c: Merged revisions 136005 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 |
	  oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines -
	  Formatting - Changing debug messages from VERBOSE to DEBUG
	  channel - Adding a few todo's - Adding a few more "XMPP"'s to
	  compliment Jabber... ........

2008-08-06 03:56 +0000 [r135901-135951]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 135950 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008)
	  | 12 lines Merged revisions 135949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008)
	  | 4 lines Fix a longstanding bug in channel walking logic, and
	  fix the explanation to make sense. (Closes issue #13124) ........
	  ................

	* /, main/translate.c: Merged revisions 135938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008)
	  | 12 lines Merged revisions 135915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008)
	  | 4 lines Since powerof() can return an error condition, it's
	  foolhardy not to detect and deal with that condition. (Related to
	  issue #13240) ........ ................

	* include/asterisk/threadstorage.h, include/asterisk/utils.h, /:
	  Merged revisions 135900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008)
	  | 12 lines Merged revisions 135899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008)
	  | 4 lines 1) Bugfix for debugging code 2) Reduce compiler
	  warnings for another section of debugging code (Closes issue
	  #13237) ........ ................

2008-08-06 00:31 +0000 [r135852]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/abstract_jb.h, main/channel.c, /,
	  main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions
	  135851 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug
	  2008) | 48 lines Merged revisions 135841,135847,135850 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug
	  2008) | 27 lines Merging the issue11259 branch. The purpose of
	  this branch was to take into account "burps" which could cause
	  jitterbuffers to misbehave. One such example is if the L option
	  to Dial() were used to inject audio into a bridged conversation
	  at regular intervals. Since the audio here was not passed through
	  the jitterbuffer, it would cause a gap in the jitterbuffer's
	  timestamps which would cause a frames to be dropped for a brief
	  period. Now ast_generic_bridge will empty and reset the
	  jitterbuffer each time it is called. This causes injected audio
	  to be handled properly. ast_generic_bridge also will empty and
	  reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
	  frame since the change in audio source could negatively affect
	  the jitterbuffer. All of this was made possible by adding a new
	  public API call to the abstract_jb called ast_jb_empty_and_reset.
	  (closes issue #11259) Reported by: plack Tested by: putnopvut
	  ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue,
	  05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel
	  that occurred when I was testing for a memory leak ........
	  r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug
	  2008) | 3 lines Remove properties that should not be here
	  ........ ................

2008-08-05 23:52 +0000 [r135822]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c,
	  include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) |
	  42 lines Merged revisions 135799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) |
	  34 lines (closes issue #12982) Reported by: bcnit Tested by: murf
	  I discovered that also, in the previous bug fixes and changes,
	  the cdr.conf 'unanswered' option is not being obeyed, so I fixed
	  this. And, yes, there are two 'answer' times involved in this
	  scenario, and I would agree with you, that the first answer time
	  is the time that should appear in the CDR. (the second 'answer'
	  time is the time that the bridge was begun). I made the necessary
	  adjustments, recording the first answer time into the peer cdr,
	  and then using that to override the bridge cdr's value. To get
	  the 'unanswered' CDRs to appear, I purposely output them, using
	  the dial cmd to mark them as DIALED (with a new flag), and
	  outputting them if they bear that flag, and you are in the right
	  mode. I also corrected one small mention of the Zap device to
	  equally consider the dahdi device. I heavily tested 10-sec-wait
	  macros in dial, and without the macro call; I tested hangups
	  while the macro was running vs. letting the macro complete and
	  the bridge form. Looks OK. Removed all the instrumentation and
	  debug. ........ ................

2008-08-05 21:38 +0000 [r135749]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500
	  (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008)
	  | 9 lines In a conversion to use ast_strlen_zero, the meaning of
	  the flag IAX_HASCALLERID was perverted. This change reverts IAX2
	  to the original meaning, which was, that the callerid set on the
	  client should be overridden on the server, even if that means the
	  resulting callerid is blank. In other words, if you set
	  "callerid=" in the IAX config, then the callerid should be
	  overridden to blank, even if set on the client. Note that there's
	  a distinction, even on realtime, between the field not existing
	  (NULL in databases) and the field existing, but set to blank
	  (override callerid to blank). ........ ................

2008-08-05 13:27 +0000 [r135599]  Sean Bright <sean.bright@gmail.com>

	* main/cli.c, /: Merged revisions 135598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug
	  2008) | 9 lines Merged revisions 135597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug
	  2008) | 1 line Use PATH_MAX for filenames ........
	  ................

2008-08-04 20:15 +0000 [r135538]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135537 | russell | 2008-08-04 15:15:27 -0500
	  (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008)
	  | 2 lines fix a config sample typo ........ ................

2008-08-04 17:12 +0000 [r135478-135486]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.mandriva.asterisk (added), Makefile,
	  contrib/init.d/rc.mandrake.asterisk (removed), /,
	  contrib/init.d/rc.mandriva.zaptel (added),
	  contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions
	  135485 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 |
	  tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines
	  Rename Mandrake scripts to Mandriva (Closes issue #13221)
	  ........

	* contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500
	  (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008)
	  | 2 lines Define ASTSBINDIR for script (Closes issue #13221)
	  ........ ................

	* apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500
	  (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008)
	  | 6 lines Memory leak on unload (closes issue #13231) Reported
	  by: eliel Patches: app_voicemail.leak.patch uploaded by eliel
	  (license 64) ........ ................

2008-08-04 16:28 +0000 [r135440-135475]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135474 | russell | 2008-08-04 11:28:07 -0500
	  (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008)
	  | 2 lines Add a minor clarification to the documentation of
	  mohinterpret and mohsuggest ........ ................

	* /, channels/chan_console.c: Merged revisions 135439 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008)
	  | 4 lines Be explicit that we don't want a result from this
	  callback. The callback would never indicate a match, so nothing
	  would have been returned anyway, but it was still a poor example
	  of proper usage. ........

2008-08-02 05:15 +0000 [r135266]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 135265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 |
	  murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines
	  (closes issue #13202) Reported by: falves11 Tested by: murf
	  falves11 == The changes I introduce here seem to clear up the
	  problem for me. However, if they do not for you, please reopen
	  this bug, and we'll keep digging. The root of this problem seems
	  to be a subtle memory corruption introduced when creating an
	  extension with an empty extension name. While valgrind cannot
	  detect it outside of DEBUG_MALLOC mode, when compiled with
	  DEBUG_MALLOC, this is certain death. The code in main/features.c
	  is a puzzle to me. On the initial module load, the code is
	  attempting to add the parking extension before the features.conf
	  file has even been opened! I just wrapped the offending call with
	  an if() that will not try to add the extension if the extension
	  name is empty. THis seems to solve the corruption, and let the
	  "memory show allocations" work as one would expect. But, really,
	  adding an extension with an empty name is a seriously bad thing
	  to allow, as it will mess up all the pattern matching algorithms,
	  etc. So, I added a statement to the add_extension2 code to return
	  a -1 if this is attempted. in 1.6.0, the changes to only
	  main/pbx.c were applicable, as apparently the code added to
	  main/features by jpeeler were not included in 1.6.0. ........

2008-08-01 19:30 +0000 [r135198]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug
	  2008) | 6 lines Remove some code that used to do something but
	  does not anymore, mainly to get rid of a shadow warning (but this
	  seemed legitimate enough to fix here instead of in my branch).
	  Thanks to putnopvut for taking a look as well. ........

2008-08-01 17:10 +0000 [r135127-135129]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 |
	  tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines
	  Picky, picky, buildbot ........

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  135126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 |
	  tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
	  SIP should use the transport type set in the Moved Temporarily
	  for the next invite. (closes issue #11843) Reported by:
	  pestermann Patches:
	  20080723__issue11843_302_ignores_transport_16branch.diff uploaded
	  by bbryant (license 36)
	  20080723__issue11843_302_ignores_transport_trunk.diff uploaded by
	  bbryant (license 36) Tested by: pabelanger ........

2008-08-01 14:43 +0000 [r135070]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
	  revisions 135067-135068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 |
	  mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13
	  lines IMAP storage functioned under the assumption that folders
	  such as "Work" and "Family" would be subfolders of the INBOX.
	  This is an invalid assumption to make, but it could be desirable
	  to set up folders in this manner, so a new option for
	  voicemail.conf, "imapparentfolder" has been added to allow for
	  this. (closes issue #13142) Reported by: jaroth Patches:
	  parentfolder.patch uploaded by jaroth (license 50) ........
	  r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug
	  2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE
	  defines... ........

2008-08-01 12:18 +0000 [r135057-135062]  Michiel van Baak <michiel@vanbaak.info>

	* /, apps/app_ices.c: Merged revisions 135059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008)
	  | 10 lines Merged revisions 135058 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008)
	  | 2 lines make app_ices compile on OpenBSD. ........
	  ................

	* /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200
	  (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008)
	  | 8 lines fix some potential deadlocks in chan_skinny (closes
	  issue #13215) Reported by: qwell Patches:
	  2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
	  Tested by: mvanbaak ........ ................

2008-07-31 22:34 +0000 [r135034]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/http.c: Merged revisions 135016 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul
	  2008) | 11 lines Merged revisions 134983 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul
	  2008) | 3 lines accomodate users who seem to lack a sense of
	  humor :-) ........ ................

2008-07-31 21:58 +0000 [r134926-134981]  Tilghman Lesher <tlesher@digium.com>

	* sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions
	  134980 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008)
	  | 16 lines Blocked revisions 134976 via svnmerge ........ r134976
	  | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9
	  lines Specify codecs in callfiles and manager, to allow video
	  calls to be set up from callfiles and AMI. (closes issue #9531)
	  Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt
	  uploaded by Corydon76 (license 14)
	  20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license
	  14) Tested by: Corydon76 ........ ................

	* res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008)
	  | 2 lines Switch command order, to meet with current specs
	  ........

2008-07-31 19:54 +0000 [r134923]  Steve Murphy <murf@digium.com>

	* /, main/features.c: Merged revisions 134922 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) |
	  63 lines Merged revisions 134883 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) |
	  51 lines (closes issue #11849) Reported by: greyvoip Tested by:
	  murf OK, a few days of debugging, a bunch of instrumentation in
	  chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook
	  pages of notes later, I have made the small tweek necc. to get
	  the start time right on the second CDR when: A Calls B B answ. A
	  hits Xfer button on sip phone, A dials C and hits the OK button,
	  A hangs up C answers ringing phone B and C converse B and/or C
	  hangs up But does not harm the scenario where: A Calls B B answ.
	  B hits xfer button on sip phone, B dials C and hits the OK
	  button, B hangs up C answers ringing phone A and C converse A
	  and/or C hangs up The difference in start times on the second CDR
	  is because of a Masquerade on the B channel when the xfer number
	  is sent. It ends up replacing the CDR on the B channel with a
	  duplicate, which ends up getting tossed out. We keep a pointer to
	  the first CDR, and update *that* after the bridge closes. But,
	  only if the CDR has changed. I hope this change is specific
	  enough not to muck up any current CDR-based apps. In my defence,
	  I assert that the previous information was wrong, and this change
	  fixes it, and possibly other similar scenarios. I wonder if I
	  should be doing the same thing for the channel, as I did for the
	  peer, but I can't think of a scenario this might affect. I leave
	  it, then, as an exersize for the users, to find the scenario
	  where the chan's CDR changes and loses the proper start time.
	  ........ ................

2008-07-31 19:41 +0000 [r134918]  Russell Bryant <russell@digium.com>

	* /, apps/app_ices.c: Merged revisions 134917 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008)
	  | 17 lines Merged revisions 134915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008)
	  | 9 lines Get app_ices working again (closes issue #12981)
	  Reported by: dlogan Patches:
	  20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
	  (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
	  bbryant (license 36) Tested by: bbryant ........ ................

2008-07-31 16:53 +0000 [r134816]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.c: Merged revisions 134815 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008)
	  | 15 lines Merged revisions 134814 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008)
	  | 7 lines In case we have some processing threads that free more
	  frames than they allocate, do not let the frame cache grow
	  forever. (closes issue #13160) Reported by: tavius Tested by:
	  tavius, russell ........ ................

2008-07-31 16:07 +0000 [r134760]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 134759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul
	  2008) | 24 lines Merged revisions 134758 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul
	  2008) | 16 lines Add more timeout checks into app_queue,
	  specifically targeting areas where an unknown and potentially
	  long time has just elapsed. Also added a check to try_calling()
	  to return early if the timeout has elapsed instead of potentially
	  setting a negative timeout for the call (thus making it have *no*
	  timeout at all). (closes issue #13186) Reported by:
	  miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
	  (license 60) Tested by: miquel_cabrespina ........
	  ................

2008-07-30 22:41 +0000 [r134651-134707]  Tilghman Lesher <tlesher@digium.com>

	* main/sched.c, /, include/asterisk/sched.h: Merged revisions
	  134703 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 |
	  tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines
	  Oops, wrong define ........

	* /, configure, configure.ac: Merged revisions 134650 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500
	  (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008)
	  | 4 lines Qwell pointed out, via IRC, that the previous fix only
	  worked when explicitly set. When nothing is set, and the option
	  is implied, it breaks, because configure sets the prefix to
	  'NONE'. Fixing. ........ ................

2008-07-30 21:06 +0000 [r134599]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 134598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul
	  2008) | 15 lines Merged revisions 134556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
	  mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
	  lines Fix the parsing of the "reason" parameter in the Diversion:
	  header. (closes issue #13195) Reported by: woodsfsg ........
	  ................

2008-07-30 20:39 +0000 [r134597]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008)
	  | 14 lines Merged revisions 134595 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008)
	  | 6 lines Reduce stack consumption by 12.5% of the max stack size
	  to fix a crash when compiled with LOW_MEMORY. (closes issue
	  #13154) Reported by: edantie ........ ................

2008-07-30 20:25 +0000 [r134561]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
	  mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
	  lines Fix the parsing of the "reason" parameter in the Diversion:
	  header. (closes issue #13195) Reported by: woodsfsg ........

2008-07-30 19:56 +0000 [r134542]  Russell Bryant <russell@digium.com>

	* funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008)
	  | 12 lines Merged revisions 134540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008)
	  | 4 lines Fix a memory leak in func_curl. Every thread that used
	  this function leaked an allocation the size of a pointer.
	  (reported by jmls in #asterisk-dev) ........ ................

2008-07-30 19:49 +0000 [r134482-134539]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 134538 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500
	  (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008)
	  | 4 lines Only override sysconfdir and mandir when prefix=/usr
	  (closes issue #13093) Reported by: pabelanger ........
	  ................

	* /, apps/app_queue.c: Merged revisions 134483 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 |
	  tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines
	  Let "roundrobin" also reference rrmemory, for the 1.6 release (as
	  described in UPGRADE-1.4.txt) (Closes issue #13181) ........

	* /, res/res_agi.c: Merged revisions 134481 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008)
	  | 13 lines Merged revisions 134480 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008)
	  | 5 lines launch_netscript sometimes returns -1, which fails to
	  set AGISTATUS. Map failure to -1, so that AGISTATUS is always
	  set. (closes issue #13199) Reported by: smw1218 ........
	  ................

2008-07-30 18:33 +0000 [r134477]  Mark Michelson <mmichelson@digium.com>

	* /, main/app.c: Merged revisions 134476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul
	  2008) | 12 lines Merged revisions 134475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul
	  2008) | 4 lines Fix a spot where a function could return without
	  bringing a channel out of autoservice. ........ ................

2008-07-30 15:34 +0000 [r134356]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 134355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul
	  2008) | 10 lines Merged revisions 134352 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul
	  2008) | 2 lines use the proper method for building version.h
	  ........ ................

2008-07-29 22:29 +0000 [r134283]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /,
	  apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c:
	  Merged revisions 134260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 |
	  kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2
	  lines build against the now-typedef-free dahdi/user.h, and remove
	  some #ifdefs for features that will always be present in DAHDI
	  ........

2008-07-28 22:16 +0000 [r134164]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500
	  (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008)
	  | 7 lines Detect when sox fails to raise the volume, because sox
	  can't read the file. (closes issue #12939) Reported by:
	  rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: rickbradley ........
	  ................

2008-07-28 19:55 +0000 [r134126]  Mark Michelson <mmichelson@digium.com>

	* /, configure, main/Makefile, configure.ac, CHANGES: Merged
	  revisions 134125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 |
	  mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27
	  lines This commit compensates for buggy poll(2) implementations.
	  Asterisk has, for a long time, had its own implementation of
	  poll(2) which just used the input arguments to call select(2). In
	  1.4, this internal implementation was used for Darwin systems.
	  This was removed in Asterisk trunk at some point, but it seems as
	  though this was not the right move to make. On Mac OS X, it
	  appears as though the poll used to gather CLI input does not
	  respond properly when connecting via a remote Asterisk console.
	  Reverting to the use of Asterisk's poll fixed the issue. Also,
	  there is now an option for the configure script,
	  --enable-internal-poll, which will allow for anyone to use
	  Asterisk's internal poll implementation in case they suspect that
	  their system's poll implementation is buggy. closes issue #11928)
	  Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded
	  by putnopvut (license 60) ........

2008-07-28 16:49 +0000 [r134087]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_parkandannounce.c, main/loader.c, sample.call,
	  contrib/scripts/autosupport, build_tools/cflags.xml,
	  main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c,
	  doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions
	  134086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 |
	  kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3
	  lines remove remaining Zaptel references in various places
	  ........

2008-07-28 16:13 +0000 [r134052]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
	  /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions
	  134050 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 |
	  mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3
	  lines merging the zap_and_dahdi_trunk branch up to trunk ........

2008-07-26 15:34 +0000 [r133942-133982]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, include/asterisk/doxyref.h, /: Include the
	  licensing page in 1.6.0 as well. Now, this page exists in 1.4,
	  trunk, and 1.6.0.

	* /: unblock 133575

	* /, main/devicestate.c: Merged revisions 133945-133946 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26
	  Jul 2008) | 6 lines ast_device_state() gets called in two
	  different ways. The first way is when called from elsewhere in
	  Asterisk to find the current state of a device. In that case, we
	  want to use the cached value if it exists. The other way is when
	  processing a device state change. In that case, we do not want to
	  check the cache because returning the last known state is counter
	  productive. ........ r133946 | russell | 2008-07-26 10:16:20
	  -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache
	  argument ........

2008-07-25 22:09 +0000 [r133863-133905]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25
	  Jul 2008) | 6 lines Update version (closes issue #13163) Reported
	  by: suretec Patches: asterisk.ldif uploaded by suretec (license
	  70) ........

2008-07-25 19:37 +0000 [r133804-133806]  Brandon Kruse <bkruse@digium.com>

	* /: Blocking revert of code changes in r133770

	* main/http.c: Include the http_decode function from trunk to
	  replace the + with a space.

2008-07-25 17:33 +0000 [r133694]  Brandon Kruse <bkruse@digium.com>

	* /: Blocking a fix from trunk for the function http_decode. 1.6.0
	  does not have this function.

2008-07-25 17:26 +0000 [r133671]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /, channels/chan_agent.c, main/devicestate.c:
	  Merged revisions 133665 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008)
	  | 16 lines Merged revisions 133649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008)
	  | 8 lines Fix some errant device states by making the devicestate
	  API more strict in terms of the device argument (only without the
	  unique identifier appended). (closes issue #12771) Reported by:
	  davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
	  (license 14) Tested by: davidw, jvandal, murf ........
	  ................

2008-07-25 15:01 +0000 [r133576-133580]  Russell Bryant <russell@digium.com>

	* /, LICENSE: Merged revisions 133579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008)
	  | 18 lines Merged revisions 133578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r133578 | russell | 2008-07-25 10:00:31 -0500
	  (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
	  | 2 lines Fix the IAX2 URI for calling Digium ........
	  ................ ................

2008-07-25 14:41 +0000 [r133571-133574]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul
	  2008) | 15 lines Merged revisions 133572 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul
	  2008) | 7 lines We need to make sure to null-terminate the "name"
	  portion of SIP URI parameters so that there are no bogus
	  comparisons. Thanks to bbryant for pointing this out. ........
	  ................

2008-07-25 13:25 +0000 [r133567-133569]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 |
	  russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines
	  Minor coding guidelines tweaks ... - Use ast_strlen_zero in one
	  place - check for successful string comparison the way most of
	  Asterisk code does it ........

2008-07-24 21:31 +0000 [r133524]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008)
	  | 11 lines Merged revisions 133488 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008)
	  | 3 lines Fix rtautoclear and rtcachefriends (Closes issue
	  #12707) ........ ................

2008-07-24 20:41 +0000 [r133487]  Russell Bryant <russell@digium.com>

	* /, channels/chan_agent.c: Merged revisions 133486 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008)
	  | 3 lines I made this change from DEVICE_STATE to
	  DEVICE_STATE_CHANGE, but I had it backwards, this is the right
	  event to subscribe to ... ........

2008-07-24 19:55 +0000 [r133449]  Mark Michelson <mmichelson@digium.com>

	* /, main/logger.c: Merged revisions 133448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 |
	  mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12
	  lines Print the correct PID in log messages. Prior to this
	  commit, only the logger thread's PID would be printed. (closes
	  issue #13150) Reported by: atis Patches: log_pid.diff uploaded by
	  putnopvut (license 60) Tested by: eliel ........

2008-07-24 05:21 +0000 [r133392-133405]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions
	  133400 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 |
	  tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines
	  Build the logrotate script according to paths (Closes issue
	  #13147) ........

	* Makefile, /: Merged revisions 133391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 |
	  tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines
	  Optionally install logrotate file (Closes issue #13148) ........

2008-07-23 22:07 +0000 [r133300]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 133299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 |
	  murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines
	  (closes issue #13144) Reported by: murf Tested by: murf For: J.
	  Geis The 'data' field in the ast_exten struct was being 'moved'
	  from the current dialplan to the replacement dialplan. This was
	  not good, as the current dialplan could have problems in the time
	  between the change and when the new dialplan is swapped in. So, I
	  modified the merge_and_delete code to strdup the 'data' field
	  (the args to the app call), and then it's freed as normal. I
	  improved a few messages; I added code to limit the number of
	  calls to the context_merge_incls_swits_igps_other_registrars() to
	  one per context. I don't think having it called multiple times
	  per context was doing anything bad, but it was inefficient. I
	  hope this fixes the problems Mr. Geiss was noting in
	  asterisk-users, see
	  http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
	  ........

2008-07-23 21:50 +0000 [r133297]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r133296 | qwell | 2008-07-23 16:50:20 -0500
	  (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul
	  2008) | 1 line inbandrelease is gone - it's now inbanddisconnect
	  ........ ................

2008-07-23 20:39 +0000 [r133218]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 |
	  bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines
	  Fix issue where tcp in sip is enabled by default, despite what it
	  says in the config sample file. Also fix "sip show settings" for
	  tcp connections. ........

2008-07-23 19:50 +0000 [r133042-133172]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
	  /: Merged revisions 133171 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul
	  2008) | 20 lines Merged revisions 133169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul
	  2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN
	  in app_chanspy should be set at load time, not at compile time,
	  since dahdi_chan_name is determined at load time. Also changed
	  the next_unique_id_to_use to have the static qualifier. Also
	  added the dahdi_chan_name_len variable so that
	  strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
	  the suggestion. ........ ................

	* apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul
	  2008) | 13 lines Merged revisions 133104 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul
	  2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is
	  twelve. The strncmp call in next_channel should account for this.
	  ........ ................

	* apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul
	  2008) | 14 lines Merged revisions 133101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul
	  2008) | 6 lines Update the "last" channel in next_channel in
	  app_chanspy so that the same pseudo channel isn't constantly
	  returned. related to issue #13124 ........ ................

	* channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500
	  (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul
	  2008) | 7 lines Small cleanup. Move the declaration of the
	  DAHDI_SPANINFO variable to the block where it is used. This
	  allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
	  Tzafrir for pointing this out on #asterisk-dev ........
	  ................

2008-07-23 17:21 +0000 [r132978-132983]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008)
	  | 6 lines Yet another conversion of '|' to ',' (closes issue
	  #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch
	  uploaded by eliel (license 64) ........

	* contrib/scripts/asterisk.logrotate (added), /: Merged revisions
	  132977 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 |
	  tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines
	  Add logrotate script for Asterisk (closes issue #13085) Reported
	  by: pabelanger Patches: logrotate uploaded by pabelanger (license
	  224) ........

2008-07-23 16:42 +0000 [r132965-132967]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r132883 | crichter | 2008-07-23 07:07:15 -0500
	  (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23
	  Jul 2008) | 1 line another Fix because of r119585, this commit
	  has broken high frequented BRI Ports, there was a possibility
	  that a channel, that was marked as in_use would be reused later,
	  the corresponding port could got stuck then. So it is recommended
	  to upgrade for chan_misdn users. ........ ................
	  r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul
	  2008) | 2 lines use correct function name... please compile with
	  --enable-dev-mode ................

	* include/asterisk/stringfields.h, /, main/utils.c: Merged
	  revisions 132964 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul
	  2008) | 10 lines Merged revisions 132872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul
	  2008) | 2 lines minor optimization for stringfields: when a field
	  is being set to a larger value than it currently contains and it
	  happens to be the most recent field allocated from the currentl
	  pool, it is possible to 'grow' it without having to waste the
	  space it is currently using (or potentially even allocate a new
	  pool) ........ ................

2008-07-23 08:18 +0000 [r132824]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 |
	  oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines
	  Well, the content of a channel variable may be longer than the
	  size of a pointer... Thanks, eliel! Reported by: eliel Patches:
	  chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
	  (closes issue #13135) ........

2008-07-22 22:20 +0000 [r132797]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul
	  2008) | 11 lines Merged revisions 132777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ Allow
	  Spiraled INVITEs to work correctly within Asterisk. Prior to this
	  change, a spiraled INVITE would cause a 482 Loop Detected to be
	  sent to the caller. With this change, if a potential loop is
	  detected, the Request-URI is inspected to see if it has changed
	  from what was originally received. If pedantic mode is on, then
	  this inspection is fully RFC 3261 compliant. If pedantic mode is
	  not on, then a string comparison is used to test the equality of
	  the two R-URIs. This has been tested by using OpenSER to rewrite
	  the R-URI and send the INVITE back to Asterisk. (closes issue
	  #7403) Reported by: stephen_dredge Modified:
	  branches/1.4/channels/chan_sip.c ........ ................

2008-07-22 22:15 +0000 [r132793]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul
	  2008) | 2 lines correct fix made in r132777... the code *did*
	  compile in dev-mode, as long as libpri was installed and enabled
	  ........

2008-07-22 21:59 +0000 [r132782]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged
	  revisions 132703 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17
	  lines Merged revisions 132645 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9
	  lines The most common question on the #asterisk iRC channel and
	  on mailing lists seems to be in regards to an error message when
	  retransmit fails. This is frequently misunderstood as a failure
	  of Asterisk, not a failure of the network to reach the other
	  party. This document tries to assist the Asterisk user in sorting
	  out these issues by explaining the logic and pointing at some
	  possible causes. Hopefully, we will get other questions now :-)
	  ........ ................

2008-07-22 21:55 +0000 [r132780]  Tilghman Lesher <tlesher@digium.com>

	* configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
	  revisions 132778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008)
	  | 18 lines Merged revisions 132713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500
	  (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
	  | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
	  ................ ................

2008-07-22 21:54 +0000 [r132779]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul
	  2008) | 3 lines Get chan_dahdi to compile in devmode ........

2008-07-22 21:23 +0000 [r132574-132729]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500
	  (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul
	  2008) | 6 lines ensure that if any alarms exist at channel
	  creation time, they are handled identically to if they occurred
	  later, so that later alarm clearing will work properly and 'make
	  sense' (closes issue #12160) Reported by: tzafrir ........
	  ................

	* /, configure, configure.ac, acinclude.m4: Merged revisions 132705
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500
	  (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul
	  2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty'
	  description of what it is doing ........ ................

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
	  configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
	  revisions 132643 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul
	  2008) | 10 lines Merged revisions 132641 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul
	  2008) | 2 lines use renamed libpri API call for controlling this
	  feature (was improperly named before) ........ ................

	* channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500
	  (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul
	  2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI
	  spans, and don't attempt to use channel 24 as a D-channel on
	  spans of unexpected sizes ........ ................

2008-07-21 21:13 +0000 [r132515]  Brett Bryant <bbryant@digium.com>

	* configs/features.conf.sample, configs/gtalk.conf.sample, /,
	  configs/jingle.conf.sample, configs/manager.conf.sample: Merged
	  revisions 132514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 |
	  bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines
	  Update configuration files to add missing options for jingle,
	  gtalk, manager.conf, and features.conf. (closes issue #13128)
	  Reported by: caio1982 Patches: missing_options1.diff uploaded by
	  caio1982 (license 22) ........

2008-07-21 21:02 +0000 [r132512-132513]  Tilghman Lesher <tlesher@digium.com>

	* main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added):
	  Merged revisions 132511 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 |
	  tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines
	  (Step 2 of 2) ........

	* main/fskmodem.c (removed), include/asterisk/fskmodem_int.h
	  (added), build_tools/cflags.xml, main/fskmodem_float.c (added),
	  /, main/tdd.c, include/asterisk/fskmodem.h (removed),
	  main/fskmodem_int.c (added), main/callerid.c,
	  include/asterisk/fskmodem_float.h (added): Merged revisions
	  132510 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 |
	  tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines
	  Optionally build integer-based routines for FSK tone decoding
	  (but default to the more accurate float-based routines). (Closes
	  issue #11679) (Step 1 of 2) ........

2008-07-21 20:55 +0000 [r132467-132509]  Brett Bryant <bbryant@digium.com>

	* /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 |
	  bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines
	  Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't
	  supported on a channel (yet _another_ useful patch by eliel).
	  (closes issue #13081) Reported by: eliel Patches:
	  app_sendtext.c.patch uploaded by eliel (license 64) Tested by:
	  eliel ........

	* /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 |
	  bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines
	  Fix bug where ast_parse_arg would inadvertantly enable sip tcp
	  when parsing a tcpbindaddr if it was disabled. (closes issue
	  #13117) Reported by: pj ........

	* /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008)
	  | 3 lines Fix an issue in iax2 where a call that's been rejected
	  still kept an open channel on the side that attempted to make the
	  call (not the side of the call that rejected the call). Changes
	  were load tested and also approved by Russell. ........

2008-07-21 15:34 +0000 [r132426]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008)
	  | 2 lines make buffers config option (chan_dahdi.conf) parsing
	  safer and added logging in case of failure ........

2008-07-21 14:48 +0000 [r132389-132391]  Russell Bryant <russell@digium.com>

	* apps/app_jack.c, include/asterisk/libresample.h (removed), /,
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, main/libresample
	  (removed), configure.ac, codecs/codec_resample.c, makeopts.in:
	  Merged revisions 132390 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 |
	  russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
	  Remove libresample from the Asterisk source tree. It is now
	  available in its own repository, and must be installed like any
	  other library for Asterisk to use. The two modules that require
	  it are codec_resample and app_jack. To install libresample: $ svn
	  co http://svn.digium.com/svn/libresample/trunk libresample $ cd
	  libresample $ ./configure $ make $ sudo make install This code is
	  currently in our own repository because the build system did not
	  include the appropriate targets for building a dynamic library or
	  for installing the library. ........

	* apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
	  132388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 |
	  russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines
	  Enable higher quality resampling, as it doesn't have a noticeable
	  performance impact on my machine .. ........

2008-07-19 16:47 +0000 [r132313]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: Merged revisions 132312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul
	  2008) | 10 lines Merged revisions 132311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul
	  2008) | 2 lines grant a license exception to allow distribution
	  of Asterisk binaries that use the UW IMAP Toolkit (which is
	  licensed under a non-GPL-compatible license) ........
	  ................

2008-07-19 10:47 +0000 [r132278]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008)
	  | 7 lines fix a couple of comments in sqlite resource driver.
	  (closes issue #13110) Reported by: gknispel_proformatique
	  Patches: res_config_sqlite_comments.patch uploaded by gknispel
	  (license 261) ........

2008-07-18 22:20 +0000 [r132245]  Brett Bryant <bbryant@digium.com>

	* main/manager.c, /: Merged revisions 132242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 |
	  bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines
	  Fixes problem where manager users loaded from users.conf would be
	  removed early (before the routine to load the configuration was
	  finished) because a variable wasn't initialized. ........

2008-07-18 20:58 +0000 [r132114-132207]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c: Merged revisions 132113 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008)
	  | 14 lines Merged revisions 132112 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008)
	  | 6 lines Fix for Taiwanese number syntax (closes issue #12319)
	  Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
	  uploaded by CharlesWang (license 444) ........ ................

2008-07-18 18:53 +0000 [r132111]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) |
	  1 line Make sure we break the poll so that messages queued will
	  be sent on the SS7 when using CLI commands for blocking and
	  blocking of CICs and linksets. ........

2008-07-18 18:51 +0000 [r132110]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c, /: Merged revisions 132109 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008)
	  | 14 lines Merged revisions 132107 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008)
	  | 6 lines Textual clarification (closes issue #13106) Reported
	  by: flefoll Patches: config.c.br14.120173.patch-unknown-directive
	  uploaded by flefoll (license 244) ........ ................

2008-07-18 17:56 +0000 [r132051]  Brett Bryant <bbryant@digium.com>

	* main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18
	  Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and
	  change cdr_radius.c to use the same keyword as the other files
	  (patch by eliel). (closes issue #13104) Reported by: eliel
	  Patches: revision.patch uploaded by eliel (license 64) ........

2008-07-18 17:40 +0000 [r131984-132047]  Tilghman Lesher <tlesher@digium.com>

	* main/sched.c, /: Merged revisions 131989 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008)
	  | 10 lines Merged revisions 131988 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008)
	  | 2 lines Oops ........ ................

	* main/sched.c, /, include/asterisk/sched.h: Merged revisions
	  131986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008)
	  | 10 lines Merged revisions 131985 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008)
	  | 2 lines Preserve ABI compatibility with last change ........
	  ................

	* main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c:
	  Merged revisions 131982 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008)
	  | 10 lines Merged revisions 131970 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008)
	  | 2 lines Make the ast_assert call within ast_sched_del report
	  something useful. ........ ................

2008-07-18 16:16 +0000 [r131924]  Kevin P. Fleming <kpfleming@digium.com>

	* main/dlfcn.c (removed), main/loader.c, /, main/Makefile,
	  include/asterisk/dlfcn-compat.h (removed): Merged revisions
	  131923 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul
	  2008) | 10 lines Merged revisions 131921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul
	  2008) | 2 lines remove the dlfcn compatibility stuff, because no
	  platforms that Asterisk currently runs on it use it, and it
	  doesn't build anyway ........ ................

2008-07-18 15:39 +0000 [r131917]  Brett Bryant <bbryant@digium.com>

	* /, main/features.c: Merged revisions 131916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008)
	  | 12 lines Merged revisions 131915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008)
	  | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER
	  variable isn't always set to the other end of the blind transfer.
	  (closes issue #12586) ........ ................

2008-07-17 22:45 +0000 [r131869]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008)
	  | 6 lines Add configuration option to chan_dahdi.conf to allow
	  buffering policy and number of buffers to be configured per
	  channel. Syntax: buffers=<num of buffers>,<policy> Where the
	  number of buffers is some non-negative integer and the policy is
	  either "full", "half", or "immediate". ........

2008-07-17 21:27 +0000 [r131830]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 |
	  mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10
	  lines Document that the duration of dtmf may be passed to the
	  SendDTMF application. Also correct the default pause between
	  digits. (closes issue #13102) Reported by: eliel Patches:
	  app_senddtmf.c.patch uploaded by eliel (license 64) ........

2008-07-17 20:38 +0000 [r131754-131792]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500
	  (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008)
	  | 7 lines Revert part of issue #5620 (revision 6965) as it
	  appears that it was in error. This should fix talk call progress
	  on analog lines. (closes issue #12178) Reported by: michael-fig
	  Patches: 20080717__bug12178.diff.txt uploaded by Corydon76
	  (license 14) ........ ................

	* res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008)
	  | 6 lines Fix memory leaks (closes issue #13099) Reported by:
	  gknispel_proformatique Patches:
	  res_config_sqlite_leak_on_error.patch uploaded by gknispel
	  (license 261) ........

2008-07-17 18:15 +0000 [r131718]  Brett Bryant <bbryant@digium.com>

	* /, main/features.c: Merged revisions 131717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 |
	  bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines
	  Fix a memory leak in register_group_feature when attempting to
	  register a feature without specifying a group or feature to
	  register. (closes issue #13101) Reported by: eliel Patches:
	  features.c.patch uploaded by eliel (license 64) ........

2008-07-17 15:46 +0000 [r131682]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008)
	  | 4 lines Fix memory leak. (Closes issue #13096) Reported by
	  gknispel_proformatique ........

2008-07-16 23:56 +0000 [r131571]  Steve Murphy <murf@digium.com>

	* /: The commit from 131570 should not be applied to 1.6.0, as it
	  is not as necessary, because log_show_lock in trunk is not
	  available in 1.6.0, and is estimated to be the only function that
	  might care about the lock_addr values.

2008-07-16 22:18 +0000 [r131493]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500
	  (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008)
	  | 6 lines Fix a bug in iax2 registration that allowed peers to
	  register with case-insensitive names (user_cmp_cb and peer_cmp_cb
	  are now both case-sensitive). (closes issue #13091) ........
	  ................

2008-07-16 21:54 +0000 [r131455-131486]  Brett Bryant <bbryant@digium.com>

	* /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008)
	  | 4 lines Fixes sysinfo operator issue also fixed elsewhere in
	  r131445. (issue #13057) ........

	* main/asterisk.c, /: Merged revisions 131445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 |
	  bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines
	  Fixes an issue with "core show sysinfo" that used the wrong
	  operator to calculate the number of bytes from a sysinfo
	  structure. unsigned long. (closes issue #13057) Reported by:
	  eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
	  ........

2008-07-16 20:48 +0000 [r131423]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r131422 | russell | 2008-07-16 15:48:27 -0500
	  (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008)
	  | 7 lines Always ensure that the channel's tech_pvt reference is
	  NULL after calling the destroy callback. (closes issue #13060)
	  Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch
	  uploaded by jpgrayson (license 492) ........ ................

2008-07-16 20:24 +0000 [r131301-131378]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 131375 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul
	  2008) | 22 lines Merged revisions 131369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul
	  2008) | 14 lines Move the init_queue call back to where it used
	  to be (changed Sept 12 last year). It was moved then to prevent a
	  memory leak. Since then, the same memory leak recurred and was
	  fixed in a better way. Now it has been found that the placement
	  of this init_queue call can cause problems if a realtime queue
	  has values changed to an empty string. The problem is that the
	  default value for that queue parameter would not be set. (closes
	  issue #13084) Reported by: elbriga ........ ................

	* res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul
	  2008) | 9 lines Don't try to dereference the dbfile pointer if we
	  know that it's NULL. (closes issue #13092) Reported by:
	  gknispel_proformatique Patches:
	  trunk_sqlite_check_vars_null.patch uploaded by gknispel (license
	  261) ........

	* /, apps/app_queue.c: Merged revisions 131358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul
	  2008) | 14 lines Merged revisions 131357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul
	  2008) | 6 lines Apparently, "thread safety" is important,
	  whatever that means. :P (Thanks Russell!) ........
	  ................

	* /, apps/app_queue.c: Merged revisions 131300 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul
	  2008) | 21 lines Merged revisions 131299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul
	  2008) | 13 lines Make absolutely certain that the transfer
	  datastore is removed from the calling channel once the caller is
	  finished in the queue. This could have weird con- sequences when
	  dialing local queue members when multiple transfers occur on a
	  single call. Also fixed a memory leak that would occur when an
	  attended transfer occurred from a queue member. (closes issue
	  #13047) Reported by: festr ........ ................

2008-07-16 18:20 +0000 [r131248]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /: Merged revisions 131243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) |
	  27 lines Merged revisions 131242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) |
	  19 lines (closes issue #13090) Reported by: murf The problem was
	  that, esoteric as it is, because the hangerupper context
	  immediately preceded the std-priv-extent macro, that the checking
	  code accidentally would fall from traversing hangerupper into the
	  std-priv-exten macro, where it would hit the hangerupper in the
	  'includes', and proceed into an infinite recursion. A small fix
	  to traverse into the statements of the context instead of the
	  context solves this issue. I also added some commented out
	  printfs for debug, which were pretty handy in the face of a dorky
	  gdb. This was a problem around since the package was first
	  written; but evidently pretty rare in turning up in the field.
	  ........ ................

2008-07-16 15:04 +0000 [r131206]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_agent.c: add missing terminator argument for
	  ast_event_subscribe(). Without it the function will randomly walk
	  on the stack possibly causing a panic

2008-07-16 00:54 +0000 [r131168]  Tilghman Lesher <tlesher@digium.com>

	* /, main/logger.c: Merged revisions 131166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 |
	  tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines
	  Fix rotate strategy (Closes issue #13086) ........

2008-07-15 23:41 +0000 [r131131]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 131129 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 |
	  murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines
	  (closes issue #12960) Reported by: mnicholson Spent most of the
	  day on this bug, and the solution was so simple. Just had to find
	  and understand the problem. The problem was, that the routine to
	  copy the existing switches, includes, and ignorepats from the old
	  context to the new one, wasn't getting called when the context is
	  already existent. (In other words, if AEL is adding a new context
	  to the mix, they get copied, but if pbx_config already defined a
	  context, then the copy wasn't happening. This made no sense, so I
	  moved the call to copy the includes & etc, no matter the case.
	  ........

2008-07-15 18:47 +0000 [r131073]  Russell Bryant <russell@digium.com>

	* /, res/res_agi.c: Merged revisions 131072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 |
	  russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines
	  Fix a couple of places in res_agi where the agi_commands lock
	  would not be released, causing a deadlock. (Reported by mvanbaak
	  in #asterisk-dev, discovered by bbryant's change to the lock
	  tracking code to yell at you if a thread exits with a lock still
	  held) ........

2008-07-15 18:29 +0000 [r131060]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged
	  revisions 131044 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008)
	  | 16 lines Merged revisions 130959 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008)
	  | 8 lines astman_send_error does not need a newline appended --
	  the API takes care of that for us. (closes issue #13068) Reported
	  by: gknispel_proformatique Patches:
	  asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
	  asterisk_trunk_astman_send.patch uploaded by gknispel (license
	  261) ........ ................

2008-07-15 18:00 +0000 [r131014]  Michiel van Baak <michiel@vanbaak.info>

	* main/cdr.c, /: Merged revisions 131013 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008)
	  | 15 lines Merged revisions 131012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008)
	  | 7 lines remove 4 lines of redundant code. (closes issue #13080)
	  Reported by: gknispel_proformatique Patches:
	  trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
	  ........ ................

2008-07-15 13:14 +0000 [r130946]  Steve Murphy <murf@digium.com>

	* utils/conf2ael.c, utils/Makefile, res/ael/pval.c,
	  channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c,
	  pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
	  utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c,
	  res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c,
	  apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /,
	  channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y,
	  channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  Merging this rev from trunk to 1.6.0 was not simple. Why? Because
	  we've enhanced trunk to do a [fast] merge-and-delete operation
	  which also solved problems with contexts having entries from
	  different registrars. Fast as in the amount of time the contexts
	  are locked down. That *is* fast, but traversing the entire
	  dialplan looking for priorities to delete takes more time
	  overall. This particular fix involved pulling in those
	  enhancements from trunk, along with all the various fixes and
	  refinements made along the way. Merging all this from trunk into
	  1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert
	  all but the prop changes c. catalog all revisions to pbx.c since
	  1.6.0 was forked (at rev 105596). d. catalog all revisions to
	  pbx.c in trunk since 1.6.0 was forked, making special note of all
	  revs that were not merged into 1.6.0. e. study each rev in trunk
	  not applied to 1.6.0, and determine if it was involved in the
	  merge_and_delete enhancements in trunk. 25 commits were done in
	  1.6.0, all but one (106306) was a merge from trunk. Trunk had 22
	  additional changes, of which 7 were involved in the
	  merge_and_delete enhancements: 106757 108894 109169 116461 123358
	  130145 130297 f. Go to trunk and collect patches, one by one, of
	  the changes made by each rev across the entire source tree, using
	  svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0,
	  and resolve all failures and compilation problems before
	  proceding to the next patch. h. test the stuff. i. profit!
	  ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul
	  2008) | 40 lines (closes issue #13041) Reported by: eliel Tested
	  by: murf (closes issue #12960) Reported by: mnicholson In this
	  'omnibus' fix, I **think** I solved both the problem in 13041,
	  where unloading pbx_ael.so caused crashes, or incomplete removal
	  of previous registrar'ed entries. And I added code to completely
	  remove all includes, switches, and ignorepats that had a matching
	  registrar entry, which should appease 12960. I also added a lot
	  of seemingly useless brackets around single statement if's, which
	  helped debug so much that I'm leaving them there. I added a
	  routine to check the correlation between the extension tree lists
	  and the hashtab tables. It can be amazingly helpful when you have
	  lots of dialplan stuff, and need to narrow down where a problem
	  is occurring. It's ifdef'd out by default. I cleaned up the code
	  around the new CIDmatch code. It was leaving hanging extens with
	  bad ptrs, getting confused over which objects to remove, etc. I
	  tightened up the code and changed the call to remove_exten in the
	  merge_and_delete code. I added more conditions to check for empty
	  context worthy of deletion. It's not empty if there are any
	  includes, switches, or ignorepats present. If I've missed
	  anything, please re-open this bug, and be prepared to supply
	  example dialplan code. ........

2008-07-15 00:00 +0000 [r130891]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500
	  (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008)
	  | 8 lines Override the callerid in all cases when the callerid is
	  set in the user, not just when a remote callerid is set. Also, if
	  not set in the user, allow the remote CallerID to pass through.
	  (closes issue #12875) Reported by: dimas Patches:
	  20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
	  ........ ................

2008-07-14 22:24 +0000 [r130795-130855]  Mark Michelson <mmichelson@digium.com>

	* main/asterisk.c, /: Merged revisions 130854 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 |
	  mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9
	  lines Fix a memory leak in the case that /dev/null cannot be
	  opened when running startup commands from cli.conf (closes issue
	  #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by
	  eliel (license 64) ........

	* apps/app_dial.c, /: Merged revisions 130794 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul
	  2008) | 16 lines Merged revisions 130792 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul
	  2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in
	  app_dial to be sure there are no audiohooks present on the
	  channels involved. This fixed a one-way audio situation I had in
	  my test setup. I couldn't find any open issues that suggested
	  one-way audio with regards to mixmonitor (or other audiohook)
	  usage, though. ........ ................

2008-07-14 17:22 +0000 [r130752]  Michiel van Baak <michiel@vanbaak.info>

	* main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008)
	  | 18 lines Merged revisions 130735 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008)
	  | 10 lines notify the user that dnsmgr refresh wont work when
	  dnsmgr is not enabled. Previously this command would
	  automagically appear and disappear. This was confusing. (closes
	  issue #12796) Reported by: chappell Patches:
	  dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
	  russell, chappell, mvanbaak ........ ................

2008-07-14 10:40 +0000 [r130636-130637]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/astobj.h: Merged revisions 129987 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129987 | russell | 2008-07-11 09:22:44 -0500
	  (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008)
	  | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........
	  ................

	* /, main/audiohook.c: Merged revisions 130635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008)
	  | 10 lines Merged revisions 130634 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008)
	  | 2 lines Bump up the debug level for a message. ........
	  ................

2008-07-13 23:20 +0000 [r130575-130582]  Michiel van Baak <michiel@vanbaak.info>

	* /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile:
	  Merged revisions 130578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 |
	  mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15
	  lines Make all sed calls Posix sed compatible. To make sure
	  nobody commits script-modified files we first make a backup of
	  asterisk.tex, run the script, generate the pdf and / or html, and
	  put the original asterisk.tex back. This will guard us for the
	  stuff that happened before that someone committed a locally
	  modified asterisk.tex, with changes done by this script. (closes
	  issue #13062) Reported by: mvanbaak Patches:
	  sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by:
	  mvanbaak Feedback from Corydon. Thanks for taking the time to go
	  through this. ........

	* main/manager.c, /: Merged revisions 130574 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008)
	  | 16 lines Merged revisions 130573 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008)
	  | 8 lines fix memory leak when originate from manager cannot
	  create a thread (closes issue #13069) Reported by:
	  gknispel_proformatique Patches:
	  asterisk_trunk_action_originate.patch uploaded by gknispel
	  (license 261) Tested by: gknispel_proformatique, mvanbaak
	  ........ ................

2008-07-13 17:59 +0000 [r130516]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500
	  (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008)
	  | 4 lines Reverting 2 changesets, as it breaks incoming IAX2
	  calls (Related to issue #12963) Reported by: mvanbaak ........
	  ................

2008-07-13 15:00 +0000 [r130480]  Michiel van Baak <michiel@vanbaak.info>

	* doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008)
	  | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This
	  got lost in commit 97634 ........

2008-07-13 02:35 +0000 [r130445]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_agent.c: Merged revisions 130444 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008)
	  | 2 lines Unlock list before returning ........

2008-07-11 21:39 +0000 [r130294]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) |
	  1 line Support new TRANSPORT definitions in libss7 ........

2008-07-11 20:04 +0000 [r130238]  Mark Michelson <mmichelson@digium.com>

	* /, main/audiohook.c: Merged revisions 130237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul
	  2008) | 11 lines Merged revisions 130236 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul
	  2008) | 3 lines Remove redundant logic ........ ................

2008-07-11 19:57 +0000 [r130231-130235]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c:
	  Merged revisions 130230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 |
	  tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines
	  Fix trunk breakage ........

2008-07-11 19:14 +0000 [r130175]  Mark Michelson <mmichelson@digium.com>

	* /, main/audiohook.c: Merged revisions 130174 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul
	  2008) | 15 lines Merged revisions 130173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul
	  2008) | 7 lines Fix a typo in audiohook_read_frame_both. While
	  this change has not been proven to fix any specific issue, it is
	  incorrect and could cause unforeseen problems. ........
	  ................

2008-07-11 18:53 +0000 [r130171]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500
	  (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008)
	  | 7 lines Ensure that a destination callno of 0 will not match
	  for frames that do not start a dialog (new, lagrq, and ping).
	  (closes issue #12963) Reported by: russellb Patches:
	  chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
	  ........ ................

2008-07-11 18:33 +0000 [r130168]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 |
	  seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1
	  line Missed one. Formatting only. ........

2008-07-11 18:14 +0000 [r130130]  Brett Bryant <bbryant@digium.com>

	* main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c,
	  channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c,
	  codecs/codec_resample.c, codecs/codec_dahdi.c,
	  apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c,
	  main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c,
	  main/threadstorage.c, utils/astman.c, main/utils.c,
	  channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions
	  130129 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 |
	  bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines
	  Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue
	  #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2
	  uploaded by pabelanger (license 224) Tested by: seanbright
	  ........

2008-07-11 17:30 +0000 [r130127]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_agent.c: Merged revisions 130126 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500
	  (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008)
	  | 9 lines Pass the devicestate from an underlying channel up
	  through the Agent channel. This should make the Agent always
	  report the correct device state, even when the underlying channel
	  is used for other purposes. (closes issue #12773) Reported by:
	  davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76
	  (license 14) Tested by: davidw ........ ................

2008-07-11 16:18 +0000 [r129936-130045]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES:
	  Merged revisions 130044 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 |
	  kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2
	  lines clean up a bunch more Zaptel-related references ........

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
	  configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
	  revisions 130040 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul
	  2008) | 12 lines Merged revisions 130039 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul
	  2008) | 4 lines add support for a configuration parameter for
	  'inband audio during RELEASE', which is currently mandatory in
	  libpri-1.4.4 but will become configurable in libpri-1.4.5 later
	  today (related to issue #13042) ........ ................

	* /, main/astmm.c: Merged revisions 129968 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul
	  2008) | 18 lines Merged revisions 129966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul
	  2008) | 5 lines fix a flaw found while experimenting with
	  structure alignment and padding; low-fence checking would not
	  work properly on 64-bit platforms, because the compiler was
	  putting 4 bytes of padding between the fence field and the
	  allocation memory block added a very obvious runtime warning if
	  this condition reoccurs, so the developer who broke it can be
	  chastised into fixing it :-) ........ r129967 | kpfleming |
	  2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify
	  calculation ........ ................

	* /, sounds/Makefile: Merged revisions 129916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul
	  2008) | 10 lines Merged revisions 129907 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul
	  2008) | 2 lines don't attempt to set user/group ownership of
	  extracted sound files (reported on asterisk-users) ........
	  ................

2008-07-11 01:01 +0000 [r129865]  Sean Bright <sean.bright@gmail.com>

	* res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged
	  revisions 129864 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 |
	  seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1
	  line Fix some usages of snprintf, and clarify a couple variable
	  names. ........

2008-07-10 22:07 +0000 [r129764-129805]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500
	  (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008)
	  | 8 lines Correctly deal with duplicate NEW frames (due to
	  retransmission). Also, fixup the destination call number matching
	  to be more strict and reliable. (closes issue #12963) Reported
	  by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by
	  jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........
	  ................

	* res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500
	  (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008)
	  | 2 lines Oops ........ ................

2008-07-10 21:05 +0000 [r129739]  Terry Wilson <twilson@digium.com>

	* Makefile, /: Merged revisions 129738 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 |
	  twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines
	  Move phoneprov config files to be installed with 'make samples'
	  so changes aren't inadvertently lost on a 'make install' ........

2008-07-10 19:14 +0000 [r129685]  Brett Bryant <bbryant@digium.com>

	* /, apps/app_queue.c: Merged revisions 129684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 |
	  bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines
	  Fixes a bug where the interface for a queue member gets reloaded
	  as the state_interface, if a state_interface was set, on reload
	  because the state_interface isn't stored in the ast_db. (closes
	  issue #13043) Reported by: jvandal Patches: app_queue.patch
	  uploaded by jvandal (license 413) ........

2008-07-10 18:20 +0000 [r129640-129647]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 |
	  seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1
	  line A couple more minor text changes ........

	* /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 |
	  seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1
	  line Remove extraneous \n. Pointed out by eliel on #asterisk-dev.
	  ........

2008-07-10 16:13 +0000 [r129570]  Russell Bryant <russell@digium.com>

	* sample.call, /: Merged revisions 129569 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008)
	  | 11 lines Merged revisions 129567 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008)
	  | 3 lines Note that pbx_spool.so is the module used for call
	  files (inspired by a question in #asterisk) ........
	  ................

2008-07-10 14:09 +0000 [r129504-129507]  Sean Bright <sean.bright@gmail.com>

	* /, main/editline: Merged revisions 129503 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 |
	  seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2
	  lines Update svn:ignore ........

2008-07-09 19:41 +0000 [r129438]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, /: Merged revisions 129437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul
	  2008) | 21 lines Merged revisions 129436 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul
	  2008) | 13 lines Fix a problem where inbound rfc2833 audio would
	  be sent to the core instead of being P2P bridged. When the core
	  regenerated the rfc2833 packet for the outbound leg, the SSRC
	  would be different than the RTP audio on the call leg causing
	  DTMF detection issues on the far end. (closes issue #12955)
	  Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
	  tsearle (license 373) Tested by: tonyredstone ........
	  ................

2008-07-09 16:01 +0000 [r129400]  Matthew Fredrickson <creslin@digium.com>

	* main/pbx.c, /: Merged revisions 129399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 |
	  mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add
	  Proceeding() application (#13025) ........

2008-07-09 13:46 +0000 [r129345]  Sean Bright <sean.bright@gmail.com>

	* main/editline/makelist (removed), main/editline/makelist.in
	  (added), /, main/editline/configure, main/editline/Makefile.in,
	  main/editline/configure.in: Merged revisions 129344 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400
	  (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul
	  2008) | 4 lines Look for the system installed awk instead of
	  assuming it's at /usr/bin/awk. Pointed out by jmls via
	  #asterisk-dev. ........ ................

2008-07-08 22:56 +0000 [r129160-129271]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul
	  2008) | 3 lines Fix compilation error when IMAP storage is
	  enabled ........

2008-07-08 21:04 +0000 [r129157]  Brett Bryant <bbryant@digium.com>

	* main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008)
	  | 6 lines Fix a bug in SRV lookups where dnsmgr would discard
	  everything but the first SRV result from DNS before processing
	  weights and priorities and dns_parse_answer wouldn't report that
	  there were no records in DNS unless a failure occured. Also fixed
	  a bug where dnsmgr_refresh would report that a entry was being
	  changed when ast_gethostbyname had failed. ........

2008-07-08 20:31 +0000 [r129049-129153]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, /, channels/chan_sip.c,
	  include/asterisk/causes.h: Merged revisions 129152 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500
	  (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008)
	  | 8 lines Cause SIP to return a 480 instead of a 404 when a sip
	  peer exists, but is not registered. (closes issue #12885)
	  Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: ibc ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500
	  (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008)
	  | 7 lines Timestamp decoding for video mini-frames is bogus,
	  because the timestamp only includes 15 bits, unlike voice frames,
	  which contain a 16-bit timestamp. (closes issue #13013) Reported
	  by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by
	  jpgrayson (license 492) ........ ................

2008-07-08 16:41 +0000 [r129041-129046]  Brett Bryant <bbryant@digium.com>

	* main/rtp.c, main/channel.c, channels/chan_dahdi.c,
	  main/manager.c, formats/format_pcm.c, main/logger.c,
	  main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c,
	  main/pbx.c, main/frame.c, /, channels/chan_sip.c,
	  apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c,
	  main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c,
	  channels/chan_iax2.c: Merged revisions 129045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 |
	  bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines
	  Janitor project to convert sizeof to ARRAY_LEN macro. (closes
	  issue #13002) Reported by: caio1982 Patches:
	  janitor_arraylen5.diff uploaded by caio1982 (license 22) ........

	* /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 |
	  bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line
	  Update transport= in sip so that the option is not broken from a
	  recent commit. ........

	* /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 |
	  bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line
	  Fix to sip_parse_host so that it passes the correct information
	  to sip_registry. ........

2008-07-08 14:18 +0000 [r129007]  Russell Bryant <russell@digium.com>

	* /, apps/app_fax.c: Merged revisions 129006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 |
	  russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines
	  Update app_fax for better compatibility with spandsp 0.0.5. Add a
	  call to t38_terminal_release, and make sure that the phase E
	  handler gets called with proper status. (closes issue #13020)
	  Reported by: dimas Patches: v1-appfax.patch uploaded by dimas
	  (license 88) ........

2008-07-08 10:06 +0000 [r128913-128952]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19
	  lines Merged revisions 128950 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11
	  lines Don't hangup the call if we can't resolve the Contact if
	  there's a proxy route set for the call. ---- This comment was
	  added a while ago and today it hit me badly. /* OEJ: Possible
	  issue that may need a check: If we have a proxy route between us
	  and the device, should we care about resolving the contact or
	  should we just send it? */ ........ ................

	* /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15
	  lines Merged revisions 128912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7
	  lines Fix issues where repeated messages where ignored, but
	  retransmitted reliably instead of unreliably. Reported by: johan
	  Patches: 12746.txt uploaded by oej (license 306) Tested by: johan
	  (issue #12746) ........ ................

2008-07-08 00:03 +0000 [r128855-128858]  Tilghman Lesher <tlesher@digium.com>

	* /: Merged revisions 128857 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008)
	  | 15 lines Merged revisions 128856 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008)
	  | 7 lines Check for non-NULL before stripping characters. (closes
	  issue #12954) Reported by: bfsworks Patches:
	  20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: deti ........ ................

	* apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500
	  (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008)
	  | 2 lines Stop using deprecated method, as requested by Kevin.
	  ........ ................

2008-07-07 22:44 +0000 [r128797]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r128796 | russell | 2008-07-07 17:42:30 -0500
	  (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008)
	  | 8 lines Fix handling of when a pvt disappears. Properly return
	  the pvt locked and don't hold the pvt lock while destroying the
	  ast_channel. (closes issue #13014) Reported by: jpgrayson
	  Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson
	  (license 492) ........ ................

2008-07-07 20:51 +0000 [r128739]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400
	  (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul
	  2008) | 9 lines Remove spurious trailing whitespace from log
	  messages and fix a spelling error in a log message. (closes issue
	  #13017) Reported by: jpgrayson Patches:
	  chan_iax2_space_after_newline.patch uploaded by jpgrayson
	  (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
	  (license 492) ........ ................

2008-07-07 20:31 +0000 [r128601-128735]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul
	  2008) | 3 lines Crap ........

	* apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul
	  2008) | 7 lines If imapfolder=foo were set in voicemail.conf,
	  then when calling VoiceMailMain, app_voicemail would attempt to
	  play a file called vm-foo instead of playing vm-INBOX to play the
	  "new" sound file. This commit fixes that issue. This may fix one
	  of the problems reported in issue #12987 ........

	* /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500
	  (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul
	  2008) | 10 lines By using the iaxdynamicthreadcount to identify a
	  thread, it was possible for thread identifiers to be duplicated.
	  By using a globally-unique monotonically- increasing integer,
	  this is now avoided. (closes issue #13009) Reported by: jpgrayson
	  Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson
	  (license 492) ........ ................

	* configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged
	  revisions 128599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 |
	  mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6
	  lines Update a few instances of "extensions reload" to "dialplan
	  reload" in the documentation. Patch provided by caio1982 (license
	  22) ........

2008-07-06 20:22 +0000 [r128288-128543]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  128524 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 |
	  oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines -
	  Fixing issues with "sip show settings" - Adding IP address for
	  TCP and/or TLS too if auto-domain is enabled and binding to a
	  different IP address - Fixing documentation in sip.conf.sample
	  ........

	* /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 |
	  oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines -
	  Remove unused variable "expiry" - Set global_outboundproxy.force
	  at reload. ........

	* doc/realtimetext.txt (added), /: The following patch with
	  references to t140red removed, since it only exists in trunk.
	  Merged revisions 128417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 |
	  oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines
	  Adding documentation on the T.140 support in Asterisk. This is a
	  function that we're the reference implementation on now. :-)
	  ........

	* /: Merged revisions 128343 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 |
	  oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines
	  Removing the CLI dumpdb command (see asterisk-dev discussion and
	  decision) ........

	* /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 |
	  oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines
	  Adding doxygen comments to missing parts, moving some #define
	  ...trying to get my head around the thoughts behind the TCP/TLS
	  stuff and figure out what needs to be done to make it useful...
	  ........

	* /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 |
	  oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines
	  Adding TCP and TLS to "sip show settings". TLS needs to have one
	  configuration per configured domain at some point. ........

	* /: Blocking changes in trunk.

2008-07-05 21:02 +0000 [r128238-128243]  Olle Johansson <oej@edvina.net>

	* /: Keep the "sip-user" structure in 1.6.0, while testing new
	  funky stuff in trunk.

	* /: Blocking the AGi changes from 1.6.0. Let's test them for a
	  while in trunk before a release.

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  128237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 |
	  oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines
	  Make TCP disabled by default (it's considered experimental)
	  ........

	* /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2
	  lines Reformatting the config sample ........

2008-07-05 15:19 +0000 [r128161]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05
	  Jul 2008) | 7 lines LDAP schema updates (closes issue #12860)
	  Reported by: flyn Patches: asterisk.ldif uploaded by suretec
	  (license 70) asterisk.schema uploaded by suretec (license 70)
	  ........

2008-07-05 03:40 +0000 [r128124-128127]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) |
	  1 line It would help if we actually parsed the ss7_explicitacm
	  option in the config file... ........

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
	  revisions 128122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 |
	  mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add
	  option to wait to be able to explicitly send ACM via the
	  Proceeding() application in the dialplan. Also minor
	  documentation update explaining how to setup multiple signalling
	  links within a linkset ........

2008-07-04 16:12 +0000 [r128028-128031]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged
	  revisions 128027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008)
	  | 16 lines Merged revisions 127973 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008)
	  | 8 lines Fix the 'dialplan remove extension' logic, so that it
	  a) works with cidmatch, and b) completes contexts correctly when
	  the extension is ambiguous. (closes issue #12980) Reported by:
	  licedey Patches: 20080703__bug12980.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: Corydon76 ........
	  ................

2008-07-03 22:23 +0000 [r127905]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged
	  revisions 127903 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul
	  2008) | 20 lines Merged revisions 127892,127895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul
	  2008) | 6 lines a couple of small Solaris-related fixes (closes
	  issue #11885) Reported by: snuffy, asgaroth ........ r127895 |
	  kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3
	  lines remove this, it has been moved to the main Makefile
	  ........ ................

2008-07-03 19:12 +0000 [r127830]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /,
	  channels/chan_sip.c, main/features.c, include/asterisk/cdr.h:
	  Merged revisions 127793 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) |
	  38 lines Merged revisions 127663 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) |
	  30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927)
	  Reported by: murf Tested by: murf, deeperror (closes issue
	  #12907) Reported by: falves11 Tested by: murf, falves11 (closes
	  issue #11849) Reported by: greyvoip As to 11849, I think these
	  changes fix the core problems brought up in that bug, but perhaps
	  not the more global problems created by the limitations of CDR's
	  themselves not being oriented around transfers. Reopen if necc,
	  but bug reports are not the best medium for enhancement
	  discussions. We need to start a second-generation CDR
	  standardization effort to cover transfers. (closes issue #11093)
	  Reported by: rossbeer Tested by: greyvoip, murf ........
	  ................

2008-07-03 16:50 +0000 [r127790-127792]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 |
	  oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make
	  sure we stop session timers as soon as we start hanging up an
	  active call. May fix issue 12919. ........

	* /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 |
	  oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines
	  Revert some logic for session timers. We do send in-dialog
	  requests that should not have session-timer require headers, like
	  MESSAGE and REFER. So in the future, only add them on requests
	  and responses that are related to INVITEs and re-INVITEs.
	  ........

2008-07-03 16:24 +0000 [r127778]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: Merged revisions 127767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 |
	  kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2
	  lines some minor fixes found while working on issue #12911 (and
	  block the rev from 1.4 since the equivalent is already here)
	  ........

2008-07-02 21:10 +0000 [r127567]  Mark Michelson <mmichelson@digium.com>

	* /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul
	  2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of
	  in-line sizeof() and division. ........

2008-07-02 20:49 +0000 [r127559-127563]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 127562 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500
	  (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul
	  2008) | 3 lines Fix thread-safety of some of the
	  pbx_builtin_getvar_helper calls ........ ................

2008-07-02 19:48 +0000 [r127467-127503]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: Merged revisions 127466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
	  tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
	  Solaris fix (closes issue #12949) Reported by: snuffy Patches:
	  bug_12949.diff uploaded by snuffy (license 35) ........

2008-07-02 14:30 +0000 [r127396-127399]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 |
	  seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1
	  line Fix a bug I noticed while doing the previous merge ........

	* cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt:
	  Merged revisions 126226,126513 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 |
	  seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8
	  lines Merge in changes from my cdr-tds-conversion branch. This
	  changes the internal implementation from using the volatile
	  libtds, to using the db-lib front end. The unintended side effect
	  of this is that we support (at least) versions 0.62 through 0.82
	  of the FreeTDS distribution without any #ifdef ugliness. (closes
	  issue #12844) Reported by: jcollie ........ r126513 | seanbright
	  | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a
	  few more strings to char *, so that we can compile cleanly
	  against FreeTDS 0.60. Update the docs to reflect that we can now
	  compile and run against all modern releases of FreeTDS (0.60
	  through 0.82) ........

	* /: Unblock some revisions so I can merge the cdr_tds changes from
	  trunk

2008-07-02 12:09 +0000 [r127364]  Russell Bryant <russell@digium.com>

	* doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008)
	  | 13 lines Add a locking section to the coding guidelines
	  document. This section covers some locking fundamentals, as well
	  as some information on locking as it is used in Asterisk. It
	  describes some of the ways that are used and could be used to
	  achieve deadlock avoidance. It also demonstrates the unfortunate
	  conclusion that with the use of recursive locks, none of the
	  constructs in use today are failsafe from deadlocks. Finally, it
	  makes some recommendations for new code being written. As proper
	  locking strategies is a complex subject, this section still has
	  room for expansion and improvement. This is a result of
	  collaboration between Luigi Rizzo and myself on the asterisk-dev
	  mailing list. ........

2008-07-02 02:49 +0000 [r127298]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 |
	  tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12
	  lines Change the global timer B to be dependent on the value of
	  the T1 timer, as recommended in RFC 3261, instead of being
	  hardcoded to 32 seconds. This is important for LANs, as it allows
	  autocongestion to occur much more quickly, if desired by the
	  local PBX administrator. It also corrects a bug: if the T1 timer
	  was increased beyond 500ms, then timer B would have been set at a
	  much lower value than recommended. (closes issue #12544) Reported
	  by: kactus Patches: 20080616__bug12544.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: kactus ........

2008-07-01 23:39 +0000 [r127246]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500
	  (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul
	  2008) | 5 lines Add error message to failed open(2) calls inside
	  the copy() function of app_voicemail. This idea came as part of
	  my work in helping to resolve issue #12764. ........
	  ................

2008-07-01 21:19 +0000 [r127163]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  127154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 |
	  bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines
	  Add a configuration option so the global outboundproxy can use
	  tcptls without it being defined by each sip user. ........

2008-07-01 21:16 +0000 [r127156-127158]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 127157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 |
	  mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8
	  lines Place the delay in __ast_answer prior to the
	  channel-specific answer callback. This change differs from commit
	  127113 in that now the channel is not set to AST_STATE_UP until
	  after the answer callback. (closes issue #12924) Reported by:
	  snyfer ........

	* main/channel.c, /: Merging Revision 127113 from trunk

2008-07-01 20:52 +0000 [r127153]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 127152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 |
	  qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines
	  Fix a typo that caused this asterisk.conf to not get correctly
	  generated. (closes issue #12966) Reported by: ibc Patches:
	  12966.patch uploaded by bkruse (license 132) ........

2008-07-01 20:29 +0000 [r127085-127149]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions
	  127143 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008)
	  | 10 lines Merged revisions 127133 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008)
	  | 2 lines Disable the old, slow search for matching callno in
	  chan_iax2 (but allow it to be reenabled for debugging) ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500
	  (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008)
	  | 8 lines Change around how we schedule pings and lagrqs, and fix
	  a reason why the jobs were not getting properly cancelled.
	  (closes issue #12903) Reported by: stevedavies Patches:
	  20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: stevedavies ........ ................

2008-07-01 16:53 +0000 [r127001]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500
	  (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008)
	  | 2 lines Suppress annoying warning by finding the remaining
	  cases where the callno is not in the hash. ........
	  ................

2008-07-01 15:05 +0000 [r126756-126904]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15
	  lines Merged revisions 126902 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7
	  lines Use domain part of SIP uri in register= configuration as
	  fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch
	  uploaded by one47 (license 23) (closes issue #12474) ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16
	  lines Merged revisions 126899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8
	  lines Handle escaped URI's in call pickups. Patch by oej and
	  IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch
	  uploaded by IgorG (license 20) Tested by: IgorG, oej (closes
	  issue #12299) ........ ................

	* /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis,
	  01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5
	  lines Clear up documentation on "domain=" setting in sip.conf
	  Reported by: davidw (closes issue #12413) ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14
	  lines Merged revisions 126789 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6
	  lines Report 200 OK to all in-dialog OPTIONs requests (to confirm
	  that the dialog exist). Don't bother checking the request URI.
	  (closes issue #11264) Reported by: ibc ........ ................

	* /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15
	  lines Merged revisions 126735 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7
	  lines Fix bad XML for hold notification. Reported by: gowen72
	  Patches: hold.patch uploaded by gowen72 (license 432) (closes
	  issue #12942) ........ ................

2008-06-30 22:34 +0000 [r126676]  Jeff Peeler <jpeeler@digium.com>

	* configs/zapata.conf.sample (removed),
	  configs/chan_dahdi.conf.sample (added), /: Merged revisions
	  126675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 |
	  jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line
	  rename zapata.conf.sample to chan_dahdi.conf.sample ........

2008-06-30 20:32 +0000 [r126638]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) |
	  1 line Add support to see MTP2 down events when the link layer
	  drops in SS7 ........

2008-06-30 16:09 +0000 [r126575]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r126574 | russell | 2008-06-30 11:07:25 -0500
	  (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008)
	  | 10 lines Fix a typo in the non-DEBUG_THREADS version of the
	  recently added DEADLOCK_AVOIDANCE() macro. This caused the lock
	  to not actually be released, and as a result, not avoid deadlocks
	  at all. This resolves the issues reported in the last while about
	  Asterisk locking up all over the place (and most commonly, in
	  chan_iax2). (closes issue #12927) (closes issue #12940) (closes
	  issue #12925) (potentially closes others ...) ........
	  ................

2008-06-30 13:07 +0000 [r126518]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) |
	  20 lines The following patch with some changes for trunk...
	  Merged revisions 126516 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) |
	  10 lines Send all responses to an INVITE reliably, so that we
	  retransmit if we don't get an ACK and also fail if we don't get
	  the very same precious ACK. Based on patch by tsearle, with my
	  own additions. (closes issue #12951) Reported by: tsearle
	  Patches: busy_retransmit.patch uploaded by tsearle (license 373)
	  ........ ................

2008-06-29 17:02 +0000 [r126362-126364]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_zapbarge.c (removed): finish converting this module

	* pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c,
	  pbx/Makefile: Merged revisions 126356 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 |
	  kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9
	  lines various minor fixes created while i worked on getting
	  *every* Asterisk module to build on laptop in dev mode: remove
	  weird pre-setting of LUA paths; they are not necessary; also use
	  the proper path for including the files in pbx_lua.c make the
	  compiler shut up about some ignored function results in
	  pbx_gtkconsole; this module is badly coded anyway ........

	* apps/app_dahdibarge.c (added): don't know how this got missed in
	  the DAHDI conversion of this branch

2008-06-29 13:20 +0000 [r126227-126322]  Sean Bright <sean.bright@gmail.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 |
	  seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6
	  lines Quote column names when inserting CDRs into postgres to
	  avoid conflicts with reserved words. (closes issue #12947)
	  Reported by: panolex ........

2008-06-28 15:58 +0000 [r126155-126188]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: update this branch to use the trunk goodness version
	  of menuselect

2008-06-27 22:43 +0000 [r126058-126112]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008)
	  | 12 lines Merged revisions 126056 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008)
	  | 4 lines When we get a 408 Timeout, don't stop trying to
	  re-register. (closes issue #12863) Reported by: ricvil ........
	  ................

2008-06-27 21:00 +0000 [r126023]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Port revisions 124661 and 123650 from trunk to
	  1.6.0 Thanks to Atis Lezdins for pointing this out on the
	  asterisk-dev mailing list

2008-06-27 19:20 +0000 [r125994]  Russell Bryant <russell@digium.com>

	* /, doc/siptls.txt: Merged revisions 125988 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 |
	  russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines
	  Fix a typo. Someone on IRC copied this literally and then
	  wondered why it wasn't working. :) ........

2008-06-27 19:06 +0000 [r125981-125985]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) |
	  1 line Revert this part of the fix. We'll fix it in libss7
	  ........

	* channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) |
	  1 line Obviously somebody didn't compile with libss7 support when
	  doing the DAHDI conversion. ........

	* channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) |
	  1 line Add support for new commands to block/unblock all CICs on
	  a linkset ........

2008-06-27 17:36 +0000 [r125948]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 |
	  bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line
	  Small error in the function that converts peer transports to a
	  string. ........

2008-06-27 16:29 +0000 [r125892]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  125891 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 |
	  bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines
	  Change the way that the transport option works for sip users.
	  transport will now take multiple arguments, the first one listed
	  will be the one used for new dialogs, and the rest listed will be
	  acceptable ways for that peer to contact us. This fixes a minor
	  bug where, because SIP TCP/UDP run on the same port, could cause
	  a TCP peer to be saved in the ast_db. There will also be warnings
	  when a transport is changed for an unexpected reason. (issue
	  #12799) ........

2008-06-27 16:19 +0000 [r125859-125863]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 125855 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 |
	  mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5
	  lines Ensure the thread-safety of the monexec variable in
	  app_queue. Thanks to Russell for pointing out the problem
	  ........

2008-06-27 16:01 +0000 [r125854]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 125853 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008)
	  | 3 lines Revert half of the fix, as this part may have been
	  unnecessary (related to issue #12914) Requested here:
	  http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html
	  ........

2008-06-27 14:57 +0000 [r125800-125852]  Mark Michelson <mmichelson@digium.com>

	* main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure
	  to only include dahdi/user.h if we have installed DAHDI.

	* channels/chan_iax2.c: I accidentally committed a change to
	  chan_iax2.c in addition to a change to app_queue.c. Reverting the
	  change to chan_iax2.c, even though it may turn out that this
	  change is necessary.

	* utils/Makefile, /: Merged revisions 125799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 |
	  mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3
	  lines Remove an unneeded target from the Makefile ........

2008-06-27 14:09 +0000 [r125742-125797]  Tilghman Lesher <tlesher@digium.com>

	* /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500
	  (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008)
	  | 2 lines In this debugging function, copy to a buffer instead of
	  using potentially unsafe pointers. ........ ................

	* channels/chan_local.c, /: Merged revisions 125741 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500
	  (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008)
	  | 7 lines Add proper deadlock avoidance. (closes issue #12914)
	  Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: ozan ........
	  ................

2008-06-27 07:41 +0000 [r125704]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions
	  125703 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 |
	  phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line
	  Fix a compile time error that occurs if OpenSSL is not installed.
	  Reported by Noel Morais on the users mailing list ........

2008-06-27 01:09 +0000 [r125648-125684]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined
	  in 1.6.0

	* /, apps/app_queue.c: Merged revisions 125666 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 |
	  mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3
	  lines Make this compile with dev-mode on ........

	* /, apps/app_queue.c: Merged revisions 125649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 |
	  mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15
	  lines The monitor-join option for queues was deprecated in favor
	  of using MixMonitor to mix audio. However, it was pointed out to
	  me that because of this, the command set for the MONITOR_EXEC
	  variable is ignored as well. This means that people can't do
	  their own custom mixing commands at the end of recordings in
	  order to make, for instance, stereo recordings of calls. With
	  this patch, app_queue will set the "joinfiles" variable for the
	  channel's monitor if MONITOR_EXEC is not zero-length. This means
	  that for normal audio mixing, MixMonitor is still the preferred
	  choice, but we allow custom mixing to be done with the two
	  Monitor streams if desired. (closes issue #12923) Reported by:
	  snyfer ........

2008-06-26 23:06 +0000 [r125592]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 125591 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 |
	  mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3
	  lines Fix a really stupid mistake ........

2008-06-26 23:05 +0000 [r125590]  Jason Parker <jparker@digium.com>

	* /, main/utils.c: Merged revisions 125589 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) |
	  9 lines Merged revisions 125587 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) |
	  1 line Make sure to unlock the lock_info lock (huh?). Possible
	  deadlock? ........ ................

2008-06-26 23:04 +0000 [r125588]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 125586 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun
	  2008) | 19 lines Merged revisions 125585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun
	  2008) | 11 lines Add the interface of a queue member to the
	  output of the "queue show" command so that it can easily be
	  associated with a queue member's name. This helps so that the
	  appropriate queue member can be removed or paused since the
	  interface is required, not the member's name. (closes issue
	  #12783) Reported by: davevg Patches: app_queue.diff uploaded by
	  davevg (license 209) with small mod from me ........
	  ................

2008-06-26 22:50 +0000 [r125584]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008)
	  | 2 lines Don't hang if the command is blank ........

2008-06-26 22:06 +0000 [r125478-125532]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 125477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun
	  2008) | 19 lines Merged revisions 125476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun
	  2008) | 11 lines Prior to this patch, the "queue show" command
	  used cached information for realtime queues instead of giving
	  up-to-date info. Now realtime is queried for the latest and
	  greatest in queue info. (closes issue #12858) Reported by: bcnit
	  Patches: queue_show.patch uploaded by putnopvut (license 60)
	  ........ ................

2008-06-26 17:07 +0000 [r125388]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12
	  lines Merged revisions 125384 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3
	  lines Add support for peer realm based auth (a few missing lines,
	  the rest is well documented but never worked) ........
	  ................

2008-06-26 15:52 +0000 [r125280-125334]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500
	  (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun
	  2008) | 5 lines ensure that (whenever possible) if we generate a
	  log message because an ioctl() call to DAHDI/Zaptel failed, that
	  we include the reason it failed by including the stringified
	  error number (issue AST-80) ........ ................

	* /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun
	  2008) | 2 lines fix compile failure found by buildbot (go,
	  buildbot!) ........

2008-06-26 11:08 +0000 [r125192-125278]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c, /: Merged revisions 125277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008)
	  | 15 lines Merged revisions 125276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008)
	  | 7 lines Check for rtcp structure before trying to delete
	  schedule. (closes issue #12872) Reported by: destiny6628 Patches:
	  20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: destiny6628 ........ ................

	* configs/agents.conf.sample, /: Merged revisions 125223 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500
	  (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008)
	  | 4 lines Document ackcall=always. (closes issue #12852) Reported
	  by: davidw ........ ................

	* configs/http.conf.sample, /: Merged revisions 125191 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008)
	  | 6 lines Update sample configuration to match what are now the
	  defaults for the prefix. (closes issue #12838, related to issue
	  #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded
	  by pabelanger (license 224) ........

2008-06-25 23:20 +0000 [r125146]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, channels/chan_dahdi.c, apps/app_flash.c,
	  configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c,
	  /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c,
	  apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h
	  (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged
	  revisions 125138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun
	  2008) | 18 lines Merged revisions 125132 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun
	  2008) | 10 lines allow tonezone to live in a different place than
	  DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate
	  packages and can be installed in different places don't include
	  tonezone.h in dahdi_compat.h, because only a couple of modules
	  need it get app_rpt building again after the DAHDI changes
	  (closes issue #12911) Reported by: tzafrir ........
	  ................

2008-06-25 01:13 +0000 [r124964-124967]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged
	  revisions 124966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008)
	  | 15 lines Merged revisions 124965 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008)
	  | 7 lines Pvt deadlock causes some channels to get stuck in
	  Reserved status. (closes issue #12621) Reported by:
	  fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: fabianoheringer ........
	  ................

	* apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500
	  (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008)
	  | 8 lines Occasionally control characters find their way into
	  CallerID. These need to be stripped prior to placing CallerID in
	  the headers of an email. (closes issue #12759) Reported by: RobH
	  Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76
	  (license 14) Tested by: RobH ........ ................

2008-06-24 17:52 +0000 [r124871-124873]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 124872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 |
	  phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines
	  Subscribe to buddy's presence only if we really need to. That is,
	  if the corresponding roster item has a subscription value set to
	  "none" or "from". Make the code more readable. ........

	* /, res/res_jabber.c: Merged revisions 124870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 |
	  phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line
	  Code simplification ........

2008-06-23 15:44 +0000 [r124708]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /: blocked revision 124707, taskprocessors are not in 1.6.0

2008-06-22 03:18 +0000 [r124542]  Steve Murphy <murf@digium.com>

	* apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) |
	  17 lines Merged revisions 124540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9
	  lines (closes issue #12910) Reported by: chris-mac Sorry, my
	  testing did not contain the simple case of forkCDR(v), I am much
	  embarrassed to admit. If I had, I would have more solidly
	  initialized the opts element for varset. ........
	  ................

2008-06-21 12:54 +0000 [r124397-124506]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008)
	  | 4 lines Reduce warning to debug, otherwise we flood the log
	  when we (legitimately) can't find a record. (Closes issue #12908)
	  ........

	* apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008)
	  | 14 lines Merged revisions 124450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008)
	  | 6 lines usleep with a value over 1,000,000 is nonportable.
	  Changing to use sleep() instead. (closes issue #12814) Reported
	  by: pputman Patches: app_rtp_sleep.patch uploaded by pputman
	  (license 81) ........ ................

	* /, main/app.c: Merged revisions 124396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008)
	  | 11 lines Merged revisions 124395 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008)
	  | 3 lines If the last character in a string to be parsed is the
	  delimiter, then we should count that final empty string as an
	  additional argument. ........ ................

2008-06-20 21:48 +0000 [r124394]  Jeff Gehlbach <jeffg@opennms.org>

	* doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions
	  124392-124393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) |
	  9 lines Merged revisions 124372 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) |
	  1 line Fix issues in digium-mib.txt and asterisk-mib.txt to
	  placate smilint - bug 12905 ........ ................ r124393 |
	  jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines
	  (Missed committing . on previous commit.....) Merged revisions
	  124372 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) |
	  1 line Fix issues in digium-mib.txt and asterisk-mib.txt to
	  placate smilint - bug 12905 ........ ................
	  ................

2008-06-20 20:18 +0000 [r124317]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 124316 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500
	  (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008)
	  | 8 lines When using a Local channel, started by a call file,
	  with a destination of an AGI script, the AGI script does not
	  always get notified of a hangup if the underlying channel hangs
	  up early. (closes issue #11833) Reported by: IgorG Patches:
	  local_hangup-v1.diff uploaded by IgorG (license 20) ........
	  ................

2008-06-20 16:31 +0000 [r124244-124279]  Mark Michelson <mmichelson@digium.com>

	* main/ast_expr2.fl, include/asterisk/doxyref.h, /,
	  main/ast_expr2f.c: Merged revisions 124278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 |
	  mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6
	  lines Change references to doc/channelvariables.txt to
	  doc/tex/channelvariables.tex. This issue came up on the
	  asterisk-dev mailing list. ........

	* /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 |
	  mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9
	  lines Add a missing "ChannelType" header to one of the
	  "PeerStatus" manager events in chan_sip (closes issue #12904)
	  Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel
	  (license 64) ........

2008-06-19 23:02 +0000 [r124184]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500
	  (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008)
	  | 7 lines It's possible for a hangup to be received, even just
	  after the initial cid spill. (closes issue #12453) Reported by:
	  Alex728 Patches: 20080604__bug12453.diff.txt uploaded by
	  Corydon76 (license 14) ........ ................

2008-06-19 20:32 +0000 [r124124]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500
	  (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun
	  2008) | 8 lines Fix IMAP forwarding so that messages are sent to
	  the proper mailbox. (closes issue #12897) Reported by: jaroth
	  Patches: destination_forward.patch uploaded by jaroth (license
	  50) ........ ................

2008-06-19 19:49 +0000 [r124065]  Brett Bryant <bbryant@digium.com>

	* /, main/utils.c: Merged revisions 124064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 |
	  bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines
	  Add errors that report any locks held by threads when they are
	  being closed. ........

2008-06-19 18:57 +0000 [r124026]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 |
	  bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines
	  Fix bug in sip registration that sets the default port to 5060
	  for tls. ........

2008-06-19 17:58 +0000 [r123871-123989]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008)
	  | 6 lines Don't change pointers that need to be later passed back
	  for deallocation. (closes issue #12572) Reported by: flyn
	  Patches: 20080613__bug12572.diff.txt uploaded by Corydon76
	  (license 14) ........

	* main/channel.c, /: Merged revisions 123931 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008)
	  | 13 lines Merged revisions 123930 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008)
	  | 5 lines Change informative messages to use the _multiple
	  variant when multiple formats are possible. (Closes issue #12848)
	  Reported by klaus3000 ........ ................

	* /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500
	  (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008)
	  | 5 lines Only process 40 arguments (20 files) at once with
	  xargs, because some older shells may force xargs to separate on
	  an odd boundary. (Closes issue #12883) Reported by Nik Soggia
	  ........ ................

	* /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500
	  (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008)
	  | 4 lines Correct description of notifyringing option. (Closes
	  issue #12890) Reported by gminet ........ ................

	* main/asterisk.c, /: Merged revisions 123870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008)
	  | 14 lines Merged revisions 123869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008)
	  | 6 lines The RDTSC instruction was introduced on the Pentium
	  line of microprocessors, and is not compatible with certain 586
	  clones, like Cyrix. Hence, asking for i386 compatibility was
	  always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes
	  issue #12886) Reported by tecnoxarxa ........ ................

2008-06-18 22:18 +0000 [r123718-123772]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged
	  revisions 123770 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008)
	  | 16 lines Merged revisions 123769 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008)
	  | 8 lines Add support for saying numbers in Hebrew. (closes issue
	  #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008
	  uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
	  uploaded by greenfieldtech (with signficant changes to the
	  spreadsheet by me) ........ ................

	* pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008)
	  | 15 lines Merged revisions 123710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008)
	  | 7 lines Set the variables top-down, so that if a script sets a
	  variable more than once, the last one will take precedence.
	  (closes issue #12673) Reported by: phber Patches:
	  20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
	  ........ ................

2008-06-18 20:08 +0000 [r123693]  Brett Bryant <bbryant@digium.com>

	* main/tcptls.c, /: Merged revisions 123692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 |
	  bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines
	  Fix a crash in tcp and tls connections related to reference
	  counts. ........

2008-06-18 15:09 +0000 [r123651-123653]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 123652 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 |
	  mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7
	  lines A portion of the code which handled the 'c' queue option
	  had been removed. No telling when it happened. Anyway, it's back
	  in now and works properly. (Based on issue reported on mailing
	  list) ........

2008-06-18 12:34 +0000 [r123646-123647]  Russell Bryant <russell@digium.com>

	* apps/app_fax.c: don't use trunk only API for frame data (closes
	  issue #12881)

	* apps/app_fax.c (added): re-add app_fax ... it got accidentally
	  removed (closes issue #12881)

2008-06-17 21:57 +0000 [r123547]  Brett Bryant <bbryant@digium.com>

	* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
	  main/http.c, include/asterisk/tcptls.h: Merged revisions 123546
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17
	  Jun 2008) | 5 lines Updates all usages of
	  ast_tcptls_session_instance to be managed by reference counts so
	  that they only get destroyed when all threads are done using
	  them, and memory does not get free'd causing strange issues with
	  SIP. This code was originally written by russellb in the
	  team/group/issue_11972/ branch. ........

2008-06-17 21:34 +0000 [r123487-123542]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun
	  2008) | 12 lines Merged revisions 123485 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun
	  2008) | 4 lines Make chan_sip build under dev mode with compilers
	  >= GCC 4.2 Thanks to jpeeler for alerting me of this ........
	  ................

2008-06-17 20:23 +0000 [r123473]  Steve Murphy <murf@digium.com>

	* /: block 123448 from trunk; it doesn't apply here.

2008-06-17 19:01 +0000 [r123394]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500
	  (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008)
	  | 3 lines Fix 3 more places where failure to lock the structure
	  could cause the wrong lock to be unlocked. (Closes issue #12795)
	  ........ ................

2008-06-17 18:28 +0000 [r123382-123387]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008)
	  | 1 line Fix some (more) variables that were forgotten to be
	  renamed, related to 117658 ........

2008-06-17 18:10 +0000 [r123335]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun
	  2008) | 19 lines Merged revisions 123333 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun
	  2008) | 11 lines Cisco BTS sends SIP responses with a tab between
	  the Cseq number and SIP request method in the Cseq: header.
	  Asterisk did not handle this properly, but with this patch, all
	  is well. (closes issue #12834) Reported by: tobias_e Patches:
	  12834.patch uploaded by putnopvut (license 60) Tested by:
	  tobias_e ........ ................

2008-06-17 18:08 +0000 [r123332]  Jeff Peeler <jpeeler@digium.com>

	* doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile,
	  doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt,
	  contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added),
	  configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c,
	  main/asterisk.c, configs/users.conf.sample, doc/ss7.txt,
	  apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt,
	  doc/tex/queues-with-callback-members.tex,
	  include/asterisk/dahdi.h (added), configs/extensions.ael.sample,
	  res/res_musiconhold.c, configs/meetme.conf.sample,
	  codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel,
	  cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex,
	  doc/tex/sla.tex, include/asterisk/dsp.h,
	  doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c,
	  contrib/scripts/autosupport, doc/manager_1_1.txt,
	  channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex,
	  doc/tex/channelvariables.tex, apps/app_getcpeid.c,
	  doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample,
	  doc/tex/security.tex, include/asterisk/zapata.h (removed),
	  doc/tex/privacy.tex, build_tools/menuselect-deps.in,
	  apps/app_flash.c, main/file.c, doc/osp.txt,
	  contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in,
	  configs/extensions.conf.sample, doc/asterisk.sgml, README,
	  contrib/init.d/rc.mandrake.asterisk, /,
	  include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added),
	  apps/app_chanisavail.c, channels/chan_iax2.c,
	  configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c
	  (added), include/asterisk/doxyref.h, configure,
	  doc/tex/backtrace.tex, apps/app_zapscan.c (removed),
	  doc/tex/app-sms.tex, apps/app_zapras.c (removed),
	  configs/extensions.lua.sample, include/asterisk/options.h,
	  contrib/init.d/rc.suse.asterisk, apps/app_dial.c,
	  apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed),
	  apps/app_dahdiras.c (added), configure.ac,
	  configs/queues.conf.sample, include/asterisk/channel.h: Goodbye
	  Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI.
	  Configuration file and dialplan backwards compatability has been
	  put in place where appropiate. Release announcement to follow.

2008-06-17 15:58 +0000 [r123276]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 123275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun
	  2008) | 20 lines Merged revisions 123274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun
	  2008) | 12 lines davidw pointed out that the holdtime calculation
	  used by app_queue does not use "boxcar" filtering as the comments
	  say. The term "boxcar" means that the number of samples used to
	  calculate stays constant, with new samples replacing the oldest
	  ones. The queue holdtime calculation uses all holdtime samples
	  collected since the queue was loaded, so the comment has been
	  changed to be accurate. (closes issue #12781) Reported by: davidw
	  ........ ................

2008-06-17 15:52 +0000 [r123273]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, /: Merged revisions 123272 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008)
	  | 12 lines Merged revisions 123271 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008)
	  | 4 lines Fix a memory leak in astobj2 that was pointed out by
	  seanbright. When a container got destroyed, the underlying bucket
	  list entry for each object that was in the container at that time
	  did not get free'd. ........ ................

2008-06-16 21:20 +0000 [r123178]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Fix some variables that were forgotten to be
	  renamed, related to 117658. Couldn't merge from trunk since the
	  chan_dahdi transition has not occurred here yet

2008-06-16 21:19 +0000 [r123173]  Steve Murphy <murf@digium.com>

	* apps/app_stack.c, apps/app_dial.c, main/pbx.c, /,
	  main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged
	  revisions 123165 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 |
	  murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines
	  (closes issue #12689) Reported by: ys Many thanks to ys for doing
	  the research on this problem. I didn't think it would be best to
	  unlock the contexts and then relock them after the
	  remove_extension2() call, so I added an extra arg to
	  remove_extension2() and set it appropriately in each call. There
	  were not that many. I considered forcing the code to lock the
	  contexts before the call to remove_extension2(), but that would
	  require a slightly greater degree of changes, especially since
	  the find_context_locked is local to pbx.c I did a simple sanity
	  test to make sure the code doesn't mess things up in general.
	  ........

2008-06-16 20:03 +0000 [r123112-123116]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c, /, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Merged revisions 123114 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008)
	  | 10 lines Merged revisions 123113 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008)
	  | 2 lines Port "hasvoicemail" change from SIP to other channel
	  drivers ........ ................

	* /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008)
	  | 16 lines Merged revisions 123110 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008)
	  | 8 lines People expect that if "hasvoicemail" is set in
	  users.conf, even if "mailbox" isn't set, that SIP will detect a
	  mailbox. (closes issue #12855) Reported by: PLL Patches:
	  20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: PLL ........ ................

2008-06-16 17:29 +0000 [r123075]  Chris Tooley <chris@tooley.com>

	* apps/app_externalivr.c: Fixes and closes bug number 12804

2008-06-16 12:32 +0000 [r122871-122921]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) |
	  14 lines Merged revisions 122919 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6
	  lines Only compare the first 15 characters so that even if the
	  charset is specified we still accept it as SDP. (closes issue
	  #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff
	  uploaded by lanzaandrea (license 496) ........ ................

	* /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) |
	  14 lines Merged revisions 122869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6
	  lines Don't send a BYE on a dialog that is already gone during a
	  REFER. (closes issue #12865) Reported by: flefoll Patches:
	  chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll
	  (license 244) ........ ................

2008-06-13 21:47 +0000 [r122715]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c, /: Merged revisions 122714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun
	  2008) | 17 lines Merged revisions 122713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun
	  2008) | 9 lines Short circuit the loop in autoservice_run if
	  there are no channels to poll. If we continued, then the result
	  would be calling poll() with a NULL pollfd array. While this is
	  fine with POSIX's poll(2) system call, those who use Asterisk's
	  internal poll mechanism (Darwin systems) would have a failed
	  assertion occur when poll is called. (related to issue #10342)
	  ........ ................

2008-06-13 14:15 +0000 [r122558]  Tilghman Lesher <tlesher@digium.com>

	* main/dial.c, /: Merged revisions 122557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 |
	  tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines
	  Convert one more delimiter to use comma. (closes issue #12850)
	  Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: bcnit ........

2008-06-13 00:18 +0000 [r122467]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_parkandannounce.c, /, main/features.c: Merged revisions
	  122433 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 |
	  jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines
	  (closes issue 0012193) Reported by: davidw Patch by: Corydon76,
	  modified by me to work properly with ParkAndAnnounce app ........

2008-06-12 18:54 +0000 [r122313]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 122312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun
	  2008) | 17 lines Merged revisions 122311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun
	  2008) | 9 lines Properly play a holdtime message if the
	  announce-holdtime option is set to "once." (closes issue #12842)
	  Reported by: ramonpeek Patches: patch001.diff uploaded by
	  ramonpeek (license 266) ........ ................

2008-06-12 18:24 +0000 [r122242-122266]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r122262 | russell | 2008-06-12 13:23:54 -0500
	  (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008)
	  | 3 lines Fix some race conditions that cause ast_assert() to
	  report that chan_iax2 tried to remove an entry that wasn't in the
	  scheduler ........ ................

2008-06-12 15:27 +0000 [r122132-122180]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008)
	  | 16 lines Merged revisions 122137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008)
	  | 8 lines Flipflop the sections for two options, since the
	  section for 'X' (exit context) may otherwise absorb keypresses
	  meant for 's' (admin/user menu). (closes issue #12836) Reported
	  by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: blitzrage ........
	  ................

	* main/channel.c, /: Merged revisions 122131 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008)
	  | 12 lines Merged revisions 122130 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008)
	  | 4 lines Occasionally, the alertpipe loses its nonblocking
	  status, so detect and correct that situation before it causes a
	  deadlock. (Reported and tested by ctooley via #asterisk-dev)
	  ........ ................

2008-06-12 15:01 +0000 [r122126-122129]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions
	  122128 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9
	  lines Merged revisions 122127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1
	  line Arkadia tried to warn me, but the code added to
	  ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot
	  it until I was resolving conflicts in trunk. Ugh. Redundant code
	  removed. It wasn't harmful. Just dumb. ........ ................

	* main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c,
	  include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu,
	  12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) |
	  37 lines (closes issue #10668) Reported by: arkadia Tested by:
	  murf, arkadia Options added to forkCDR() app and the CDR() func
	  to remove some roadblocks for CDR applications. The "show
	  application ForkCDR" output was upgraded to more fully explain
	  the inner workings of forkCDR. The A option was added to forkCDR
	  to force the CDR system to NOT change the disposition on the
	  original CDR, after the fork. This involves ast_cdr_answer,
	  _busy, _failed, and so on. The T option was added to forkCDR to
	  force obedience of the cdr LOCKED flag in the ast_cdr_end, all
	  the disposition changing funcs (ast_cdr_answer, etc), and in the
	  ast_cdr_setvar func. The CHANGES file was updated to explain ALL
	  the new options added to satisfy this bug report (and some
	  requests made verbally and via email, irc, etc, over the past
	  months/year) The 's' option was added to the CDR() func, to force
	  it to skip LOCKED cdr's in the chain. Again, the new options
	  should be totally transparent to existing apps! Current behavior
	  of CDR, forkCDR, and the rest of the CDR system should not change
	  one little bit. Until you add the new options, at least! ........
	  ................

2008-06-11 18:57 +0000 [r121915]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 |
	  mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix
	  pseudo channel allocation errors on startup when using SS7
	  ........

2008-06-11 18:20 +0000 [r121872]  Tilghman Lesher <tlesher@digium.com>

	* main/sched.c, main/channel.c, /, channels/chan_agent.c,
	  main/abstract_jb.c: Merged revisions 121867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008)
	  | 11 lines Merged revisions 121861 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008)
	  | 3 lines Make calls to ast_assert() actually test something, so
	  that the error message printed is not nonsensical (reported by
	  mvanbaak via #asterisk-bugs). ........ ................

2008-06-11 17:59 +0000 [r121858]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 |
	  mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line
	  Make sure we hangup any calls we have and NULL out the ss7call
	  value when we get a reset circuit message. Fixes crash bug
	  ........

2008-06-11 17:45 +0000 [r121856]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt,
	  include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 |
	  tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines
	  Expand CDR uniqueid field to 150 chars, to account for maximum
	  systemname. (Closes issue #12831) ........

2008-06-11 16:13 +0000 [r121806]  Jeff Peeler <jpeeler@digium.com>

	* /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008)
	  | 9 lines Merged revisions 121804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008)
	  | 1 line add instructions for logging gdb output via set logging
	  on ........ ................

2008-06-10 18:36 +0000 [r121598]  Sean Bright <sean.bright@gmail.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400
	  (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun
	  2008) | 6 lines Fixes a problem with some buggy versions of GNU
	  awk (3.1.3) not liking carriage returns in scripts. (closes issue
	  #12749) Reported by: alinux Tested by: Laureano (on
	  #asterisk-dev), juggie ........ ................

2008-06-10 12:55 +0000 [r121445]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 121444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) |
	  12 lines Merged revisions 121442 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4
	  lines Update BRIDGEPEER variable before we do a generic bridge in
	  case we just broke out of a native bridge and fell through to
	  generic. (closes issue #12815) Reported by: ramonpeek ........
	  ................

2008-06-10 00:53 +0000 [r121404-121408]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008)
	  | 2 lines Bump up the debug level of a couple of messages
	  ........

2008-06-09 16:37 +0000 [r121283]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 121282 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008)
	  | 18 lines Merged revisions 121280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008)
	  | 10 lines Do not attempt to do emulation if an END digit is
	  received and the length is less than the defined minimum digit
	  length, and the other end only wants END digits (SIP INFO, for
	  example). (closes issue #12778) Reported by: tsearle Patches:
	  12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle
	  ........ ................

2008-06-09 16:36 +0000 [r121281]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 121279 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 |
	  tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines
	  Implement FINDLABEL matching for the new extension matching
	  engine. (closes issue #12800) Reported by: chris-mac Patches:
	  20080608__bug12800.diff.txt uploaded by Corydon76 (license 14)
	  ........

2008-06-09 15:10 +0000 [r121231]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 121230 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500
	  (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that
	  this is being merged to trunk/1.6.0 because it may affect
	  non-callback agents with ackcall set) ........ r121229 |
	  mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16
	  lines A unique situation of timeouts brought forth a failure
	  situation for autologoff in chan_agent. If using
	  AgentCallbackLogin-style agents, then if the timeout specified by
	  the Dial() to reach the agent's phone was shorter than the
	  timeout specified in queues.conf, then autologoff would only work
	  if the caller hung up while the agent's phone was ringing. This
	  patch allows autologoff to work in this situation when the call
	  in queue transfers to the next available agent (as it would have
	  if the timeout in queues.conf were less than the timeout in the
	  Dial()). (closes issue #12754) Reported by: Rodrigo Patches:
	  12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo
	  ........ ................

2008-06-08 01:43 +0000 [r121138-121164]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_console.c: Merged revisions 121163 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008)
	  | 4 lines This was accidentally reverted. Fixes a bug where if a
	  stream monitor thread was not created (caused from failure of
	  opening or starting the stream) pthread_cancel was called with an
	  invalid thread ID. ........

	* apps/app_parkandannounce.c, /: Merged revisions 121131 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07
	  Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce.
	  Also, loop made more efficient as announce template only needs to
	  be checked until the number of colon separated arguments run out,
	  not the entire pointer storage array. Was done in a similiar
	  fashion in 1.4, but here we're using less variables. ........

2008-06-07 14:19 +0000 [r121080]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /, channels/chan_agent.c: Merged revisions
	  121079 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008)
	  | 15 lines Merged revisions 121078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008)
	  | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes
	  issue #12807) Reported by: ys Patches: chan_agent_local.diff
	  uploaded by ys (license 281) ........ ................

2008-06-06 20:25 +0000 [r121011-121047]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 121010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 |
	  tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines
	  Make extension match characters case-insensitive. (closes issue
	  #12777) Reported by: jsmith Patches:
	  lower_case_patterns-trunk-v1.patch uploaded by jsmith (license
	  15) ........

2008-06-06 18:31 +0000 [r120907-120961]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008)
	  | 9 lines Merged revisions 120959 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008)
	  | 1 line add another LOW_MEMORY define I forgot ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008)
	  | 9 lines Merged revisions 120908 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008)
	  | 1 line only define thread storage variable if necessary for
	  LOW_MEMORY ........ ................

	* channels/chan_sip.c: Merged revisions 120906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008)
	  | 16 lines Merged revisions 120863,120885 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008)
	  | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two
	  allocations of the ast_rtp struct that were previously allocated
	  on the stack have been modified to use thread local storage
	  instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500
	  (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make
	  sure proper destructor function is called as well define two
	  thread storage local variables. ........ ................

2008-06-06 17:35 +0000 [r120864-120905]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_exec.c: Merged revisions 120904 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 |
	  tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines
	  For the purpose of making the changed syntax to ExecIf easier to
	  transition, allow the deprecated syntax (fixed for jmls on -dev).
	  ........

2008-06-05 21:39 +0000 [r120829]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 120828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 |
	  murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a
	  small fix for a crash that occurs when compiling AEL with global
	  vars ........

2008-06-05 17:17 +0000 [r120677]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 120676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008)
	  | 10 lines Merged revisions 120675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008)
	  | 2 lines Ignore appended resource when comparing JIDs. ........
	  ................

2008-06-05 16:42 +0000 [r120643-120674]  Brett Bryant <bbryant@digium.com>

2008-06-05 16:01 +0000 [r120566-120603]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged
	  revisions 120602 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 |
	  tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines
	  Conditionally load the AGI command gosub, depending on whether or
	  not res_agi has been loaded, fix a return value in the loader,
	  and ensure that the help workhorse header does not print on load.
	  ........

	* /, UPGRADE.txt: Merged revisions 120567 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 |
	  tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines
	  Add info on the [compat] section of asterisk.conf. ........

	* apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793)

2008-06-04 22:08 +0000 [r120515]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 120514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun
	  2008) | 14 lines Merged revisions 120513 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun
	  2008) | 6 lines Make sure that the string we set will survive the
	  unref of the queue member. Thanks to Russell, who pointed this
	  out. ........ ................

2008-06-04 20:35 +0000 [r120478]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 120477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 |
	  tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines
	  MSet doesn't necessarily need chan to be set ........

2008-06-04 15:38 +0000 [r120338]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 |
	  file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We
	  like tabs. ........

2008-06-04 14:13 +0000 [r120287]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 120286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun
	  2008) | 15 lines Merged revisions 120285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun
	  2008) | 7 lines Tab completion when removing a member should give
	  the member's interface, not the name, since the interface is what
	  is expected for the command. (closes issue #12783) Reported by:
	  davevg ........ ................

2008-06-04 13:34 +0000 [r120284]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) |
	  14 lines Merged revisions 120282 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6
	  lines Fix a log message and add a message for when the dialplan
	  is done reloading. (closes issue #12716) Reported by: chappell
	  Patches: dialplan_reload_2.diff uploaded by chappell (license 8)
	  ........ ................

2008-06-03 23:18 +0000 [r120228-120234]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008)
	  | 16 lines Merged revisions 120226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008)
	  | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD()
	  macro the loopback switch cannot perform any translation on the
	  extension number before searching for it in the target context.
	  (closes issue #12473) Reported by: chappell Patches:
	  pbx_loopback.c.diff uploaded by chappell (license 8) ........
	  ................

2008-06-03 22:18 +0000 [r120178]  Jeff Peeler <jpeeler@digium.com>

	* main/config.c: Merged revisions 120174 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008)
	  | 14 lines Merged revisions 120173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008)
	  | 6 lines (closes issue #11594) Reported by: yem Tested by: yem
	  This change decreases the buffer size allocated on the stack
	  substantially in config_text_file_load when LOW_MEMORY is turned
	  on. This change combined with the fix from revision 117462
	  (making mkintf not copy the zt_chan_conf structure) was enough to
	  prevent the crash. ........ ................

2008-06-03 22:08 +0000 [r120172]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/options.h, main/asterisk.c, Makefile,
	  main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c,
	  configs/pbx_realtime.conf (removed): Merged revisions 120171 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03
	  Jun 2008) | 5 lines Move compatibility options into
	  asterisk.conf, default them to on for upgrades, and off for new
	  installations. This includes the translation from pipes to commas
	  for pbx_realtime and the EXEC command for AGI, as well as the
	  change to the Set application not to support multiple variables
	  at once. ........

2008-06-03 21:35 +0000 [r120170]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r120169 | russell | 2008-06-03 16:35:11 -0500
	  (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008)
	  | 4 lines Fix another place where peer->callno could change at a
	  very bad time, and also fix a place where a peer was used after
	  the reference was released. (inspired by rev 120001) ........
	  ................

2008-06-03 16:24 +0000 [r120034]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500
	  (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008)
	  | 9 lines Save the callno when we're poking, because our peer
	  structure could change during destruction (and thus we unlock the
	  wrong callno, causing a cascade failure). (closes issue #12717)
	  Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: gewfie ........
	  ................

2008-06-03 15:57 +0000 [r119931-120000]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-vtest21,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue,
	  03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8
	  lines Updated the regressions on AEL. Hadn't updated this for the
	  changes I made to preserve ${EXTEN} in switches, which affected
	  several tests because it adds extra priorities, and at least one
	  needed to be updated because of the removal of the empty
	  extension warning message. ........ ................

	* res/ael/pval.c, /: Merged revisions 119930 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) |
	  24 lines Merged revisions 119929 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) |
	  16 lines as per
	  http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
	  which is a message from Philipp Kempgen, requesting that the
	  WARNING that an extension is empty be reduced to a NOTICE or
	  less, as empty extensions are syntactically possible, and no big
	  deal. With which I agree, and have removed that WARNING message
	  entirely. I think it is not necessary to see this message. It
	  didn't state that a NoOp() was inserted automatically on your
	  behalf, and really, as users, who cares? Why freak out dialplan
	  writers with unnecessary warnings? The details of the
	  machinations a compiler goes thru to produce working assembly
	  code is of little interest to most programmers-- we will follow
	  the unix principal of doing our work silently. ........
	  ................

2008-06-03 14:48 +0000 [r119928]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) |
	  10 lines Merged revisions 119926 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2
	  lines Treat ECONNREFUSED as an error that will stop further
	  retransmissions. (issue #AST-58, patch from Switchvox) ........
	  ................

2008-06-03 13:30 +0000 [r119745-119893]  Russell Bryant <russell@digium.com>

	* /, main/logger.c: Merged revisions 119892 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 |
	  russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines
	  Do a deep copy of file and function strings to avoid a potential
	  crash when modules are unloaded. (closes issue #12780) Reported
	  by: ys Patches: logger.diff uploaded by ys (license 281) --
	  modified by me for coding guidelines ........

	* /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119839 | russell | 2008-06-02 15:08:24 -0500
	  (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008)
	  | 7 lines Revert a change made for issue #12479. This change
	  caused a regression such that a dial string such as (IAX2/foo)
	  did not automatically fall back to dialing the 's' extension
	  anymore. (closes issue #12770) Reported by: dagmoller ........
	  ................

	* /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008)
	  | 4 lines Add app_fax from asterisk-addons, with some additional
	  changes to resolve compiler warnings, as well as update to the
	  APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under
	  the LGPL, so we can move this module into the main tree. ........

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 119799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 |
	  russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines
	  After determining that the version of spandsp installed is an
	  acceptable version, do a build and link test to ensure that the
	  library is usable, and that libtiff is also available ........

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  Merged revisions 119795 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 |
	  russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines
	  Add a configure script check for spandsp ........

	* main/manager.c, /: Merged revisions 119744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008)
	  | 13 lines Merged revisions 119742 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008)
	  | 5 lines Improve CLI command blacklist checking for the command
	  manager action. Previously, it did not handle case or whitespace
	  properly. This made it possible for blacklisted commands to get
	  executed anyway. (closes issue #12765) ........ ................

2008-06-02 14:40 +0000 [r119743]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c,
	  res/res_jabber.c: Merged revisions 119741 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 |
	  phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13
	  lines Do not link the guest account with any configured XMPP
	  client (in jabber.conf). The actual connection is made when a
	  call comes in Asterisk. Apply this fix to Jingle too. Fix the
	  ast_aji_get_client function that was not able to retrieve an XMPP
	  client from its JID. (closes issue #12085) Reported by: junky
	  Tested by: phsultan ........

2008-06-02 12:32 +0000 [r119532-119690]  Russell Bryant <russell@digium.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
	  revisions 119586,119637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008)
	  | 9 lines Merged revisions 119585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008)
	  | 1 line Added counter for unhandled_bmsg Print, this prevents
	  the logs to be flooded to fast and save CPU in this error
	  scenario. Added 'last_used' element to bc structure, when a
	  bchannel changes from used to free this exact time will be marked
	  in last_used. When a new channel is requested the find_free_chan
	  function will check if the new empty channel was used within the
	  last second, if yes it will search for the next channel, if no it
	  will return this channel. This simple mechanism has prooven to
	  prevent race conditions where the NT and TE tried to allocate the
	  exact same channel at the same time (RELEASE cause: 44). ........
	  ................ r119637 | crichter | 2008-06-02 04:35:04 -0500
	  (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02
	  Jun 2008) | 1 line fixed compile issue when dev-mode is enabled
	  ........ ................

	* /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119688 | russell | 2008-06-02 07:30:42 -0500
	  (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008)
	  | 3 lines Even of the first PING or LAGRQ doesn't get sent
	  because it comes up too soon, make sure to reschedule so it gets
	  sent later. ........ ................

	* /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119534 | russell | 2008-06-01 20:08:16 -0500
	  (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008)
	  | 2 lines Change a debug message to an actual debug message
	  ........ ................

	* apps/app_dial.c, /: Merged revisions 119531 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008)
	  | 10 lines Merged revisions 119530 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008)
	  | 2 lines Fix another typo in documentation ........
	  ................

2008-06-01 21:59 +0000 [r119529]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_dial.c, /: Merged revisions 119479 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008)
	  | 10 lines Merged revisions 119478 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008)
	  | 2 lines small typo fix 'retires' => 'retries' ........
	  ................

2008-05-30 21:24 +0000 [r119420]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: Merged revisions 119419 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008)
	  | 14 lines Merged revisions 119404 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008)
	  | 6 lines When joinempty=strict, it only failed on join if there
	  were busy members. If all members were logged out OR paused, then
	  it (incorrectly) let callers join the queue. (closes issue
	  #12451) Reported by: davidw ........ ................

2008-05-30 19:48 +0000 [r119356]  Joshua Colp <jcolp@digium.com>

	* main/autoservice.c, /: Merged revisions 119355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) |
	  10 lines Merged revisions 119354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2
	  lines Fix a bug I found while testing for another issue. ........
	  ................

2008-05-30 17:13 +0000 [r119304]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Oops, broke 1.6 (thanks MattF)

2008-05-30 16:57 +0000 [r119303]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk, /,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200
	  (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008)
	  | 14 lines dont use a bashism way to check the $VERSION variable.
	  The rc/init.d scripts, and safe_asterisk work on normal sh now
	  again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy
	  (me and loloski) FC9 (loloski) (closes issue #12687) Reported by:
	  loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt
	  uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak
	  ........ ................

2008-05-30 16:40 +0000 [r119297-119300]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c, /: Merged revisions 119299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 |
	  tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines
	  Suppress warning about pbx structure already existing ........

	* apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /,
	  CHANGES: Merged revisions 119296 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 |
	  tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines
	  Add native AGI command GOSUB, as invoking Gosub with EXEC does
	  not work properly. (closes issue #12760) Reported by: Corydon76
	  Patches: 20080530__bug12760.diff.txt uploaded by Corydon76
	  (license 14) Tested by: tim_ringenbach, Corydon76 ........

2008-05-30 13:01 +0000 [r119158-119240]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119239 | russell | 2008-05-30 07:59:11 -0500
	  (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r119238 | russell | 2008-05-30 07:55:36 -0500
	  (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008)
	  | 7 lines - Instead of only enforcing destination call number
	  checking on an ACK, check all full frames except for PING and
	  LAGRQ, which may be sent by older versions too quickly to contain
	  the destination call number. (As suggested by Tim Panton on the
	  asterisk-dev list) - Merge changes from
	  team/russell/iax2-frame-race, which prevents PING and LAGRQ from
	  being sent before the destination call number is known. ........
	  ................ ................

	* main/autoservice.c, /: Merged revisions 119157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008)
	  | 18 lines Merged revisions 119156 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008)
	  | 10 lines Fix a race condition in channel autoservice. There was
	  still a small window of opportunity for a DTMF frame, or some
	  other deferred frame type, to come in and get dropped. (closes
	  issue #12656) (closes issue #12656) Reported by: dimas Patches:
	  v3-12656.patch uploaded by dimas (license 88) -- with some
	  modifications by me ........ ................

2008-05-29 20:26 +0000 [r119073]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008)
	  | 15 lines Merged revisions 119071 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008)
	  | 7 lines Call waiting tone occurs too often, because it's
	  getting serviced by both subchannels. (closes issue #11354)
	  Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
	  by Corydon76 (license 14) ........ ................

2008-05-29 19:06 +0000 [r118960-119014]  Russell Bryant <russell@digium.com>

	* apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119013 | russell | 2008-05-29 14:05:33 -0500
	  (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008)
	  | 4 lines - Fix a typo in the argument to Playtones - use
	  ast_safe_sleep() instead of calling the wait application (thanks
	  to tilghman for pointing these out!) ........ ................

	* /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r119010 | russell | 2008-05-29 13:54:11 -0500
	  (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r119009 | russell | 2008-05-29 13:49:12 -0500
	  (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008)
	  | 7 lines Merge changes from
	  team/russell/iax2-another-fix-to-the-fix As described in the
	  following post to the asterisk-dev mailing list, only enforce
	  destination call numbers when processing an ACK.
	  http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
	  (closes issue #12631) ........ ................ ................

	* apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r118962 | russell | 2008-05-29 12:52:00 -0500
	  (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008)
	  | 3 lines - Mark app_milliwatt dependent on res_indications
	  (thanks to jsmith) - fix a typo in a log message (thanks to
	  qwell) ........ ................

	* apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r118959 | russell | 2008-05-29 12:46:04 -0500
	  (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008)
	  | 3 lines Change milliwatt to use the proper tone by default
	  (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz
	  for anyone that might want it. ........ ................

2008-05-29 17:42 +0000 [r118958]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c, channels/chan_zap.c, /,
	  channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
	  include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions
	  118955,118957 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008)
	  | 11 lines Merged revisions 118953 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008)
	  | 3 lines Add some debugging code that ensures that when we do
	  deadlock avoidance, we don't lose the information about how a
	  lock was originally acquired. ........ ................ r118957 |
	  tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10
	  lines Merged revisions 118954 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008)
	  | 2 lines Define also when not DEBUG_THREADS ........
	  ................

2008-05-29 04:11 +0000 [r118909]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed,
	  28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) |
	  46 lines (closes issue #10668) (closes issue #11721) (closes
	  issue #12726) Reported by: arkadia Tested by: murf These changes:
	  1. revert the changes made via bug 10668; I should have known
	  that such changes, even tho they made sense at the time, seemed
	  like an omission, etc, were actually integral to the CDR system
	  via forkCDR. It makes sense to me now that forkCDR didn't
	  natively end any CDR's, but rather depended on natively closing
	  them all at hangup time via traversing and closing them all,
	  whether locked or not. I still don't completely understand the
	  benefits of setvar and answer operating on locked cdrs, but I've
	  seen enough to revert those changes also, and stop messing up
	  users who depended on that behavior. bug 12726 found reverting
	  the changes fixed his changes, and after a long review and
	  working on forkCDR, I can see why. 2. Apply the suggested
	  enhancements proposed in 10668, but in a completely compatible
	  way. ForkCDR will behave exactly as before, but now has new
	  options that will allow some actions to be taken that will
	  slightly modify the outcome and side-effects of forkCDR. Based on
	  conversations I've had with various people, these small tweaks
	  will allow some users to get the behavior they need. For
	  instance, users executing forkCDR in an AGI script will find the
	  answer time set, and DISPOSITION set, a situation not covered
	  when the routines were first written. 3. A small problem in the
	  cdr serializer would output answer and end times even when they
	  were not set. This is now fixed. ........ ................

2008-05-28 18:07 +0000 [r118781]  Michiel van Baak <michiel@vanbaak.info>

	* /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008)
	  | 2 lines remove unused astobj.h header file from chan_skinny.c
	  ........

2008-05-28 14:31 +0000 [r118648]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged
	  revisions 118647 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) |
	  12 lines Merged revisions 118646 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4
	  lines Add an option to use the source IP address of RTP as the
	  destination IP address of UDPTL when a specific option is
	  enabled. If the remote side is properly configured (ports
	  forwarded) then UDPTL will flow. (closes issue #10417) Reported
	  by: cstadlmann ........ ................

2008-05-28 14:13 +0000 [r118615-118645]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged
	  revisions 118644 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 |
	  phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10
	  lines Changed to temporary namespaces to match with latest XEPs.
	  As soon as Jingle is completely standardized, we can set those
	  namespaces to their final values. Added two attributes to the
	  jingle_pvt struct to store the content name attributes. Reported
	  by Robert McQueen on Telepathy's framework mailing list :
	  http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html
	  Keeping working on our Jingle stack! ........

	* channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008)
	  | 1 line Code simplification ........

2008-05-27 19:35 +0000 [r118561]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) |
	  12 lines Merged revisions 118558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4
	  lines Fix an issue where codec preferences were not set on
	  dialogs that were not authenticated via a user or peer and allow
	  framing to work without rtpmap in the SDP. (closes issue #12501)
	  Reported by: slimey ........ ................

2008-05-27 19:28 +0000 [r118557]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/compat.h: Merged revisions 118556 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27
	  May 2008) | 6 lines Add printf format attribute for vasprintf().
	  (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff
	  uploaded by snuffy (license 35) ........

2008-05-27 19:22 +0000 [r118555]  Tilghman Lesher <tlesher@digium.com>

	* main/cli.c, /: Merged revisions 118554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008)
	  | 14 lines Merged revisions 118551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008)
	  | 6 lines When showing an error message for a command, don't
	  shorten the command output, as it tends to confuse the user (it's
	  fine for suggesting other commands, however). Reported by:
	  seanbright (on #asterisk-dev) Fixed by: me ........
	  ................

2008-05-27 19:09 +0000 [r118518]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May
	  2008) | 19 lines Merged revisions 118509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May
	  2008) | 11 lines Russell noted to me that in the case that
	  separate threads use their own addressing system, the fix I made
	  for issue 12376 does not guarantee uniqueness to the datastores'
	  uids. Though I know of no system that works this way, I am going
	  to change this right now to prevent trying to track down some
	  future bug that may occur and cause untold hours of debugging
	  time to track down. The change involves using a global counter
	  which increases with each new chanspy_ds which is created. This
	  guarantees uniqueness. ........ ................

2008-05-27 18:59 +0000 [r118471]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 118466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008)
	  | 16 lines Merged revisions 118465 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008)
	  | 8 lines NULL character should terminate only commands back to
	  the core, not log messages to the console. (closes issue #12731)
	  Reported by: seanbright Patches: 20080527__bug12731.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: seanbright ........
	  ................

2008-05-27 17:25 +0000 [r118418]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_voicemail.c: small update to the g() option of
	  app_voicemail to note that gain changes only work on zap channels
	  right now. issue #12578 shows it's not clear right now.

2008-05-27 16:48 +0000 [r118378-118382]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May
	  2008) | 22 lines Merged revisions 118365 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May
	  2008) | 14 lines Add a unique id to the datastore allocated in
	  app_chanspy since it is possible that multiple spies may be
	  listening to the same channel. (closes issue #12376) Reported by:
	  DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
	  (license 60) Tested by: destiny6628 (closes issue #12243)
	  Reported by: atis ........ ................

	* /: Hmm, I apparently forgot to commit the block of revision
	  118175. Now I'm doing it.

2008-05-27 15:47 +0000 [r118360]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/queues.conf.sample: Merged revisions 118359 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500
	  (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008)
	  | 3 lines Add a note that pbx_config.so is needed for Local
	  channels. (Closes issue #12671) ........ ................

2008-05-27 14:51 +0000 [r118331]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/compat.h: Merged revisions 118328 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27
	  May 2008) | 2 lines Add printf attribute to asprintf ........

2008-05-27 13:30 +0000 [r118301-118303]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008)
	  | 6 lines When binding anonymously, credentials are still needed.
	  (closes issue #12601) Reported by: suretec Patches:
	  res_config_ldap.c.patch uploaded by suretec (license 70) ........

	* /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 |
	  tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines
	  In compat14 mode, don't translate pipes inside expressions, as
	  they aren't argument delimiters, but rather 'or' symbols. (Closes
	  issue #12723) ........

2008-05-25 16:20 +0000 [r118253]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008)
	  | 20 lines Merged revisions 118251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008)
	  | 12 lines Realtime flag affects construction in multiple ways,
	  so consulting whether rtcachefriends was set was done too soon
	  (needed to be done inside build_peer, not just as a flag to
	  build_peer). Also, fullcontact needed to be reconstructed,
	  because realtime separates the embedded ';' into multiple fields.
	  (closes issue #12722) Reported by: barthpbx Patches:
	  20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: barthpbx (Much of the discussion happened on
	  #asterisk-dev for diagnosing this issue) ........
	  ................

2008-05-24 01:15 +0000 [r118177-118179]  Jeff Peeler <jpeeler@digium.com>

	* doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23
	  May 2008) | 1 line add document describing API changes from 1.4.0
	  to 1.6.0 ........

2008-05-23 21:37 +0000 [r118168]  Brett Bryant <bbryant@digium.com>

	* main/manager.c, /, main/http.c, include/asterisk/manager.h:
	  Merged revisions 118161 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 |
	  bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines
	  Add new functionality to http server that requires manager
	  authentication for any path that includes a directory named
	  'private'. This patch also requires manager authentication for
	  any POST's being sent to the server as well to help secure
	  uploads. ........

2008-05-23 21:31 +0000 [r118165]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Merged revisions 118164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008)
	  | 9 lines Merged revisions 118163 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008)
	  | 1 line Fix a few things I missed to ensure zt_chan_conf
	  structure is not modified in mkintf ........ ................

2008-05-23 18:15 +0000 [r118130]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, /: Merged revisions 118129 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 |
	  tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines
	  Protect the object from changing while the 'odbc show' CLI
	  command is running (Closes issue #12704) ........

2008-05-23 13:00 +0000 [r118054]  Tilghman Lesher <tlesher@digium.com>

	* doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008)
	  | 11 lines Merged revisions 118052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008)
	  | 3 lines Add information on using the Asterisk console,
	  including tab command line completion. (Closes issue #12681)
	  ........ ................

2008-05-23 12:37 +0000 [r118050]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, /, main/utils.c: Merged revisions
	  118049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008)
	  | 17 lines Merged revisions 118048 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008)
	  | 9 lines Don't declare a function that takes variable arguments
	  as inline, because it's not valid, and on some compilers, will
	  emit a warning.
	  http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
	  issue #12289) Reported by: francesco_r Patches by Tilghman, final
	  patch by me ........ ................

2008-05-23 11:02 +0000 [r118021]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions
	  118020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 |
	  phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15
	  lines - remove whitespaces between tags in received XML packets
	  before giving them to the parser ; - report Gtalk error messages
	  from a buddy to the console. This patch makes Asterisk "Google
	  Jingle" (chan_gtalk) implementation work with Empathy. Note that
	  this is only true for audio streams, not video. Thank you to PH
	  for his great help! (closes issue #12647) Reported by: PH
	  Patches: trunk-12647-1.diff uploaded by phsultan (license 73)
	  Tested by: phsultan, PH ........

2008-05-22 21:43 +0000 [r117984-117987]  Tilghman Lesher <tlesher@digium.com>

	* /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged
	  revisions 117986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 |
	  tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines
	  Add a compatibility option for upgrading realtime extensions
	  ........

2008-05-22 18:55 +0000 [r117901]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 117900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008)
	  | 10 lines Merged revisions 117899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008)
	  | 2 lines Also remove preamble from asynchronous events (reported
	  by jsmith on #asterisk-dev) ........ ................

2008-05-22 15:51 +0000 [r117793]  Sean Bright <sean.bright@gmail.com>

	* /, configs/jabber.conf.sample: Merged revisions 117792 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu,
	  22 May 2008) | 1 line Minor text fix. roster -> resource.
	  ........

2008-05-22 13:41 +0000 [r117757]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, /, build_tools/make_buildopts_h: Merged
	  revisions 117756 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 |
	  russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines
	  Store build-time options as a string in AST_BUILDOPTS in
	  buildopts.h. Also, display this information in the "core show
	  settings" CLI command. This is useful if you want to verify that
	  you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc.
	  ........

2008-05-21 22:01 +0000 [r117659-117660]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Merged revisions 117658 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008)
	  | 10 lines Merged revisions 117582 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008)
	  | 2 lines Ensure that passed in zt_chan_conf structure is not
	  modified in mkintf. ........ ................

	* channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008)
	  | 12 lines Merged revisions 117462 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008)
	  | 3 lines Pass a pointer for the conf parameter to the function
	  mkintf rather than the whole zt_chan_conf structure. Another
	  commit is following to make sure the zt_chan_conf structure is
	  not modified. ........ ................

2008-05-21 19:45 +0000 [r117576]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) |
	  10 lines Merged revisions 117574 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2
	  lines Apply the autoframing setting to dialogs that do not get
	  matched against a user or peer. ........ ................

2008-05-21 18:44 +0000 [r117522]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 117520 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008)
	  | 11 lines Merged revisions 117519 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008)
	  | 3 lines Strip the preamble from the output also when -rx is not
	  being used (Related to issue #12702) ........ ................

2008-05-21 18:29 +0000 [r117486-117516]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, /: Merged revisions 117515 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008)
	  | 12 lines Merged revisions 117514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008)
	  | 4 lines Don't filter the magic character in the network
	  verboser. It gets filtered once it reaches the client. (related
	  to issue #12702, pointed out by tilghman) ........
	  ................

	* main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r117508 | russell | 2008-05-21 13:20:11 -0500
	  (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008)
	  | 7 lines 1) Don't print the verbose marker in front of every
	  message from ast_verbose() being sent to remote consoles. 2) Fix
	  pbx_gtkconsole to filter out the verbose marker. (related to
	  issue #12702) ........ ................

	* main/asterisk.c, /: Merged revisions 117481 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008)
	  | 14 lines Merged revisions 117479 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008)
	  | 6 lines Don't display the verbose marker for calls to
	  ast_verbose() that do not include a VERBOSE_PREFIX in front of
	  the message. (closes issue #12702) Reported by: johnlange Patched
	  by me ........ ................

2008-05-21 02:21 +0000 [r117368]  Mark Michelson <mmichelson@digium.com>

	* main/config.c, /: Merged revisions 117367 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 |
	  mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19
	  lines Be sure that we cache included files for each source file
	  which loads a configuration file. As it was, only the first did
	  so. This led to a problem if the included file was changed (but
	  not the configuration file which includes it) and the second
	  source file attempted to reload the configuration. It would not
	  see that the included file had changed. In this particular
	  example, res_phoneprov and chan_sip both loaded sip.conf, which
	  included a file call sip.peers.conf. Since res_phoneprov was the
	  first to load sip.conf, only it cached the fact that sip.conf
	  included sip.peers.conf. If sip.peers.conf were changed and
	  sip.conf were not and a sip reload were issued (meaning that
	  chan_sip attempts to reload sip.conf only if it and its included
	  files have changed) the changes made to sip.peers.conf would not
	  be seen and therefore no action would be taken. (closes issue
	  #12693) Reported by: marsosa ........

2008-05-21 01:20 +0000 [r117365]  Steve Murphy <murf@digium.com>

	* /, utils/ael_main.c: Merged revisions 117335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 |
	  murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines
	  These changes were made via the comments atis_work made at 4:30am
	  (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He
	  noted that a backslash was being inserted before commas in app
	  call arguments in the extensions.conf.aeldump file that you get
	  from aelparse with the -w arg. This was being generated from code
	  left over from 1.4, where commas were substituted with '|', and
	  any remaining commas needed to be escaped. Many thanks to atis
	  for his comment; please let us know if these changes break
	  anything! ........

2008-05-19 16:58 +0000 [r117134-117137]  Joshua Colp <jcolp@digium.com>

	* res/res_smdi.c, /: Merged revisions 117136 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) |
	  14 lines Merged revisions 117135 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6
	  lines Use the right pthread lock and condition when waiting.
	  (closes issue #12664) Reported by: tomo1657 Patches:
	  res_smdi.c.patch uploaded by tomo1657 (license 484) ........
	  ................

2008-05-19 16:07 +0000 [r117089]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500
	  (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008)
	  | 2 lines The addition of usleep(2) within ast_assert requires
	  the inclusion of the unistd.h header ........ ................

2008-05-19 16:05 +0000 [r117083-117087]  Joshua Colp <jcolp@digium.com>

	* /, main/logger.c: Merged revisions 117085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 |
	  file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The
	  logger closes the files it is logging to when reloading so we
	  have to read in the logger configuration even if it has not
	  changed so that the logs get opened again. (closes issue #12665)
	  Reported by: DennisD ........

	* /, channels/h323/ast_h323.cxx: Merged revisions 117082 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon,
	  19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6
	  lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682)
	  Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby
	  (license 430) ........ ................

2008-05-19 03:44 +0000 [r116980]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r116979 | russell | 2008-05-18 22:44:28 -0500
	  (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008)
	  | 4 lines Avoid access of uninitialized memory. This caused a
	  bunch of crashes for me while doing load testing of development
	  branch where I'm working on some performance improvements.
	  ........ ................

2008-05-18 21:18 +0000 [r116949]  Tilghman Lesher <tlesher@digium.com>

	* /, utils/astcanary.c: Merged revisions 116948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 |
	  tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines
	  Add a set of text to the file astcanary uses to communicate back
	  the main Asterisk process, which explains the purpose for the
	  file being there. This should assist people who find the file and
	  wonder why it exists. ........

2008-05-18 19:59 +0000 [r116922]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 |
	  russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines
	  Remove duplicate colon on Reason header (closes issue #12678)
	  ........

2008-05-17 19:40 +0000 [r116849-116885]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri,
	  16 May 2008) | 12 lines Merged revisions 116799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4
	  lines Check to make sure an RTP structure exists before calling
	  ast_rtp_new_source on it. (closes issue #12669) Reported by:
	  sbisker ........ ................

2008-05-16 20:03 +0000 [r116798]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 |
	  mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try
	  to see if we can make our ringback situation a little better
	  ........

2008-05-15 22:07 +0000 [r116636-116695]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/utils.h, /, include/asterisk/strings.h: Merged
	  revisions 116694 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 |
	  tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines
	  Add an extra check in ast_strlen_zero, and make ast_assert() not
	  print the file, line, and function name twice. (Closes issue
	  #12650) ........

	* cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 |
	  tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines
	  Don't unload config on reload, when config has not changed.
	  (Closes issue #12652) ........

2008-05-14 21:41 +0000 [r116470]  Russell Bryant <russell@digium.com>

	* main/rtp.c, main/sched.c, main/channel.c, main/udptl.c,
	  include/asterisk/utils.h, /, channels/chan_agent.c,
	  main/abstract_jb.c, include/asterisk/channel.h: Merged revisions
	  116469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008)
	  | 12 lines Merged revisions 116463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008)
	  | 4 lines Add ast_assert(), which can be used to handle fatal
	  errors. It is only compiled in if dev-mode is enabled, and only
	  aborts if DO_CRASH is defined. (inspired by issue #12650)
	  ........ ................

2008-05-14 21:39 +0000 [r116468]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 116467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008)
	  | 15 lines Merged revisions 116466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008)
	  | 7 lines Avoid zombies when the channel exits before the AGI.
	  (closes issue #12648) Reported by: gkloepfer Patches:
	  20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: gkloepfer ........ ................

2008-05-14 20:43 +0000 [r116408-116411]  Jason Parker <jparker@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 116410 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r116410 | qwell | 2008-05-14 15:43:26 -0500
	  (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May
	  2008) | 1 line Document exitcontext in app_voicemail sample
	  config ........ ................

	* apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) |
	  9 lines Voicemail "* exit" should not require an exitcontext to
	  be specified. The behavior in 1.4 was that it would use the
	  current context if an exitcontext existed. (closes issue #12605)
	  Reported by: kenjreno Patches: 12605-starexit.diff uploaded by
	  qwell (license 4) Tested by: file ........

2008-05-14 18:54 +0000 [r116351-116354]  Joshua Colp <jcolp@digium.com>

	* /, main/Makefile: Merged revisions 116353 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) |
	  12 lines Merged revisions 116352 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4
	  lines Add linux-gnueabi in. (closes issue #12529) Reported by:
	  tzafrir ........ ................

	* /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4
	  lines Make the ldap version setting work without having both
	  version and protocol set. (closes issue #12613) Reported by:
	  suretec ........

2008-05-14 17:01 +0000 [r116319]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500
	  (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008)
	  | 2 lines Detect another way for a connection to have gone away.
	  (closes issue #12618) Reported by: ctooley Patches:
	  1.4-externalivr-test_fd.diff uploaded by ctooley (license 136)
	  trunk-externalivr-test_fd.diff uploaded by ctooley (license 136)
	  ........ ................

2008-05-14  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta9 released.

2008-05-14 13:13 +0000 [r116236]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11
	  lines Merged revisions 116230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3
	  lines Accept text messages even with Content-Type:
	  text/plain;charset=Södermanländska ........ ................

2008-05-14 00:20 +0000 [r116096-116139]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /, include/asterisk/lock.h: Merged revisions
	  116089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May
	  2008) | 20 lines Merged revisions 116088 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May
	  2008) | 12 lines A change to the way channel locks are handled
	  when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock,
	  it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in
	  menuselect, the actual origin of channel locks is obscured by the
	  fact that all channel locks appear to happen in the function
	  ast_channel_lock(). This code change redefines ast_channel_lock
	  to be a macro which maps to __ast_channel_lock(), which then
	  relays the proper file name, line number, and function name
	  information to the core lock functions so that this information
	  will be displayed in the case that there is some sort of locking
	  error or core show locks is issued. ........ ................

2008-05-13 21:19 +0000 [r116020-116040]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /: Merged revisions 116039 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r116039 | russell | 2008-05-13 16:18:55 -0500
	  (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008)
	  | 24 lines Fix a deadlock involving channel autoservice and
	  chan_local that was debugged and fixed by mmichelson and me. We
	  observed a system that had a bunch of threads stuck in
	  ast_autoservice_stop(). The reason these threads were waiting
	  around is because this function waits to ensure that the channel
	  list in the autoservice thread gets rebuilt before the stop()
	  function returns. However, the autoservice thread was also
	  locked, so the autoservice channel list was never getting
	  rebuilt. The autoservice thread was stuck waiting for the channel
	  lock on a local channel. However, the local channel was locked by
	  a thread that was stuck in the autoservice stop function. It
	  turned out that the issue came down to the local_queue_frame()
	  function in chan_local. This function assumed that one of the
	  channels passed in as an argument was locked when called.
	  However, that was not always the case. There were multiple cases
	  in which this channel was not locked when the function was
	  called. We fixed up chan_local to indicate to this function
	  whether this channel was locked or not. The previous assumption
	  had caused local_queue_frame() to improperly return with the
	  channel locked, where it would then never get unlocked. (closes
	  issue #12584) (related to issue #12603) ........ ................

	* main/autoservice.c, /: Merged revisions 116001 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008)
	  | 13 lines Merged revisions 115990 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008)
	  | 5 lines Fix an issue that I noticed in autoservice while
	  mmichelson and I were debugging a different problem. I noticed
	  that it was theoretically possible for two threads to attempt to
	  start the autoservice thread at the same time. This change makes
	  the process of starting the autoservice thread, thread-safe.
	  ........ ................

2008-05-13 20:30 +0000 [r115946]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue,
	  13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4
	  lines Use the right flag to open the audio in non-blocking.
	  (closes issue #12616) Reported by: nicklewisdigiumuser ........
	  ................

2008-05-13 20:19 +0000 [r115940-115942]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 |
	  mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line
	  Need to clear calling_party_cat variable after we retrieve it
	  ........

	* channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 |
	  mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add
	  support for receiving calling party category ........

2008-05-13 18:38 +0000 [r115887]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 115886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008)
	  | 11 lines Merged revisions 115884 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008)
	  | 3 lines If the socket dies (read returns 0=EOF), return
	  immediately. (Closes issue #12637) ........ ................

2008-05-13 17:48 +0000 [r115848-115851]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c, /: Merged revisions 115847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 |
	  russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines
	  Initialize the start time in smdi_msg_wait. Somehow this code got
	  lost in trunk. ........

2008-05-12 17:57 +0000 [r115738]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: Merged revisions 115737 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May
	  2008) | 15 lines Merged revisions 115735 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May
	  2008) | 7 lines If a thread holds no locks, do not print any
	  information on the thread when issuing a core show locks command.
	  This will help to de-clutter output somewhat. Russell said it
	  would be fine to place this improvement in the 1.4 branch, so
	  that's why it's going here too. ........ ................

2008-05-12 16:36 +0000 [r115706]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 115705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 |
	  qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line
	  Correctly document state interface for AddQueueMember. Discovered
	  while looking at issue #12626. ........

2008-05-12 15:18 +0000 [r115672]  Brett Bryant <bbryant@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008)
	  | 3 lines A small change to fix iax2 native bridging. ........

2008-05-11 03:27 +0000 [r115599-115601]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
	  115600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 |
	  mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add
	  Zap MTP2 support to chan_zap ........

	* channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 |
	  mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line
	  Open up audio channel when we get ACM on SS7 event ........

2008-05-10 14:22 +0000 [r115597]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 |
	  tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines
	  Ensure that "calldate" is acceptable for a column name. ........

2008-05-09 16:38 +0000 [r115581]  Joshua Colp <jcolp@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 115580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) |
	  10 lines Merged revisions 115579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2
	  lines Improve res_ninit and res_ndestroy autoconf logic on the
	  Darwin platform. ........ ................

2008-05-08 19:21 +0000 [r115553-115570]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115569 | russell | 2008-05-08 14:20:35 -0500
	  (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008)
	  | 2 lines Remove debug output. ........ ................

	* /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115566 | russell | 2008-05-08 14:17:04 -0500
	  (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r115565 | russell | 2008-05-08 14:15:25 -0500
	  (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008)
	  | 25 lines Fix a race condition that bbryant just found while
	  doing some IAX2 testing. He was running Asterisk trunk running
	  IAX2 calls through a few Asterisk boxes, however, the audio was
	  extremely choppy. We looked at a packet trace and saw a storm of
	  INVAL and VNAK frames being sent from one box to another. It
	  turned out that what had happened was that one box tried to send
	  a CONTROL frame before the 3 way handshake had completed. So,
	  that frame did not include the destination call number, because
	  it didn't have it yet. Part of our recent work for security
	  issues included an additional check to ensure that frames that
	  are supposed to include the destination call number have the
	  correct one. This caused the frame to be rejected with an INVAL.
	  The frame would get retransmitted for forever, rejected every
	  time ... This race condition exists in all versions that got the
	  security changes, in theory. However, it is really only likely
	  that this would cause a problem in Asterisk trunk. There was a
	  control frame being sent (SRCUPDATE) at the _very_ beginning of
	  the call, which does not exist in 1.2 or 1.4. However, I am
	  fixing all versions that could potentially be affected by the
	  introduced race condition. These changes are what bbryant and I
	  came up with to fix the issue. Instead of simply dropping control
	  frames that get sent before the handshake is complete, the code
	  attempts to wait a little while, since in most cases, the
	  handshake will complete very quickly. If it doesn't complete
	  after yielding for a little while, then the frame gets dropped.
	  ........ ................ ................

	* /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008)
	  | 11 lines Merged revisions 115561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008)
	  | 3 lines Don't give up on attempting an outbound registration if
	  we receive a 408 Timeout. (closes issue #12323) ........
	  ................

	* /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions
	  115558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008)
	  | 11 lines Merged revisions 115557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008)
	  | 3 lines remove postgres_cdr.sql, as the CDR schema is in
	  realtime_pgsql.sql, as well (closes issue #9676) ........
	  ................

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115555 | russell | 2008-05-08 10:32:48 -0500
	  (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008)
	  | 3 lines Don't exit the script if Asterisk is not running.
	  (closes issue #12611) ........ ................

	* main/pbx.c, /: Merged revisions 115552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008)
	  | 12 lines Merged revisions 115551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008)
	  | 4 lines Don't use a channel before checking for channel
	  allocation failure. (closes issue #12609) Reported by: edantie
	  ........ ................

2008-05-08 15:08 +0000 [r115549]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 |
	  mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line
	  Remove unused code as well as demote an error message to a debug
	  message ........

2008-05-08 14:41 +0000 [r115538-115547]  Russell Bryant <russell@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115546 | russell | 2008-05-08 09:41:12 -0500
	  (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008)
	  | 4 lines Use the same method for executing Asterisk as the rest
	  of the script. (closes issue #12611) Reported by: b_plessis
	  ........ ................

2008-05-07 18:35 +0000 [r115514-115524]  Russell Bryant <russell@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008)
	  | 6 lines Only save a password if a username exists. (closes
	  issue #12600) Reported By: suretec Patch by me ........

	* /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008)
	  | 7 lines Use the default that the log output claims will be used
	  for the basedn (closes issue #12599) Reported by: suretec
	  Patches: 12599.patch uploaded by juggie (license 24) ........

	* /, channels/chan_h323.c: Merged revisions 115519 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008)
	  | 2 lines Let chan_h323 build in dev mode ........

	* /, include/asterisk/dlinkedlists.h (removed),
	  channels/chan_iax2.c: Merged revisions 115513 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008)
	  | 19 lines Merged revisions 115512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r115512 | russell | 2008-05-07 11:24:09 -0500
	  (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008)
	  | 3 lines Remove remnants of dlinkedlists. I didn't actually use
	  them in the final version of my IAX2 improvements. ........
	  ................ ................

2008-05-07 13:49 +0000 [r115510]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07
	  May 2008) | 2 lines Update typos in description fields (closes
	  issue #12598) Reported by: suretec Patches:
	  asterisk_schema_changes.patch uploaded by suretec (license 70)
	  ........

2008-05-06 19:56 +0000 [r115420-115424]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115423 | qwell | 2008-05-06 14:55:45 -0500
	  (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r115422 | qwell | 2008-05-06 14:55:29 -0500
	  (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
	  7 lines read requires an argument on some non-bash shells (closes
	  issue #12593) Reported by: bkruse Patches:
	  getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
	  ........ ................ ................

	* /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115419 | qwell | 2008-05-06 14:38:44 -0500
	  (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) |
	  7 lines Switch to using ast_random() rather than just rand().
	  This does not fix the bug reported, but I believe it is correct.
	  (from issue #12446) Patches: bug_12446.diff uploaded by snuffy
	  (license 35) ........ ................

2008-05-06 19:33 +0000 [r115417]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 115416 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008)
	  | 10 lines Merged revisions 115415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008)
	  | 2 lines Don't print the terminating NUL. (Closes issue #12589)
	  ........ ................

2008-05-06 13:57 +0000 [r115343]  Joshua Colp <jcolp@digium.com>

	* /, configure, configure.ac: Merged revisions 115342 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue,
	  06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2
	  lines Add in missing argument. ........ ................

2008-05-05 23:01 +0000 [r115335]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /, main/logger.c: Merged revisions 115334 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500
	  (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008)
	  | 7 lines Separate verbose output from CLI output, by using a
	  preamble. (closes issue #12402) Reported by: Corydon76 Patches:
	  20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76
	  (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt
	  uploaded by Corydon76 (license 14) ........ ................

2008-05-05 22:17 +0000 [r115331]  Joshua Colp <jcolp@digium.com>

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac: Merged revisions 115328 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) |
	  10 lines Merged revisions 115327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2
	  lines Make sure that either the main speex library contains
	  preprocess functions or that speexdsp does. If both fail then
	  speex stuff can not be built. ........ ................

2008-05-05 22:14 +0000 [r115330]  Mark Michelson <mmichelson@digium.com>

	* main/config.c, /: Merged revisions 115329 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 |
	  mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15
	  lines #execing the same file multiple times led to warning
	  messages saying that the same file was being #included twice.
	  This was due to the fact that #exec created a temporary file
	  which was then #included. The name of the temporary file was the
	  name of the #exec'd file, with the Unix timestamp and thread ID
	  concatenated. The issue was that if multiple #exec statements of
	  the same file were reached in the same second, then the result
	  was that the temporary files would have duplicate names. To
	  resolve this, the temporary file now has microsecond resolution
	  for the timestamp portion. (closes issue #12574) Reported by:
	  jmls Patches: 12574.patch uploaded by putnopvut (license 60)
	  Tested by: jmls, putnopvut ........

2008-05-05 21:44 +0000 [r115322]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 115321 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May
	  2008) | 21 lines Merged revisions 115320 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May
	  2008) | 13 lines Don't consider a caller "handled" until the
	  caller is bridged with a queue member. There was too much of an
	  opportunity for the member to hang up (either during a delay,
	  announcement, or overly long agi) between the time that he
	  answered the phone and the time when he actually was bridged with
	  the caller. The consequence of this was that if the member hung
	  up in that interval, then proper abandonment details would not be
	  noted in the queue log if the caller were to hang up at any point
	  after the member hangup. (closes issue #12561) Reported by:
	  ablackthorn ........ ................

2008-05-05 20:28 +0000 [r115316]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008)
	  | 2 lines Remove my rant, since I have now replaced the rant with
	  code. ........

2008-05-05 19:58 +0000 [r115310]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, /: Merged revisions 115309 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500
	  (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008)
	  | 2 lines Err, the documentation on the return value of
	  ast_odbc_backslash_is_escape is exactly backwards. ........
	  ................

2008-05-05 19:50 +0000 [r115306]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008)
	  | 13 lines Merged revisions 115304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008)
	  | 5 lines Avoid putting opaque="" in Digest authentication. This
	  patch came from switchvox. It fixes authentication with Primus in
	  Canada, and has been in use for a very long time without causing
	  problems with any other providers. (closes issue AST-36) ........
	  ................

2008-05-05 19:43 +0000 [r115303]  Tilghman Lesher <tlesher@digium.com>

	* /, UPGRADE.txt: Merged revisions 115302 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 |
	  tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines
	  Note change for ExecIf syntax (caught by jmls on IRC) ........

2008-05-05 10:55 +0000 [r115289]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: Merged revisions 115288 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 |
	  kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2
	  lines clarify wording ........

2008-05-05 03:26 +0000 [r115287]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk, /,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500
	  (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008)
	  | 7 lines When starting Asterisk, bug out if Asterisk is already
	  running. (closes issue #12525) Reported by: explidous Patches:
	  20080428__bug12525.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: mvanbaak ........ ................

2008-05-04 02:12 +0000 [r115278-115284]  Joshua Colp <jcolp@digium.com>

	* /, configure, acinclude.m4: Merged revisions 115283 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat,
	  03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2
	  lines Expand the test function for GCC attributes so that more
	  complex attributes are properly recognized. ........
	  ................

	* /, include/asterisk/compiler.h: Merged revisions 115280 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat,
	  03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2
	  lines For my next trick I will make these work with what our
	  autoconf header file gives us. ........ ................

	* /, configure, acinclude.m4: Merged revisions 115277 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat,
	  03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2
	  lines Treat warnings as errors when checking if a GCC attribute
	  exists. We have to do this as GCC will just ignore the attribute
	  and pop up a warning, it won't actually fail to compile. ........
	  ................

2008-05-03 04:25 +0000 [r115269-115275]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /: block voicemail mwi notification subscriptions taskprocessor

	* /: block pbx taskprocessor

	* /: block app_queue taskprocessor

	* /: blocked taskprocessors

2008-05-02 14:55 +0000 [r115198-115200]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500
	  (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May
	  2008) | 6 lines Clarify a comment that was, well, just wrong. It
	  turns out that ignoring the way that macros expand. Instead, I
	  have clarified in the comment why the macro will work even if the
	  scheduler id for the task to be deleted changes during the
	  execution of the macro. ........ ................

2008-05-02 02:57 +0000 [r115107-115160]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, /: Merged revisions 115104 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500
	  (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008)
	  | 2 lines Change the comment of deprecated to an actual compiler
	  deprecation ........ ................

2008-05-01 19:01 +0000 [r115020]  Tilghman Lesher <tlesher@digium.com>

	* /, main/utils.c: Merged revisions 115018 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008)
	  | 14 lines Merged revisions 115017 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008)
	  | 6 lines '#' is another reserved character for URIs that also
	  needs to be escaped. (closes issue #10543) Reported by: blitzrage
	  Patches: 20080418__bug10543.diff.txt uploaded by Corydon76
	  (license 14) ........ ................

2008-05-01 17:28 +0000 [r114932]  Russell Bryant <russell@digium.com>

	* /, UPGRADE.txt: Merged revisions 114931 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 |
	  russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines
	  Clarify the deprecation notice about Macro() to note that it will
	  not be removed for the sake of backwards compatibility, since it
	  is a non-trivial task to convert existing large dialplans that
	  depend on Macro() to use GoSub(), instead. ........

2008-05-01 16:52 +0000 [r114923]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 |
	  qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines
	  Allow dringXrange to properly default to 10, as was done in 1.4.
	  dringXrange is a new feature that was added, and it attempted to
	  default, but only when the option was specified. (closes issue
	  #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded
	  by qwell (license 4) Tested by: bjm ........



2008-04-30 20:20 +0000 [r114909]  Russell Bryant <russell@digium.com>

	* include/asterisk/dlinkedlists.h (added): Add the dlinkedlists
	  implementation from trunk

2008-04-30 20:17 +0000 [r114907-114908]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Make 1.6.0 compile

2008-04-30 17:06 +0000 [r114900]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15
	  lines Merged revisions 114890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7
	  lines Don't crash on bad SIP replys. Fix created in Huntsville
	  together with Mark M (putnopvut) (closes issue #12363) Reported
	  by: jvandal Tested by: putnopvut, oej ........ ................

2008-04-30 16:41 +0000 [r114893]  Russell Bryant <russell@digium.com>

	* /, channels/chan_console.c, channels/chan_iax2.c: Merged
	  revisions 114892 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008)
	  | 36 lines Merged revisions 114891 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008)
	  | 28 lines Merge changes from team/russell/iax2_find_callno and
	  iax2_find_callno_1.4 These changes address a critical performance
	  issue introduced in the latest release. The fix for the latest
	  security issue included a change that made Asterisk randomly
	  choose call numbers to make them more difficult to guess by
	  attackers. However, due to some inefficient (this is by far, an
	  understatement) code, when Asterisk chose high call numbers,
	  chan_iax2 became unusable after just a small number of calls. On
	  a small embedded platform, it would not be able to handle a
	  single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
	  more than about 16 IAX2 channels. Ouch. These changes address
	  some performance issues of the find_callno() function that have
	  bothered me for a very long time. On every incoming media frame,
	  it iterated through every possible call number trying to find a
	  matching active call. This involved a mutex lock and unlock for
	  each call number checked. So, if the random call number chosen
	  was 20000, then every media frame would cause 20000 locks and
	  unlocks. Previously, this problem was not as obvious since
	  Asterisk always chose the lowest call number it could. A second
	  container for IAX2 pvt structs has been added. It is an astobj2
	  hash table. When we know the remote side's call number, the pvt
	  goes into the hash table with a hash value of the remote side's
	  call number. Then, lookups for incoming media frames are a very
	  fast hash lookup instead of an absolutely insane array traversal.
	  In a quick test, I was able to get more than 3600% more IAX2
	  channels on my machine with these changes. ........
	  ................

2008-04-30 16:15 +0000 [r114889]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_console.c: Merged revisions 114888 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008)
	  | 3 lines Fixes a bug where if a stream monitor thread was not
	  created (caused from failure of opening or starting the stream)
	  pthread_cancel was called with an invalid thread ID. ........

2008-04-30 14:55 +0000 [r114877-114886]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500
	  (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr
	  2008) | 2 lines use the ARRAY_LEN macro for indexing through the
	  iaxs/iaxsl arrays so that the size of the arrays can be adjusted
	  in one place, and change the size of the arrays from 32768 calls
	  to 2048 calls when LOW_MEMORY is defined ........
	  ................

	* /, Makefile.rules: Merged revisions 114876 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr
	  2008) | 10 lines Merged revisions 114875 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr
	  2008) | 2 lines pay attention to *all* header files for
	  dependency tracking, not just the local ones (inspired by r578 of
	  asterisk-addons by tilghman) ........ ................

2008-04-29 22:55 +0000 [r114867]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/iax2-provision.c: Merged revisions 114866 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29
	  Apr 2008) | 2 lines Fixes a problem where all the templates were
	  marked as dead no matter what. The templates should only be
	  marked as dead if a configuration file has been successfully
	  loaded and has changes. Bug found while making API documentation
	  for 1.6.0. ........

2008-04-29 21:09 +0000 [r114850-114858]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 114849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr
	  2008) | 22 lines Merged revisions 114848 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr
	  2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel
	  variables instead of the channel's macrocontext and macroexten
	  fields. This is needed because if macros are daisy-chained, the
	  incorrect context and extension are placed on the new channel. I
	  also added locking to the channel prior to accessing these
	  variables as noted in trunk's janitor project file. (closes issue
	  #12549) Reported by: darren1713 Patches:
	  app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
	  (with modifications from me) Tested by: putnopvut ........
	  ................

2008-04-29 19:04 +0000 [r114846]  Kevin P. Fleming <kpfleming@digium.com>

	* /: bug is not present in this branch

2008-04-29 17:11 +0000 [r114831]  Jason Parker <jparker@digium.com>

	* res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114830 | qwell | 2008-04-29 12:10:55 -0500
	  (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr
	  2008) | 1 line Change warning message to debug, since there are
	  cases where 0 results is perfectly fine. ........
	  ................

2008-04-29 12:55 +0000 [r114825]  Kevin P. Fleming <kpfleming@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500
	  (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500
	  (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
	  2008) | 2 lines stop script from appending source code if run
	  multiple times ........ ................ ................

2008-04-28 17:04 +0000 [r114777]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 |
	  mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix
	  deadlock issue in chan_zap with libss7 due to channel variables
	  being set with the channel pvt lock being held. #12512 ........

2008-04-28 13:44 +0000 [r114714]  Joshua Colp <jcolp@digium.com>

	* /, configure, configure.ac: Merged revisions 114713 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2
	  lines Update autoconf logic with latest API change for libss7.
	  ........

2008-04-28 04:54 +0000 [r114707-114710]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 114709 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008)
	  | 13 lines Merged revisions 114708 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008)
	  | 5 lines When modules are embedded, they take on a different
	  name, without the ".so" extension. Specifically check for this
	  name, when we're checking if a module is loaded. (Closes issue
	  #12534) ........ ................

2008-04-27 15:20 +0000 [r114701]  Michiel van Baak <michiel@vanbaak.info>

	* /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6
	  because it fixes a crash. ........ r114700 | mvanbaak |
	  2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI
	  in chan_skinny event based modeled after chan_zap and chan_mgcp.
	  (closes issue #12214) Reported by: DEA Patches:
	  chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested
	  by: DEA and me ........

2008-04-27 01:30 +0000 [r114697]  Sean Bright <sean.bright@gmail.com>

	* /, configure, configure.ac: Merged revisions 114696 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400
	  (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr
	  2008) | 5 lines When we don't explicitly pass a path to the
	  --with-tds configure option, we may end up finding tds.h in
	  /usr/local/include instead of /usr/include. If this happens, the
	  grep that looks for the version (from tdsver.h) will fail and
	  we'll have some problems during the build. ........
	  ................

2008-04-26 15:09 +0000 [r114684-114693]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/vmail.cgi: Merged revisions 114690 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500
	  (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008)
	  | 6 lines Clicking forward without selecting a message leaves an
	  errant .lock file. (closes issue #12528) Reported by: pukepail
	  Patches: patch.diff uploaded by pukepail (license 431) ........
	  ................

2008-04-25 22:05 +0000 [r114671-114677]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 |
	  russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines
	  Lock the channel around datastore access (closes issue #12527)
	  Reported by: mnicholson Patches: pbx_lua4.diff uploaded by
	  mnicholson (license 96) ........

	* /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114674 | russell | 2008-04-25 17:00:35 -0500
	  (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008)
	  | 3 lines Use consistent logic for checking to see if a call
	  number has been chosen yet. Also, remove some redundant logic I
	  recently added in a fix. ........ ................

2008-04-25 19:34 +0000 [r114664]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr
	  2008) | 12 lines Merged revisions 114662 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr
	  2008) | 4 lines Move the unlock of the spyee channel to outside
	  the start_spying() function so that the channel is not unlocked
	  twice when using whisper mode. ........ ................

2008-04-25 16:26 +0000 [r114652]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr
	  2008) | 4 lines Fix a memory leak and protect against potential
	  dereferences of a NULL pointer. ........

2008-04-24 22:14 +0000 [r114636]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 |
	  file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey
	  look, it builds. (closes issue #12519) Reported by: falves11
	  ........

2008-04-24 21:36 +0000 [r114626-114634]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr
	  2008) | 19 lines Merged revisions 114632 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr
	  2008) | 11 lines Re-invite RTP during a masquerade so that, for
	  instance, an AMI redirect of two channels which are natively
	  bridged will preserve audio on both channels. This prevents a
	  problem with Asterisk not re-inviting due to one of the channels
	  having being a zombie. (closes issue #12513) Reported by:
	  mneuhauser Patches:
	  asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
	  mneuhauser (license 425) ........ ................

	* /, apps/app_queue.c: Merged revisions 114629 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr
	  2008) | 16 lines Merged revisions 114628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr
	  2008) | 8 lines Output of channel variables when
	  eventwhencalled=vars was set was being truncated two characters.
	  This patch corrects the problem. (closes issue #12493) Reported
	  by: davidw ........ ................

	* channels/chan_local.c, /: Merged revisions 114625 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500
	  (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr
	  2008) | 10 lines Resolve a deadlock in chan_local by releasing
	  the channel lock temporarily. (closes issue #11712) Reported by:
	  callguy Patches: 11712.patch uploaded by putnopvut (license 60)
	  Tested by: acunningham ........ ................

2008-04-24 19:55 +0000 [r114619-114623]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 114622 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500
	  (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008)
	  | 4 lines Ensure that when we set the accountcode, it actually
	  shows up in the CDR. (Fix for AMI Originate) (Closes issue
	  #12007) ........ ................

	* /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 |
	  tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines
	  Fix DST calculation, and fix bug in calculation of whether conf
	  has started yet or not (Closes issue #12292) Reported by: DEA
	  Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license
	  3) ........

2008-04-24 16:48 +0000 [r114613]  Jason Parker <jparker@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114612 | qwell | 2008-04-24 11:47:01 -0500
	  (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #12496) Reported by: daniele Patches:
	  misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
	  Tested by: daniele Technically, I didn't use the patch above
	  except to find out what revision to merge - but it's the same
	  thing as this revision. ........ r51989 | crichter | 2007-01-24
	  06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899
	  ........ ................

2008-04-24 15:57 +0000 [r114610]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114609 | russell | 2008-04-24 10:56:55 -0500
	  (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008)
	  | 4 lines Fix a silly mistake in a change I made yesterday that
	  caused chan_iax2 to blow up very quickly. (issue #12515) ........
	  ................

2008-04-24 15:00 +0000 [r114607]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Merged revisions 114606 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11
	  lines Merged revisions 114603 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3
	  lines Only have one max-forwards header in outbound REFERs.
	  Discovered in the Asterisk SIP Masterclass in Orlando. Thanks
	  Joe! ........ ................

2008-04-24 14:56 +0000 [r114599-114605]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 |
	  russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines
	  Change a verbose message to debug. (closes issue #12514) ........

	* /, main/http.c: Merged revisions 114601 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008)
	  | 14 lines Merged revisions 114600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008)
	  | 6 lines Improve some broken cookie parsing code. Previously,
	  manager login over HTTP would only work if the mansession_id
	  cookie was first. Now, the code builds a list of all of the
	  cookies in the Cookie header. This fixes a problem observed by
	  users of the Asterisk GUI. (closes AST-20) ........
	  ................

	* apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008)
	  | 18 lines Merged revisions 114597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008)
	  | 10 lines Fix an issue that caused getting the correct next
	  channel to not always work. Also, remove setting the amount of
	  time to wait for a digit from 5 seconds back down to 1/10 of a
	  second. I believe this was so the beep didn't get played over and
	  over really fast, but a while back I put in another fix for that
	  issue. (closes issue #12498) Reported by: jsmith Patches:
	  app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license
	  15) ........ ................

2008-04-23 18:34 +0000 [r114596]  Jason Parker <jparker@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114595 | qwell | 2008-04-23 13:33:28 -0500
	  (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) |
	  8 lines Fix reload/unload for res_musiconhold module. (closes
	  issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff
	  uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded
	  by jamesgolovich (license 176) ........ ................

2008-04-23 18:01 +0000 [r114589-114593]  Russell Bryant <russell@digium.com>

	* main/manager.c, /, include/asterisk/manager.h: Merged revisions
	  114592 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008)
	  | 13 lines Merged revisions 114591 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008)
	  | 5 lines Store the manager session ID explicitly as 4 byte ID
	  instead of a ulong. The mansession_id cookie is coded to be
	  limited to 8 characters of hex, and this could break logins from
	  64-bit machines in some cases. (inspired by AST-20) ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114588 | russell | 2008-04-23 12:18:29 -0500
	  (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008)
	  | 2 lines Fix find_callno_locked() to actually return the callno
	  locked in some more cases. ........ ................

2008-04-23 16:57 +0000 [r114586]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Merged revisions 114585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10
	  lines Merged revisions 114584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2
	  lines Add 502 support for both directions, not only one... (see
	  r114571) ........ ................

2008-04-23 14:56 +0000 [r114581]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, /: Merged revisions 114580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) |
	  12 lines Merged revisions 114579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4
	  lines Instead of stopping dialplan execution when SayNumber
	  attempts to say a large number that it can not print out a
	  message informing the user and continue on. (closes issue #12502)
	  Reported by: bcnit ........ ................

2008-04-23 01:00 +0000 [r114576-114578]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr
	  2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail
	  changes This makes IMAP_STORAGE include the proper headers if you
	  have specified the "system" option for --with-imap when running
	  the configure script and your IMAP-related headers exist in
	  /usr/include/c-client. This change is due to a hasty merge of a
	  1.4 change I made. ........

2008-04-22 23:59 +0000 [r114573]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008)
	  | 10 lines Merged revisions 114571 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008)
	  | 2 lines Treat a 502 just like a 503, when it comes to
	  processing a response code ........ ................

2008-04-22  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta8 released.

2008-04-22 22:18 +0000 [r114560]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114559 | russell | 2008-04-22 17:17:31 -0500
	  (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008)
	  | 5 lines When we receive a full frame that is supposed to
	  contain our call number, ensure that it has the correct one.
	  (closes issue #10078) (AST-2008-006) ........ ................

2008-04-22 22:04 +0000 [r114556]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 114553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 |
	  murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines
	  (closes issue #12469) Reported by: triccyx I had a bit a problem
	  reproducing this in my setup (trying not to disturb my other
	  stuff) but finally, I got it. The problem appears to be that the
	  extension is being added in replace mode, which kinda assumes
	  that the pattern trie has been formed, when in fact, in this
	  case, it was not. The checks being done are not nec. when the
	  tree is not yet formed, as changes like this will be summarized
	  when the trie is formed in the future. I tested the fix, and the
	  crash no longer happens. Feel free to open the bug again if this
	  fix doesn't cure the problem. ........

2008-04-22 21:16 +0000 [r114544-114552]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 114548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 |
	  russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines
	  re-add a fix that got lost with a recent change ........

2008-04-22 18:14 +0000 [r114541]  Jason Parker <jparker@digium.com>

	* main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged
	  revisions 114540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 |
	  qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines
	  Allow setqueuevar=yes (et al) to work, after changes to
	  pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit
	  Patches: 12490-queuevars-3.diff uploaded by qwell (license 4)
	  Tested by: qwell ........

2008-04-22 18:06 +0000 [r114534-114539]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114538 | russell | 2008-04-22 13:04:39 -0500
	  (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008)
	  | 9 lines If the dial string passed to the call channel callback
	  does not indicate an extension, then consider the extension on
	  the channel before falling back to the default. (closes issue
	  #12479) Reported by: darren1713 Patches:
	  exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
	  116) ........ ................

2008-04-22 15:46 +0000 [r114524-114528]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 114527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 |
	  russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines
	  Correct action_ping() and action_events() with regards to Manager
	  1.1 documentation. Also, fix a bug in xml_translate(). (closes
	  issue #11649) Reported by: ys Patches: trunk_manager.c.diff
	  uploaded by ys (license 281) ........

2008-04-21 20:23 +0000 [r114422]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 |
	  mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add
	  support for generic name transmission (#12484) on SS7 in chan_zap
	  ........

2008-04-21 15:38 +0000 [r114328]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008)
	  | 2 lines This removes an invalid warning message for an
	  incorrectly entered pin, but more importantly removes an
	  inapplicable check. If the first argument passed to
	  app_authenticate does not contain a '/', the argument should be
	  treated as the sole fixed "password" to match against and that is
	  all. (Previous behavior was attempting to open a file based on
	  the pin.) ........

2008-04-21 14:42 +0000 [r114321-114324]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) |
	  12 lines Merged revisions 114322 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4
	  lines Only drop audio if we receive it without a progress
	  indication. We allow other frames through such as DTMF because
	  they may be needed to complete the call. (closes issue #12440)
	  Reported by: aragon ........ ................

	* /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6
	  lines Only print out the error message if ldap_modify_ext_s
	  actually returns an error, and not success. (closes issue #12438)
	  Reported by: gservat Patches: res_config_ldap.c-patch-code
	  uploaded by gservat (license 466) ........

2008-04-19 17:00 +0000 [r114304]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: SS7:Added - Generic Name / Access Transport
	  / Redirecting Number handling. #12425

2008-04-18 21:51 +0000 [r114277-114286]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 114285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008)
	  | 10 lines Merged revisions 114284 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008)
	  | 2 lines Don't destroy a manager session if poll() returns an
	  error of EAGAIN. ........ ................

	* Makefile, /: Merged revisions 114279 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008)
	  | 10 lines Merged revisions 114278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008)
	  | 2 lines ensure directories are created before we try to install
	  stuff into them ........ ................

	* Makefile, /: Merged revisions 114276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008)
	  | 10 lines Merged revisions 114275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008)
	  | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for
	  bininstall ........ ................

2008-04-18 19:36 +0000 [r114262-114272]  Joshua Colp <jcolp@digium.com>

	* channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4
	  lines Make sure ADSI is marked as unavailable on Unistim channels
	  so voicemail does not try to do some ADSI jazz. (closes issue
	  #12460) Reported by: PerryB ........

2008-04-18 18:04 +0000 [r114260]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500
	  (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr
	  2008) | 6 lines Clearing up error messages so they make a bit
	  more sense. Also removing a redundant error message. Issue AST-15
	  ........ ................

2008-04-18 16:12 +0000 [r114255]  Joshua Colp <jcolp@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4
	  lines If the parsing of the config file fails make sure we unlock
	  ldap_lock. (closes issue #12477) Reported by: IgorG ........

2008-04-18 13:40 +0000 [r114247]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: Merged revisions 114246 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr
	  2008) | 9 lines Merged revisions 114245 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr
	  2008) | 1 line Only complete the SIP channel name once for 'sip
	  show channel <channel>' ........ ................

2008-04-18 06:54 +0000 [r114244]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500
	  (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008)
	  | 3 lines For consistency sake, ensure that the values that
	  ${CALLINGPRES} returns are valid as an input to SetCallingPres.
	  (Closes issue #12472) ........ ................

2008-04-17 23:09 +0000 [r114232-114241]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10
	  lines Merged revisions 114148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2
	  lines Handle subscribe queues in all situations... Thanks to
	  festr_ on irc for telling me about this bug. ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 |
	  oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines
	  Adding chanvar to SIPPEER from 1.4 branch ........

	* main/autoservice.c, /: Merged revisions 114233 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008)
	  | 14 lines Merged revisions 114230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008)
	  | 6 lines Remove redundant safety net. The check for the
	  autoservice channel list state accomplishes the same goal in a
	  better way. (issue #12470) Reported By: atis ........
	  ................

2008-04-17 21:05 +0000 [r114228]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr
	  2008) | 17 lines Merged revisions 114226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr
	  2008) | 9 lines Declaration of the peer channel in this scope was
	  making it so the peer variable defined in the outer scope was
	  never set properly, therefore making iterating through the
	  channel list always restart from the beginning. This bug would
	  have affected anyone who called chanspy without specifying a
	  first argument. (closes issue #12461) Reported by: stever28
	  ........ ................

2008-04-17 16:51 +0000 [r114210-114213]  Mark Michelson <mmichelson@digium.com>

	* main/dsp.c, main/frame.c, /, include/asterisk/dsp.h,
	  include/asterisk/frame.h: Merged revisions 114208 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500
	  (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr
	  2008) | 12 lines It was possible for a reference to a frame which
	  was part of a freed DSP to still be referenced, leading to memory
	  corruption and eventual crashes. This code change ensures that
	  the dsp is freed when we are finished with the frame. This change
	  is very similar to a change Russell made with translators back a
	  month or so ago. (closes issue #11999) Reported by: destiny6628
	  Patches: 11999.patch uploaded by putnopvut (license 60) Tested
	  by: destiny6628, victoryure ........ ................

2008-04-17 16:26 +0000 [r114206]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 114205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008)
	  | 11 lines Merged revisions 114204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008)
	  | 3 lines Fix the bininstall target to install from subdirs, as
	  well. (closes issue AST-8, patch from bmd at switchvox) ........
	  ................

2008-04-17 15:17 +0000 [r114203]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008)
	  | 2 lines fileio.h does not exist; io.h does, though. ........

2008-04-17 13:55 +0000 [r114200]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 114199 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008)
	  | 10 lines Merged revisions 114198 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008)
	  | 2 lines Use keepalives effectively in order diagnose bug
	  #12432. ........ ................

2008-04-17 12:59 +0000 [r114197]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 114196 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008)
	  | 16 lines Merged revisions 114195 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008)
	  | 8 lines Add special case for when the agi cannot be executed,
	  to comply with the documentation that we return failure in that
	  case. (closes issue #12462) Reported by: fmueller Patches:
	  20080416__bug12462.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: fmueller ........ ................

2008-04-17 10:56 +0000 [r114193]  Sean Bright <sean.bright@gmail.com>

	* apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr
	  2008) | 9 lines Merged revisions 114191 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr
	  2008) | 1 line Make sure we have enough room for the recording's
	  filename. ........ ................

2008-04-16 20:48 +0000 [r114186]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr
	  2008) | 14 lines Merged revisions 114184 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr
	  2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by
	  initializing the structure to all zeroes in case it contains
	  fields that we don't write values into (which it does as of
	  Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
	  ........ ................

2008-04-15 20:53 +0000 [r114153]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 |
	  tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines
	  Oops, buffer wasn't long enough for query ........

2008-04-15 20:09 +0000 [r114147]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 114146 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 |
	  murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines
	  These changes: a. fix a self-found problem with SPAWN-ing an
	  extension, where matches were not being found b. correct some
	  wording in a comment c. Add some debug for future debugging.
	  ........

2008-04-15 17:22 +0000 [r114132-114142]  Jason Parker <jparker@digium.com>

	* channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) |
	  8 lines Shorten the mac address pattern, since some phones use
	  different identifiers (such as the i2050 softphone). (closes
	  issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff
	  uploaded by c (license 460) Tested by: c_hans ........

	* contrib/scripts/autosupport, /: Merged revisions 114139 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114139 | qwell | 2008-04-15 12:17:37 -0500
	  (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) |
	  7 lines Update Digium autosupport script, for more useful
	  information. (closes issue #12452) Reported by: angler Patches:
	  autosupport.diff uploaded by angler (license 106) ........
	  ................

	* /, apps/app_queue.c: Merged revisions 114134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) |
	  16 lines Merged revisions 114133 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) |
	  8 lines Allow autofill to work in the general section of
	  queues.conf. Additionally, don't try to (re)set options when they
	  have empty values in realtime (all unset columns would have an
	  empty value). (closes issue #12445) Reported by: atis Patches:
	  12445-autofill.diff uploaded by qwell (license 4) ........
	  ................

2008-04-14 18:34 +0000 [r114122]  Jason Parker <jparker@digium.com>

	* /, channels/chan_h323.c: Merged revisions 114121 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114121 | qwell | 2008-04-14 13:34:17 -0500
	  (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) |
	  7 lines The call_token on the pvt can occasionally be NULL,
	  causing a crash. If it is NULL, we can skip this channel, since
	  it can't the one we're looking for. (closes issue #9299) Reported
	  by: vazir ........ ................

2008-04-14 17:42 +0000 [r114119]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 114118 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr
	  2008) | 19 lines Merged revisions 114117 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr
	  2008) | 11 lines Increase the retry count when attempting to show
	  channels. This apparently cleared an issue someone was seeing
	  when attempting to show channels when the load was high. (closes
	  issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
	  by russell (license 2) Tested by: falves11 ........
	  ................

2008-04-14 16:33 +0000 [r114116]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008)
	  | 2 lines Make tab-completion work for all cases ........

2008-04-14 16:25 +0000 [r114114]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500
	  (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr
	  2008) | 9 lines If the datastore has been moved to another
	  channel due to a masquerade, then freeing the datastore here
	  causes an eventual double free when the new channel hangs up. We
	  should only free the datastore if we were able to successfully
	  remove it from the channel we are referencing (i.e. the datastore
	  was not moved). (closes issue #12359) Reported by: pguido
	  ........ ................

2008-04-14 15:02 +0000 [r114108]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 114107 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr
	  2008) | 13 lines Merged revisions 114106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr
	  2008) | 5 lines Save a local copy of the generate callback prior
	  to unlocking the channel in case the generate callback goes NULL
	  on us after the channel is unlocked. Thanks to Russell for
	  pointing this need out to me. ........ ................

2008-04-14 14:54 +0000 [r114102-114105]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) |
	  12 lines Merged revisions 114103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4
	  lines It is possible for the remote side to say they want T38 but
	  not give any capabilities. (closes issue #12414) Reported by: MVF
	  ........ ................

	* main/rtp.c, /: Merged revisions 114101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) |
	  12 lines Merged revisions 114100 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4
	  lines Don't change the SSRC when a new source comes into play,
	  this might happen quite often and depending on the remote side...
	  they might not like this. (closes issue #12353) Reported by:
	  dimas ........ ................

2008-04-14 02:59 +0000 [r114097-114099]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008)
	  | 3 lines Add tab command-line completion (Closes issue #12428)
	  ........

	* /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 |
	  tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines
	  Use ast_mkdir instead of mkdir (Closes issue #12430) ........

2008-04-12 16:22 +0000 [r114094-114095]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure linkset is locked exiting
	  ss7_start_call

	* channels/chan_zap.c: Make sure we start incoming calls on SS7
	  with echo cancellation enabled. Also make sure when completing a
	  COT we call ss7_start_call with the proper locks held. Lastly,
	  make sure if we fail to get a channel from zt_new that we don't
	  assume it's there.

2008-04-11 23:27 +0000 [r114089-114091]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 |
	  tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines
	  If any field is not null, but has no default, then it must be set
	  or the insert will fail. (Closes issue #12285) ........

	* /, configs/res_ldap.conf.sample: Merged revisions 114088 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11
	  Apr 2008) | 3 lines Make the sample config match the contributed
	  LDAP schema (Closes issue #12421) ........

2008-04-11 23:21 +0000 [r114087]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r114084 | twilson | 2008-04-11 17:48:52 -0500
	  (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008)
	  | 7 lines Several places in the code called find_callno() (which
	  releases the lock on the pvt structure) and then immediately
	  locked the call and did things with it. Unfortunately, the call
	  can disappear between the find_callno and the lock, causing Bad
	  Stuff(tm) to happen. Added find_callno_locked() function to
	  return the callno withtout unlocking for instances that it is
	  needed. (issue #12400) Reported by: ztel ........
	  ................

2008-04-11 23:13 +0000 [r114086]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008)
	  | 7 lines Use the correct function for free'ing objects, and
	  maybe we won't crash. (closes issue #12163) Reported by: gservat
	  Patches: 20080411__bug12163.diff.txt uploaded by Corydon76
	  (license 14) Tested by: gservat ........

2008-04-11 15:51 +0000 [r114065]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: Merged revisions 114064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr
	  2008) | 19 lines Merged revisions 114063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr
	  2008) | 11 lines Fix a race condition that may happen between a
	  sip hangup and a "core show channel" command. This patch adds
	  locking to prevent the resulting crash. (closes issue #12155)
	  Reported by: tsearle Patches: show_channels_crash2.patch uploaded
	  by tsearle (license 373) Tested by: tsearle ........
	  ................

2008-04-11 14:56 +0000 [r114062]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008)
	  | 6 lines Errors are all greater than 0 (closes issue #12422)
	  Reported by: nito Patches:
	  res_config_ldap_result_check_patch.diff uploaded by nito (license
	  340) ........

2008-04-10 22:23 +0000 [r114056]  Mark Michelson <mmichelson@digium.com>

	* utils/conf2ael.c, utils/check_expr.c, utils/Makefile,
	  main/manager.c, /, utils/astman.c, utils/hashtest.c,
	  main/utils.c, include/asterisk/lock.h, utils/ael_main.c,
	  utils/hashtest2.c: Merged revisions 114052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr
	  2008) | 11 lines Merged revisions 114051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr
	  2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled.
	  ........ ................

2008-04-10 19:59 +0000 [r114047]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr
	  2008) | 14 lines Merged revisions 114045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr
	  2008) | 6 lines Be sure that we're not about to set bridgepvt
	  NULL prior to dereferencing it. (closes issue #11775) Reported
	  by: fujin ........ ................

2008-04-10 19:09 +0000 [r114043]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008)
	  | 7 lines The hydra grows yet another head... (closes issue
	  #12401) Reported by: davevg Patches: astcli.diff2 uploaded by
	  davevg (license 209) Tested by: davevg, Corydon76 ........

2008-04-10 17:27 +0000 [r114037]  Jason Parker <jparker@digium.com>

	* /, main/file.c: Merged revisions 114036 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) |
	  18 lines Merged revisions 114035 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) |
	  10 lines Only try to prefix language if we are not using an
	  absolute path (suffix it otherwise).
	  en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
	  issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
	  uploaded by qwell (license 4) Tested by: kuj, qwell ........
	  ................

2008-04-10 16:00 +0000 [r114023-114034]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) |
	  14 lines Merged revisions 114029 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6
	  lines Create the directory where name recordings will go if it
	  does not exist. (closes issue #12311) Reported by: rkeene
	  Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........
	  ................

	* apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6
	  lines Don't hardcode ru into the digits filename so that
	  languageprefix can work. (closes issue #12404) Reported by: IgorG
	  Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG
	  (license 20) ........

	* main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c:
	  Merged revisions 114024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 |
	  file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix
	  spelling of existent in a few places. (closes issue #12409)
	  Reported by: candlerb ........

	* /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) |
	  14 lines Merged revisions 114021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6
	  lines Don't add custom URI options if they don't exist OR they
	  are empty. (closes issue #12407) Reported by: homesick Patches:
	  uri_options-1.4.diff uploaded by homesick (license 91) ........
	  ................

2008-04-09 22:34 +0000 [r113929-113982]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 113980 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 |
	  mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8
	  lines Fix a crash that happened due to accessing free'd memory
	  (closes issue #12396) Reported by: tcalosi Patches: 12396.patch
	  uploaded by putnopvut (license 60) Tested by: tcalosi ........

	* /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr
	  2008) | 16 lines Merged revisions 113927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr
	  2008) | 8 lines We need to set the persistant_route [sic]
	  parameter for the sip_pvt during the initial INVITE, no matter if
	  we're building the route set from an INVITE request or response.
	  (closes issue #12391) Reported by: benjaminbohlmann Tested by:
	  benjaminbohlmann ........ ................

2008-04-09 19:02 +0000 [r113876]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions
	  113875 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008)
	  | 12 lines Merged revisions 113874 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008)
	  | 4 lines If the [csv] section does not exist in cdr.conf, then
	  an unload/load sequence is needed to correct the problem. Track
	  whether the load succeeded with a variable, so we can fix this
	  with a simple reload event, instead. ........ ................

2008-04-09 17:56 +0000 [r113839]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) |
	  2 lines Fix a small file handle "leak" pointed out by jjshoe on
	  #asterisk. ........

2008-04-09 17:50 +0000 [r113837]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 113836 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 |
	  mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14
	  lines There was a subtle logical difference between 1.4 and trunk
	  with regards to how timeouts were handled. In 1.4, if the
	  absolute timeout were reached on a call, no matter what the
	  return value of ast_spawn_extension was, the pbx would attempt to
	  go to the 'T' extension or hangup otherwise. The rearrangement of
	  this function in trunk made this check only happen in the case
	  that ast_spawn_extension returned 0. If ast_spawn_extension
	  returned 1, then the fact that the timeout expired resulted in a
	  no-op, and would cause an infinite loop to occur in
	  __ast_pbx_run. This change fixes this problem. Now timeouts will
	  behave as they did in 1.4 (closes issue #11550) Reported by: pj
	  Tested by: putnopvut ........

2008-04-09 16:53 +0000 [r113786]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed,
	  09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4
	  lines If we receive an AUTHREQ from the remote server and we are
	  unable to reply (for example they have a secret configured, but
	  we do not) then queue a hangup frame on the Asterisk channel.
	  This will cause the channel to hangup and a HANGUP to be sent via
	  IAX2 to the remote side which is the proper thing to do in this
	  scenario. (closes issue #12385) Reported by: viraptor ........
	  ................

2008-04-09 14:42 +0000 [r113683]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr
	  2008) | 17 lines Merged revisions 113681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr
	  2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a
	  reinvite), then we should not send a BYE. (closes issue #12392)
	  Reported by: fnordian Patches: chan_sip.patch uploaded by
	  fnordian (license 110) with small modification from me ........
	  ................

2008-04-09 13:56 +0000 [r113648-113650]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008)
	  | 6 lines Additional enhancements (closes issue #12390) Reported
	  by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir
	  (license 46) ........

2008-04-09 01:40 +0000 [r113598]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113597 | twilson | 2008-04-08 20:36:58 -0500
	  (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008)
	  | 2 lines Initialize fr->cacheable to make valgrind happy
	  ........ ................

2008-04-08 21:34 +0000 [r113560]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astcli (added): Merged revisions 113559 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08
	  Apr 2008) | 6 lines Add commandline tool for doing CLI commands
	  through AMI (instead of using asterisk -rx) (closes issue #12389)
	  Reported by: davevg Patches: astcli uploaded by davevg (license
	  209) ........

2008-04-08 18:49 +0000 [r113404-113506]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113505 | qwell | 2008-04-08 13:49:21 -0500
	  (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr
	  2008) | 1 line Add a little more that is required for previously
	  added devices. ........ ................

	* /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113455 | qwell | 2008-04-08 13:08:35 -0500
	  (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) |
	  4 lines Add support for several new(ish) devices - most notably,
	  7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing
	  me the required information. ........ ................

	* main/asterisk.c, /: Merged revisions 113403 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) |
	  9 lines Merged revisions 113402 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) |
	  1 line Work around some silliness caused by sys/capability.h -
	  this should fix compile errors a number of users have been
	  experiencing. ........ ................

2008-04-08 16:56 +0000 [r113350-113401]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astgenkey.8: Merged revisions 113400 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500
	  (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008)
	  | 6 lines Add security note on astgenkey's manpage. (closes issue
	  #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt
	  uploaded by Corydon76 (license 14) ........ ................

	* /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008)
	  | 15 lines Merged revisions 113348 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008)
	  | 7 lines Move check for still-bridged channels out a little
	  further, to avoid possible deadlocks. (Closes issue #12252)
	  Reported by: callguy Patches: 20080319__bug12252.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: callguy ........
	  ................

2008-04-08 15:10 +0000 [r113298-113299]  Joshua Colp <jcolp@digium.com>

	* /, main/audiohook.c: Merged revisions 113297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) |
	  12 lines Merged revisions 113296 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4
	  lines If audio suddenly gets fed into one side of a channel after
	  a lapse of frames flush the other factory so that old audio does
	  not remain in the factory causing the sync code to not execute.
	  (closes issue #12296) Reported by: jvandal ........
	  ................

2008-04-07 22:17 +0000 [r113246]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/manager.conf.sample: Merged revisions 113245 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07
	  Apr 2008) | 2 lines Additional note ........

2008-04-07 21:49 +0000 [r113244]  Jason Parker <jparker@digium.com>

	* /, configs/manager.conf.sample: Merged revisions 113243 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr
	  2008) | 1 line Document 'originate' permission in manager sample
	  config. ........

2008-04-07 21:36 +0000 [r113242]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008)
	  | 23 lines Merged revisions 113013 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
	  | 15 lines Merged revisions 113012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
	  | 7 lines (closes issue #12362) (closes issue #12372) Reported
	  by: vinsik Tested by: tecnoxarxa This one line change makes an if
	  inside a for loop (in realtime_peer) check all the ast_variables
	  the loop was intending to test rather than just the first one.
	  ........ ................ ................

2008-04-07 19:10 +0000 [r113174]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged
	  revisions 113119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) |
	  16 lines Merged revisions 113118 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) |
	  8 lines Allow playback with noanswer (and add earlyrtp option).
	  (closes issue #9077) Reported by: pj Patches: earlyrtp.diff
	  uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA,
	  wedhorn ........ ................

2008-04-07 19:08 +0000 [r113173]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 113172 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500
	  (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008)
	  | 3 lines Force ast_mktime() to check for DST, since strptime(3)
	  does not. (Closes issue #12374) ........ ................

2008-04-07 16:13 +0000 [r113067]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 113066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr
	  2008) | 21 lines Merged revisions 113065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr
	  2008) | 13 lines This fix prevents a deadlock that was
	  experienced in chan_local. There was deadlock prevention in place
	  in chan_local, but it would not work in a specific case because
	  the channel was recursively locked. By unlocking the channel
	  prior to calling the generator's generate callback in
	  ast_read_generator_actions(), we prevent the recursive locking,
	  and therefore the deadlock. (closes issue #12307) Reported by:
	  callguy Patches: 12307.patch uploaded by putnopvut (license 60)
	  Tested by: callguy ........ ................

2008-04-07 15:28 +0000 [r113042]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
	  | 15 lines Merged revisions 113012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
	  | 7 lines (closes issue #12362) (closes issue #12372) Reported
	  by: vinsik Tested by: tecnoxarxa This one line change makes an if
	  inside a for loop (in realtime_peer) check all the ast_variables
	  the loop was intending to test rather than just the first one.
	  ........ ................

2008-04-05 13:30 +0000 [r112973-112975]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 112972 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 |
	  tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines
	  AsyncAGI should not close the manager session on error. (closes
	  issue #12370) Reported by: srt Patches: asterisk-12370.diff
	  uploaded by srt (license 378) ........

2008-04-04 19:30 +0000 [r112786-112822]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200
	  (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04
	  Apr 2008) | 1 line Free newly allocated channel before returning
	  ........ ................

	* /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200
	  (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008)
	  | 7 lines Prevent call connections when codecs don't match.
	  (closes issue #10604) Reported by: keepitcool Patches:
	  branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
	  by: phsultan ........ ................

2008-04-04 01:08 +0000 [r112715]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/asterisk.c, /: Merged revisions 112653,112656,112714 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03
	  Apr 2008) | 1 line add a Zaptel timer check to verify the timer
	  is responding when Zaptel support is compiled into Asterisk and
	  Zaptel drivers are loaded. This will help people not waste their
	  valuable time debugging side effects. ........ r112656 | dhubbard
	  | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy
	  buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500
	  (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel
	  timer error message to display before exit ........

2008-04-04 00:54 +0000 [r112713]  Joshua Colp <jcolp@digium.com>

	* /, main/Makefile: Merged revisions 112712 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) |
	  10 lines Merged revisions 112711 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2
	  lines Pass in the path to Zaptel for systems that install Zaptel
	  headers in a separate location. ........ ................

2008-04-03 14:42 +0000 [r112601]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr
	  2008) | 17 lines Merged revisions 112599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr
	  2008) | 9 lines Fix the testing of the "res" variable so that it
	  is more logically correct and makes the correct warning and debug
	  messages print. (closes issue #12361) Reported by: one47 Patches:
	  chan_zap_deferred_digit.patch uploaded by one47 (license 23)
	  ........ ................

2008-04-02 17:37 +0000 [r112470]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 112469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr
	  2008) | 21 lines Merged revisions 112468 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr
	  2008) | 13 lines Fix a race condition in the manager. It is
	  possible that a new manager event could be appended during a
	  brief time when the manager is not waiting for input. If an event
	  comes during this period, we need to set an indicator that there
	  is an event pending so that the manager doesn't attempt to wait
	  forever for an event that already happened. (closes issue #12354)
	  Reported by: bamby Patches: manager_race_condition.diff uploaded
	  by bamby (license 430) (comments added by me) ........
	  ................

2008-04-02 15:27 +0000 [r112436]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 |
	  file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines
	  Since the SIP request structure gets reused multiple times with
	  TCP handling we have to clear the debug state or else we will
	  keep spitting out debug even after it has been turned off.
	  (closes issue #12169) Reported by: pj Patches:
	  12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj
	  ........

2008-04-02 14:33 +0000 [r112395]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 112394 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr
	  2008) | 14 lines Merged revisions 112393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr
	  2008) | 6 lines Ensure that there is no timeout if none is
	  specified. (closes issue #12349) Reported by: johnlange ........
	  ................

2008-04-01 22:48 +0000 [r112359]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 112357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 |
	  murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line
	  Bumped across another test set for the new exten pattern matcher,
	  which revealed a problem with the CANMATCH/MATCHMORE modes.
	  Direct matches were getting in the way. Fixed. ........

2008-04-01 20:20 +0000 [r112299]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 112289 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 |
	  murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines
	  (closes issue #12298) Reported by: falves11 Patches: 12298.patch1
	  uploaded by murf (license 17) Tested by: murf I have hopes that
	  the changes made over the last few days will finalize and
	  solidify this code. While there are bound to be small tweaks
	  still needed, I feel that the job (at last) is somewhat
	  completed. Finally, I had a chance to comprehend how the scoring
	  of extension patterns was done in the previous version, and I've
	  come very close to using the exact same criteria in the new
	  pattern matching code. The left-right sorting is now replicated
	  in the trie structure itself, such that the first match found
	  will the 'best' match. Compared the results against 1.4 for
	  several extensions. Replicated falves11's setup and it works.
	  Used some devious patterns provided by jsmith, supplemented with
	  a few of my own. Looks good. ........

2008-04-01 18:09 +0000 [r112211]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 112210 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) |
	  12 lines Merged revisions 112209 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4
	  lines Disable Packet2Packet bridging when we need to feed DTMF
	  frames into the core. Some implementations do not like how we
	  switch between things. (closes issue #12212) Reported by: bamby
	  ........ ................

2008-04-01 17:52 +0000 [r112170-112206]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) |
	  12 lines Merged revisions 112204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4
	  lines Do not pass audio until the remote side has indicated they
	  are providing early media, or if the channel has been answered.
	  (closes issue #11823) Reported by: SDamm ........
	  ................

2008-04-01 17:25 +0000 [r112157]  Mark Michelson <mmichelson@digium.com>

	* main/dns.c, /: Merged revisions 112148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr
	  2008) | 18 lines Merged revisions 112138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr
	  2008) | 10 lines Initialize the __res_state structure used for
	  dns purposes to all 0's prior to using it. This is due to
	  valgrind's complaints on issue #12284 as well as an excerpt found
	  in "Description" portion of the online man page found here:
	  http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV
	  (pertains to issue #12284 but does not necessarily close it)
	  ........ ................

2008-04-01 16:57 +0000 [r112127]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged
	  revisions 112126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) |
	  13 lines Merged revisions 112125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5
	  lines Ensure that we do not exceed the hold's maximum size with a
	  single frame. (closes issue #12047) Reported by: fabianoheringer
	  Tested by: fabianoheringer ........ ................

2008-03-31 22:17 +0000 [r112070-112072]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r112069 | qwell | 2008-03-31 16:48:30 -0500
	  (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) |
	  5 lines Fix a silly infinite loop when choosing an invalid
	  option. (closes issue #12315) Reported by: jmls ........
	  ................

2008-03-31 21:03 +0000 [r112034-112036]  Terry Wilson <twilson@digium.com>

	* /, main/http.c: Merged revisions 112033 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 |
	  twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines
	  Handle blank prefix= in http.conf ........

2008-03-31 17:15 +0000 [r111997-111999]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 111998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 |
	  russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines
	  Ensure configure gets run on a clean checkout. (closes issue
	  #12197) Reported by: juggie Patches: 12197.diff uploaded by
	  juggie (license 24) ........

2008-03-31 14:22 +0000 [r111962]  Joshua Colp <jcolp@digium.com>

	* res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4
	  lines Initialize all these here tmp pointers at declaration. They
	  confused some compilers a wee bit. (closes issue #12333) Reported
	  by: ovi ........

2008-03-29  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta7.1 released.  
	
	  Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch.

2008-03-28 21:46 +0000 [r111858]  Jason Parker <jparker@digium.com>

	* codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111857 | qwell | 2008-03-28 16:46:02 -0500
	  (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) |
	  12 lines Allow gsm to compile correctly on x86 with gcc4
	  optimizations. (closes issue #11243) Reported by: whiskerp
	  Patches: 11243-maybe-asm.diff uploaded by qwell (license 4)
	  Tested by: Seggy (IRC) Note: While I did write this patch, I
	  would not have found this if fossil had not reported and fixed
	  issue #12253. A huge thanks to him for helping to (indirectly)
	  find the problem here. ........ ................

2008-03-28 19:11 +0000 [r111722-111776]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111721 | qwell | 2008-03-28 12:57:12 -0500
	  (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar
	  2008) | 1 line Remove unimplemented softkeys. Prompted by issue
	  #12325. ........ ................

2008-03-28 16:21 +0000 [r111660]  Jason Parker <jparker@digium.com>

	* /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111659 | qwell | 2008-03-28 11:20:59 -0500
	  (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) |
	  8 lines The file size of WAV49 does not need to be an even
	  number. (closes issue #12128) Reported by: mdu113 Patches:
	  12128-noevenlength.diff uploaded by qwell (license 4) Tested by:
	  qwell, mdu113 ........ ................

2008-03-28 14:43 +0000 [r111607-111608]  Tilghman Lesher <tlesher@digium.com>

	* doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008)
	  | 11 lines Merged revisions 111605 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008)
	  | 3 lines Update debugging text, since Valgrind eliminated the
	  --log-file-exactly option. (Closes issue #12320) ........
	  ................

2008-03-28 00:56 +0000 [r111566]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 111565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 |
	  file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines
	  Forgetting to unregister a manager action is bad, mmmk? ........

2008-03-28 00:17 +0000 [r111534]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 111533 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 |
	  mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10
	  lines Fix a crash that would happen when attempting to unload the
	  app_queue module. The problem was that when the refcount on the
	  queue hit 0, the destructor was called, and inside the
	  destructor, another function was called which would increase the
	  refcount back to 1 again and then decrease it again back to 0 for
	  every member in the queue. This meant that the destructor was
	  being recursively called, leading to a double free of the queue.
	  This is now fixed by making sure to unlink the queue from the
	  queues container prior to the final unref of the queue. ........

2008-03-27 21:28 +0000 [r111498]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 111497 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 |
	  murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line
	  comment cleanup and iron out a really dumb mistake in handling
	  the '.'-wildcard in the new exten pattern matcher. ........

2008-03-27 19:30 +0000 [r111444]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: Merged revisions 111443 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008)
	  | 14 lines Merged revisions 111442 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008)
	  | 6 lines For FreeBSD, at least, the ifa_addr element could be
	  NULL. (closes issue #12300) Reported by: festr Patches:
	  acl.c.patch uploaded by festr (license 443) ........
	  ................

2008-03-27 13:42 +0000 [r111361-111411]  Steve Murphy <murf@digium.com>

	* apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu,
	  27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9
	  lines These small documentation updates made in response to a
	  query in asterisk-users, where a user was using Playback, but
	  needed the features of Background, and had no idea that
	  Background existed, or that it might provide the features he
	  needed. I thought the best way to avert these kinds of queries
	  was to provide "See Also" references in all three of
	  "Background", "Playback", "WaitExten". Perhaps a project to do
	  this with all related apps is in order. ........ ................

	* res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c,
	  include/asterisk/ael_structs.h: Merged revisions 111360 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed,
	  26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) |
	  15 lines (closes issue #12302) Reported by: pj Tested by: murf
	  These changes will set a channel variable ~~EXTEN~~ just before
	  generating code for a switch, with the value of ${EXTEN}. The
	  exten is marked as having a switch, and ever after that, till the
	  end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~}
	  instead in application arguments; (and the ${EXTEN: also). The
	  reason for this, is that because switches are coded using
	  separate extensions to provide pattern matching, and jumping
	  to/from these switch extensions messes up the ${EXTEN} value,
	  which blows the minds of users. ........ ................

2008-03-27 00:36 +0000 [r111247-111339]  Jason Parker <jparker@digium.com>

	* main/frame.c, /: Merged revisions 111285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) |
	  9 lines Merged revisions 111280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) |
	  1 line Put this flag back so we don't change the API. ........
	  ................

	* main/frame.c, /: Merged revisions 111246 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) |
	  17 lines Merged revisions 111245 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) |
	  9 lines Remove excessive smoother optimization that was causing
	  audio glitches (small "pops") after (about 200ms later) an
	  "incorrectly" sized frame was received. While it would be very
	  nice to keep this as optimized as possible, it makes no sense for
	  the smoother to be dropping random bits of audio like this. Isn't
	  that the whole point of a smoother? Closes issue #12093. ........
	  ................

2008-03-26 19:57 +0000 [r111131]  Joshua Colp <jcolp@digium.com>

	* contrib/scripts/autosupport, /: Merged revisions 111130 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed,
	  26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6
	  lines Update autosupport script. (closes issue #12310) Reported
	  by: angler Patches: autosupport.diff uploaded by angler (license
	  106) ........ ................

2008-03-26 19:53 +0000 [r111128]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: Merged revisions 111127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar
	  2008) | 18 lines Merged revisions 111126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500
	  (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
	  2008) | 2 lines update UPGRADE notes to document usage of the
	  script ........ ................ ................

2008-03-26 19:41 +0000 [r111124]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500
	  (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar
	  2008) | 4 lines This code change is made just for clarification.
	  It does exactly the same thing as before. It just doesn't look as
	  wrong. ........ ................

2008-03-26 19:27 +0000 [r111072]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500
	  (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar
	  2008) | 9 lines Add a lock to the vm_state structure and use the
	  lock around mail_open calls to prevent concurrent access of the
	  same mailstream. This, along with trunk's ability to configure
	  TCP timeouts for IMAP storage will help to prevent crashes and
	  hangs when using voicemail with IMAP storage. (closes issue
	  #10487) Reported by: ewilhelmsen ........ ................

2008-03-26 19:08 +0000 [r111026]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
	  Merged revisions 111025 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar
	  2008) | 18 lines Merged revisions 111024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500
	  (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
	  2008) | 2 lines add a script to make getting the iLBC source code
	  simple for end users ........ ................ ................

2008-03-26 19:06 +0000 [r111018-111023]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) |
	  12 lines Merged revisions 111020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4
	  lines If we are requested to authenticate a reinvite make sure
	  that it contains T38 SDP if need be. (closes issue #11995)
	  Reported by: fall ........ ................

	* /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed,
	  26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
	  lines Add an option (transmit_silence) which transmits silence
	  during both Record() and DTMF generation. The reason this is an
	  option is that in order to transmit silence we have to setup a
	  translation path. This may not be needed/wanted in all cases.
	  (closes issue #10058) Reported by: tracinet ........
	  ................

2008-03-26 17:44 +0000 [r110964]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: Merged revisions 110963 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar
	  2008) | 10 lines Merged revisions 110962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar
	  2008) | 2 lines add note that the user will need to enable
	  codec_ilbc to get it to build ........ ................

2008-03-26 17:35 +0000 [r110959]  Donny Kavanagh <donnyk@gmail.com>

	* /, doc/snmp.txt: Merged revisions 110911 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 |
	  juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines
	  update documentation to reflect the changes in the way configure
	  detects net-snmp. (closes issue #12067) Reported by: juggie
	  Patches: 12067_snmp_doc.patch uploaded by juggie (license 24)
	  Tested by: juggie ........

2008-03-26 17:15 +0000 [r110882]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h
	  (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile,
	  codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
	  codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
	  (removed), codecs/ilbc/iCBSearch.h (removed),
	  codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
	  codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
	  (removed), codecs/ilbc/hpOutput.h (removed),
	  codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
	  codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c
	  (removed), codecs/ilbc/StateSearchW.h (removed),
	  codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
	  (removed), codecs/ilbc/syntFilter.h (removed),
	  codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c
	  (removed), codecs/ilbc/packing.h (removed),
	  codecs/ilbc/libilbc.vcproj (removed),
	  codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c
	  (removed), codecs/ilbc/getCBvec.c (removed),
	  codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed),
	  codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h
	  (removed), codecs/ilbc/LPCdecode.h (removed),
	  codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c
	  (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h
	  (removed), codecs/ilbc/iLBC_encode.h (removed),
	  codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c
	  (removed), codecs/ilbc/doCPLC.c (removed),
	  codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h
	  (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h
	  (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
	  codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c
	  (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged
	  revisions 110881 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar
	  2008) | 18 lines Merged revisions 110880 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700
	  (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
	  2008) | 2 lines due to licensing restrictions, we cannot
	  distribute the source code for iLBC encoding and decoding... so
	  remove it, and add instructions on how the user can obtain it
	  themselves ........ ................ ................

2008-03-26 15:33 +0000 [r110866-110868]  Joshua Colp <jcolp@digium.com>

	* /: Merged revisions 110726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 |
	  jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines
	  This one line change makes an if inside a for loop (in
	  realtime_peer) check all the ast_variables the loop was intending
	  to test rather than just the first one. ........

2008-03-26 00:03 +0000 [r110832]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 110831 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 |
	  mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6
	  lines This ensures that the manager interface is not enabled by
	  default. Prior to this change, it was possible to start Asterisk
	  with the manager interface enabled, then either comment out the
	  enabled option or make manager.conf unopenable and the manager
	  interface would still be enabled. ........

2008-03-25 22:52 +0000 [r110781]  Jason Parker <jparker@digium.com>

	* cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) |
	  14 lines Merged revisions 110779 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) |
	  6 lines Make file access in cdr_custom similar to cdr_csv. Fixes
	  issue #12268. Patch borrowed from r82344 ........
	  ................

2008-03-25 22:11 +0000 [r110778]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: This one line change makes an if inside a
	  for loop (in realtime_peer) check all the ast_variables the loop
	  was intending to test rather than just the first one.

2008-03-25 17:47 +0000 [r110690-110692]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample, /, configs/voicemail.conf.sample:
	  Merged revisions 110691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 |
	  tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines
	  Update sample configurations to make virtual hosting more
	  obvious. (closes issue #11969) Reported by: pprindeville Patches:
	  acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
	  ........

	* configs/extensions.conf.sample, /: Merged revisions 110689 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25
	  Mar 2008) | 6 lines Update the sample configuration, to use Macro
	  less (since it's now deprecated). (closes issue #12293) Reported
	  by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by
	  pprindeville (license 347) ........

2008-03-25 15:43 +0000 [r110637-110638]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Oops.

	* /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar
	  2008) | 15 lines Merged revisions 110635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar
	  2008) | 7 lines When reverting a commit, I accidentally left in
	  this bit which was an experiment to see what would happen. It
	  passed the compile test, and I didn't notice I had left this
	  change in too. So this is a revert of a revert...sort of.
	  ........ ................

2008-03-25 15:39 +0000 [r110630-110634]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/options.h, main/asterisk.c, Makefile, /,
	  main/app.c: Merged revisions 110629 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) |
	  12 lines Merged revisions 110628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
	  lines Add an option (transmit_silence) which transmits silence
	  during both Record() and DTMF generation. The reason this is an
	  option is that in order to transmit silence we have to setup a
	  translation path. This may not be needed/wanted in all cases.
	  (closes issue #10058) Reported by: tracinet ........
	  ................

2008-03-24 20:14 +0000 [r110620-110622]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar
	  2008) | 23 lines Merged revisions 110618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar
	  2008) | 15 lines This is a revert for revision 108288. The reason
	  is that that revision was not for an actual bug fix per se, and
	  so it really should not have been in 1.4 in the first place.
	  Plus, people who compile with DO_CRASH are more likely to
	  encounter a crash due to this change. While I think the usage of
	  DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
	  beyond the scope of 1.4 and should be done instead in a developer
	  branch based on trunk so that all scheduler functions are fixed
	  at once. I also am reverting the change to trunk and 1.6 since
	  they also suffer from the DO_CRASH potential. (closes issue
	  #12272) Reported by: qq12345 ........ ................

2008-03-24 17:36 +0000 [r110616]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r110615 | russell | 2008-03-24 12:36:04 -0500
	  (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008)
	  | 2 lines Turn a NOTICE into a DEBUG message. ........
	  ................

2008-03-24 15:29 +0000 [r110611]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 |
	  file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines
	  Only print out the set_address_from_contact host verbose message
	  if debugging is enabled on the dialog. (closes issue #12280)
	  Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain
	  (license 226) ........

2008-03-21 21:52 +0000 [r110579]  Jason Parker <jparker@digium.com>

	* /, sounds/Makefile: Merged revisions 110578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 |
	  qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line
	  Update to 1.4.11 core sounds. ........

2008-03-21 15:25 +0000 [r110501]  Russell Bryant <russell@digium.com>

	* /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21
	  Mar 2008) | 3 lines Note that the TCP and TLS support is
	  currently considered experimental and is subject to change while
	  we work out the remaining issues. ........

2008-03-21 14:36 +0000 [r110476]  Jason Parker <jparker@digium.com>

	* /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) |
	  15 lines Merged revisions 110474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) |
	  7 lines Don't attempt to do optimizations of gsm on mips
	  platforms either. (closes issue #12270) Reported by: zandbelt
	  Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33)
	  ........ ................

2008-03-20 23:14 +0000 [r110304-110397]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: Merged revisions 110396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008)
	  | 17 lines Merged revisions 110395 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008)
	  | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms
	  in the autoservice thread. This really should not make a
	  difference except in very rare cases. That case would be that all
	  of the channels in autoservice are not generating any frames. In
	  that case, this change reduces the potential amount of time that
	  a thread waits in ast_autoservice_stop() for the autoservice
	  thread to wrap back around to the beginning of its loop. (closes
	  issue #12266, reported by dimas) ........ ................

	* codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 |
	  russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines
	  Use the correct buffer for g722tolin16_sample. This shouldn't
	  have caused any problems, but Qwell noticed the typo here.
	  ........

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  110337 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008)
	  | 22 lines Merged revisions 110336 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r110336 | russell | 2008-03-20 16:54:58 -0500
	  (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
	  | 6 lines Fix some very broken code that was introduced in 1.2.26
	  as a part of the security fix. The dnsmgr is not appropriate
	  here. The dnsmgr takes a pointer to an address structure that a
	  background thread continuously updates. However, in these cases,
	  a stack variable was passed. That means that the dnsmgr thread
	  would be continuously writing to bogus memory. ........
	  ................ ................

	* /, main/file.c: Merged revisions 110303 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 |
	  russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines
	  Fix a bug when using zaptel timing for playing back files that
	  have a sample rate other than 8 kHz. The issue here is that
	  format modules give a "whennext" sample value, which is used to
	  calculate when to set a timer for to retrieve the next frame.
	  However, the zaptel timer operates on 8 kHz samples, so this must
	  be taken into account. (another part of issue #12164, reported by
	  milazzo and jsmith, patch by me) ........

2008-03-20 18:02 +0000 [r110273]  Mark Michelson <mmichelson@digium.com>

	* main/dial.c, /: Merged revisions 110272 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 |
	  mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3
	  lines Add missing unlock ........

2008-03-20 17:45 +0000 [r110269-110271]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20
	  Mar 2008) | 27 lines Add some fixes that I made in regards to
	  wideband codec handling to get G.722 music on hold working for
	  me. (issue #12164, reported by milazzo and jsmith, patches by me)
	  res/res_musiconhold.c: - I moved a single line so that the sample
	  queue update happened before ast_write(). The reason that this
	  was a bug is that the G.722 frame originally says it has 320
	  samples in it (which is correct). However, when the frame is
	  written to a channel that uses RTP, main/rtp.c modifies the frame
	  to cut the number of samples in half before it sends it on the
	  wire. This is to account for the stupid incorrect G.722 spec that
	  makes it so we have to lie about the number of samples with RTP.
	  I should probably go and re-work the RTP code so it doesn't
	  modify the frame so that a bug like this won't happen in the
	  future. However, this change to MOH is harmless. main/channel.c:
	  - I made two fixes in regards to generator timing. Generators use
	  samples for timing. However, this code assumed 8 kHz samples. In
	  one case, it was a hard coded 160 samples, that is now written as
	  the sample rate / 50. The other place was dealing with timing a
	  generator based on frames coming from the other direction.
	  However, that would have only worked if the sample rates for the
	  formats in both directions were the same. The code now takes into
	  account that the sample rates may differ, and scales the
	  generator samples accordingly. ........

2008-03-19 23:00 +0000 [r110165]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008)
	  | 13 lines Merged revisions 110163 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008)
	  | 5 lines Fix a bug where when calls on the trunk side hang up
	  while on hold, the state is not properly reflected. (closes issue
	  #11990, reported by anakaoka, patched by me) ........
	  ................

2008-03-19 21:06 +0000 [r110088]  Jeff Peeler <jpeeler@digium.com>

	* /: marking rev 110087 from trunk as not applying

2008-03-19 20:37 +0000 [r110085]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar
	  2008) | 12 lines Merged revisions 110083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar
	  2008) | 4 lines Add a missing unlock in the case that memory
	  allocation fails in app_chanspy. Thanks to Russell for confirming
	  that this was an issue. ........ ................

2008-03-19 19:14 +0000 [r110037]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed,
	  19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4
	  lines Add sanity checking for position resuming. We *have* to
	  make sure that the position does not exceed the total number of
	  files present, and we have to make sure that the position's
	  filename is the same as previous. These values can change if a
	  music class is reloaded and give unpredictable behavior. (closes
	  issue #11663) Reported by: junky ........ ................

2008-03-19 19:00 +0000 [r110024-110032]  Russell Bryant <russell@digium.com>

	* Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
	  (added), /: Merged revisions 109974 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) |
	  13 lines Merged revisions 109973 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) |
	  5 lines People report bugs about Asterisk crashing with DO_CRASH
	  enabled was getting a little silly... Now we only show certain
	  cflags when you run configure with --enable-dev-mode
	  (corresponding menuselect change to follow) ........
	  ................

2008-03-19 18:26 +0000 [r109971-110021]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 110020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) |
	  14 lines Merged revisions 110019 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6
	  lines Make sure that the mark bit does not incorrectly cause
	  video frame timestamps to be calculated as if they are audio
	  frames. (closes issue #11429) Reported by: sperreault Patches:
	  11429-frametype.diff uploaded by qwell (license 4) ........
	  ................

2008-03-19 16:46 +0000 [r109969]  Steve Murphy <murf@digium.com>

	* main/config.c, /: Merged revisions 109942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) |
	  80 lines Merged revisions 109908 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) |
	  72 lines (closes issue #11442) Reported by: tzafrir Patches:
	  11442.patch uploaded by murf (license 17) Tested by: murf I
	  didn't give tzafrir very much time to test this, but if he does
	  still have remaining issues, he is welcome to re-open this bug,
	  and we'll do what is called for. I reproduced the problem, and
	  tested the fix, so I hope I am not jumping by just going ahead
	  and committing the fix. The problem was with what file_save does
	  with templates; firstly, it tended to print out multiple options:
	  [my_category](!)(templateref) instead of
	  [my_category](!,templateref) which is fixed by this patch.
	  Nextly, the code to suppress output of duplicate declarations
	  that would occur because the reader copies inherited declarations
	  down the hierarchy, was not working. Thus: [master-template](!)
	  mastervar = bar [template](!,master-template) tvar = value
	  [cat](template) catvar = val would be rewritten as: ;! ;!
	  Automatically generated configuration file ;! Filename:
	  experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
	  Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
	  [master-template](!) mastervar = bar
	  [template](!,master-template) mastervar = bar tvar = value
	  [cat](template) mastervar = bar tvar = value catvar = val This
	  has been fixed. Since the config reader 'explodes' inherited vars
	  into the category, users may, in certain circumstances, see
	  output different from what they originally entered, but it should
	  be both correct and equivalent. ........ ................

2008-03-19 04:06 +0000 [r109834-109840]  Russell Bryant <russell@digium.com>

	* /, main/utils.c: Merged revisions 109839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008)
	  | 10 lines Merged revisions 109838 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008)
	  | 2 lines Tweak spacing in a recent change because I'm very
	  picky. ........ ................

	* apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008)
	  | 11 lines Merged revisions 109763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008)
	  | 3 lines Fix one place where the chanspy datastore isn't removed
	  from a channel. (issue #12243, reported by atis, patch by me)
	  ........ ................

2008-03-18 23:23 +0000 [r109779]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged
	  revisions 109775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 |
	  tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines
	  Change back to using ldap_initialize() and let the user specify a
	  URL directly, instead of trying to piece it together, badly.
	  ........

2008-03-18 21:03 +0000 [r109716]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 109714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar
	  2008) | 20 lines Merged revisions 109713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar
	  2008) | 12 lines This patch makes it so that all queue member
	  status changes are handled through device state code. This
	  removes several problems people were seeing where their queue
	  members would get into an "unknown" state. Huge props go to atis
	  on this one since he was the one who found the code section that
	  was causing the problem and proposed the solution. I just wrote
	  what he suggested :) (closes issue #12127) Reported by: atis
	  Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
	  by: atis, jvandal ........ ................

2008-03-18 20:14 +0000 [r109684]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008)
	  | 4 lines Set protocol version, port number correctly. (closes
	  issue #12211, closes issue #12209) Reported by: sylvain ........

2008-03-18 19:24 +0000 [r109654]  Jason Parker <jparker@digium.com>

	* /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) |
	  15 lines Merged revisions 109648 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) |
	  7 lines Allow codecs that use log2comp (g726) to compile
	  correctly on x86 with gcc4 optimizations. (closes issue #12253)
	  Reported by: fossil Patches: log2comp.patch uploaded by fossil
	  (license 140) ........ ................

2008-03-18 19:00 +0000 [r109546-109622]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 109576 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500
	  (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar
	  2008) | 6 lines Make sure an agent doesn't try to send dtmf to a
	  NULL channel closes issue #12242 Reported by Yourname ........
	  ................

	* include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
	  2008) | 3 lines Add format attribute to printf-style functions in
	  astmm.h ........

2008-03-18  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta6 released.

2008-03-18 17:01 +0000 [r109546]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
	  2008) | 3 lines Add format attribute to printf-style functions in
	  astmm.h ........

2008-03-18 16:26 +0000 [r109487]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar
	  2008) | 2 lines fix up various warnings found via the addition of
	  format string checking... some of these were really, really bad
	  code ........

2008-03-18 15:58 +0000 [r109454-109459]  Russell Bryant <russell@digium.com>

	* Makefile, channels/chan_misdn.c, include/asterisk/strings.h,
	  res/res_indications.c, utils/extconf.c, main/asterisk.c,
	  apps/app_voicemail.c, utils/check_expr.c,
	  cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /,
	  res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c,
	  utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c,
	  include/asterisk/astobj.h, res/res_agi.c, main/features.c,
	  apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c,
	  include/asterisk/utils.h, channels/chan_sip.c,
	  apps/app_festival.c, main/translate.c, main/jitterbuf.c,
	  utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged
	  revisions 109447 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 |
	  twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
	  Go through and fix a bunch of places where character strings were
	  being interpreted as format strings. Most of these changes are
	  solely to make compiling with -Wsecurity and -Wformat=2 happy,
	  and were not actual problems, per se. I also added format
	  attributes to any printf wrapper functions I found that didn't
	  have them. -Wsecurity and -Wmissing-format-attribute added to
	  --enable-dev-mode. ........

	* configs/sip_notify.conf.sample, /: Merged revisions 109111 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar
	  2008) | 10 lines Add sample events for aastra phones.
	  aastra-check-cfg is the same as the other check-cfg entries, and
	  aastra-xml is to load a pre-configured xml script. (closes issue
	  #12229) Reported by: gowen72 Patches: aastra.patch uploaded by
	  gowen72 (license 432) ........

2008-03-18 15:50 +0000 [r109453]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, acinclude.m4:
	  Merged revisions 109451 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 |
	  kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2
	  lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work
	  properly ........

2008-03-18 15:50 +0000 [r109448-109452]  Russell Bryant <russell@digium.com>

	* main/dial.c, /: Merged revisions 108962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008)
	  | 15 lines Merged revisions 108961 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008)
	  | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes
	  issue #12228) Reported by: andrew Patches: SRC.patch uploaded by
	  andrew (license 240) ........ ................

2008-03-18 15:16 +0000 [r109398]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, /, main/logger.c: Merged revisions 109396 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar
	  2008) | 3 lines Make sure values are interpreted as character
	  strings and not format strings. (AST-2008-004) ........

2008-03-18 15:14 +0000 [r109397]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-ntest23 (added),
	  pbx/ael/ael-test/ael-ntest23/t1/a.ael,
	  pbx/ael/ael-test/ael-ntest23/t1/b.ael,
	  pbx/ael/ael-test/ael-ntest23/t1/c.ael,
	  pbx/ael/ael-test/ael-ntest23/t2/d.ael,
	  pbx/ael/ael-test/ael-ntest23/t2/e.ael,
	  pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c,
	  pbx/ael/ael-test/ref.ael-ntest23 (added),
	  pbx/ael/ael-test/ael-ntest23/t3/g.ael,
	  pbx/ael/ael-test/ael-ntest23/t3/h.ael,
	  pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex,
	  pbx/ael/ael-test/ael-ntest23/t3/j.ael,
	  pbx/ael/ael-test/ael-ntest23/qq.ael,
	  pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2,
	  pbx/ael/ael-test/ael-ntest23/t3, /,
	  pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions
	  109357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) |
	  25 lines Merged revisions 109309 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) |
	  17 lines (closes issue #11903) Reported by: atis Many thanks to
	  atis for spotting this problem and reporting it. The fix was to
	  straighten out how items are placed on and removed from the file
	  stack. Regressions as well as the provided test case helped to
	  straighten out all code paths. valgrind was used to make sure all
	  memory allocated was freed. Sorry for not solving this earlier. I
	  got distracted. Added the ntest23 regression test, which is
	  mainly a copy of ntest22, but with a few juicy errors thrown in,
	  to replicate the kind of error that atis spotted. ........
	  ................

2008-03-18 15:11 +0000 [r109395]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 |
	  qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do
	  not return with a successful authentication if the From header
	  ends up empty. (AST-2008-003) ........

2008-03-18 15:09 +0000 [r109392]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue,
	  18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3
	  lines Put a maximum limit on the number of payloads accepted, and
	  also make sure a given payload does not exceed our maximum value.
	  (AST-2008-002) ........ ................

2008-03-18 00:40 +0000 [r109283]  Sean Bright <sean.bright@gmail.com>

	* /, configure, configure.ac: Merged revisions 109282 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar
	  2008) | 1 line Fix a typo ........

2008-03-17 22:24 +0000 [r109254]  Terry Wilson <twilson@digium.com>

	* build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
	  configure, include/asterisk/autoconfig.h.in, main/Makefile,
	  configure.ac, main/http.c, main/minimime (removed),
	  build_tools/make_buildopts_h, makeopts.in: Merged revisions
	  109229 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 |
	  twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines
	  Replace minimime with superior GMime library so that the entire
	  contents of an http post are not read into memory. This does
	  introduce a dependency on the GMime library for handling HTTP
	  POSTs, but it is available in most distros. If the library is
	  present, then the compile flag for ENABLE_UPLOADS is enabled by
	  default in menuselect. ........

2008-03-17 22:07 +0000 [r109228]  Mark Michelson <mmichelson@digium.com>

	* /, main/utils.c: Merged revisions 109227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar
	  2008) | 20 lines Merged revisions 109226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar
	  2008) | 12 lines Fix a logic flaw in the code that stores lock
	  info which is displayed via the "core show locks" command. The
	  idea behind this section of code was to remove the previous lock
	  from the list if it was a trylock that had failed. Unfortunately,
	  instead of checking the status of the previous lock, we were
	  referencing the index immediately following the previous lock in
	  the lock_info->locks array. The result of this problem, under the
	  right circumstances, was that the lock which we currently in the
	  process of attempting to acquire could "overwrite" the previous
	  lock which was acquired. While this does not in any way affect
	  typical operation, it *could* lead to misleading "core show
	  locks" output. ........ ................

2008-03-17 18:11 +0000 [r109175]  Michiel van Baak <michiel@vanbaak.info>

	* /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008)
	  | 11 lines Update the directory of placed calls on skinny phones
	  when dialing a channel that does not provide progress (analog ZAP
	  lines) The phone does handle the double update on calls to
	  channels that do provide progress and wont insert duplicate items
	  (closes issue #12239) Reported by: DEA Patches:
	  chan_skinny-call-log.txt uploaded by DEA (license 3) ........

2008-03-17 17:42 +0000 [r109167]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, configure, configure.ac, acinclude.m4: Merged
	  revisions 109166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 |
	  kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3
	  lines don't define Zaptel features as libraries, they aren't, and
	  we don't want '--with-zaptel-<foo>' configure options for them
	  also some minor cleanups ........

2008-03-17 16:47 +0000 [r109109-109114]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) |
	  12 lines Merged revisions 109107 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4
	  lines 200 OKs in response to a reinvite need to be sent reliably.
	  If the remote side does not receive one the dialog will be torn
	  down. (closes issue #12208) Reported by: atrash ........
	  ................

2008-03-17 14:21 +0000 [r109027]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar
	  2008) | 14 lines Merged revisions 109012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar
	  2008) | 6 lines Make sure that we release the lock on the spyee
	  channel if the spyee or spy has hung up (closes issue #12232)
	  Reported by: atis ........ ................

2008-03-16 17:56 +0000 [r108928-108930]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008)
	  | 7 lines Fix polling for mailbox changes in mailboxes that are
	  not in the default vm context. (closes issue #12223) Reported by:
	  DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3)
	  ........

2008-03-15 16:21 +0000 [r108741-108895]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 108799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 |
	  russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines
	  Make sure configure is run before menuselect on a clean checkout
	  (closes issue #12197) Reported by: juggie Patches: 12197.diff
	  uploaded by juggie (license 24) ........

	* channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008)
	  | 13 lines Merged revisions 108796 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008)
	  | 5 lines Fix a channel name issue. chan_oss registers the
	  "Console" channel type, but it created channels with an "OSS"
	  prefix. (closes issue #12194, reported by davidw, patched by me)
	  ........ ................

	* contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r108793 | russell | 2008-03-14 15:04:56 -0500
	  (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008)
	  | 4 lines Update the SuSE init script to start networking before
	  asterisk, as well. (closes issue #12200, reported by and change
	  suggested by reinerotto) ........ ................

	* /, configure, acinclude.m4: Merged revisions 108740 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008)
	  | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we
	  have all the required libs reported by the tool. (closes issue
	  #12067, reported by Juggie, patched by me) ........

2008-03-14 16:54 +0000 [r108739]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar
	  2008) | 41 lines Merged revisions 108737 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar
	  2008) | 33 lines Fix a race condition in the SIP packet scheduler
	  which could cause a crash. chan_sip uses the scheduler API in
	  order to schedule retransmission of reliable packets (such as
	  INVITES). If a retransmission of a packet is occurring, then the
	  packet is removed from the scheduler and retrans_pkt is called.
	  Meanwhile, if a response is received from the packet as
	  previously transmitted, then when we ACK the response, we will
	  remove the packet from the scheduler and free the packet. The
	  problem is that both the ACK function and retrans_pkt attempt to
	  acquire the same lock at the beginning of the function call. This
	  means that if the ACK function acquires the lock first, then it
	  will free the packet which retrans_pkt is about to read from and
	  write to. The result is a crash. The solution: 1. If the ACK
	  function fails to remove the packet from the scheduler and the
	  retransmit id of the packet is not -1 (meaning that we have not
	  reached the maximum number of retransmissions) then release the
	  lock and yield so that retrans_pkt may acquire the lock and
	  operate. 2. Make absolutely certain that the ACK function does
	  not recursively lock the lock in question. If it does, then
	  releasing the lock will do no good, since retrans_pkt will still
	  be unable to acquire the lock. (closes issue #12098) Reported by:
	  wegbert (closes issue #12089) Reported by: PTorres Patches:
	  12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
	  by: jvandal ........ ................

2008-03-14 14:33 +0000 [r108684]  Jason Parker <jparker@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r108683 | qwell | 2008-03-14 09:32:55 -0500
	  (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) |
	  4 lines Fix a potential segfault if chan (or chan->music_state)
	  is NULL. Closes issue #12210, credit to edantie for pointing this
	  out. ........ ................

2008-03-13 21:48 +0000 [r108587]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 108586 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 |
	  mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3
	  lines Make this compile ........

2008-03-13 21:41 +0000 [r108585]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c, main/channel.c, /,
	  include/asterisk/channel.h: Merged revisions 108584 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r108584 | russell | 2008-03-13 16:40:43 -0500
	  (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008)
	  | 11 lines Fix another issue that was causing crashes in chanspy.
	  This introduces a new datastore callback, called chan_fixup().
	  The concept is exactly like the fixup callback that is used in
	  the channel technology interface. This callback gets called when
	  the owning channel changes due to a masquerade. Before this was
	  introduced, if a masquerade happened on a channel being spyed on,
	  the channel pointer in the datastore became invalid. (closes
	  issue #12187) (reported by, and lots of testing from atis) (props
	  to file for the help with ideas) ........ ................

2008-03-13 21:31 +0000 [r108582]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 108529 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 |
	  mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11
	  lines Fixing a potential buffer overflow in the manager command
	  ModuleCheck. Though this overflow is exploitable remotely, we are
	  NOT issuing a security advisory for this since in order to
	  exploit the overflow, the attacker would have to establish an
	  authenticated manager session AND have the system privilege. By
	  gaining this privilege, the attacker already has more powerful
	  weapons at his disposal than overflowing a buffer with a
	  malformed manager header, so the vulnerability in this case
	  really lies with the authentication method that allowed the
	  attacker to gain the system privilege in the first place.
	  ........

2008-03-13 21:07 +0000 [r108347-108532]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008)
	  | 18 lines Merged revisions 108530 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008)
	  | 10 lines Make a tweak that gets the LEDs on polycom phones to
	  blink when an extension that has been subscribed to goes on hold.
	  Otherwise, they just stay on like it does when an extension is in
	  use. (closes issue #11263) Reported by: russell Patches:
	  notify_hold.rev1.txt uploaded by russell (license 2) Tested by:
	  russell ........ ................

	* apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008)
	  | 2 lines Fix a place where configuration values could cause an
	  overflow of a buffer. ........

	* /, apps/app_followme.c: Merged revisions 108472 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008)
	  | 12 lines Merged revisions 108469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008)
	  | 4 lines Fix a couple uses of sprintf. The second one could
	  actually cause an overflow of a stack buffer. It's not a security
	  issue though, it only depends on your configuration. ........
	  ................

	* /, main/features.c: Merged revisions 107465 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 |
	  file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines
	  Clarify comment about masquerading and playback of the parking
	  slot. (closes issue #12180) Reported by: davidw ........

	* /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 |
	  file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If
	  we receive a 488 on a T38 request reinvite back to audio. As well
	  reinvite across a bridge back to audio if one side doesn't
	  negotiate to T38. (closes issue #8677) Reported by: alex-911
	  ........

	* /: Merged revisions 106892 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 |
	  mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line
	  Make sure we don't start a call when we have already done so in
	  response to a COT message ........

	* /, main/editline/Makefile.in: Merged revisions 106843 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r106843 | qwell | 2008-03-07 16:15:20 -0600
	  (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) |
	  5 lines Fix hardcoded grep in editline, were GNU grep is
	  required. (closes issue #12124) Reported by: dmartin ........
	  ................

	* include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
	  channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
	  include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008)
	  | 3 lines Rename ast_tcptls_server_instance to session_instance,
	  since this pertains to server and client usage. ........

	* /, main/http.c: Merged revisions 108346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 |
	  russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines
	  Make the default prefix empty, like it was in Asterisk 1.4.
	  (closes issue #12198, reported by bkruse, patched by me) ........

2008-03-12 22:10 +0000 [r108246-108294]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 |
	  mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3
	  lines Let's get this to compile ........

	* /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar
	  2008) | 22 lines Merged revisions 108288 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar
	  2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for
	  autocongestion in chan_sip. The scheduler callback will always
	  return 0. This means that this id is never rescheduled, so it
	  makes no sense to loop trying to delete the id from the scheduler
	  queue. If we fail to remove the item from the queue once, it will
	  fail every single time. (Yes I realize that in this case, the
	  macro would exit early because the id is set to -1 in the
	  callback, but it still makes no sense to use that macro in favor
	  of calling ast_sched_del once and being done with it) This is the
	  first of potentially several such fixes. ........
	  ................

	* /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500
	  (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar
	  2008) | 12 lines Added a large comment before the AST_SCHED_DEL
	  macro to explain its purpose as well as when it is appropriate
	  and when it is not appropriate to use it. I also removed the part
	  of the debug message that mentions that this is probably a bug
	  because there are some perfectly legitimate places where
	  ast_sched_del may fail to delete an entry (e.g. when the
	  scheduler callback manually reschedules with a new id instead of
	  returning non-zero to tell the scheduler to reschedule with the
	  same idea). I also raised the debug level of the debug message in
	  AST_SCHED_DEL since it seems like it could come up quite
	  frequently since the macro is probably being used in several
	  places where it shouldn't be. Also removed the redundant line,
	  file, and function information since that is provided by ast_log.
	  ........ ................

2008-03-12 20:29 +0000 [r108205]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar
	  2008) | 14 lines Merged revisions 108086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar
	  2008) | 6 lines if we receive an INVITE with a Content-Length
	  that is not a valid number, or is zero, then don't process the
	  rest of the message body looking for an SDP closes issue #11475
	  Reported by: andrebarbosa ........ ................

2008-03-12 19:59 +0000 [r108138]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r108137 | russell | 2008-03-12 14:59:05 -0500
	  (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008)
	  | 40 lines (closes issue #12187, reported by atis, fixed by me
	  after some brainstorming on the issue with mmichelson) - Update
	  copyright info on app_chanspy. - Fix a race condition that caused
	  app_chanspy to crash. The issue was that the chanspy datastore
	  magic that was used to ensure that spyee channels did not
	  disappear out from under the code did not completely solve the
	  problem. It was actually possible for chanspy to acquire a
	  channel reference out of its datastore to a channel that was in
	  the middle of being destroyed. That was because datastore
	  destruction in ast_channel_free() was done near the end. So, this
	  left the code in app_chanspy accessing a channel that was
	  partially, or completely invalid because it was in the process of
	  being free'd by another thread. The following sort of shows the
	  code path where the race occurred:
	  =============================================================================
	  Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
	  --------------------------------------||-------------------------------------
	  ast_channel_free() || - remove channel from channel list || -
	  lock/unlock the channel to ensure || that no references retrieved
	  from || the channel list exist. ||
	  --------------------------------------||-------------------------------------
	  || channel_spy() - destroy some channel data || - Lock chanspy
	  datastore || - Retrieve reference to channel || - lock channel ||
	  - Unlock chanspy datastore
	  --------------------------------------||-------------------------------------
	  - destroy channel datastores || - call chanspy datastore d'tor ||
	  which NULL's out the ds' || - Operate on the channel ...
	  reference to the channel || || - free the channel || || || -
	  unlock the channel
	  --------------------------------------||-------------------------------------
	  =============================================================================
	  ........ ................

2008-03-12 18:31 +0000 [r108085]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, /, include/asterisk/audiohook.h,
	  main/audiohook.c: Merged revisions 108084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) |
	  12 lines Merged revisions 108083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4
	  lines Add a trigger mode that triggers on both read and write.
	  The actual function that returns the combined audio frame though
	  will wait until both sides have fed in audio, or until one side
	  stops (such as the case when you call Wait). (closes issue
	  #11945) Reported by: xheliox ........ ................

2008-03-12 17:03 +0000 [r108033]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 108032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008)
	  | 12 lines Merged revisions 108031 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008)
	  | 4 lines Destroy the channel lock after the channel datastores.
	  (inspired by issue #12187) ........ ................

2008-03-12 07:44 +0000 [r107879-107999]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 |
	  tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines
	  Deadlock fixes (closes issue #12143) Reported by: kactus Patches:
	  20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: kactus ........

	* main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged
	  revisions 107960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 |
	  tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines
	  Revert several changes from revision 102525, as the changes were
	  not compatible, and, in fact, introduced regressions. (Closes
	  issue #12190) ........

	* contrib/scripts/iax-friends.sql, /,
	  contrib/scripts/sip-friends.sql: Merged revisions 107878 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500
	  (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008)
	  | 2 lines Document all of the possible realtime fields ........
	  ................

2008-03-11 23:38 +0000 [r107828]  Jason Parker <jparker@digium.com>

	* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r107827 | qwell | 2008-03-11 18:38:00 -0500
	  (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) |
	  7 lines Update documentation for pgsql ODBC voicemail. (closes
	  issue #12186) Reported by: jsmith Patches:
	  vm_pgsql_doc_update.patch uploaded by jsmith (license 15)
	  ........ ................

2008-03-11 22:59 +0000 [r107723-107793]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c, main/config.c, res/res_config_curl.c,
	  res/res_config_pgsql.c, res/res_config_odbc.c, /,
	  include/asterisk/config.h, res/res_config_ldap.c: Merged
	  revisions 107791 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 |
	  tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines
	  An offhand comment from Russell made me realize that the
	  configuration file caching would not work properly for users.conf
	  and any other file read from more than one place. I needed to add
	  the filename which requested the config file to get it to work
	  properly. ........

2008-03-11 20:54 +0000 [r107720]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 107718 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) |
	  13 lines Merged revisions 107714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) |
	  5 lines Copy voicemail dependency logic for res_adsi to
	  chan_gtalk and chan_jingle (for jabber). (closes issue #12014)
	  Reported by: junky ........ ................

2008-03-11 20:51 +0000 [r107716]  Kevin P. Fleming <kpfleming@digium.com>

	* /, Makefile.rules, channels/Makefile: Merged revisions 107715 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500
	  (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar
	  2008) | 2 lines get chan_vpb to build properly in dev mode
	  ........ ................

2008-03-11 20:37 +0000 [r107584-107711]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_page.c: Merged revisions 107710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 |
	  file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines
	  Dial a device even if it's state is unknown. (closes issue
	  #12184) Reported by: bluecrow76 Patches:
	  asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by
	  bluecrow76 (license 270) ........

	* /, main/features.c: Merged revisions 107659 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) |
	  12 lines Merged revisions 107646 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4
	  lines Make sure the visible indication is on the right channel so
	  when the masquerade happens the proper indication is enacted.
	  (closes issue #11707) Reported by: iam ........ ................

	* /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) |
	  12 lines Merged revisions 107637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4
	  lines Add an additional check for setting conference parameter
	  when using the marked user options. It was possible for it to
	  return to a no listen/no talk state if a masquerade happened.
	  (closes issue #12136) Reported by: aragon ........
	  ................

2008-03-11 15:39 +0000 [r107374-107526]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar
	  2008) | 2 lines fix another potential bug found by gcc 4.3
	  ........

	* apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /,
	  apps/app_sms.c: Merged revisions 107466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar
	  2008) | 10 lines Merged revisions 107464 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar
	  2008) | 2 lines fix various other problems found by gcc 4.3
	  ........ ................

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  apps/app_sms.c: Merged revisions 107462 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar
	  2008) | 10 lines Merged revisions 107461 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar
	  2008) | 2 lines stop checking for mktime() in the configure
	  script... we don't use it, and the test is buggy under gcc 4.3
	  ........ ................

	* /, configure, main/Makefile, configure.ac, makeopts.in: Merged
	  revisions 107409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar
	  2008) | 13 lines Merged revisions 107408 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar
	  2008) | 5 lines check for compiler support for
	  -fno-strict-overflow before using it (tested with Debian's gcc
	  4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview
	  ........ ................

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 107406 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar
	  2008) | 10 lines Merged revisions 107405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar
	  2008) | 2 lines fix small bug in IMAP toolkit testing ........
	  ................

	* main/udptl.c, utils/Makefile, /, main/Makefile,
	  main/editline/readline.c, res/Makefile: Merged revisions 107373
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500
	  (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar
	  2008) | 11 lines fix up various compiler warnings found with
	  gcc-4.3: - the output of flex includes a static function called
	  'input' that is not used, so for the moment we'll stop having the
	  compiler tell us about unused variables in the flex source files
	  (a better fix would be to improve our flex post-processing to
	  remove the unused function) - main/stdtime/localtime.c makes
	  assumptions about signed integer overflow, and gcc-4.3's improved
	  optimizer tries to take advantage of handling potential overflow
	  conditions at compile time; for now, suppress these optimizations
	  until we can fiure out if the code needs improvement -
	  main/udptl.c has some references to uninitialized variables; in
	  one case there was no bug, but in the other it was certainly
	  possibly for unexpected behavior to occur -
	  main/editline/readline.c had an unused variable ........
	  ................

2008-03-11 01:27 +0000 [r107336]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008)
	  | 10 lines Merged revisions 107290 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008)
	  | 2 lines If we fail to alloc a channel, we should re-lock the
	  pvt structure before returning. ........ ................

2008-03-10 23:46 +0000 [r107289]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 107019 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 |
	  murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way
	  back in July, in r.75706, a fix was made ot the strftime usages,
	  which was good, but in this case, the check for a nil time was
	  accidentally removed, and now it is restored, to keep timevals
	  like '1969-12-31 17:00:00' from showing up in the cdrs. No idea
	  what databases will do with this. No bugs filed as yet, but it
	  felt like a bug. ........

2008-03-10 20:29 +0000 [r107180]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) |
	  13 lines Merged revisions 107173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) |
	  5 lines Make sure to reenable echo can after a "failed"
	  (canceled, etc) three-way call. (closes issue #11335) Reported
	  by: rebuild ........ ................

2008-03-10 20:18 +0000 [r107101-107163]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 107162 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008)
	  | 16 lines Merged revisions 107161 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008)
	  | 8 lines Fix another bug specifically related to asynchronous
	  call origination. Once the PBX is started on the channel using
	  ast_pbx_start(), then the ownership of the channel has been
	  passed on to another thread. We can no longer access it in this
	  code. If the channel gets hung up very quickly, it is possible
	  that we could access a channel that has been free'd. (inspired by
	  BE-386) ........ ................

	* main/pbx.c, /: Merged revisions 107159 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008)
	  | 17 lines Merged revisions 107158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008)
	  | 9 lines Fix some bugs related to originating calls. If the code
	  failed to start a PBX on the channel (such as if you set a call
	  limit based on the system's load average), then there were cases
	  where a channel that has already been free'd using ast_hangup()
	  got accessed. This caused weird memory corruption and crashes to
	  occur. (fixes issue BE-386) (much debugging credit goes to
	  twilson, final patch written by me) ........ ................

	* main/channel.c, /: Merged revisions 107103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008)
	  | 10 lines Merged revisions 107102 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008)
	  | 2 lines Resolve a compiler warning. ........ ................

	* main/channel.c, /: Merged revisions 107100 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008)
	  | 11 lines Merged revisions 107099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008)
	  | 3 lines Fix a race condition where the generator can go away
	  (closes issue #12175, reported by edantie, patched by me)
	  ........ ................

2008-03-10 15:46 +0000 [r107069]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 107068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 |
	  mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10
	  lines app_queue has now been doxygenified thanks to snuffy! The
	  ony thing I changed was the way that locks are referenced, since
	  the old 1.2 names were still used in the comments. (closes issue
	  #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff
	  uploaded by snuffy (license 35) ........

2008-03-10 14:38 +0000 [r107018]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged
	  revisions 107017 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) |
	  15 lines Merged revisions 107016 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7
	  lines Move where unanswered CDRs are dropped to the CDR core, not
	  everything uses app_dial. (closes issue #11516) Reported by: ys
	  Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested
	  by: anest, jcapp, dartvader ........ ................

2008-03-08 17:54 +0000 [r106997]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we don't start a call on a channel
	  that has already started a call

2008-03-08 16:14 +0000 [r106947]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar
	  2008) | 10 lines Merged revisions 106945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar
	  2008) | 2 lines don't generate D-Channel "up" and "down" messages
	  unless the channel state is actually changing; also, generate the
	  "up" message when an implicit "up" occurs due to reception of a
	  normal event when we thought the channel was "down" ........
	  ................

2008-03-07 22:53 +0000 [r106897]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008)
	  | 10 lines Merged revisions 106895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008)
	  | 2 lines Only start the SLA thread if SLA has actually been
	  configured. ........ ................

2008-03-07 19:34 +0000 [r106790]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 106789 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) |
	  12 lines Merged revisions 106788 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4
	  lines Ignore source update control frame. (closes issue #12168)
	  Reported by: plack ........ ................

2008-03-07 17:18 +0000 [r106686-106713]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r106707 | russell | 2008-03-07 11:17:30 -0600
	  (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008)
	  | 8 lines Change a warning message to a debug message. This is
	  happening quite frequently, and it is not worth spamming users
	  with these messages unless we are pretty confident that it should
	  never happen. As it stands today, it _will_ and _does_ happen and
	  until that gets cleaned up a reasonable amount on the development
	  side, let's not spam the logs of everyone else. (closes issue
	  #12154) ........ ................

	* doc/smdi.txt, /: Merged revisions 106684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 |
	  russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines
	  fix example usage ........

2008-03-07 16:27 +0000 [r106554-106662]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600
	  (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008)
	  | 3 lines Warn the user when a temporary greeting exists (Closes
	  issue #11409) ........ ................

	* main/rtp.c, /: Merged revisions 106607 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008)
	  | 11 lines Merged revisions 106606 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008)
	  | 3 lines Properly initialize rtp->schedid (Closes issue #12154)
	  ........ ................

	* apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
	  apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
	  funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /,
	  channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c,
	  utils/extconf.c: Merged revisions 106553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008)
	  | 14 lines Merged revisions 106552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008)
	  | 6 lines Safely use the strncat() function. (closes issue
	  #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
	  uploaded by Corydon76 (license 14) ........ ................

2008-03-07 01:19 +0000 [r106502-106520]  Russell Bryant <russell@digium.com>

	* doc/smdi.txt, /: Merged revisions 106518 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 |
	  russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line
	  minor text changes ........

	* doc/smdi.txt, /: Merged revisions 106507 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 |
	  russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines
	  Add updated SMDI documentation that I had only sitting in my
	  email ... oops ........

	* main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c,
	  main/file.c: Merged revisions 106501 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 |
	  russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines
	  Merge changes from team/russell/g722-sillyness ... Fix a number
	  of other places where the number of samples in a G722 frame was
	  not properly handled because of various reasons. main/rtp.c: -
	  When a G722 frame is read from the smoother, the number of
	  samples in the frame must be divided by 2 before being sent out
	  over the network. Even though G722 is 16 kHz, an error in some
	  previous spec has made it so that we have to list the number of
	  samples such as if it was 8 kHz. main/file.c: - When scheduling
	  the next time to expect a frame, take into account that the
	  format of the file we're reading from may not be 8 kHz.
	  codecs/codec_g722.c: - When converting from G722 to slinear,
	  g722_decode() expects its samples parameter to be in the silly
	  (real samples / 2) format. Make it so. - When converting from
	  slinear to G722, properly set the number of samples in the frame
	  to be the number of bytes of output * 2. formats/format_pcm.c: -
	  This format module handles G722, among a number of other formats.
	  However, the read() and seek() functions did not account for the
	  fact that G722 has 2 samples per byte. (closes issue #12130,
	  reported by rickross, patched by me) ........

2008-03-06 22:16 +0000 [r106442]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 106438 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar
	  2008) | 16 lines Merged revisions 106437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar
	  2008) | 8 lines Quell an annoying message that is likely to print
	  every single time that ast_pbx_outgoing_app is called. The reason
	  is that __ast_request_and_dial allocates the cdr for the channel,
	  so it should be expected that the channel will have a cdr on it.
	  Thanks to joetester on IRC for pointing this out ........
	  ................

2008-03-06 22:15 +0000 [r106440]  Jason Parker <jparker@digium.com>

	* /, main/file.c: Merged revisions 106439 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 |
	  qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines
	  Fix file playback in many cases. (closes issue #12115) Reported
	  by: pj Patches: v2-fileexists.patch uploaded by dimas (license
	  88) (with modifications by me) Tested by: dimas, qwell, russell
	  ........

2008-03-06 20:39 +0000 [r106433]  Donny Kavanagh <donnyk@gmail.com>

	* /, res/res_agi.c: Merged revisions 106399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 |
	  juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines
	  trivial fix for an agi error when attempting to use EAGI on a
	  dead/hungup channel, we now print an error that makes sense given
	  our removal of deadagi as an actual application. (closes issue
	  #12161) Reported by: explidous Patches: res_agi_12161.patch
	  uploaded by juggie (license 24) Tested by: juggie ........

2008-03-06 05:25 +0000 [r106330-106359]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008)
	  | 7 lines Missing braces, fix parsing (closes issue #12112)
	  Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008
	  uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76
	  ........

	* /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600
	  (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008)
	  | 2 lines Upgrade to the next release of sounds ........
	  ................

2008-03-06 00:23 +0000 [r106299-106320]  Russell Bryant <russell@digium.com>

	* channels/chan_oss.c, main/rtp.c, main/channel.c,
	  channels/chan_phone.c, main/dial.c, channels/chan_skinny.c,
	  main/file.c, channels/chan_h323.c, channels/chan_alsa.c,
	  include/asterisk/frame.h, channels/chan_mgcp.c,
	  channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /,
	  channels/chan_sip.c, channels/chan_console.c,
	  apps/app_followme.c: Merged revisions 106239 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) |
	  12 lines Merged revisions 106235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4
	  lines Add a control frame to indicate the source of media has
	  changed. Depending on the underlying technology it may need to
	  change some things. (closes issue #12148) Reported by: jcomellas
	  ........ ................

	* /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r106238 | russell | 2008-03-05 16:40:58 -0600
	  (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008)
	  | 3 lines Fix a potential deadlock and a few different potential
	  crashes. (closes issue #12145, reported by thiagarcia, patched by
	  me) ........ ................

	* /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008)
	  | 7 lines document var_metric usage to prevent bugreports that
	  are actually configuration issues (closes issue #12151) Reported
	  by: caio1982 Patches: DB_metric3.diff uploaded by caio1982
	  (license 22) ........

	* main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged
	  revisions 105933 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008)
	  | 13 lines Merged revisions 105932 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008)
	  | 5 lines Fix a bug that I just noticed in the RTP code. The
	  calculation for setting the len field in an ast_frame of audio
	  was wrong when G.722 is in use. The len field represents the
	  number of ms of audio that the frame contains. It would have set
	  the value to be twice what it should be. ........
	  ................

	* funcs/func_global.c, /: Merged revisions 105899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 |
	  russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines
	  Fix the SHARED() read callback to properly unlock the channel.
	  This function could not have worked, as it left the channel
	  locked in all cases. ........

	* main/manager.c, /: Merged revisions 105864 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 |
	  mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5
	  lines There are several places in manager.c where BUFSIZ is used
	  for a buffer which will contain nowhere near that amount of data.
	  This makes these buffers more reasonably sized. ........

	* main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c,
	  apps/app_queue.c: Merged revisions 105841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 |
	  tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines
	  Fix minor misuses of snprintf ........

	* main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c,
	  main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c,
	  main/frame.c, /, main/say.c, main/utils.c, main/enum.c,
	  main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c,
	  main/loader.c, main/term.c, main/cli.c, main/channel.c,
	  main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c,
	  main/features.c, main/logger.c, main/app.c, main/image.c,
	  main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c:
	  Merged revisions 105840 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 |
	  tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines
	  Whitespace changes only ........

	* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
	  main/http.c, include/asterisk/tcptls.h: Merged revisions 105804
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04
	  Mar 2008) | 2 lines add a destroy API call for a server instance
	  ........

	* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
	  main/http.c, include/asterisk/tcptls.h: Merged revisions 105785
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04
	  Mar 2008) | 2 lines More public API name changes to use an
	  appropriate ast_ prefix ........

	* include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
	  channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
	  include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008)
	  | 2 lines Rename public object server_instance to
	  ast_tcptls_server_instance ........

	* /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 |
	  russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
	  Fix some bugs in the SIP tcp helper thread. - fix a spot where a
	  lock wouldn't get unlocked in an error condition - call
	  ast_mutex_destroy() on the lock before freeing its memory
	  (related to issue #11972) ........

	* /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 |
	  twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines
	  Set username to default to the category name if it isn't
	  overridden by a usernmae= setting in users.conf ........

	* main/rtp.c, /: Merged revisions 105677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) |
	  10 lines Merged revisions 105676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2
	  lines In addition to setting the marker bit let's change our ssrc
	  so they know for sure it is a different source. ........
	  ................

	* main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
	  Merged revisions 105675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) |
	  16 lines Merged revisions 105674 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8
	  lines When a new source of audio comes in (such as music on hold)
	  make sure the marker bit gets set. (closes issue #10355) Reported
	  by: wdecarne Patches: 10355.diff uploaded by file (license 11)
	  (closes issue #11491) Reported by: kanderson ........
	  ................

2008-03-05 17:42 +0000 [r106140]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008)
	  | 3 lines Should check these values for non-NULL before scanning.
	  (Closes issue #12147) ........

2008-03-05 15:43 +0000 [r106041]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar
	  2008) | 15 lines Merged revisions 106038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar
	  2008) | 7 lines when a PRI call must be moved to a different B
	  channel at the request of the other endpoint, ensure that any DSP
	  active on the original channel is moved to the new one (closes
	  issue #11917) Reported by: mavetju Tested by: mavetju ........
	  ................

2008-03-05 15:31 +0000 [r106037]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, include/asterisk/sched.h: Merged
	  revisions 106036 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008)
	  | 15 lines Merged revisions 106015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008)
	  | 7 lines Correctly initialize retransid in SIP, and ensure that
	  the warning when failing to delete a schedule entry can actually
	  hit the log. (closes issue #12140) Reported by: slavon Patches:
	  sch2.patch uploaded by slavon (license 288) (Patch slightly
	  modified by me) ........ ................

2008-03-04  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta5 released.

2008-03-04 16:55 +0000 [r105574-105597]  Russell Bryant <russell@digium.com>

	* CHANGES: Update CHANGES heading

	* funcs/func_version.c: Simplify a trivial snprintf() with
	  ast_copy_string()

	* main/hashtab.c: Make it so you don't have to cast away const in a
	  couple places

	* main/hashtab.c: remove unnecessary casts

	* main/pbx.c: - Add curly braces around the while loop - Properly
	  break out of the loop on error when an included context is not
	  found

	* main/pbx.c: Use ast_copy_string() instead of strncpy(), and use
	  sizeof() instead of a magic number

	* channels/chan_zap.c: Fix some code that was improperly changed in
	  revision 104866 from issue #12079. (closes issue #12129, reported
	  by elguero, patched by me)

2008-03-03 18:08 +0000 [r105573]  Jason Parker <jparker@digium.com>

	* /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) |
	  7 lines Fix types for astNumChannels and astConfigCallsProcessed.
	  (closes issue #12114) Reported by: jeffg Patches: 12114.patch
	  uploaded by jeffg (license 192) ........

2008-03-03 17:17 +0000 [r105564-105571]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /: Merged revisions 105570 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03
	  Mar 2008) | 3 lines In the case of an ast_channel allocation
	  failure, take the local_pvt out of the pvt list before destroying
	  it. ........

	* channels/chan_local.c, /: Merged revisions 105568 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03
	  Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt
	  struct when ast_channel allocation fails. Also, in passing,
	  centralize the code necessary to destroy a local_pvt. ........

	* main/autoservice.c, /: Merged revisions 105565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008)
	  | 3 lines Update the copyright information for autoservice. Most
	  of the code in this file now is stuff that I have written
	  recently ... ........

	* main/channel.c, main/autoservice.c, /,
	  include/asterisk/_private.h, main/asterisk.c: 3) In addition to
	  merging the changes below, change trunk back to a regular LIST
	  instead of an RWLIST. The way this list works makes it such that
	  a RWLIST provides no additional benefit. Also, a mutex is needed
	  for use with the thread condition. Merged revisions 105563 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008)
	  | 24 lines Merge in some changes from
	  team/russell/autoservice-nochans-1.4 These changes fix up some
	  dubious code that I came across while auditing what happens in
	  the autoservice thread when there are no channels currently in
	  autoservice. 1) Change it so that autoservice thread doesn't keep
	  looping around calling ast_waitfor_n() on 0 channels twice a
	  second. Instead, use a thread condition so that the thread
	  properly goes to sleep and does not wake up until a channel is
	  put into autoservice. This actually fixes an interesting bug, as
	  well. If the autoservice thread is already running (almost always
	  is the case), then when the thread goes from having 0 channels to
	  have 1 channel to autoservice, that channel would have to wait
	  for up to 1/2 of a second to have the first frame read from it.
	  2) Fix up the code in ast_waitfor_nandfds() for when it gets
	  called with no channels and no fds to poll() on, such as was the
	  case with the previous code for the autoservice thread. In this
	  case, the code would call alloca(0), and pass the result as the
	  first argument to poll(). In this case, the 2nd argument to
	  poll() specified that there were no fds, so this invalid pointer
	  shouldn't actually get dereferenced, but, this code makes it
	  explicit and ensures the pointers are NULL unless we have valid
	  data to put there. (related to issue #12116) ........

2008-03-03 15:30 +0000 [r105558-105561]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 105560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7
	  lines It is possible for no audio to pass between the current
	  digit and next digit so expand logic that clears emulation to
	  AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg
	  Patches: v1-11911.patch uploaded by dimas (license 88) Tested by:
	  tbsky ........

	* /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6
	  lines Add a comment to describe some logic. (closes issue #12120)
	  Reported by: flefoll Patches:
	  chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
	  244) ........

2008-03-01 03:59 +0000 [r105509]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Add support for 16KHz signed linear.

2008-03-01 02:03 +0000 [r105479]  Tilghman Lesher <tlesher@digium.com>

	* /: Drop bad property

2008-03-01 01:30 +0000 [r105477]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, include/asterisk/app.h,
	  main/global_datastores.c, /, main/features.c, main/app.c,
	  include/asterisk/global_datastores.h: Asterisk, when parking can
	  drop rights a caller when a parking timeout occurs. Also, when
	  doing built-in attended transfers, sometimes incorrectly passes
	  rights from the transferrer to the transferee. This patch tries
	  to fixes the parking issue and lays some groundwork for later
	  fixing the transfer issue. (closes issue #11520) Reported by:
	  pliew Tested by: otherwiseguy

2008-03-01 00:53 +0000 [r105461]  Russell Bryant <russell@digium.com>

	* CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI
	  command to control custom device states. Also, do some additional
	  code cleanup and improvement in passing. (closes issue #12106)
	  Reported by: nizon Patches: devstate-patch.txt uploaded by nizon
	  (license 415) -- Updated to trunk, and tab completion added by me

2008-02-29 23:53 +0000 [r105411]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Convert to use ast_str

2008-02-29 23:36 +0000 [r105410]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: Merged revisions 105409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008)
	  | 23 lines Fix a major bug in autoservice. There was a race
	  condition in the handling of the list of channels in autoservice.
	  The problem was that it was possible for a channel to get removed
	  from autoservice and destroyed, while the autoservice thread was
	  still messing with the channel. This led to memory corruption,
	  and caused crashes. This explains multiple backtraces I have seen
	  that have references to autoservice, but do to the nature of the
	  issue (memory corruption), could cause crashes in a number of
	  areas. (fixes the crash in BE-386) (closes issue #11694) (closes
	  issue #11940) The following issues could be related. If you are
	  the reporter of one of these, please update to include this fix
	  and try again. (potentially fixes issue #11189) (potentially
	  fixes issue #12107) (potentially fixes issue #11573) (potentially
	  fixes issue #12008) (potentially fixes issue #11189) (potentially
	  fixes issue #11993) (potentially fixes issue #11791) ........

2008-02-29 18:34 +0000 [r105378]  Joshua Colp <jcolp@digium.com>

	* configs/sip.conf.sample: Add documentation for setting
	  username/password in SIP dial string. (closes issue #11587)
	  Reported by: sobomax Patches: dialstring_doc.diff uploaded by
	  sobomax (license 359)

2008-02-29 14:50 +0000 [r105263-105327]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 105326 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008)
	  | 1 line Fix a potential memory leak ........

	* channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c:
	  Remove unnecessary if statements before calling iks_delete
	  (redundant check is done inside iks_delete), thus making the code
	  conform with coding guidelines.

2008-02-29 13:55 +0000 [r105262]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb
	  2008) | 4 lines Bump up the size of the uniqueid variable.
	  (closes issue #12107) Reported by: asgaroth ........

2008-02-29 13:12 +0000 [r105210]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Automatically create new buddy upon reception
	  of a presence stanza of type subscribed. (closes issue #12066)
	  Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
	  phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
	  (license 73) Tested by: ffadaie, phsultan

2008-02-29 01:15 +0000 [r105176]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008)
	  | 7 lines Update init script for LSB compat (closes issue #9843)
	  Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by
	  ibc (license 211) Tested by: paravoid ........

2008-02-28 22:39 +0000 [r105144]  Russell Bryant <russell@digium.com>

	* /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c:
	  Merged revisions 105116 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008)
	  | 8 lines Fix a bug in the lock tracking code that was discovered
	  by mmichelson. The issue is that if the lock history array was
	  full, then the functions to mark a lock as acquired or not would
	  adjust the stats for whatever lock is at the end of the array,
	  which may not be itself. So, do a sanity check to make sure that
	  we're updating lock info for the proper lock. (This explains the
	  bizarre stats on lock #63 in BE-396, thanks Mark!) ........

2008-02-28 20:14 +0000 [r105060-105061]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 105059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb
	  2008) | 6 lines When using autofill, members who are in use
	  should be counted towards the number of available members to call
	  if ringinuse is set to yes. Thanks to jmls who brought this issue
	  up on IRC ........

	* main/dial.c, /: Merged revisions 104841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb
	  2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels
	  a lockable list. This is because in some cases, the ast_dial may
	  exist in multiple threads due to asynchronous execution of its
	  application, and I found some cases where race conditions could
	  exist. 2. Check in ast_dial_join to be sure that the channel
	  still exists before attempting to lock it, since it could have
	  gotten hung up but the is_running_app flag on the
	  ast_dial_channel may not have been cleared yet. (closes issue
	  #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
	  putnopvut (license 60) Tested by: jvandal ........

2008-02-28 19:21 +0000 [r105006]  Jason Parker <jparker@digium.com>

	* main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb
	  2008) | 9 lines Make pbx_exec pass an empty string into
	  applications, if we get NULL. This protects against possible
	  segfaults in applications that may try to use data before
	  checking length (ast_strdupa'ing it, for example) (closes issue
	  #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
	  uploaded by qwell (license 4) ........

2008-02-28 14:42 +0000 [r104974]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_vpb.cc: Fix crash when configuration does not match
	  hardware detection. (closes issue #12096) Reported by: mmickan
	  Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)

2008-02-28 04:37 +0000 [r104921]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb
	  2008) | 2 lines According to a video at www.cisco.com, the 7921G
	  supports 6 line appearances. ........

2008-02-28 00:11 +0000 [r104869]  Tilghman Lesher <tlesher@digium.com>

	* /, main/Makefile, build_tools/strip_nonapi: Merged revisions
	  104868 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008)
	  | 7 lines Compatibility fix for PPC64 (closes issue #12081)
	  Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch
	  uploaded by jcollie (license 412) Tested by: jcollie, Corydon76
	  ........

2008-02-27 23:58 +0000 [r104866]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: reduce indentation in alloc_sub (issue
	  #12079) Reported by: tzafrir Patches: alloc_sub uploaded by
	  tzafrir (license 46)

2008-02-27 21:02 +0000 [r104788]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2
	  lines Don't loop around infinitely trying to spy on our own
	  channel, and don't forget to free/detach the datastore upon
	  hangup of the spy. ........

2008-02-27 20:37 +0000 [r104784]  Mark Michelson <mmichelson@digium.com>

	* /, main/file.c: Merged revisions 104783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb
	  2008) | 4 lines Bump a couple of more buffers up by 2 so that
	  annoying warnings aren't generated like crazy on every
	  fileexists_core call. ........

2008-02-27 19:36 +0000 [r104756]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Remove useless 's' and 'key' variables, in
	  favor of 'val', which serves the exact same purpose.

2008-02-27 18:20 +0000 [r104705]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 104704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008)
	  | 2 lines Ensure the session ID can't be 0. ........

2008-02-27 17:45 +0000 [r104687]  Joshua Colp <jcolp@digium.com>

	* /, main/file.c: Merged revisions 104665 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2
	  lines Bump up the buffer by 2. ........

2008-02-27 17:36 +0000 [r104643]  Russell Bryant <russell@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008)
	  | 4 lines Fix a problem in ChanSpy where it could get stuck in an
	  infinite loop without being able to detect that the calling
	  channel hung up. (closes issue #12076, reported by junky, patched
	  by me) ........

2008-02-27 17:31 +0000 [r104617]  Jason Parker <jparker@digium.com>

	* /, main/features.c: Merged revisions 104598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) |
	  8 lines Inherit language from the transfering channel on a blind
	  transfer. (closes issue #11682) Reported by: caio1982 Patches:
	  local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22)
	  Tested by: caio1982, victoryure ........

2008-02-27 17:12 +0000 [r104595-104597]  Joshua Colp <jcolp@digium.com>

	* /, main/loader.c: Merged revisions 104596 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4
	  lines Use the lock (which already existed, it just wasn't used)
	  on the updaters list to protect the contents instead of the
	  overall module list lock. (closes issue #12080) Reported by:
	  ChaseVenters ........

	* channels/chan_sip.c: After further discussion revert my previous
	  commit for this. Currently in order to ensure devicestate is the
	  expected value in another module (such as app_queue) then
	  chan_sip must be loaded before hand.

2008-02-27 16:54 +0000 [r104594]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/file.c: Merged revisions 104593 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb
	  2008) | 8 lines fallback to standard English prompts properly
	  when using new prompt directory layout (closes issue #11831)
	  Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG
	  (license 20) (modified by me to improve code and conform rest of
	  function to coding guidelines) ........

2008-02-27 16:26 +0000 [r104537-104539]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When queueing up a device state change when
	  the peer is loaded from the configuration give it a state of not
	  in use. We have to do this because the channel technology may not
	  yet be registered so the state could not be queried and would be
	  considered invalid. (closes issue #12087) Reported by: liorm

	* res/res_smdi.c, /: Merged revisions 104536 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4
	  lines Only stop the MWI monitor thread if it was actually
	  started. (closes issue #12086) Reported by: francesco_r ........

2008-02-27 15:34 +0000 [r104534]  Tilghman Lesher <tlesher@digium.com>

	* utils/astcanary.c: open(2) needs a mode argument when O_CREAT is
	  specified. (Closes issue #12083)

2008-02-27 15:31 +0000 [r104533]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression
	  introduced by state changes. (closes issue #12078) Reported by:
	  dimas Patches: v1-12078.patch uploaded by dimas (license 88)
	  (closes issue #12074) Reported by: Ivan

2008-02-27 08:20 +0000 [r104502]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_vpb.cc, configs/vpb.conf.sample,
	  include/asterisk/module.h: Bring Voicetronix driver up to date
	  with current drivers (closes issue #12084) Reported by: mmickan
	  Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
	  module.h.diff uploaded by mmickan (license 400) vpb.conf.sample
	  uploaded by mmickan (license 400)

2008-02-27 04:42 +0000 [r104419-104473]  Russell Bryant <russell@digium.com>

	* doc/janitor-projects.txt: note that the chan_sip conversion is
	  already in progress

	* doc/janitor-projects.txt: add another janitor project

	* doc/janitor-projects.txt: Add the stuff from the janitor projects
	  page that is still relevant. I figure that if we keep this in the
	  tree, it will be much easier to keep up to date. The page on
	  asterisk.org just links to this on svn.digium.com/view

2008-02-27 03:52 +0000 [r104418]  Jason Parker <jparker@digium.com>

	* doc/janitor-projects.txt (added): Create placeholder file...for
	  now.

2008-02-27 02:05 +0000 [r104388]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Whitespace changes only

2008-02-27 01:16 +0000 [r104333-104335]  Russell Bryant <russell@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008)
	  | 3 lines Avoid some recursion in the cleanup code for the
	  chanspy datastore (closes issue #12076, reported by junky,
	  patched by me) ........

2008-02-26 22:14 +0000 [r104301]  Steve Murphy <murf@digium.com>

	* res/snmp/agent.c: small change to allow this file to compile. No
	  problem if you don't install the libsnmp package.

2008-02-26 20:33 +0000 [r104244-104270]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: I swear I compiled this ... *cough*

	* res/res_phoneprov.c: fix this module, too

	* funcs/func_version.c: fix this module

	* Makefile, include/asterisk, build_tools/make_version_h (added):
	  Re-add the automatically generated version.h, so that modules can
	  include for making build time decisions for cross asterisk
	  version compatibility

	* main/manager.c, channels/chan_sip.c, include/asterisk/version.h
	  (removed), build_tools/make_version_c, res/res_agi.c,
	  main/http.c, include/asterisk/ast_version.h (added): Rename
	  version.h to ast_version.h. Next, I will be re-adding version.h
	  as an automatically generated file like it used to be. This still
	  needs to be there for modules that have to check it to compile
	  against multiple asterisk versions.

2008-02-26 19:14 +0000 [r104215]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an
	  'e' option to ResetCDR which re-enables a CDR that has been
	  disabled. (closes issue #11170) Reported by: kratzers Patches:
	  ResetCDR.1.diff uploaded by kratzers (license 307)

2008-02-26 18:40 +0000 [r104176]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while,
	  even around single statements. 2) Revise the argument parsing
	  section, showing use of the standard macros. 3) Fix a typo.

2008-02-26 18:27 +0000 [r104140-104142]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 104141 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) |
	  1 line Add badshell to .PHONY target (thanks Kevin) ........

	* Makefile, /: Merged revisions 104139 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) |
	  2 lines Since all shells aren't as awesome as bash, we have to
	  fail if somebody tries to use a literal "~" in DESTDIR. ........

2008-02-26 16:51 +0000 [r104137]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting and doxygen while waiting on an
	  airport...

2008-02-26 16:36 +0000 [r104133-104136]  Jason Parker <jparker@digium.com>

	* /, sounds/Makefile: Merged revisions 104135 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) |
	  5 lines Revert previous abspath change. ...abspath is new in GNU
	  make 3.81. I feel so...defeated. Must find new fix! ........

	* /, sounds/Makefile: Merged revisions 104132 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) |
	  9 lines Fix a very bizarre issue we were seeing with our buildbot
	  when using a DESTDIR that wasn't an absolute path (such as
	  DESTDIR=~/asterisk-1.4). Apparently what was happening, was that
	  some of the targets were being expanded to the full path, so $@
	  ended up being /root/asterisk-1.4/[...]/ rather than
	  ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
	  in GNU make. (*cough*
	  http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
	  ........

2008-02-26 14:51 +0000 [r104127]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Remove more hardcoded pipe symbols and replace
	  with commas. (closes issue #12072) Reported by: SimonSharman
	  Patches: features.patch uploaded by SimonSharman (license 410)
	  Tested by: SimonSharman

2008-02-26 06:43 +0000 [r104125]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Use the readhandle for reads (closes issue
	  #12069)

2008-02-26 00:38 +0000 [r104120-104124]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c: Add a \todo to convert this module to the event
	  system

	* CHANGES: Update CHANGES for SMDI stuff

	* channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample,
	  include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions
	  104119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008)
	  | 33 lines Merge changes from team/russell/smdi-1.4 This commit
	  brings in a significant set of changes to the SMDI support in
	  Asterisk. There were a number of bugs in the current
	  implementation, most notably being that it was very likely on
	  busy systems to pop off the wrong message from the SMDI message
	  queue. So, this set of changes fixes the issues discovered as
	  well as introducing some new ways to use the SMDI support which
	  are required to avoid the bugs with grabbing the wrong message
	  off of the queue. This code introduces a new interface to SMDI,
	  with two dialplan functions. First, you get an SMDI message in
	  the dialplan using SMDI_MSG_RETRIEVE() and then you access
	  details in the message using the SMDI_MSG() function. A side
	  benefit of this is that it now supports more than just chan_zap.
	  For example, with this implementation, you can have some FXO
	  lines being terminated on a SIP gateway, but the SMDI link in
	  Asterisk. Another issue with the current implementation is that
	  it is quite common that the station ID that comes in on the SMDI
	  link is not necessarily the same as the Asterisk voicemail box.
	  There are now additional directives in the smdi.conf
	  configuration file which let you map SMDI station IDs to Asterisk
	  voicemail boxes. Yet another issue with the current SMDI support
	  was related to MWI reporting over the SMDI link. The current code
	  could only report a MWI change when the change was made by
	  someone calling into voicemail. If the change was made by some
	  other entity (such as with IMAP storage, or with a web interface
	  of some kind), then the MWI change would never be sent. The SMDI
	  module can now poll for MWI changes if configured to do so. This
	  work was inspired by and primarily done for the University of
	  Pennsylvania. (also related to issue #9260) ........

2008-02-25 23:56 +0000 [r104103-104110]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option
	  in favor of dialplan substring variable syntax. (closes issue
	  #12060)

	* /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008)
	  | 10 lines This patch fixes some pretty significant problems with
	  how app_chanspy handles pointers to channels that are being spied
	  upon. It was very likely that a crash would occur if the channel
	  being spied upon hung up. This was because the current
	  ast_channel handling _requires_ that the object is locked or else
	  it could disappear at any time (except in the owning channel
	  thread). So, this patch uses some channel datastore magic on the
	  spied upon channel to be able to detect if and when the channel
	  goes away. (closes issue #11877) (patch written by me, but thanks
	  to kpfleming for the idea, and to file for review) ........

	* /, main/utils.c: Merged revisions 104102 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008)
	  | 7 lines Improve the lock tracking code a bit so that a bunch of
	  old locks that threads failed to lock don't sit around in the
	  history. When a lock is first locked, this checks to see if the
	  last lock in the list was one that was failed to be locked. If it
	  is, then that was a lock that we're no longer sitting in a
	  trylock loop trying to lock, so just remove it. (inspired by
	  issue #11712) ........

2008-02-25 23:04 +0000 [r104097-104101]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be
	  saved in Postgres. Note that these changes are
	  backward-compatible, so no changes to UPGRADE.txt are necessary.
	  (closes issue #9279) Reported by: rottenroddy Patches:
	  20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76

	* funcs/func_global.c: Shared space for variables (instead of
	  letting other channels muck with your own) (closes issue #11943)
	  Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: jmls

	* /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25
	  Feb 2008) | 5 lines If the destination folder is full, don't
	  delete a message when exiting. (closes issue #12065) Reported by:
	  selsky Patch by: (myself) ........

2008-02-25 21:40 +0000 [r104096]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6
	  lines Make it so a users.conf user creates both a SIP peer and a
	  SIP user. The user will be used for inbound authentication for
	  the device, and peer will be used for placing calls to the
	  device. (closes issue #9044) Reported by: queuetue Patches:
	  sip-gui-friend.diff uploaded by qwell (license 4) ........

2008-02-25 20:50 +0000 [r104093]  Jason Parker <jparker@digium.com>

	* /, main/config.c: Merged revisions 104092 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) |
	  11 lines Allow the use of #include and #exec in situations where
	  the max include depth was only 1. Specifically, this fixes using
	  #include and #exec in extconfig.conf. This was basically caused
	  because the config file itself raises the include level to 1. I
	  opted not to raise the include limit, because recursion here
	  could cause very bizarre behavior. Pointed out, and tested by
	  jmls (closes issue #12064) ........

2008-02-25 19:02 +0000 [r104089]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Instead of outputting a verbose message
	  every so often let's make it a debug message.

2008-02-25 19:00 +0000 [r104088]  Brett Bryant <bbryant@digium.com>

	* doc/siptls.txt, configs/sip.conf.sample: Adding more tls
	  configuration details to sip.conf sample, with a list of valid
	  ciphers provided in both files. .. First commit since July, woot

2008-02-25 18:38 +0000 [r104087]  Russell Bryant <russell@digium.com>

	* /, channels/chan_agent.c: Merged revisions 104086 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25
	  Feb 2008) | 4 lines Ensure that the channel doesn't disappear in
	  agent_logoff(). If it does, it could cause a crash. (fixes the
	  crash reported in BE-396) ........

2008-02-25 16:18 +0000 [r104081-104085]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6
	  lines If a resubscription comes in for a dialog we no longer know
	  about tell the remote side that the dialog does not exist so they
	  subscribe again using a new dialog. (closes issue #10727)
	  Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file
	  (license 11) ........

	* /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6
	  lines Due to recent changes tag will no longer be NULL if not
	  present so we have to use ast_strlen_zero to see if it's actually
	  blank. (closes issue #12061) Reported by: flefoll Patches:
	  chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll
	  (license 244) ........

	* res/res_config_pgsql.c: Fix building of trunk. dbpass is always
	  going to exist.

2008-02-24 02:37 +0000 [r104073-104074]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: Enforce a space between function args as per
	  code review.

	* res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned
	  on, and pgsql installed, I get warnings that stops the compile.
	  They are fixed now.

2008-02-22 23:56 +0000 [r104045]  Doug Bailey <dbailey@digium.com>

	* channels/chan_zap.c, configure, configure.ac: Add protection to
	  chan_zap build when NEONMWI events are not defined

2008-02-22 22:55 +0000 [r104036-104039]  Tilghman Lesher <tlesher@digium.com>

	* doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES,
	  include/asterisk/manager.h: Move Originate to a separate
	  privilege and require the additional System privilege to call out
	  to a subshell.

	* /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008)
	  | 6 lines Backwards debug message. (closes issue #12052) Reported
	  by: flefoll Patches: chan_sip.c.br14.patch_found-notfound
	  uploaded by flefoll (license 244) ........

	* res/res_config_pgsql.c: Allow database password to be NULL and
	  several other cleanups. (closes issue #12048) Reported by: bukaj
	  Patches: 20080222__bug12048.diff.txt uploaded by Corydon76
	  (license 14) Tested by: bukaj

2008-02-21 21:27 +0000 [r104031]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix a typo

2008-02-21 21:09 +0000 [r104025-104029]  Mark Michelson <mmichelson@digium.com>

	* res/res_agi.c: Instead of a notice, make the message about a
	  hung-up channel a debug message, and revert the original logic on
	  the if statement. Thanks to Juggie for bringing this to my
	  attention.

2008-02-21 17:38 +0000 [r104024]  Doug Bailey <dbailey@digium.com>

	* channels/chan_zap.c: Added configuration distinction between neon
	  and fsk mwi detection Add the detection for neon MWI events got
	  rid of extraneous handle_init_event call in monitor loop

2008-02-21 16:46 +0000 [r104020]  Mark Michelson <mmichelson@digium.com>

	* res/res_agi.c: Don't print the fact that we are using dead mode
	  in AGI if called from the 'h' extension since it is well-known
	  that it will be running in dead mode. (closes issue #12046)
	  Reported by: explidous

2008-02-21 16:44 +0000 [r104019]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Disable epoll as it has caused more obscure issues then any of my
	  previous code. I will continue to work on it in a separate branch
	  to make it stable for a release and test it against the following
	  issues. (closes issue #11253) Reported by: falves11 (closes issue
	  #11657) Reported by: davevg (closes issue #11033) Reported by:
	  falves11

2008-02-21 14:44 +0000 [r104016]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c, /: Merged revisions 104015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb
	  2008) | 2 lines reduce the likelihood that HTTP Manager session
	  ids will consist of primarily '1' bits ........

2008-02-21 05:21 +0000 [r104014]  Tilghman Lesher <tlesher@digium.com>

	* utils/astman.c: Ignore some more unused generated events. (closes
	  issue #12042) Reported by: junky Patches: astman_events.diff
	  uploaded by junky (license 177)

2008-02-20  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta4 released.

2008-02-20 22:34 +0000 [r103957]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 103956 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb
	  2008) | 8 lines Clear up confusion when viewing the
	  QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
	  user's perspective, the queue does exist, we shouldn't tell them
	  we couldn't find the queue. Instead since it is a dead queue,
	  report a 0 waiting count This issue was brought up on IRC by jmls
	  ........

2008-02-20 22:29 +0000 [r103954-103955]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Try to do Packet2Packet bridging with
	  chan_h323 if reinviting isn't enabled. (closes issue #11901)
	  Reported by: pj

	* channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6
	  lines Don't wait for additional digits when overlap dialing is
	  enabled if the setup message contains the sending_complete
	  information element. (closes issue #11785) Reported by: klaus3000
	  Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt
	  uploaded by klaus3000 (license 65) ........

2008-02-20 21:41 +0000 [r103908]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c, /: Merged revisions 103904 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed,
	  20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL
	  while attempting to lock it. (closes issue #12039) Reported by:
	  danpwi ........

2008-02-20 21:36 +0000 [r103903]  Jason Parker <jparker@digium.com>

	* include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone
	  detection routines. Separate fax detection from digit detected.
	  Added CED (called) tone detection for fax (previously, only CNG
	  (calling) was supported). Separate DTMF/MF code paths where
	  appropriate. Allow detection of arbitary tones. (closes issue
	  #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch
	  uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache

2008-02-20 21:08 +0000 [r103902]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a crash due to the wrong variable being
	  used when building a directory string. (closes issue #12027)
	  Reported by: jaroth Patches: forward.patch uploaded by jaroth
	  (license 50) Tested by: jaroth

2008-02-20 18:29 +0000 [r103846-103847]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/sched.h: Add some documentation fixups

	* /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20
	  Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022)
	  Reported by: asgaroth Patches: 20080219__bug12022.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: asgaroth ........

2008-02-20 15:21 +0000 [r103844]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't
	  been converted to ',' (closes issue #12034) Reported by: kowalma

2008-02-20 03:52 +0000 [r103838-103842]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c: *mumble*

	* main/audiohook.c: file not found.

	* main/audiohook.c: Minor test...

2008-02-20 00:49 +0000 [r103833]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: When using IMAP storage, if the folder you
	  attempt to save to does not exist, create it first. (closes issue
	  #12032) Reported by: jaroth Patches: createfolder.patch uploaded
	  by jaroth (license 50) Tested by: jaroth

2008-02-19 22:35 +0000 [r103831-103832]  Jason Parker <jparker@digium.com>

	* main/channel.c: Make sure to mask out non-audio first as well

	* main/channel.c: Maybe we should set the value before we test it?
	  Fixes an issue people have been seeing (unreported?) with file
	  playback not working.

2008-02-19 21:54 +0000 [r103824-103828]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: Add a log message that appears when you try to
	  unload a module that isn't loaded. (closes issue #12033) Reported
	  by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by
	  jamesgolovich (license 176)

	* main/file.c: Only output a log message saying the format does not
	  exist if it actually does not exist, not if the file itself could
	  not be opened. (closes issue #11828) Reported by: IgorG Patches:
	  readfile.v1.diff uploaded by IgorG (license 20)

	* /, channels/h323/ast_h323.cxx: Merged revisions 103823 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6
	  lines Send CallerID Name in setup message. (closes issue #11241)
	  Reported by: tusar Patches: h323id_as_callerid_name.patch
	  uploaded by tusar (license 344) ........

2008-02-19 20:06 +0000 [r103822]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /: Merged revisions 103821 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19
	  Feb 2008) | 8 lines Account for the fact that the "other" channel
	  can disappear while the local pvt is not locked. (fixes a problem
	  introduced in rev 100581) (closes issue #12012) Reported by:
	  stevedavies Patch by me ........

2008-02-19 19:27 +0000 [r103819-103820]  Joshua Colp <jcolp@digium.com>

	* apps/app_authenticate.c: len already contains the position we
	  want to examine, if we move one left again we'll actually
	  probably be looking at a digit. (issue #12030) Reported by:
	  alligosh

	* apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add
	  CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan
	  application. This will either be set to NOCHANNEL if the given
	  channel was not found or SUCCESS if it worked. (closes issue
	  #11553) Reported by: johan Patches:
	  UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
	  CHANGES.channelredirect.patch uploaded by johan (license 334)
	  app_channelredirect-20080219.patch uploaded by johan (license
	  334)

2008-02-19 18:14 +0000 [r103818]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: (closes issue #11864) Reported by: julianjm
	  Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm
	  (license 99) Patch fixes problem of device state incorrectly
	  reporting idle before PBX answers incoming call on FXO channel.
	  Device status is updated now during new channel creation.

2008-02-19 17:33 +0000 [r103808-103813]  Joshua Colp <jcolp@digium.com>

	* /, configure, configure.ac: Merged revisions 103812 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb
	  2008) | 4 lines Don't look for launchd when cross compiling.
	  (closes issue #12029) Reported by: ovi ........

2008-02-19 00:59 +0000 [r103805]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Change verbosity into debug for Hebrew (and various
	  whitespace fixes) (Closes issue #12011)

2008-02-18 23:58 +0000 [r103798-103802]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 103801 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) |
	  10 lines Ensure that emulated DTMFs do not get interrupted by
	  another begin frame. (closes issue #11740) Reported by: gserra
	  Patches: v1-11740.patch uploaded by dimas (license 88) (closes
	  issue #11955) Reported by: tsearle (closes issue #10530) Reported
	  by: xmarksthespot ........

	* main/channel.c, main/frame.c, channels/chan_sip.c,
	  include/asterisk/channel.h, include/asterisk/frame.h: Add a
	  non-invasive API for application level manipulation of T38 on a
	  channel. This uses control frames (so they can even pass across
	  IAX2) to negotiate T38 and provided a way of getting the current
	  status of T38 using queryoption. This should by no means cause
	  any issues and if it does I will take responsibility for it.
	  (closes issue #11873) Reported by: dimas Patches:
	  v4-t38-api.patch uploaded by dimas (license 88)

	* main/frame.c: Add some missing control frames.

2008-02-18 22:33 +0000 [r103796]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) |
	  1 line Fix previous commit so that we actually disable
	  echocanbridged if echocancel is off. ........

2008-02-18 21:57 +0000 [r103794]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Commit chan_zap portion of #11964: add the
	  ability to get ORIG_CALLED_NUM

2008-02-18 21:30 +0000 [r103791]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) |
	  4 lines Correct a message when echocancelwhenbridged is on, but
	  echocancel is not. Closes issue #12019 ........

2008-02-18 20:58 +0000 [r103788]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure EC is enabled when SS7 call comes
	  in. Also add support for multiple DPCs per linkset. #11779

2008-02-18 20:53 +0000 [r103787]  Mark Michelson <mmichelson@digium.com>

	* /, main/app.c: Merged revisions 103786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb
	  2008) | 10 lines There was an invalid assumption when calculating
	  the duration of a file that the filestream in question was
	  created properly. Unfortunately this led to a segfault in the
	  situation where an unknown format was specified in voicemail.conf
	  and a voicemail was recorded. Now, we first check to be sure that
	  the stream was written correctly or else assume a zero duration.
	  (closes issue #12021) Reported by: jakep Tested by: putnopvut
	  ........

2008-02-18 19:47 +0000 [r103783]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: make the output of 'core show settings' a bit
	  nicer. (closes issue #12020) Reported by: seanbright Patches:
	  asterisk.c.patch uploaded by seanbright (license 71)

2008-02-18 17:45 +0000 [r103781]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008)
	  | 9 lines When a SIP channel is being auto-destroyed, it's
	  possible for it to still be in bridge code. When that happens, we
	  crash. Delay the RTP destruction until the bridge is ended.
	  (closes issue #11960) Reported by: norman Patches:
	  20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: norman ........

2008-02-18  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta3 released.

2008-02-18 17:12 +0000 [r103772]  Olle Johansson <oej@edvina.net>

	* main/channel.c, channels/chan_sip.c: Make sure we can set up
	  calls without audio (text+video). And ... it works!

2008-02-18 16:40 +0000 [r103771]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb
	  2008) | 10 lines Fix a linked list corruption that under the
	  right circumstances could lead to a looped list, meaning it will
	  traverse forever. (closes issue #11818) Reported by: michael-fig
	  Patches: 11818.patch uploaded by putnopvut (license 60) Tested
	  by: michael-fig ........

2008-02-18 16:13 +0000 [r103764-103769]  Joshua Colp <jcolp@digium.com>

	* apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h:
	  Add an API call (ast_async_parseable_goto) which parses a goto
	  string and does an async goto instead of an explicit goto.
	  (closes issue #11753) Reported by: johan

	* /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2
	  lines Don't care if the extension given doesn't exist for
	  subscription based MWI. ........

2008-02-18 10:10 +0000 [r103755]  Olle Johansson <oej@edvina.net>

	* CHANGES, channels/chan_iax2.c: - No space in manager event names,
	  please - Add new event to CHANGES

2008-02-18 04:43 +0000 [r103754]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, main/channel.c, main/pbx.c,
	  funcs/func_channel.c, include/asterisk/channel.h, CHANGES,
	  main/cli.c: Context tracing for channels (closes issue #11268)
	  Reported by: moy Patches:
	  chantrace-datastored-encapsulated-rev94934.patch uploaded by moy
	  (license 222)

2008-02-16 21:22 +0000 [r103750]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: move two ast_log calls to ast_debug. Now
	  monitoring chan_skinny port with nagios or zabbix wont generate
	  noise on the console. @ok tilghman

2008-02-15 23:32 +0000 [r103742]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15
	  Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race
	  condition (closes issue #11780) Reported by: guillecabeza
	  Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
	  380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
	  380) ........

2008-02-15 23:20 +0000 [r103740]  Mark Michelson <mmichelson@digium.com>

	* CHANGES: Document GotoIfTime change from svn revision 103738

2008-02-15 23:14 +0000 [r103739]  Russell Bryant <russell@digium.com>

	* include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related
	  to the use of AES encryption. 1024 was used instead of 128 when
	  using AES from OpenSSL. Many thanks to d1mas for figuring this
	  one out! (closes issue #11946) Reported by: bbhoss Patches:
	  v1-11946.patch uploaded by dimas (license 88)

2008-02-15 23:07 +0000 [r103737-103738]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Add proper "false" case behavior to GotoIfTime
	  (closes issue #11719) Reported by: kshumard Patches:
	  gotoiftime.twobranches.patch uploaded by kshumard (license 92)
	  Tested by: kshumard

	* apps/app_voicemail.c: Fix redeclaration of variables when using
	  IMAP storage (closes issue #11988) Reported by: jaroth Patches:
	  variable_cleanup.patch uploaded by jaroth (license 50)

2008-02-15 19:50 +0000 [r103727-103729]  Russell Bryant <russell@digium.com>

	* /, main/loader.c: Merged revisions 103728 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008)
	  | 4 lines In the case that you try to directly reload a module
	  has returned AST_MODULE_LOAD_DECLINE, log a message indicating
	  that the module is not fully initialized and must be initialized
	  using "module load". ........

	* /, main/loader.c: Merged revisions 103726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008)
	  | 6 lines Don't attempt to execute the reload callback for a
	  module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
	  that was reported against chan_console in trunk. (closes issue
	  #11953, reported by junky, fixed by me) ........

2008-02-15 17:32 +0000 [r103725]  Mark Michelson <mmichelson@digium.com>

	* doc/tex/imapstorage.tex, /, configure, configure.ac: Merged
	  revisions 103722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb
	  2008) | 8 lines Final round of changes for configure script logic
	  for IMAP Now if a directory is specified, then we will search
	  that directory for a source installation of the IMAP toolkit. If
	  none is found, then we will use that directory as the basis for
	  detecting a package installation of the IMAP c-client. If that
	  check fails, then configure will fail. ........

2008-02-15 17:29 +0000 [r103723]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c,
	  include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c,
	  apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c,
	  main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h,
	  main/ast_expr2f.c, channels/misdn/ie.c,
	  channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c,
	  include/asterisk/strings.h, res/res_config_ldap.c,
	  include/asterisk/time.h: Fix up some doxygen issues. (closes
	  issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy
	  (license 35)

2008-02-15 15:45 +0000 [r103716]  Tilghman Lesher <tlesher@digium.com>

	* utils/conf2ael.c: Remove extraneous copy (closes issue #12002)
	  Reported by: junky Patches: conf2ael.diff uploaded by junky
	  (license 177)

2008-02-15 15:11 +0000 [r103699-103715]  Mark Michelson <mmichelson@digium.com>

	* configure, configure.ac: Merging of changes from 1.4 revision
	  103713.

	* doc/tex/imapstorage.tex, configure, configure.ac: Same changes as
	  made to 1.4 in revision 103710

	* doc/tex/imapstorage.tex: Trunk version of 1.4's imap
	  documentation updates

	* configure, configure.ac: See commit message for svn revision
	  103698. This behavior is the same as what is described there. The
	  difference is that trunk already had the --with-imap=system
	  option, but it only checked the include path for headers in the
	  imap directory and not also the c-client directory.

2008-02-14 21:21 +0000 [r103694]  Jason Parker <jparker@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Modify
	  ldap autoconf function, so that an incorrect ldap library is not
	  found on Solaris (it is incompatible). Also removes second check
	  for awk, which causes Solaris to find an incompatible version of
	  awk. (closes issue #11829) Reported by: snuffy Patches:
	  bug-11829.diff uploaded by snuffy (license 35)

2008-02-14 21:04 +0000 [r103687-103691]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu,
	  14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........

	* /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu,
	  14 Feb 2008) | 9 lines Fix the new message count if delete=yes
	  when using IMAP storage. (closes issue #11406) Reported by:
	  jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
	  50) Tested by: jaroth ........

	* configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change
	  the queue holdtime announcement to happen at any interval (not
	  just greater than two minutes). Remove the saying of less-than
	  for holdtime announcements since it can lead to awkward holdtime
	  announcements. Using '1' as a queue-round-seconds value is no
	  longer valid. (closes issue #9736) Reported by: caio1982 Patches:
	  queue_announce5.diff uploaded by caio1982 (license 22) Tested by:
	  caio1982, putnopvut

2008-02-14 19:52 +0000 [r103685]  Jason Parker <jparker@digium.com>

	* /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) |
	  5 lines Document the 'l' option to the CDR() function. (Thanks
	  voipgate for pointing out the option, and Leif for providing text
	  for it.) Closes issue #11695. ........

2008-02-14 19:47 +0000 [r103682]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_externalivr.c: a few syntax changes and safer code

2008-02-14 18:39 +0000 [r103677]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Add periodic jitter stats to CLI and
	  manager. (closes issue #8188) Reported by: stevedavies Patches:
	  jblogging-trunk.patch uploaded by stevedavies
	  jblogging-trunk_wmgrevent.patch uploaded by johann8384
	  updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384
	  (license 190) (with additional changes by me) Tested by:
	  stevedavies, johann8384

2008-02-14 10:19 +0000 [r103668]  Olle Johansson <oej@edvina.net>

	* res/res_agi.c, apps/app_externalivr.c: Formatting fixes

2008-02-13 21:04 +0000 [r103662]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_externalivr.c: (closes issue #11825) Reported by:
	  ctooley Patches: additional_eivr_commands.patch uploaded by
	  ctooley (license 136) Tested by: ctooley

2008-02-13 15:47 +0000 [r103658]  Mark Michelson <mmichelson@digium.com>

	* UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold
	  and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold
	  (closes issue #11904) Reported by: dimas Patches: v2-moh.patch
	  uploaded by dimas (license 88) Tested by: dimas

2008-02-13 00:55 +0000 [r103559]  Mark Michelson <mmichelson@digium.com>

	* main/event.c: Fix a small logic error in ast_event_iterator_next.
	  The previous logic allowed for the iterator to indicate there was
	  more data than there really was, causing the iterator read beyond
	  the end of the event structure. This led to invalid memory reads
	  and potential crashes.

2008-02-12 22:26 +0000 [r103447-103506]  Jason Parker <jparker@digium.com>

	* main/manager.c: Even more sane permissions. This should be
	  handled via a umask, like in many other places.

	* main/manager.c: Use slight more sane permissions

2008-02-12 15:39 +0000 [r103387-103388]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Remove development version notice.

	* main/manager.c: Fix build on *BSD. These permissions constants
	  are not available there.

2008-02-12 15:13 +0000 [r103386]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4
	  lines Even if no CallerID name or number has been provided by the
	  remote party still use the configured sip.conf ones. (closes
	  issue #11977) Reported by: pj ........

2008-02-12 14:08 +0000 [r103341]  Philippe Sultan <philippe.sultan@gmail.com>

	* include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags
	  structure in aji_client and aji_buddy rather than an integer.
	  Modify calls to various ast_*_flag macros accordingly.

2008-02-12 00:24 +0000 [r103331]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, include/asterisk/config.h, CHANGES,
	  main/config.c: Requested changes from Pari, reviewed by Russell.
	  Added ability to retrieve list of categories in a config file.
	  Added ability to retrieve the content of a particular category.
	  Added ability to empty a context. Created new action to create a
	  new file. Updated delete action to allow deletion by line number
	  with respect to category. Added new action insert to add new
	  variable to category at specified line. Updated action newcat to
	  allow new category to be inserted in file above another existing
	  category.

2008-02-11 22:10 +0000 [r103317-103325]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4
	  lines If entering a conference with the 'w' option ensure that we
	  can't listen or speak until the marked user appears. (closes
	  issue #11835) Reported by: alanmcmillan ........

	* res/res_agi.c: Remove ast_module_user usage from res_agi. This is
	  taken care of in the core.

	* main/pbx.c, main/manager.c, main/translate.c, main/logger.c,
	  main/app.c, main/utils.c, main/indications.c, main/asterisk.c,
	  main/rtp.c: Just some minor coding style cleanup...

	* main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue
	  #10661) Reported by: junky

2008-02-11 17:09 +0000 [r103316]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configs/zapata.conf.sample: Merged revisions 103315 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb
	  2008) | 2 lines improve 2BCT documentation a bit (thanks Jared)
	  ........

2008-02-11 16:17 +0000 [r103313-103314]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, channels/chan_iax2.c: Add support for allowing a
	  native bridge to happen when the L option is enabled. The RTP
	  bridging could already handle this, it just needed to be enabled
	  in the main bridging code. (issue #10647) Reported by: samdell3

	* channels/chan_skinny.c: Change chan_skinny to use debug messages
	  as appropriate. (closes issue #11967) Reported by: mvanbaak
	  Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak
	  (license 7)

2008-02-11 06:05 +0000 [r103306]  James Golovich <james@gnuinter.net>

	* channels/chan_sip.c: Don't wipe out transport and fd in chan_sip
	  on reload (issue #11930)

2008-02-11 03:03 +0000 [r103282-103284]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix improper indentation. Thanks again to
	  snuffy for pointing it out.

	* apps/app_queue.c: Add a couple of comments to clarify the
	  unreffing of queues. Thanks to snuffy for the idea.

	* main/event.c: Fix a problem regarding network vs. host byte order
	  in the event API. ast_event_iterator_get_ie_type should return
	  the ie type in host byte order. Furthermore, ast_event_get_ie_raw
	  should already have its ie type argument in host byte order since
	  it could be called externally (and it in fact is called in this
	  way by ast_event_get_cached).

2008-02-09 11:27 +0000 [r103249]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c,
	  apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c,
	  apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c,
	  apps/app_channelredirect.c, apps/app_directed_pickup.c,
	  apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c,
	  apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c:
	  whitespace fixes only.

2008-02-09 06:33 +0000 [r103198]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09
	  Feb 2008) | 4 lines Commit fix for being unable to send voicemail
	  from VoiceMailMain Reported by: William F Acker (via the -users
	  mailing list) Patch by: Corydon76 (license 14) ........

2008-02-08 21:26 +0000 [r103171]  Russell Bryant <russell@digium.com>

	* main/udptl.c, main/pbx.c, channels/chan_sip.c,
	  channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c,
	  main/rtp.c, channels/chan_usbradio.c, main/cdr.c,
	  channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c,
	  pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c,
	  apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c,
	  apps/app_voicemail.c: Merge changes from
	  team/mvanbaak/cli-command-audit (closes issue #8925) About a year
	  ago, as Leif Madsen and Jim van Meggelen were going over the CLI
	  commands in Asterisk 1.4 for the next version of their book, they
	  documented a lot of inconsistencies. This set of changes
	  addresses all of these issues and has been reviewed by Leif.
	  While this does introduce even more changes to the CLI command
	  structure, it makes everything consistent, which is the most
	  important thing. Thanks to all that helped with this one!

2008-02-08 18:58 +0000 [r103071-103122]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes
	  different arguments in trunk than 1.4.

	* /, apps/app_queue.c: Merged revisions 103120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb
	  2008) | 10 lines Prevent a potential three-thread deadlock. Also
	  added a comment block to explicitly state the locking order
	  necessary inside app_queue. (closes issue #11862) Reported by:
	  flujan Patches: 11862.patch uploaded by putnopvut (license 60)
	  Tested by: flujan ........

	* /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri,
	  08 Feb 2008) | 6 lines Yield the thread and return -1 if the
	  ioctl fails for Zaptel timing device. (closes issue #11891)
	  Reported by: tzafrir ........

2008-02-08 16:49 +0000 [r103044]  Russell Bryant <russell@digium.com>

	* UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At
	  the request of ManxPower, include the UPGRADE.txt from 1.2 and
	  1.4, as well. This way, if people need to go back and review what
	  was deprecated in previous major releases, it is readily
	  available to them. Thanks for the suggestion!

2008-02-08 15:31 +0000 [r102969-103018]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a network byte order issue and ensure
	  when creating an outgoing dialog that the socket always contains
	  information such as type and port. (closes issue #11916) Reported
	  by: mnnojd

	* /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb
	  2008) | 4 lines Make sure the presence of dbsecret is factored
	  into user scoring. (closes issue #11952) Reported by: bbhoss
	  ........

2008-02-07 21:37 +0000 [r102933]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: This is a combination new feature/bug fix for
	  app_chanspy. New feature: Add the 'e' option, which takes as an
	  argument a list of interfaces separated by colons. This way, you
	  will only be able to spy on this limited list of interfaces. Bug
	  fix: change some pointer checks to ast_strlen_zero so that spying
	  would work properly even if no channel was specified as the first
	  argument to chanspy. (closes issue #10072) Reported by:
	  xmarksthespot Patches:
	  bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by
	  xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak

2008-02-07 21:08 +0000 [r102906-102908]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_adsiprog.c: whitespace fixes only

	* apps/app_alarmreceiver.c: There she goes! First commit from me to
	  trunk \o/ Make app_alarmreceiver honor code guidelines and fix
	  whitespace errors. No functional changes.

2008-02-07 20:02 +0000 [r102859]  Jason Parker <jparker@digium.com>

	* /, main/features.c: Merged revisions 102858 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) |
	  7 lines Specify which digit string was matched in debug message.
	  (closes issue #11949) Reported by: dimas Patches:
	  v1-feature-debug.patch uploaded by dimas (license 88) ........

2008-02-07 16:47 +0000 [r102808]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configs/zapata.conf.sample: Merged revisions 102807 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb
	  2008) | 2 lines document usage of 'transfer' configuration option
	  for ISDN PRI switch-side transfers ........

2008-02-06 20:12 +0000 [r102777]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Add the channel's unique id to the AgentCalled
	  manager event to make it more consistent with other manager
	  events.

2008-02-06 18:01 +0000 [r102726]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2
	  lines Only consider a T.38-only INVITE compatible if we have both
	  a joint capability between us and them and if they provided T.38.
	  ........

2008-02-06 16:23 +0000 [r102700]  Terry Wilson <twilson@digium.com>

	* funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY
	  dialplan functions provided by sergee. I just added the ability
	  to set multiple fields at once after discussions with Tilghman
	  and Russell. Currently limited to 30 fields. (closes issue
	  #11887) Reported by: sergee Patches:
	  rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy
	  (license 396) Tested by: sergee, otherwiseguy

2008-02-06 15:20 +0000 [r102652]  Russell Bryant <russell@digium.com>

	* /, configs/features.conf.sample: Merged revisions 102651 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008)
	  | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets
	  inherited by outbound channels. (due to a discussion between me
	  and a user via email) ........

2008-02-06 03:05 +0000 [r102602]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05
	  Feb 2008) | 4 lines Move around some defines to unbreak ODBC
	  storage. (closes issue #11932) Reported by: snuffy ........

2008-02-06 00:08 +0000 [r102501-102550]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Remove an extra debug message I left in

	* channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
	  apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c,
	  apps/app_followme.c, main/loader.c, channels/chan_usbradio.c,
	  main/tcptls.c, res/res_agi.c, apps/app_minivm.c,
	  apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c,
	  main/astmm.c: Get rid of any remaining ast_verbose calls in the
	  code in favor of ast_verb (closes issue #11934) Reported by:
	  mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by
	  mvanbaak (license 7)

	* apps/app_voicemail.c: Change verbose messages to use the ast_verb
	  macro. (closes issue #11931) Reported by: snuffy Patches:
	  bug-11931.diff uploaded by snuffy (license 35)

2008-02-05 20:51 +0000 [r102500]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Change where priority of a goto is adjusted.
	  Partially reverts 102272. Closes issue #11929 (credit to file for
	  fix suggestion - we still <3 you)

2008-02-05 20:03 +0000 [r102454]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue,
	  05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes
	  issue #11919) Reported by: eferro Patches:
	  mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
	  Tested by: eferro ........

2008-02-05 19:58 +0000 [r102379-102452]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot
	  me.

	* /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3
	  lines If a REGISTER attempt comes in that is a retransmission of
	  a previous REGISTER do not create a new nonce value. (issue
	  #BE-381) ........

	* /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb
	  2008) | 4 lines Perform dialing asynchronously when using the
	  originate CLI command so the CLI does not appear to block.
	  (closes issue #11927) Reported by: bbhoss ........

2008-02-04 21:15 +0000 [r102329]  Tilghman Lesher <tlesher@digium.com>

	* utils/muted.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, main/asterisk.c: Merged revisions 102323 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008)
	  | 7 lines Cross-platform fix: OS X now deprecates the use of the
	  daemon(3) API. (closes issue #11908) Reported by: oej Patches:
	  20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76 ........

2008-02-04 18:39 +0000 [r102297]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Add line numbers to warning/error messages
	  (and pretty up some existing ones). (closes issue #11894)
	  Reported by: jmls Patches: chan_zap.patch uploaded by jmls
	  (license 141)

2008-02-04 15:16 +0000 [r102272]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Update handling of asyncgoto so it properly works on
	  channels that are currently executing a PBX. (closes issue
	  #11914) Reported by: arnd (closes issue #11753) Reported by:
	  johan

2008-02-04 14:37 +0000 [r102262]  Jason Parker <jparker@digium.com>

	* configs/extensions.ael.sample, configs/extensions.lua.sample:
	  Change examples to use G here also. Closes issue #11875

2008-02-04 05:32 +0000 [r102190-102238]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 102214 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03
	  Feb 2008) | 6 lines Missing braces. (closes issue #11912)
	  Reported by: dimas Patches: sprintf.patch uploaded by dimas
	  (license 88) ........

	* main/manager.c: CoreSettings and CoreStatus are missing the
	  terminating "\r\n". Also, some miscellaneous spacing and
	  initialization issues. (closes issue #11909) Reported by: srt
	  Patches: patch-11909-2.diff uploaded by srt (license 378) Tested
	  by: srt

2008-02-03 16:46 +0000 [r102091-102143]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8
	  lines Use the same CSEQ on CANCEL as on INVITE (according to RFC
	  3261) (closes issue #9492) Reported by: kryptolus Patches:
	  bug9492.txt uploaded by oej (license 306) Tested by: oej ........

	* /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8
	  lines Handle ACK and CANCEL in an invite transaction - even if we
	  get INFO transactions during the actual call setup. (closes issue
	  #10567) Reported by: jacksch Tested by: oej Patch by: oej
	  inspired by suggestions from neutrino88 in the bug tracker
	  ........

2008-02-03 06:43 +0000 [r102064]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Change the version number in the
	  configure script from 1.4 to 1.6

2008-02-02 06:10 +0000 [r101990-102037]  Russell Bryant <russell@digium.com>

	* include/asterisk/event.h: The documentation page has to be in its
	  own comment block to work, apparently. Fix it up!

	* /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008)
	  | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that
	  even though G.722 is 16 kHz, it is supposed to specified as 8 kHz
	  in the RTP, and RTP timestamps are supposed to be calculated
	  based on 8 kHz. (Apparently this is due to a bug in a spec, but
	  people follow it anyway, because it's the spec ...) ........

2008-02-01 22:12 +0000 [r101873-101943]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01
	  Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded
	  voicemail, and fixed several other bugs while I'm in the area.
	  (closes issue #11615) Reported by: jamessan Patches:
	  20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76, jamessan ........

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 101894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008)
	  | 2 lines Change detection of getifaddrs to use
	  AST_C_COMPILE_CHECK, backported from trunk (as suggested by
	  kpfleming) ........

	* res/res_config_curl.c: Fix multi, when using the LIKE query.
	  (closes issue #11889) Reported by: jmls Patches:
	  res_config_curl.patch uploaded by jmls (license 141) Tested by:
	  jmls

2008-02-01 18:24 +0000 [r101869]  Jason Parker <jparker@digium.com>

	* apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak.

2008-02-01 18:08 +0000 [r101824]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling
	  functionality by changing the config file keyword

2008-02-01 17:44 +0000 [r101823]  Jason Parker <jparker@digium.com>

	* /, apps/app_authenticate.c: Move an feof() call to before the
	  fgets(). This would have exited the loop early if you had an
	  authentication file with no newline at the end.

2008-02-01 17:28 +0000 [r101819-101821]  Russell Bryant <russell@digium.com>

	* /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01
	  Feb 2008) | 4 lines Don't overwrite the last character of a line
	  if it's not a newline. This would happen if the last line in the
	  file doesn't have a newline. (pointed out by Qwell) ........

2008-02-01 16:01 +0000 [r101773]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/acl.c: Merged revisions 101772 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008)
	  | 2 lines Compatibility fix for OpenWRT (reported by Brian
	  Capouch via the mailing list) ........

2008-02-01 06:27 +0000 [r101694-101746]  Russell Bryant <russell@digium.com>

	* apps/app_authenticate.c: simplify some code, tweak formatting,
	  and reduce indentation

	* apps/app_authenticate.c: reduce a level of indentation

	* apps/app_channelredirect.c: Get rid of a goto where there was no
	  extra cleanup happening at the exit point

	* /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31
	  Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial
	  strings for the case that no peer or hostname was provided, which
	  is the one part of the dial string that is absolutely required.
	  If it's not there, bail out. (closes issue #11897) Reported by
	  sokhapkin Patch by me ........

2008-02-01 00:08 +0000 [r101650]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_amd.c: Merged revisions 101649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan
	  2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle
	  periods of no channel activity" (closes issue #9256) Reported by:
	  cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
	  uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
	  rjain ........

2008-01-31 23:14 +0000 [r101611]  Russell Bryant <russell@digium.com>

	* /, main/translate.c, main/file.c: Merged revisions 101601 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008)
	  | 12 lines Fix a couple of places where ast_frfree() was not
	  called on a frame that came from a translator. This showed itself
	  by g729 decoders not getting released. Since the flag inside the
	  translator frame never got unset by freeing the frame to indicate
	  it was no longer in use, the translators never got destroyed, and
	  thus the g729 licenses were not released. (closes issue #11892)
	  Reported by: xrg Patches: 11892.diff uploaded by russell (license
	  2) Tested by: xrg, russell ........

2008-01-31 22:12 +0000 [r101578-101580]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Forgot an !

	* apps/app_queue.c: A change I made to accommodate the "linear"
	  strategy in trunk caused queue strategies to not be loaded from
	  realtime queues. This commit fixes that. Thanks to jmls for
	  pointing this problem out to me on IRC. This also contains some
	  changes to S_OR where it should be used. Thanks to Qwell for
	  pointing these out.

2008-01-31 21:33 +0000 [r101577]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a simple deadlock that was introduced
	  _right_ before this code got merged into trunk. (closes issue
	  #11895, reported by pj, patched by me)

2008-01-31 21:31 +0000 [r101532-101576]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Handle the case of a NULL state_interface when
	  checking a realtime member. Thanks to jmls for finding this
	  issue.

	* /, res/res_monitor.c: Merged revisions 101531 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan
	  2008) | 10 lines 1. Prevent the addition of an extra '/' to the
	  beginning of an absolute pathname. 2. If ast_monitor_change_fname
	  is called and the new filename is the same as the old, then exit
	  early and don't set the filename_changed field in the monitor
	  structure. Setting it in this case was causing ast_monitor_stop
	  to erroneously delete them. (closes issue #11741) Reported by:
	  garlew Tested by: putnopvut ........

2008-01-31 19:54 +0000 [r101483]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  101482 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) |
	  4 lines Solaris compat fixes for struct in_addr funkiness. Issue
	  #11885, patch by snuffy. ........

2008-01-31 19:43 +0000 [r101481]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 101480 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1
	  line closes issue #11845; that's the one where there's a 1004
	  byte cdr leak with every AMI Redirect to a zap channel ........

2008-01-31 19:20 +0000 [r101416-101449]  Russell Bryant <russell@digium.com>

	* /, channels/chan_agent.c: Merged revisions 101433 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31
	  Jan 2008) | 2 lines Add more missing locking of the agents list
	  ... ........

	* /, channels/chan_agent.c: Merged revisions 101413-101414 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008)
	  | 2 lines Add missing locking to the find_agent() function.
	  ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31
	  Jan 2008) | 3 lines Move the locking from find_agent() into the
	  agent dialplan function handler to ensure that the agent doesn't
	  disappear while we're looking at it. ........

2008-01-31 15:36 +0000 [r101393]  Joshua Colp <jcolp@digium.com>

	* funcs/func_realtime.c: Add missing braces. (closes issue #11886)
	  Reported by: sergee Patches: func_realtime_fix-r101392.diff
	  uploaded by sergee (license 138)

2008-01-31 05:28 +0000 [r101373]  Russell Bryant <russell@digium.com>

	* CHANGES: remove entry that is no longer in the tree

2008-01-30 23:10 +0000 [r101344]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: The deprecation of "username" in favor of
	  "defaultuser" for SIP peers unfortunately broke realtime
	  configurations which still used the "username" field. This was
	  taken care of properly when reading from realtime but was not
	  handled properly when updating a realtime peer. This change also
	  adds a deprecation NOTICE CLI message that will print if using
	  the deprecated "username" field. (closes issue #11880) Reported
	  by: cabal95 Patches: 11880.patch uploaded by putnopvut (license
	  60) Tested by: cabal95

2008-01-30 20:08 +0000 [r101322]  Olle Johansson <oej@edvina.net>

	* configs/cli.conf.sample: Clarify configuration file that can be
	  misunderstood

2008-01-30 19:03 +0000 [r101296]  Jason Parker <jparker@digium.com>

	* apps/app_controlplayback.c: Allow disabling the default
	  ffwd/rewind keys in the ControlPlayback application. This is done
	  in a backward compat way. If the "default" key for ffwd/rew is
	  used for another option (such as stop), the "default" is removed.
	  (closes issue #11754) Reported by: johan Patches:
	  app_controlplayback.c.option3.patch uploaded by johan (license
	  334) Tested by: johan, qwell

2008-01-30 17:12 +0000 [r101271]  Olle Johansson <oej@edvina.net>

	* configs/rtppage.conf.sample (removed), apps/app_rtppage.c
	  (removed): Removing applications that wasn't ready for svn trunk,
	  as trunk now has pre-release status.

2008-01-30 16:54 +0000 [r101269]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Make the VoicemailUsersList AMI command
	  consistent with other manager list functions. (closes issue
	  #11874) Reported by: srt Patches: voicemail_ami-11847.patch
	  uploaded by srt (license 378)

2008-01-30 16:39 +0000 [r101267-101268]  Olle Johansson <oej@edvina.net>

	* include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change
	  function to void because it always returned the same value and no
	  one read it.

	* main/rtp.c: Formatting fixes

2008-01-30 15:42 +0000 [r101224]  Mark Michelson <mmichelson@digium.com>

	* apps/app_rtppage.c: Get trunk to compile

2008-01-30 15:42 +0000 [r101223]  Joshua Colp <jcolp@digium.com>

	* /, main/slinfactory.c: Merged revisions 101222 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4
	  lines Fix an issue where if a frame of higher sample size
	  preceeded a frame of lower sample size and ast_slinfactory_read
	  was called with a sample size of the combined values or higher a
	  crash would happen. (closes issue #11878) Reported by: stuarth
	  ........

2008-01-30 15:36 +0000 [r101221]  Olle Johansson <oej@edvina.net>

	* CHANGES: Update CHANGES with rtppage

2008-01-30 15:35 +0000 [r101220]  Jason Parker <jparker@digium.com>

	* /, configs/extensions.conf.sample: Merged revisions 101219 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37
	  -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use
	  descending channel order of groups, rather than ascending. Fixes
	  a potential source of confusion in glare-type situations. Issue
	  11875, reported by JimVanM. ........

2008-01-30 15:30 +0000 [r101218]  Olle Johansson <oej@edvina.net>

	* configs/rtppage.conf.sample (added), apps/app_rtppage.c (added):
	  Add rtppage() application to do multicast or unicast RTP paging
	  to SIP phones. (closes issue #11797) Reported by: macbrody
	  Patches: app_rtppage-20080130.c uploaded by macbrody (license
	  352)

2008-01-30 15:27 +0000 [r101217]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 101216 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan
	  2008) | 5 lines Fix a logic error with regards to autofill. Prior
	  to this change, it was possible for a caller to go out of turn if
	  autofill were enabled and callers ahead in the queue were
	  attempting to call a member. This change fixes this. ........

2008-01-30 12:48 +0000 [r101196]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: simplify this code and eliminate the return
	  value cast that is no longer necessary

2008-01-30 11:27 +0000 [r101153-101154]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, include/asterisk/channel.h: Constifying the
	  interface to get pvt_ids in the bridge, based on suggestion from
	  (const char *) Kevin. Thanks!

	* /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7
	  lines Stop musiconhold on attended transfer. (closes issue
	  #11872) Reported by: gareth Patches: svn-101018.patch uploaded by
	  gareth (license 208) ........

2008-01-30 00:58 +0000 [r101126]  Jason Parker <jparker@digium.com>

	* CHANGES: Fix a typo

2008-01-30 00:04 +0000 [r101082]  Russell Bryant <russell@digium.com>

	* CHANGES, apps/app_speech_utils.c: Add the 'n' option to
	  SpeechBackground, which has the application not answer the
	  channel if it has not already been answered. (closes SPD-51)

2008-01-29 23:59 +0000 [r101081]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, build_tools/make_version: Merged revisions 101080 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29
	  Jan 2008) | 1 line updated build_tools to handle the autotag
	  directory structure changes; changes related to BE-353. Patch by
	  The Russell and reviewed by The Me. ........

2008-01-29 23:02 +0000 [r101036]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 101035 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan
	  2008) | 3 lines Remove a memory leak from updating realtime
	  queues ........

2008-01-29 22:04 +0000 [r101018]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c: Oops, a sizeof error

2008-01-29 19:41 +0000 [r100974]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 100973 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan
	  2008) | 6 lines Fixing an erroneous return value returned when
	  attempting to pause or unpause a queue member fails. Fixes
	  BE-366, thanks to John Bigelow for writing the patch. ........

2008-01-29 17:44 +0000 [r100933]  Russell Bryant <russell@digium.com>

	* /, main/Makefile: Merged revisions 100932 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008)
	  | 4 lines Fix the last couple of issues related to building from
	  a path that contains spaces. (closes issue #11834) ........

2008-01-29 17:42 +0000 [r100931]  Jason Parker <jparker@digium.com>

	* /, channels/misdn_config.c: Merged revisions 100930 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan
	  2008) | 6 lines Initialize an array to 0s if config option not
	  specified. (closes issue #11860) Patches:
	  misdn_get_config.v1.diff uploaded by IgorG (license 20) ........

2008-01-29 17:22 +0000 [r100900-100928]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 100922 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008)
	  | 3 lines Use GNU make magic instead of shell magic to escape
	  spaces in the working directory. (related to issue #11834)
	  ........

	* Makefile, /: Merged revisions 100882 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008)
	  | 6 lines Fix building Asterisk when the working path has spaces
	  in it. (closes issue #11834) Reported by: spendergrass Patched
	  by: me ........

2008-01-29 16:14 +0000 [r100843]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) |
	  5 lines Allow zap groups above 30 to work properly. (closes issue
	  #11590) Reported by: tbsky ........

2008-01-29 15:30 +0000 [r100833]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make externip work as documented. If no port
	  is specified it will use the value of bindport instead of always
	  being 5060. (closes issue #11858) Reported by: hmodes

2008-01-29 10:50 +0000 [r100794-100795]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29
	  Jan 2008) | 1 line fixed potential segfault in misdn show
	  channels CLI command ........

	* channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03
	  Jan 2008) | 1 line make sure frame is completely clean, before we
	  send it to asterisk as DTMF. If we don't make it clean, it
	  happens that one way audio occurs.. ........

2008-01-29 09:18 +0000 [r100741-100767]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8
	  lines (closes issue #11736) Reported by: MVF Patches:
	  bug11736-2.diff uploaded by oej (license 306) Tested by: oej,
	  MVF, revolution (russellb: This was the showstopper for the
	  release.) ........

	* channels/chan_sip.c: Removing code that wasn't supposed to be
	  there at all, only at an experimental stage before I found
	  another solution. Thanks Kevin, for reminding me.

2008-01-28  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta2 released.

2008-01-28 21:11 +0000 [r100679]  Jason Parker <jparker@digium.com>

	* build_tools/menuselect-deps.in, configs/vpb.conf.sample (added),
	  doc/tex/channelvariables.tex, makeopts.in: Reintroduce more
	  chan_vpb stuff that was removed in r100421 and r100422

2008-01-28 21:07 +0000 [r100678]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc (added), configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  channels/Makefile: Re-inserting chan_vpb into trunk.

2008-01-28 21:05 +0000 [r100677]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 100675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008)
	  | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went
	  to 't' instead of 'T') ........

2008-01-28 21:02 +0000 [r100676]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43
	  -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make
	  sure we put greeting files into the database like we do with the
	  others. Issue #11795 Reported by: dimas Patches: vmgreet.patch
	  uploaded by dimas (license 88) ........

2008-01-28 20:40 +0000 [r100632-100671]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix up some T38 state change issues. (closes
	  issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch
	  uploaded by dimas (license 88)

	* channels/chan_sip.c: Fix up two scheduling issues. In one
	  instance a scheduled item was not deleted when it should have
	  been and in the other it was scheduled again when it shouldn't
	  have been.

2008-01-28 18:41 +0000 [r100630-100631]  Russell Bryant <russell@digium.com>

	* main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge
	  of this commit was a NoOp, since res_features doesn't exist in
	  trunk. Thanks to qwell for pointing it out!

	* /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008)
	  | 5 lines For some reason, the use of this strdupa() is leading
	  to memory corruption on freebsd sparc64. This trivial workaround
	  fixes it. (closes issue #10300, closes issue #11857, reported by
	  mattias04 and Home-of-the-Brave) ........

2008-01-28 18:27 +0000 [r100628]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/logger.c: Normalize the detection for execinfo, so that
	  Linux (glibc) and other platforms with libexecinfo will generate
	  inline stack backtraces correctly.

2008-01-28 18:27 +0000 [r100627]  Russell Bryant <russell@digium.com>

	* /: Merged revisions 100626 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008)
	  | 7 lines Fix a crash in ast_masq_park_call() (issue #11342)
	  Reported by: DEA Patches: res_features-park.txt uploaded by DEA
	  (license 3) ........

2008-01-28 18:24 +0000 [r100625]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) |
	  1 line Correct a comment which made little/no sense. ........

2008-01-28 17:21 +0000 [r100565-100582]  Russell Bryant <russell@digium.com>

	* main/channel.c, channels/chan_local.c, /,
	  include/asterisk/channel.h: Merged revisions 100581 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28
	  Jan 2008) | 9 lines Make some deadlock related fixes. These bugs
	  were discovered and reported internally at Digium by Steve Pitts.
	  - Fix up chan_local to ensure that the channel lock is held
	  before the local pvt lock. - Don't hold the channel lock when
	  executing the timing function, as it can cause a deadlock when
	  using chan_local. This actually changes the code back to be how
	  it was before the change for issue #10765. But, I added some
	  other locking that I think will prevent the problem reported
	  there, as well. ........

	* main/pbx.c: Clean up some formatting, and simplify a bit of code
	  using ast_str

2008-01-28 13:57 +0000 [r100549]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't do a network byte order conversion
	  when setting the socket's port variable to that of bindaddr's. It
	  is already in the correct network byte order. (closes issue
	  #11800) Reported by: hmodes

2008-01-28 04:43 +0000 [r100514-100533]  Russell Bryant <russell@digium.com>

	* main/channel.c: Make a couple more uses of ARRAY_LEN, and convert
	  some spaces to tabs

	* main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few
	  little formatting changes

	* main/channel.c: These readlocks always fail for me on my mac, and
	  I saw it happen again today on another mac. We ignore the return
	  value of locking operations almost everywhere in Asterisk. So,
	  ignore these, as well, so Asterisk will actually work on systems
	  where this is occurring while I look into what the issue is.

2008-01-27 23:14 +0000 [r100488-100497]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, include/asterisk/sched.h,
	  channels/chan_iax2.c: With the switch to the ast_sched_replace*
	  API in trunk, we lose the correction that was just merged from
	  1.4, so this is a changeover to those APIs to use the macro
	  versions, so that we properly detect errors from ast_sched_del,
	  instead of simply ignoring the return values.

	* main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /,
	  channels/chan_sip.c, channels/chan_h323.c,
	  include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c,
	  channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged
	  revisions 100465 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008)
	  | 11 lines When deleting a task from the scheduler, ignoring the
	  return value could possibly cause memory to be accessed after it
	  is freed, which causes all sorts of random memory corruption.
	  Instead, if a deletion fails, wait a bit and try again (noting
	  that another thread could change our taskid value). (closes issue
	  #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan,
	  stuarth` ........

2008-01-25 22:54 +0000 [r100421-100422]  Jason Parker <jparker@digium.com>

	* doc/tex/channelvariables.tex: Get rid of that last little bit.

	* build_tools/menuselect-deps.in, configs/vpb.conf.sample
	  (removed), makeopts.in: Remove more remnants of chan_vpb

2008-01-25 22:39 +0000 [r100419-100420]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc (removed), configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  channels/Makefile, .cleancount: Removing chan_vpb from the tree

2008-01-25 21:26 +0000 [r100379]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) |
	  2 lines This would have never been true, since we're passing
	  (sizeof(req.data) - 1) as the len to recvfrom(). ........

2008-01-25 20:51 +0000 [r100361]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_rpt.c: correct a real problem and silence an annoying
	  compiler warning

2008-01-25 14:53 +0000 [r100344]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Insure that we are not going to pass a NULL
	  pointer to add_to_interfaces. (closes issue #11840) Reported by:
	  junky

2008-01-25 02:52 +0000 [r100325]  Joshua Colp <jcolp@digium.com>

	* main/dial.c, include/asterisk/dial.h: Add an API call that steals
	  the answered channel so that a destruction of the dialing
	  structure does not hang it up.

2008-01-24 22:58 +0000 [r100307]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as
	  the default, too

2008-01-24 22:36 +0000 [r100305-100306]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/app.h: ummm... might be good if this macro
	  argument was actually used :-)

	* include/asterisk/app.h: add the ability to define a structure
	  type for argument parsing when it would be useful to be able to
	  pass it between functions

2008-01-24 22:02 +0000 [r100266]  James Golovich <james@gnuinter.net>

	* channels/chan_sip.c: Fix simple whitespace issue

2008-01-24 22:01 +0000 [r100265]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/app.h, /: Merged revisions 100264 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24
	  Jan 2008) | 2 lines make these macros not assume that the only
	  other field in the structure is 'argc'... this is true when
	  someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
	  to define your own structure as long as it has the right fields
	  ........

2008-01-24 20:32 +0000 [r100245]  Joshua Colp <jcolp@digium.com>

	* main/features.c: Minor cosmetic change...

2008-01-24 18:35 +0000 [r100224]  James Golovich <james@gnuinter.net>

	* main/astmm.c: Increase the size of filenames stored when astmm is
	  used. If the path length was long they would be truncated and
	  grouped together with whatever matches

2008-01-24 17:47 +0000 [r100206]  Joshua Colp <jcolp@digium.com>

	* configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp
	  branch. This adds a strictrtp option to rtp.conf which drops
	  packets that do not come from the remote party. (closes issue
	  #8952) Reported by: amorsen

2008-01-24 17:24 +0000 [r100169]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.c: Merged revisions 100164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008)
	  | 2 lines Update main Asterisk copyright info to 2008 ........

2008-01-24 16:47 +0000 [r100121-100139]  Jason Parker <jparker@digium.com>

	* /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) |
	  6 lines Fix compilation on Solaris. (closes issue #11832)
	  Patches: bug-11832.diff uploaded by snuffy (license 35) ........

	* channels/chan_sip.c, main/features.c: Move chan_local dependency
	  into places (only one) that previously depended on res_features,
	  and used local channels

2008-01-24 15:54 +0000 [r100076-100112]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
	  channels/chan_mgcp.c: Remove dependency on res_features from some
	  channel drivers. It is now part of the core and no longer exists
	  as a module.

	* main/channel.c: Some more cosmetic changes.

	* main/channel.c: Add some spacing.

	* main/dial.c: Test hopefully over.

	* main/dial.c: Testing something...

2008-01-24 00:04 +0000 [r100057]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix flag bit definitions to make code from
	  issue #11049 actually work; along the way, clarify comments and
	  add some dummy flag definitions for other multi-bit flags to
	  hopefully stop this from happening in the future (closes issue
	  #11049)

2008-01-23 23:09 +0000 [r100039]  Jason Parker <jparker@digium.com>

	* res/res_features.c (removed), main/Makefile, main/features.c
	  (added), include/asterisk/_private.h, CHANGES, .cleancount,
	  main/asterisk.c, main/loader.c, include/asterisk/features.h: Move
	  code from res_features into (new file) main/features.c

2008-01-23 22:00 +0000 [r100021]  Russell Bryant <russell@digium.com>

	* CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your
	  contributions!

2008-01-23 21:11 +0000 [r99979-99980]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7
	  lines Second attempt. Don't change invitestate when receiving 18x
	  messages in CANCEL state. (issue #11736) Reported by: MVF Patch
	  by oej. ........

	* /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9
	  lines Make sure we don't cancel destruction on calls in CANCEL
	  state, even if we get 183 while waiting for answer on our CANCEL.
	  (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by
	  oej (license 306) Tested by: MVF ........

2008-01-23 20:26 +0000 [r99976]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23
	  Jan 2008) | 3 lines Fixing a typo. ........

2008-01-23 17:48 +0000 [r99922-99924]  Russell Bryant <russell@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) |
	  8 lines ChanSpy issues a beep when it starts at the beginning of
	  a list of channels to potentially spy on. However, if there were
	  no matching channels, it would beep at you over and over, which
	  is pretty annoying. Now, it will only beep once in the case that
	  there are no channels to spy on, but it will still beep again
	  once it reaches the beginning of the channel list again. (closes
	  issue #11738, patched by me) ........

	* main/tcptls.c: Fix tcptls build when openssl isn't installed
	  (closes issue #11813) Reported by: tzafrir Patches:
	  asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176)

2008-01-23 17:27 +0000 [r99920]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: since echo canceler parameters in Zaptel are
	  now signed integers, allow them during parsing

2008-01-23 15:23 +0000 [r99860]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_h323.c: Progress messages don't work (closes issue
	  #10497) Reported by: pj Patches: h323-announces-r99483.diff
	  uploaded by sergee (license 138) Tested by: pj

2008-01-23 10:18 +0000 [r99839]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure
	  that we are aware of a pending INVITE even if we're using TCP

2008-01-23 05:29 +0000 [r99696-99818]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Coding guidelines fixups

	* /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008)
	  | 8 lines When we reset the password via an external command, we
	  should also reset the password stored in the in-memory list, too
	  (otherwise it doesn't really take effect). (closes issue #11809)
	  Reported by: davetroy Patches: fix_externpass.diff uploaded by
	  davetroy (license 384) ........

	* /, res/res_odbc.c: Merged revisions 99775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008)
	  | 2 lines Oops, should have checked for a NULL obj, here, too
	  ........

	* res/res_config_ldap.c: Coding guidelines cleanup

	* /, main/acl.c: Merged revisions 99718 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008)
	  | 2 lines Just confirmed that all current platforms need this
	  header file ........

	* /: Oops

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, doc/ldap.txt (added),
	  configure.ac, configs/res_ldap.conf.sample (added),
	  res/res_config_ldap.c (added), CHANGES, makeopts.in,
	  contrib/scripts/asterisk.ldap-schema (added),
	  contrib/scripts/asterisk.ldif (added): Add res_config_ldap for
	  realtime LDAP engine. (closes issue #5768) Reported by: mguesdon
	  Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon
	  (license 121) res_ldap.conf.sample uploaded by suretec (license
	  70) asterisk-v3.1.4.ldif uploaded by suretec (license 70)
	  asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested
	  by: oej, mguesdon, suretec, cthorner

2008-01-22 21:09 +0000 [r99647-99653]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4
	  lines Thanks to Russell's education I realize that BUFSIZ has
	  changed since I learned the C language over 20 years ago...
	  Resetting chan_sip to the size of BUFSIZ that I expected in my
	  old head to avoid too heavy memory allocations on some systems.
	  ........

	* doc/tex/channelvariables.tex, CHANGES: Documentation updates for
	  BRIDGEPVTCALLID

2008-01-22 20:42 +0000 [r99646]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: Merged revisions 99643 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008)
	  | 2 lines Fix the defines for OS X (and Solaris, too) ........

2008-01-22 20:41 +0000 [r99645]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Make sure the command is not just present but is
	  also configured to be executed

2008-01-22 20:35 +0000 [r99644]  Olle Johansson <oej@edvina.net>

	* main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
	  Add a generic function to set the bridged call PVT unique id
	  string as a channel variable BRIDGEPVTCALLID This is important
	  for call tracing in log files and CDRs, so that the SIP callID
	  can be traced along servers. The CHANNEL dialplan function won't
	  work here, since the outbound channel is gone when we need the
	  Call-ID. Other channel drivers may now implement the same
	  function :-), but this patch only supports chan_sip.so. Inspired
	  by (issue #11816) Reported by: ctooley Patch by oej

2008-01-22 20:33 +0000 [r99642]  Russell Bryant <russell@digium.com>

	* configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change
	  the Asterisk CLI startup commands feature to read commands to run
	  from cli.conf after a discussion on the -dev list.

2008-01-22 17:46 +0000 [r99595-99596]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c, /, res/res_features.c,
	  channels/chan_agent.c, apps/app_followme.c: Merged revisions
	  99594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3
	  lines Add more dependencies on chan_local and add a note to the
	  description of chan_local so that people don't disable it in
	  menuselect just to clean up. ........

	* apps/app_dial.c, /: Merged revisions 99592 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5
	  lines Add dependency on chan_local to app_dial. Dial still runs
	  without chan_local, but will be missing forwarding functionality.
	  ........

2008-01-22 17:15 +0000 [r99559]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: Merged revisions 99540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008)
	  | 7 lines Ensure that we can get an address even when we don't
	  have a default route. (closes issue #9225) Reported by: junky
	  Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
	  (license 14) Tested by: oej, loloski, sergee ........

2008-01-22 16:55 +0000 [r99542]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Point out a bug in some debug counter
	  handling

2008-01-22 15:25 +0000 [r99464-99521]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add authentication options to the SIP
	  dialstring. Documentation follows separately (issue #11587)
	  Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by
	  sobomax (license 359)

	* configs/sip.conf.sample: Documentation updates

	* doc/siptls.txt: Small fixes

	* main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c,
	  include/asterisk/tcptls.h: Doxygen updates

2008-01-21 23:56 +0000 [r99427]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c, /: Merged revisions 99426 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21
	  Jan 2008) | 12 lines Fixing an issue wherein monitoring local
	  channels was not possible. During a channel masquerade, the
	  monitors on the two channels involved are swapped. In 99% of the
	  cases this results in the desired effect. However, if monitoring
	  a local channel, this caused the monitor which was on the local
	  channel to get moved onto a channel which is immediately hung up
	  after the masquerade has completed. By swapping the monitors
	  prior to the masquerade, we avoid the problem by tricking the
	  masquerade into placing the monitor back onto the channel where
	  we want it. During the investigation of the issue, the channel's
	  monitor was the only thing that was swapped in such a manner
	  which did not make sense to have done. All other variable
	  swapping made sense. ........

2008-01-21 23:25 +0000 [r99424]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Fix distinctive ring detection. Reported by:
	  milazzo Patches: drings.diff uploaded by milazzo (license 383)
	  Closes issue #11799

2008-01-21 22:32 +0000 [r99406]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Adding the
	  QUEUENAME variable to the variables set using the setqueuevar
	  option in queues.conf. Suggestion comes from Shaun2222 on IRC.

2008-01-21 21:11 +0000 [r99382-99384]  Olle Johansson <oej@edvina.net>

	* channels/chan_console.c: Remove compiler warning for
	  uninitialized variable

	* channels/chan_sip.c: Doxygen updates. The TCP/TLS code was
	  committed without any doxygen obviously. Tss tss.

	* channels/chan_sip.c: Updating doxygen

2008-01-21 18:15 +0000 [r99350]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, /, res/res_odbc.c,
	  configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21
	  Jan 2008) | 8 lines Permit the user to specify number of seconds
	  that a connection may remain idle, which fixes a crash on
	  reconnect with the MyODBC driver. (closes issue #11798) Reported
	  by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: mvanbaak ........

2008-01-21 16:02 +0000 [r99302]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4
	  lines Bump the buffer size for Via headers up to 512. There are
	  some exceptionally large Via headers out there. (closes issue
	  #11783) Reported by: ofirroval ........

2008-01-21 07:02 +0000 [r99280]  Olle Johansson <oej@edvina.net>

	* CREDITS: Update

2008-01-21 03:54 +0000 [r99265]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Change over to using ast_debug so these
	  debug messages don't always show up.

2008-01-20 07:28 +0000 [r99166-99248]  Russell Bryant <russell@digium.com>

	* channels/chan_console.c: Add a "console active" CLI command,
	  which lets you find out which console device is currently active
	  for the Asterisk CLI, or to set it. Also, knock multiple device
	  support off of the to-do list.

	* configs/console.conf.sample: correct the name of a CLI command
	  for getting available device names

	* configs/console.conf.sample, channels/chan_console.c: Merge
	  changes from team/russell/console_devices - Add support for
	  multiple devices. All devices are configured in console.conf. -
	  Add "console list devices" CLI command to show configured
	  devices. Also, changed the old "list devices" to be "list
	  available", which queries PortAudio for all audio devices that
	  are available for use.

	* /, main/slinfactory.c: Merged revisions 99187 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) |
	  4 lines Fix a couple of memory leaks with frame handling.
	  Specifically, ast_frame_free() needed to be called on the frame
	  that came from the translator to signed linear. ........

	* README: Add Cygwin as an "other" platform that is supported

	* README: Various README updates

2008-01-18  Russell Bryant  <russell@digium.com>

	* Asterisk 1.6.0-beta1 released.

2008-01-18 22:04 +0000 [r99080-99085]  Russell Bryant <russell@digium.com>

	* CREDITS, include/asterisk/http.h, main/tcptls.c (added),
	  main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
	  main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
	  configs/sip.conf.sample, CHANGES: Merge changes from
	  team/group/sip-tcptls This set of changes introduces TCP and TLS
	  support for chan_sip. There are various new options in
	  configs/sip.conf.sample that are used to enable these features.
	  Also, there is a document, doc/siptls.txt that describes some
	  things in more detail. This code was implemented by Brett Bryant
	  and James Golovich. It was reviewed by Joshua Colp and myself. A
	  number of other people participated in the testing of this code,
	  but since it was done outside of the bug tracker, I do not have
	  their names. If you were one of them, thanks a lot for the help!
	  (closes issue #4903, but with completely different code that what
	  exists there.)

	* main/frame.c, /, include/asterisk/translate.h: Merged revisions
	  99081 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) |
	  9 lines Revert adding the packed attribute, as it really doesn't
	  make sense why that would do any good. Fix the real bug, which is
	  to do the check to see if the frame came from a translator at the
	  beginning of ast_frame_free(), instead of at the end. This
	  ensures that it always gets checked, even if none of the parts of
	  the frame are malloc'd, and also ensures that we aren't looking
	  at free'd memory in the case that it is a malloc'd frame. (closes
	  issue #11792, reported by explidous, patched by me) ........

	* /, include/asterisk/translate.h: Merged revisions 99079 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) |
	  4 lines Since we're relying on the offset between the frame and
	  the beginning of the translator pvt struct, set the packed
	  attribute to make sure we get to the right place. (potential fix
	  for issue #11792) ........

2008-01-18 16:58 +0000 [r99026]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: This should at least temporarily fix a
	  problem where the 't' Dial option is incorrectly passed to the
	  transferee when built-in attended transfers are used. There is
	  still a problem with 'T', but better to fix some problems than no
	  problems while we work on it. (closes issue #7904) Reported by:
	  k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee
	  (license 138) Tested by: sergee, otherwiseguy

2008-01-18 06:58 +0000 [r99015-99018]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for
	  speed (closes issue #10723) Reported by: mnicholson Patches:
	  func-odbc-direct-execute1.diff uploaded by mnicholson (license
	  96) Tested by: Corydon76, mnicholson, falves11

	* res/res_odbc.c: Permit username and password to be NULL (which
	  enables pass-through from the layer above). Reported by: lurcher
	  Patch by: tilghman (Closes issue #11739)

	* funcs/func_cut.c: Reset default CUT delimiter back to '-'

2008-01-17 23:28 +0000 [r99006-99011]  Russell Bryant <russell@digium.com>

	* channels/chan_console.c: Make the output of "console list
	  devices" a bit prettier.

	* channels/chan_console.c: List which devices are inputs and
	  outputs in "console list devices"

	* main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for
	  ast_best_codec()

	* main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
	  Merged revisions 99004 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) |
	  10 lines Have IAX2 optimize the codec translation path just like
	  chan_sip does it. If the caller's codec is in our codec list,
	  move it to the top to avoid transcoding. (closes issue #10500)
	  Reported by: stevedavies Patches: iax-prefer-current-codec.patch
	  uploaded by stevedavies (license 184)
	  iax-prefer-current-codec.1.4.patch uploaded by stevedavies
	  (license 184) Tested by: stevedavies, pj, sheldonh ........

2008-01-17 22:22 +0000 [r99002]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing trunk IMAP build (closes issue
	  #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded
	  by DEA (license 3)

2008-01-17 20:51 +0000 [r98998]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/cflags.xml, channels/chan_zap.c,
	  main/dsp.c, configs/zapata.conf.sample: Add several busy
	  detection related defines to menuselect. Allow better busy detect
	  debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and
	  BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches:
	  busydetect_enhancement.patch uploaded by agx (license 298)
	  busydetect-r94975.diff uploaded by sergee (license 138)
	  Additional changes/cleanup by me.

2008-01-17 16:33 +0000 [r98993-98994]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: state_interface could be NULL, so use the
	  never-NULL cur->state_interface for this check

	* apps/app_queue.c: Get the device state of the state interface
	  instead of the interface when creating a new queue member. Thanks
	  to Atis Lezdins for bringing this up on the Asterisk-Dev mailing
	  list.

2008-01-17 16:21 +0000 [r98992]  Jason Parker <jparker@digium.com>

	* /, configs/zapata.conf.sample: Merged revisions 98991 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600
	  (Thu, 17 Jan 2008) | 4 lines Add a clarification about the
	  immediate= option of zapata.conf Issue 11784, patch by klaus3000.
	  ........

2008-01-17 16:17 +0000 [r98989-98990]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: major
	  reliability and performance improvement in VWMI monitoring for
	  FXO ports (code by markster, me and dbailey)

	* res/res_config_curl.c: resolve (valid) compiler warning about
	  variable that could be used before being initialized

2008-01-17 03:09 +0000 [r98988]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c, doc/tex/phoneprov.tex,
	  configs/phoneprov.conf.sample: Update res_phoneprov to default to
	  setting the SERVER variable to the IP the HTTP request for the
	  config came in on and the SERVER_PORT to the bindport setting in
	  sip.conf. I've left in the ability to override these options,
	  because I can't always guess how someone might decide to do
	  something weird with what is available to them--although needing
	  to is pretty unlikely. Documentation was updated to reflect
	  preference for not setting serveraddr, serveriface, or
	  serverport. Tested on Linux and OS X.

2008-01-17 00:13 +0000 [r98987]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Change the way the new filter feature
	  works, by allowing it to be a column NOT logged into the
	  database. This will allow more granularity of a decision
	  evaluated in the dialplan, then takes effect when posting the
	  CDR.

2008-01-17 00:05 +0000 [r98986]  Russell Bryant <russell@digium.com>

	* CHANGES, main/asterisk.c: Add support for an easy way to
	  automatically execute some Asterisk CLI commands immediately at
	  startup. Any commands in the startup_commands file in the
	  Asterisk config diretory will get executed. (closes issue #11781)
	  Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt
	  uploaded by jamesgolovich (license 176) -- With some changes by
	  me.

2008-01-16 23:08 +0000 [r98985]  Jason Parker <jparker@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build
	  against <package>_LIB, per recommendations from Russell.

2008-01-16 22:36 +0000 [r98984]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Info about res_config_curl

2008-01-16 22:20 +0000 [r98981]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c (added), main/utils.c: New module
	  res_config_curl (closes issue #11747) Reported by: Corydon76
	  Patches: res_config_curl.c uploaded by Corydon76 (license 14)
	  20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: jmls

2008-01-16 21:53 +0000 [r98978]  Russell Bryant <russell@digium.com>

	* CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the
	  changes from issue #10665 from the team/group/sip_session_timers
	  branch. This set of changes introduces SIP session timers support
	  (RFC 4028). In short, this prevents stuck SIP sessions that were
	  not properly torn down due to network or endpoint failures during
	  an established SIP session. To quote some of the documentation
	  supplied with the patch: "The SIP Session-Timers is an extension
	  of the SIP protocol that allows end-points and proxies to refresh
	  a session periodically. The sessions are kept alive by sending a
	  RE-INVITE or UPDATE request at a negotiated interval. If a
	  session refresh fails then all the entities that support Session-
	  Timers clear their internal session state. In addition, UAs
	  generate a BYE request in order to clear the state in the proxies
	  and the remote UA (this is done for the benefit of SIP entities
	  in the path that do not support Session-Timers)." (closes issue
	  #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by
	  rjain (license 226) chan_sip.c.diff uploaded by rjain (license
	  226) sip.conf.sample.diff uploaded by rjain (license 226)
	  proc_422_rsp_comment.diff uploaded by rjain (license 226)
	  chan_sip.c.cache.diff uploaded by rjain (license 226)
	  chan_sip.memalloc uploaded by rjain (license 226)
	  chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches
	  tracked in team/group/sip_session_timers, with some additional
	  fixes by russell and oej. Tested by: jtodd, rjain, loloski

2008-01-16 19:41 +0000 [r98968-98971]  Jason Parker <jparker@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Partially revert r93898, because it broke the way netsnmp was
	  being detected. rizzo, do you want to discuss so we can rethink
	  this, or do you have another way?

	* CHANGES: Add note about new update.log to CHANGES, by request of
	  jmls and further prodding by jsmith.

	* Makefile, /: Add logging for 'make update' command (also fixes
	  updates in some places). Issue #11766, initial patch by jmls.

2008-01-16 17:51 +0000 [r98967]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6
	  lines Add missing NULLs at end of two ast_load_realtimes. (closes
	  issue #11769) Reported by: tequ Patches: chaniax.patch uploaded
	  by dimas (license 88) ........

2008-01-16 17:21 +0000 [r98965]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c, /: Merged revisions 98964 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16
	  Jan 2008) | 10 lines Fix a deadlock in chan_local in
	  local_hangup. There was contention because the local_pvt was held
	  and it was attempting to lock a channel, which is the incorrect
	  locking order. (closes issue #11730) Reported by: UDI-Doug
	  Patches: 11730.patch uploaded by putnopvut (license 60) Tested
	  by: UDI-Doug ........

2008-01-16 16:06 +0000 [r98962]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Make users list static

2008-01-16 15:09 +0000 [r98954-98961]  Joshua Colp <jcolp@digium.com>

	* main/dial.c, /: Merged revisions 98960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6
	  lines Introduce a lock into the dialing API that protects it when
	  destroying the structure. (closes issue #11687) Reported by:
	  callguy Patches: 11687.diff uploaded by file (license 11)
	  ........

	* /, main/rtp.c: Merged revisions 98958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4
	  lines Add two more SDP names for ulaw and alaw. (closes issue
	  #11777) Reported by: tootai ........

	* /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6
	  lines Don't drop the old record route information when dealing
	  with packets related to a reinvite. (closes issue #11545)
	  Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by
	  kebl0155 (license 356) ........

	* channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This
	  seems to come from way back when and I can't think of a reason
	  for it being here, plus it could cause needless DNS lookups.
	  (closes issue #10983) Reported by: jtodd

2008-01-16 01:35 +0000 [r98953]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found
	  this problem with running the expr2 parser on OSX. Make the
	  #defines come out the same between the parser & lexer.

2008-01-16 01:17 +0000 [r98952]  Joshua Colp <jcolp@digium.com>

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac, makeopts.in: Merged revisions 98951 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan
	  2008) | 4 lines Add autoconf logic for speexdsp. Later versions
	  use a separate library for some things so we need to use it if
	  present in codec_speex. (closes issue #11693) Reported by: yzg
	  ........

2008-01-15 23:53 +0000 [r98948]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) |
	  11 lines Change a buffer in check_auth() to be a thread local
	  dynamically allocated buffer, instead of a massive buffer on the
	  stack. This fixes a crash reported by Qwell due to running out of
	  stack space when building with LOW_MEMORY defined. On a very
	  related note, the usage of BUFSIZ in various places in chan_sip
	  is arbitrary and careless. BUFSIZ is a system specific define. On
	  my machine, it is 8192, but by definition (according to google)
	  could be as small as 256. So, this buffer in check_auth was 16
	  kB. We don't even support SIP messages larger than 4 kB! Further
	  usage of this define should be avoided, unless it is used in the
	  proper context. ........

2008-01-15 23:52 +0000 [r98947]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
	  Add the "filter" keyword

2008-01-15 23:35 +0000 [r98944-98945]  Russell Bryant <russell@digium.com>

	* main/translate.c, include/asterisk/translate.h: Clean up
	  something I did for ABI compatability in 1.4

	* main/frame.c, /, main/translate.c, main/abstract_jb.c,
	  channels/chan_iax2.c, codecs/codec_zap.c,
	  include/asterisk/frame.h, main/rtp.c,
	  include/asterisk/translate.h: Merged revisions 98943 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15
	  Jan 2008) | 25 lines Commit a fix for some memory access errors
	  pointed out by the valgrind2.txt output on issue #11698. The
	  issue here is that it is possible for an instance of a translator
	  to get destroyed while the frame allocated as a part of the
	  translator is still being processed. Specifically, this is
	  possible anywhere between a call to ast_read() and
	  ast_frame_free(), which is _a lot_ of places in the code. The
	  reason this happens is that the channel might get masqueraded
	  during this time. During a masquerade, existing translation paths
	  get destroyed. So, this patch fixes the issue in an API and ABI
	  compatible way. (This one is for you, paravoid!) It changes an
	  int in ast_frame to be used as flag bits. The 1 bit is still used
	  to indicate that the frame contains timing information. Also, a
	  second flag has been added to indicate that the frame came from a
	  translator. When a frame with this flag gets released and has
	  this flag, a function is called in translate.c to let it know
	  that this frame is doing being processed. At this point, the flag
	  gets cleared. Also, if the translator was requested to be
	  destroyed while its internal frame still had this flag set, its
	  destruction has been deffered until it finds out that the frame
	  is no longer being processed. Admittedly, this feels like a hack.
	  But, it does fix the issue, and I was not able to think of a
	  better solution ... ........

2008-01-15 20:10 +0000 [r98895-98935]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4
	  lines Based on the boundary found move over the correct amount.
	  (closes issue #11750) Reported by: tasker ........

	* /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4
	  lines Accept "; boundary=" not just ";boundary=" in the multipart
	  mixed content type. (closes issue #11750) Reported by: tasker
	  ........

2008-01-14 22:19 +0000 [r98889]  Jason Parker <jparker@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
	  backupdeleted option to app_voicemail (closes issue #10740)
	  Reported by: ruffle Patches: app_voicemail.diff uploaded by
	  ruffle (license 201) 10740-voicemail.diff uploaded by qwell
	  (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak
	  (license 7) Tested by: blitzrage, mvanbaak, qwell

2008-01-14 22:11 +0000 [r98850-98888]  Mark Michelson <mmichelson@digium.com>

	* apps/app_directory.c: Big improvement for app_directory. This
	  patch breaks the do_directory function up so that it is more
	  easily parsed by the human brain. It also fixes some errors. I'll
	  quote dimas from the original bug description: "app_directory
	  contained some duplicate code even before addition of 'm' option.
	  Addition of that option doubled amount of that code. Worst of
	  all, there are minor differences between these code block and
	  bugs caused by these differences. 1. There is a memory leak. In
	  the 'menu' mode, result of the convert(pos) function is not freed
	  while it should be. 2. In the 'menu' mode check for
	  OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result,
	  application works in the mode opposite to what user expect
	  (checking last name when user wants the first nd vice versa). 3.
	  select_item function plays message for user using res = func1()
	  || func2() || func3()... construct. This construct loses the
	  actual value returned by ast_waitstream() for example so at the
	  end, res does not contain digit user dialed while listening to
	  the message. 4. (also in 1.4) application announces entries from
	  voicemail.conf/realtime separately from entries from users.conf.
	  I see no reason why doing so instead of building combined list.
	  5. Alot of duplicated code as already mentioned." This was tested
	  by dimas and I (I tested under valgrind). A word of caution: any
	  bug fixes that happen in app_directory in 1.4 will almost
	  certainly not merge cleanly into trunk as a result of this, but
	  it is well worth it. Huge thanks to dimas for this wonderful
	  submission. (closes issue #11744) Reported by: dimas Patches:
	  dir3.patch uploaded by dimas (license 88) Tested by: putnopvut,
	  dimas

2008-01-14 20:01 +0000 [r98830]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Make sure the user's manager secret exists, even
	  if it is blank. (closes issue #11749) Reported by: srt

2008-01-14 18:42 +0000 [r98811]  Terry Wilson <twilson@digium.com>

	* CHANGES: Add description of TOUPPER and TOLOWER dialplan
	  functions to CHANGES.

2008-01-14 17:40 +0000 [r98776]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Add proper call forwarding (all and busy)
	  support for chan_skinny. Note: NoAnswer support is currently not
	  implemented, as it would take a significant amount of work to
	  figure out how to do correctly. Closes issue #11310, patches,
	  testing, and support by DEA, mvanbaak, and myself.

2008-01-14 17:39 +0000 [r98775]  Russell Bryant <russell@digium.com>

	* /, main/translate.c: Merged revisions 98774 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) |
	  3 lines Revert a change that introduces an unacceptable
	  performance hit and is causing memory leaks ... (from rev 97973)
	  ........

2008-01-14 17:18 +0000 [r98773]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix for potential crash with vmexten

2008-01-14 16:36 +0000 [r98735-98738]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Merged revisions 98737 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan
	  2008) | 3 lines Fixing another compilation error. I'm a bit off
	  today :( ........

	* /, apps/app_queue.c: Merged revisions 98733 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan
	  2008) | 8 lines Adding explicit defaults for missing options to
	  init_queue. This is necessary because if a user either removes or
	  comments one of these options and reloads their queues, the
	  option will not reset to its default, instead maintaining the
	  value from prior to the reload. Thanks to John Bigelow for
	  pointing this error out to me. ........

2008-01-14 15:07 +0000 [r98695-98714]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Print out a warning when spaces are used in the
	  variable name in Set and MSet. It is extremely hard to debug this
	  issue so this should make it easier. (closes issue #11759)
	  Reported by: caio1982 Patches: setvar_space_warning1.diff
	  uploaded by caio1982 (license 22)

	* apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update
	  documentation. (closes issue #11763) Reported by: IgorG Patches:
	  docupd.v1.diff uploaded by IgorG (license 20)

2008-01-14 04:53 +0000 [r98558-98676]  Russell Bryant <russell@digium.com>

	* apps/app_jack.c: Add another small option for the JACK app and
	  JACK_HOOK function. The 'n' option tells JACK not to start jackd
	  automatically if it is not already running. Otherwise, the
	  default is that jackd will get started for you if it isn't
	  running already.

	* CHANGES: - Break up the Misc. section a bit with a new section
	  for Misc. New Modules - Change spacing a bit in some places for
	  consistent indentation

	* CHANGES, apps/app_jack.c (added): Bring in the code from
	  team/russell/jack/. Add a new module, app_jack, which provides
	  interfaces to JACK, the Jack Audio Connection Kit
	  (http://www.jackaudio.org/). Two interfaces are provided; there
	  is a JACK() application, and a JACK_HOOK() function. Both
	  interfaces create an input and output JACK port. The application
	  makes these ports the endpoint of the call. The audio coming from
	  the channel goes out the output port and whatever comes back in
	  on the input port is what gets sent to the channel. The
	  JACK_HOOK() function turns on a JACK audiohook on the channel.
	  This lets you run the audio coming from a channel through JACK,
	  and whatever comes back in is what gets forwarded on as the
	  channel's audio. This is very useful for building custom vocoders
	  or doing recording or analysis of the channel's audio in another
	  application. In case anyone is curious, the platform that
	  inspired me to write this is PureData (http://puredata.info/). I
	  wrote these JACK interfaces so that I could use Pd to do
	  interesting things with the audio of phone calls ...

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  configure script check for JACK.

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  Remove KDE configure script check that isn't used

	* main/audiohook.c: Remove a duplicate lock of the audiohook lock
	  when destroying manipulate audiohooks. This causes an error when
	  we attempt to destroy the lock later when freeing the audiohook.

	* main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar",
	  which allows you to set a channel variable (or function) on an
	  active channel from the CLI.

2008-01-12 18:12 +0000 [r98536]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Conversion to load manager.conf into memory did
	  not convert the password functions correctly. (Closes issue
	  #11749)

2008-01-12 05:13 +0000 [r98514]  Pari Nannapaneni <paripurnachand@digium.com>

	* /, main/http.c: merging a comment added in 1.4

2008-01-12 00:20 +0000 [r98488]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and
	  ensure that the AMI DNDState event always gets generated. (closes
	  issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded
	  by tzafrir (modified by me) (license 46)

2008-01-12 00:17 +0000 [r98487]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 98467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008)
	  | 4 lines Add a connection timeout attribute, as that was what
	  was intended with the login timeout, but ODBC divides it up into
	  2 different timeouts. (Closes issue #11745) ........

2008-01-11 23:57 +0000 [r98454]  Russell Bryant <russell@digium.com>

	* configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some
	  extra checking to help out with a potential error when trying to
	  run "make asterisk.pdf" when not all of the right packages are
	  installed. (closes issue #10763) Reported by: Corydon76 Patches:
	  20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76

2008-01-11 23:10 +0000 [r98436]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
	  'auto' signalling mode for Zaptel channels. (closes issue #11690)
	  Reported by: tzafrir Patches: signaling_to_signalling.diff
	  uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded
	  by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir
	  (license 46) zap_no_default_sig.diff uploaded by tzafrir (license
	  46) zap_signal_auto.diff uploaded by tzafrir (license 46)

2008-01-11 23:09 +0000 [r98424-98435]  Joshua Colp <jcolp@digium.com>

	* main/event.c: Goodbye again drumkilla.

	* main/event.c: drumkilla ftw.

	* main/audiohook.c: I am no longer Rockin'

	* main/audiohook.c: Testing something...

2008-01-11 22:52 +0000 [r98400]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) |
	  9 lines Fix up setting the EID on BSD based systems. (closes
	  issue #11646) Reported by: caio1982 Patches:
	  dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
	  dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
	  by: caio1982, mvanbaak ........

2008-01-11 19:53 +0000 [r98318-98334]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 98325 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6
	  lines If the incoming RTP stream changes codec force the bridge
	  to break if the other side does not support it. (closes issue
	  #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch
	  uploaded by tsearle (license 373) ........

	* /, res/res_agi.c: Merged revisions 98317 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6
	  lines If the channel is hungup during RECORD FILE send a result
	  code of -1 to be uniform with everything else. (closes issue
	  #11743) Reported by: davevg Patches: res_agi.diff uploaded by
	  davevg (license 209) ........

2008-01-11 19:12 +0000 [r98316]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 98315 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan
	  2008) | 5 lines Properly report the hangup cause as no answer
	  when someone does not answer (closes issue #10574, reported by
	  boch, patched by moy) ........

2008-01-11 19:05 +0000 [r98270-98308]  Russell Bryant <russell@digium.com>

	* codecs/codec_resample.c: Kevin noted that the thing that I
	  _actually_ changed here was that I converted a value from a
	  double, to a float, back to a double. Sure enough, when I changed
	  my interim variable back to a double, it still blows up.
	  Switching all of these to a float fixes the problem. This seems
	  like a compiler bug where a double passed as an argument isn't
	  getting properly aligned, so I'll have to see if I can replicate
	  it with a small test program. (related to issue #11725)

	* codecs/codec_resample.c: Fix a bus error that happened when
	  asterisk was built with optimizations on with platforms that
	  explode on unaligned access. I'm not exactly sure why this fixes
	  it, but it fixed it on the machine I was testing on. If it makes
	  sense to you, feel free to enlighten me. :) (closes issue #11725,
	  patched by me)

2008-01-11 18:35 +0000 [r98268-98269]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to
	  adaptive_odbc, too

	* cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix

2008-01-11 17:27 +0000 [r98220]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_followme.c: Merged revisions 98219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4
	  lines Ensure the return value of ast_bridge_call is passed back
	  up as the application return value. This is needed for transfers
	  to function so the PBX core knows to continue execution. (closes
	  issue #10327) Reported by: kkiely ........

2008-01-11 17:17 +0000 [r98218]  Russell Bryant <russell@digium.com>

	* codecs/codec_g722.c: At one point during working on this module,
	  I had the lin/lin16 versions of the framein callbacks different.
	  However, they are now the same again, so remove the duplicate
	  code and use the same functions for the lin/lin16 versions.

2008-01-11 16:08 +0000 [r98152-98193]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008)
	  | 2 lines Back out changes from revision 97077, since it wasn't
	  perfect ........

	* doc/manager_1_1.txt: Documentation updates

2008-01-11 12:51 +0000 [r98124]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Ascom phones send Flash events as SIP INFO
	  using '!' as the 'digit'

2008-01-11 03:40 +0000 [r98081-98083]  Russell Bryant <russell@digium.com>

	* codecs/codec_g722.c, main/frame.c: - Fix the last set of places
	  where incorrect assumptions were made about the sample length
	  with g722. It is _2_ samples per byte, not 1. This was all over
	  the place, and I believed it, and it is what caused me to take so
	  long to figure out what was broken. - Update copyright
	  information on codec_g722.

2008-01-11 00:54 +0000 [r98047]  Mark Michelson <mmichelson@digium.com>

	* main/translate.c: Fix "core show translation" to not output
	  information for "unknown" codecs. This fix was made in favor of
	  the proposed patch since it doesn't involve changing a core codec
	  define. (closes issue #11722, reported and initially patched by
	  caio1982, final patch by me)

2008-01-11 00:38 +0000 [r98024-98027]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new
	  global and per-peer option to chan_sip, qualifyfreq, which allows
	  you to set the qualify frequency. (closes issue #11597) Reported
	  by: wilder Patches: qualifyfreq5.patch uploaded by wilder
	  (license 362) -- with some mods by me

	* main/translate.c: Simplify this code with a suggestion from Luigi
	  on the asterisk-dev list. Instead of using is16kHz(), implement a
	  format_rate() function.

2008-01-10 23:40 +0000 [r97978]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, main/translate.c: Merged revisions 97973
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008)
	  | 6 lines 1) When we get a translated frame out, clone it,
	  because if the translator pvt is freed before we use the frame,
	  bad things happen. 2) Getting a failure from ast_sched_delete
	  means that the schedule ID is currently running. Don't just
	  ignore it. (Closes issue #11698) ........

2008-01-10 23:33 +0000 [r97974-97977]  Russell Bryant <russell@digium.com>

	* /, main/translate.c: Merged revisions 97976 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) |
	  3 lines Fix various timing calculations that made assumptions
	  that the audio being processed was at a sample rate of 8 kHz.
	  ........

	* codecs/codec_g722.c: Fix various issues in codec_g722. - The most
	  common fix being made here is to fix all of the places where the
	  number of output samples and output bytes gets updated in the
	  translator state structure. - Fix a number of other places where
	  the number of samples provided as an initialization value to a
	  struct was incorrect.

	* codecs/codec_resample.c: Fix the buffer_samples value. For signed
	  linear, the number of samples needed to fill the buffer is half
	  the buffer size.

2008-01-10 21:58 +0000 [r97933]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan
	  2008) | 6 lines Let us leave a voicemail for ourself if we have
	  logged into VoiceMailMain and chosen to leave a message. (closes
	  issue #11735, reported and patched by jamessan) ........

2008-01-10 21:46 +0000 [r97850-97890]  Steve Murphy <murf@digium.com>

	* /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged
	  revisions 97889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1
	  line Applied the same fixes for ael.flex as was done in 97849 for
	  ast_expr2.fl; overrode the normally generate yyfree func with our
	  own version that checks the pointer for non-null before passing
	  to free(). Also takes care of a little problem with 2.5.33 and
	  the use of the __STDC_VERSION__ macro. ........

	* /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged
	  revisions 97849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1
	  line This is a fix for 2 things: a problem Terry was having in
	  OSX with null pointers, which was my fault, as I probably forgot
	  to run the sed script last time I made mods. So, I moved the fix
	  into the flex input itself. Then, I found when I used flex
	  2.5.33, that it was using __STDC_VERSION__, and that's not real
	  good; so I added back in a DIFFERENT sed script to fix that
	  little mess. Tested everything, a couple different ways. Hope I
	  did no harm, at the least. ........

2008-01-10 20:13 +0000 [r97848]  Jason Parker <jparker@digium.com>

	* /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan
	  2008) | 1 line Fix a comment that is no longer true. ........

2008-01-10 20:05 +0000 [r97846]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Use the appropriate line ending for the
	  X-Asterisk-VM-Message-Type header. (closes issue #11734, reported
	  and patched by jaroth)

2008-01-10 19:07 +0000 [r97825-97826]  Terry Wilson <twilson@digium.com>

	* main/ast_expr2f.c: heh, remove patch to generated file.

	* main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was
	  getting WARNINGS under MALLOC_DEBUG)

2008-01-10 17:38 +0000 [r97805]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_odbc.c: Fix problem with timestr going out of scope
	  (Closes issue #11726, closes issue #11731)

2008-01-10 17:30 +0000 [r97745-97804]  Russell Bryant <russell@digium.com>

	* formats/format_sln16.c: minor formatting changes

	* main/translate.c: spaces to tabs

	* configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK
	  check again. I changed this to an inline implementation to fix a
	  small bug, but after a discussion with rizzo, I went to change it
	  back. Also, it turns out that the implementation of the macro
	  already supported what was needed to fix the problem.

	* pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample,
	  pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) |
	  2 lines Remove other remnants of pbx_kdeconsole ........

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
	  pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) |
	  4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in
	  ages, and nobody has complained. (closes issue #11706, reported
	  by caio1982) ........

2008-01-10 15:12 +0000 [r97698]  Joshua Colp <jcolp@digium.com>

	* funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan
	  2008) | 6 lines Don't try to copy the category from the group if
	  no category exists. (closes issue #11724) Reported by: IgorG
	  Patches: group_count.v1.patch uploaded by IgorG (license 20)
	  ........

2008-01-10 00:54 +0000 [r97657]  Russell Bryant <russell@digium.com>

	* include/asterisk.h: These prototypes are not supposed to be in
	  asterisk.h. They are already in version.h.

2008-01-10 00:50 +0000 [r97656]  Steve Murphy <murf@digium.com>

	* include/asterisk.h, channels/console_video.c, utils/astman.c,
	  channels/console_board.c, channels/vgrabbers.c: The fixes in this
	  commit are mainly to allow compiling of trunk with
	  --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly,
	  the version.c needs to be in the OBJS line; asterisk.h was chosen
	  to have the prototypes for ast_get_version, ast_get_version_num;
	  and the ASTERISK_FILE_VERSION macro needs to be used after
	  including asterisk.h in a few files. I hope I did the right
	  thing. If not, let me know.

2008-01-10 00:39 +0000 [r97655]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: oops, missed the case of a 0 permission (which
	  should mean everybody is allowed, not nobody)

2008-01-10 00:22 +0000 [r97653]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Attempt at making lookup_iface work under
	  FreeBSD. Not yet tested, but it compiles under OS X. And still
	  works under linux.

2008-01-10 00:17 +0000 [r97652]  Russell Bryant <russell@digium.com>

	* codecs/Makefile: Fix this so it doesn't force codec_g722 to get
	  relinked every time

2008-01-10 00:12 +0000 [r97651]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, main/manager.c, channels/chan_sip.c,
	  res/res_features.c, pbx/pbx_realtime.c,
	  configs/manager.conf.sample, CHANGES, channels/chan_iax2.c,
	  include/asterisk/manager.h, apps/app_stack.c, main/db.c,
	  apps/app_voicemail.c: Several manager changes: 1) Add the
	  Dialplan class, for NewExten and VarSet events, which should cut
	  down on the volume of traffic in the Call class. 2) Permit some
	  commands to be run from multiple classes, such as allowing DBGet
	  to be run from either the System or the Reporting class. 3)
	  Heavily document each class in the sample config, as there were
	  several that made no sense to be in the write= line, and two that
	  made no sense to be in the read= line (since they controlled no
	  permissions there). (Closes issue #10386)

2008-01-10 00:11 +0000 [r97641-97650]  Russell Bryant <russell@digium.com>

	* codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the
	  files changes

	* /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) |
	  2 lines Strip terminal sequences from the verbose messages
	  ........

	* configure: re-gen configure

	* configure.ac: re-add check for gtk1, which is used for
	  pbx_gtkconsole (related to issue #11706)

	* /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) |
	  3 lines Make pbx_gtkconsole build ... but doesn't actually load
	  on my system still (related to issue #11706) ........

2008-01-09 21:37 +0000 [r97634]  Terry Wilson <twilson@digium.com>

	* phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml,
	  phoneprov/polycom.xml, res/res_phoneprov.c (added),
	  funcs/func_strings.c, phoneprov/000000000000-phone.cfg,
	  configs/modules.conf.sample, main/acl.c,
	  include/asterisk/localtime.h, CHANGES,
	  configs/phoneprov.conf.sample (added), Makefile, phoneprov
	  (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c,
	  doc/tex/asterisk.tex: Added a new module, res_phoneprov, which
	  allows auto-provisioning of phones based on configuration
	  templates that use Asterisk dialplan function and variable
	  substitution. It should be possible to create phone profiles and
	  templates that work for the majority of phones provisioned over
	  http. It is currently only intended to provision a single user
	  account per phone. An example profile and set of templates for
	  Polycom phones is provided. NOTE: Polycom firmware is not
	  included, but should be placed in AST_DATA_DIR/phoneprov/configs
	  to match up with the included templates.

2008-01-09 20:30 +0000 [r97620-97623]  Jason Parker <jparker@digium.com>

	* /, main/cli.c: Merged revisions 97622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600
	  (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a
	  command could not be found. Also fix a comment which may have led
	  to this happening. Issue 11718, reported by kshumard. ........

	* /, main/cli.c: Merged revisions 97618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1
	  line Fix some locking and return value funkiness. We really
	  shouldn't be unlocking this lock inside of a function, unless we
	  locked it there too. ........

2008-01-09 18:53 +0000 [r97577]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 97575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan
	  2008) | 3 lines Part 2 of app_queue doxygen improvements. Some
	  smaller functions this time ........

2008-01-09 18:12 +0000 [r97532-97533]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: remove a wrong 'const'

	* images/kpad2.jpg: add annotations for the two message windows we
	  use.

2008-01-09 18:04 +0000 [r97531]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 97529 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) |
	  2 lines Fix saying the parking space number to the caller doing
	  the parking ... ........

2008-01-09 18:03 +0000 [r97530]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_board.c,
	  channels/console_video.h: Two changes: - support scrolling of
	  message window; - simplify the code for creating a message
	  window, and try it using a second one in the top of the keypad
	  (where we echo the dialed number). The 'skin' that supports these
	  two windows will be committed separately.

2008-01-09 17:30 +0000 [r97495]  Kevin P. Fleming <kpfleming@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008)
	  | 2 lines report the same message whether Zaptel does not have
	  transcoder support loaded or no transcoders were found ........

2008-01-09 16:59 +0000 [r97490]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09
	  Jan 2008) | 7 lines Set the caller id within the gtalk_alloc
	  function. As underlined in issue #10437 by Josh, we need to
	  prevent a possible memory leak. We only set the name part of the
	  caller id, the number part is not relevant when dealing with
	  JIDs. Closes issue #11549. ........

2008-01-09 16:44 +0000 [r97488]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c,
	  channels/console_board.c, channels/console_video.h: Implement
	  keyboard handling, and use it to enter a number to dial in the
	  'message' area under the keypad. Now you can make calls using the
	  keypad as a regular phone (or the keyboard for chars not present
	  on the keypad)

2008-01-09 16:13 +0000 [r97451]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6
	  lines Don't do conferencing totally in Zaptel if Monitor is
	  running on the channel. (closes issue #11709) Reported by:
	  BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license
	  371) ........

2008-01-09 15:45 +0000 [r97421-97449]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008)
	  | 2 lines pass the right variable to get an error string... oops
	  ........

	* channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008)
	  | 2 lines add error number output to ioctl failure messages to
	  help with debugging ........

2008-01-09 12:23 +0000 [r97389-97390]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_video.c, channels/console_video.h: implement the
	  "console startgui" and "console stopgui" commands so you can
	  start and stop the gui even outside of a call. This is convenient
	  for testing, and also for using the keypad to pick up a call, and
	  to dial a number (the latter not yet implemented, but should be
	  close).

	* channels/chan_oss.c: make get_video_desc() return the active
	  console if passed a null argument (channel).

2008-01-09 00:58 +0000 [r97364-97365]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: New option in trunk, needs strdupa to be safe,
	  too

	* /, main/editline/readline.c, main/cli.c: Merged revisions 97350
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008)
	  | 5 lines Allow filename completion on zero-length modules,
	  remove a memory leak, remove a file descriptor leak, and make
	  filename completion thread-safe. Patched and tested by tilghman.
	  (Closes issue #11681) ........

2008-01-09 00:18 +0000 [r97307-97309]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 97308 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan
	  2008) | 3 lines use the \retval doxygen command properly ........

	* /, apps/app_queue.c: Merged revisions 97304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan
	  2008) | 5 lines Part 1 of N of adding doxygen comments to
	  app_queue. I picked some of the most common functions used (which
	  also happen to be some the biggest/ugliest functions too) to
	  document first. I'm pretty new to doxygen so criticism is
	  welcome. ........

2008-01-08 23:51 +0000 [r97305]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Add a new flag 'd' (with optional context)
	  permitting any extension within that context to be entered as a
	  new extension during the playback of a voicemail greeting. Patch
	  inspired by bluecrow76, by tilghman. (Closes issue #7063)

2008-01-08 23:35 +0000 [r97280-97303]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_board.c: add copyright (most of this code was
	  written by Marta Carbone), remove some unused code, add/clarify
	  some comments.

	* images/kpad2.jpg: Add the annotation for the textarea used for
	  messages, and also change the background from white to something
	  different to show that we can make use of fonts with transparent
	  background.

	* images/font.png (added): add a font suitable for use with the
	  console GUI. The background of this particular image is
	  transparent so we can preserve the original background when we
	  draw strings.

	* channels/console_gui.c, channels/console_video.c,
	  channels/console_board.c (added), channels/Makefile: add support
	  for textareas, used for various dialog windows on the gui. The
	  main code to implement the textarea is in console_board.c, and
	  uses a simple png image with the font, blitting characters on the
	  designated areas of the main screen. Additionally we provide some
	  annotations in the image used as a skin to indicate which areas
	  are used for text messages. (images will be committed
	  separately). At the moment the dialog area is only used to
	  display a running counter, just as a proof of concept.

2008-01-08 21:56 +0000 [r97248]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c: Initialize new variable to NULL

2008-01-08 21:28 +0000 [r97203-97208]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
	  option of specifying a second interface in a member definition
	  for a queue. app_queue will monitor this second device's state
	  for the member, even though it actually calls the first
	  interface. This ability has been added for statically defined
	  queue members, realtime queue members, and dynamic queue members
	  added through the CLI, dialplan, or manager. (closes issue
	  #11603, reported by acidv)

2008-01-08 21:01 +0000 [r97199-97200]  Olle Johansson <oej@edvina.net>

	* channels/chan_console.c: Change reference to external library so
	  it appears on the extref listing
	  http://www.asterisk.org/doxygen/trunk/extref.html

	* res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is
	  out with bug fixes.

2008-01-08 20:56 +0000 [r97198]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c, /, main/utils.c: Merged revisions 97194 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008)
	  | 3 lines Increase constants to where we're less likely to hit
	  them while debugging. (Closes issue #11694) ........

2008-01-08 20:52 +0000 [r97196-97197]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: One line documentation ftw!

	* /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6
	  lines Fix various DTMF issues in chan_mgcp. (closes issue #11443)
	  Reported by: eferro Patches:
	  dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
	  337) ........

2008-01-08 20:45 +0000 [r97193]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan
	  2008) | 9 lines Making some changes designed to not allow for a
	  corrupted mailstream for a vm_state. 1. Add locking to the
	  vm_state retrieval functions so that no linked list corruption
	  occurs. 2. Make sure to always grab the persistent vm_state when
	  mailstream access is necessary. 3. Correct an incorrect return
	  value in the init_mailstream function. (closes issue #11304,
	  reported by dwhite) ........

2008-01-08 20:06 +0000 [r97153-97154]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Move common code for setting T38
	  capabilities and fix a bug with fax detection in the SIP RTP read
	  callback. It's still sort of silly... but more on that later.
	  (closes issue #11239) Reported by: dimas Patches:
	  sipt38prop.patch uploaded by dimas (license 88)

	* funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan
	  2008) | 4 lines If no group has been provided to the GROUP_COUNT
	  dialplan function then use the first one specific to the channel.
	  (closes issue #11077) Reported by: m4him ........

2008-01-08 19:06 +0000 [r97125]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008)
	  | 3 lines Apply multiple crash fixes, found in issue #11386, but
	  not completely closing that issue. ........

2008-01-08 18:42 +0000 [r97041-97103]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 97093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4
	  lines Make app_queue calls work with directed pickup. (closes
	  issue #11700) Reported by: jbauer ........

	* utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically
	  appear in extconf. (closes issue #11703) Reported by: dmartin

2008-01-07 23:03 +0000 [r96988]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: add support for cropping the keypad image
	  while displaying it. This way it can contain additional elements
	  (e.g. fonts, buttons, widgets) without having to use a zillion
	  files to store them.

2008-01-07 22:31 +0000 [r96987]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Explicitly make literal constants long
	  where they are expected to be.

2008-01-07 21:12 +0000 [r96936]  Jason Parker <jparker@digium.com>

	* main/config.c: Display a message if no config mappings are found
	  with "core show config mappings". Closes issue #11704, patch by
	  kshumard.

2008-01-07 21:10 +0000 [r96934-96935]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Document some weird casting magic that's
	  necessary to interface with the c-client

	* doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding
	  user-configurable TCP timeout settings to IMAP voicemail. This
	  could go a long way towards preventing unexplainable hangs
	  experienced by people. In the case of MWI hangs, this also will
	  mean that the SIP port isn't blocked anymore. (closes issue
	  #11665, reported by yehavi)

2008-01-07 20:48 +0000 [r96885-96933]  Russell Bryant <russell@digium.com>

	* /, configs/extensions.conf.sample: Merged revisions 96932 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r96932 | russell | 2008-01-07 14:47:52 -0600
	  (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07
	  Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com
	  ........ ................

	* configs/http.conf.sample: Add a note about viewing the default
	  set of documentation using the built-in http server

	* Makefile: If the HTML documentation exists, install it in the
	  static-http/docs directory so that it can be viewed through the
	  Asterisk http server if it is turned on.

	* build_tools/prep_tarball: Build the HTML version of the doc files
	  for tarballs, as well

	* res/res_smdi.c, /: Merged revisions 96884 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) |
	  3 lines Don't crash if something happens when setting up an SMDI
	  interface and it gets destroyed before the SMDI port handling
	  thread gets created. ........

2008-01-07 16:17 +0000 [r96862]  Kevin P. Fleming <kpfleming@digium.com>

	* formats/format_sln16.c (added): add a file-format driver for
	  16KHz signed linear... which may or may not work

2008-01-07 15:52 +0000 [r96858]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck
	  manager commands from loader.c to manager.c. Previously they
	  would get registered twice because of the way manager.c operates.
	  (closes issue #11699) Reported by: caio1982 Patches:
	  manager_module_commands1.diff uploaded by caio1982 (license 22)

2008-01-07 15:06 +0000 [r96776-96836]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: update comments to reflect reality (or at
	  least planned behaviour). minor code cleanups

	* channels/console_gui.c: resolve a load-time problem avoiding a
	  call to console_do_answer. On passing, fix dialling from the
	  keypad.

2008-01-05 23:05 +0000 [r96645-96743]  Russell Bryant <russell@digium.com>

	* res/snmp/agent.c: Convert this file over the new method of
	  getting the Asterisk version. (I don't have this building on this
	  machine, so caio1982 on IRC is going to test it for me. :) )

	* Makefile, funcs/func_version.c, main/manager.c,
	  channels/chan_sip.c, main/Makefile, build_tools/make_version_c
	  (added), include/asterisk/version.h (added), res/res_agi.c, main,
	  main/http.c, build_tools/make_version_h (removed),
	  include/asterisk, main/asterisk.c: Now that the version.h file
	  was getting properly regenerated every time the svn revision
	  changed, every module that used the version was getting rebuilt
	  after every svn update. This severly annoyed me pretty quickly,
	  so I have improved the situation. Now, instead of generating
	  version.h, main/version.c is generated. version.c includes the
	  version information, as well as a couple of API calls for modules
	  to retrieve the version. So now, only version.c will get rebuilt,
	  and the main asterisk binary relinked, which is must faster than
	  rebuilding http.c, manager.c, asterisk.c, relinking the asterisk
	  binary, chan_sip.c, func_version.c, res_agi ... The only minor
	  change in behavior here is that the version information reported
	  by chan_sip, for example, is the version of the Asterisk core,
	  and not necessarily the Asterisk version that the chan_sip module
	  came from.

	* main/pbx.c: Print out the name of a function being registered in
	  color, just like the name of applications when they get
	  registered.

	* UPGRADE.txt: Add a note about changing modules.conf since another
	  console channel driver is now present that can not be used at the
	  same time as chan_alsa or chan_oss.

	* channels/chan_console.c: Add the URL to the home page for
	  portaudio. Also add the location of the svn repository to check
	  out portaudio v19.

	* /, main/devicestate.c: Merged revisions 96644 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) |
	  2 lines Don't pass an empty string as the device name. ........

2008-01-05 01:05 +0000 [r96621]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_usbradio.c: improve chan_usbradio to use
	  indications just like chan_alsa/chan_oss do now

2008-01-04 23:12 +0000 [r96576]  Tilghman Lesher <tlesher@digium.com>

	* /, main/devicestate.c: Merged revisions 96575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008)
	  | 7 lines Fix the problem of notification of a device state
	  change to a device with a '-' in the name. Could probably do with
	  a better fix in trunk, but this bug has been open way too long
	  without a better solution. Reported by: stevedavies Patch by:
	  tilghman (Closes issue #9668) ........

2008-01-04 22:57 +0000 [r96574]  Jason Parker <jparker@digium.com>

	* /, res/res_features.c: Merged revisions 96573 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600
	  (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if
	  using PARKINGEXTEN and the slot is full. Issue 11237, patch by
	  me. ........

2008-01-04 19:35 +0000 [r96547]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008)
	  | 4 lines If you change the bindaddr in sip.conf to a non-bound
	  address and reload, sip goes kablooie. Reported and patched by:
	  one47 (Closes issue #11535) ........

2008-01-04 17:21 +0000 [r96500]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, acinclude.m4: [commit message] (closes issue
	  #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff
	  uploaded by tzafrir (license 46) (modified by me and added
	  configure script support)

2008-01-04 17:19 +0000 [r96499]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Use SASL DIGEST-MD5 authentication over
	  unsecured network connections only. This authentication mechanism
	  is implemented under the iksemel API, which makes use of GnuTLS,
	  whereas we use OpenSSL. Note : there's ongoing dicsussion at the
	  SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
	  http://ietfreport.isoc.org/ids-wg-sasl.html.

2008-01-04 16:21 +0000 [r96450]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) |
	  7 lines Make use of the temporary channel pointer while the pvt
	  is unlocked. (closes issue #11675) Reported by: flefoll Patches:
	  chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll
	  (license 244) ........

2008-01-03 23:14 +0000 [r96397-96398]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: we have to *always* use a completely silent 'make'
	  invocation for generating the module embedding rules

	* Makefile: there was no reason to add this define for non-Solaris
	  platforms

2008-01-03 22:46 +0000 [r96395]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) |
	  3 lines Don't crash if the iax2 pvt structure has been destroyed
	  before we get to this point (closes issue #11672, reported by
	  snuffy, patched by me) ........

2008-01-03 21:58 +0000 [r96301-96368]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/channel.h: Document recent API addition

	* res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03
	  Jan 2008) | 4 lines Missed initialization caused crash. Reported
	  and fixed by: tiziano (Closes issue #11671) ........

	* main/channel.c: Allow the uniqueid to be used for searching for a
	  channel in the list. Reported and initially patched by:
	  michael-fig (Closes issue #11340)

2008-01-03 20:04 +0000 [r96245-96272]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, tests/Makefile (added), tests/test_skel.c (added),
	  tests (added): add some simple infrastructure for modules to be
	  used for testing parts of Asterisk

	* channels/answer.h (removed), channels/ring10.h (removed),
	  channels/busy.h (removed), channels/ringtone.h (removed),
	  channels/Makefile, channels/chan_oss.c, channels/gentone.c
	  (removed), channels: eliminiate sound_thread() and other stuff
	  from chan_oss since Asterisk indications can handle it remove
	  gentone and all the headers containing tones that are no longer
	  needed

	* channels/chan_alsa.c: coding guidelines cleanup remove background
	  thread and all sound generation mechanisms, as the built-in
	  indications can handle everything that is needed

2008-01-03 14:47 +0000 [r96221]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03
	  Jan 2008) | 1 line when overlapdial was used and no number was
	  dialed, the call was dropped, now we just jump into the s
	  extension, which makes a lot more sense. ........

2008-01-03 06:16 +0000 [r96147-96174]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Add coordination between AMI and AGI applications,
	  with an asyncagi method Feature proposed and patched by: moy
	  (Closes issue #11282)

	* apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility
	  fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669)

2008-01-02 23:48 +0000 [r96103]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 96102 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan
	  2008) | 4 lines We need to reset the membername to NULL on each
	  iteration of this loop, otherwise the result is that multiple
	  members can have the same name, since the variable was not reset
	  on each iteration of the loop. ........

2008-01-02 23:22 +0000 [r96076-96079]  Russell Bryant <russell@digium.com>

	* channels/chan_console.c: Add support for generating a ringing
	  sound on an incoming call. This is a bit of a hack. It just asks
	  the core to generate the same tone that it would when you hear
	  ringback when making an outbound call. But hey, it works, and you
	  get the localized ring tone for the appropriate language set on
	  the channel.

	* channels/chan_console.c: Note that this module doesn't actually
	  play a ringing sound for an incoming call ... oops

	* channels/chan_console.c: Show the correct CLI command to answer
	  the call

2008-01-02 22:41 +0000 [r96073]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: actually parse and store echocan parameters
	  from zapata.conf... this *should* work <G>

2008-01-02 22:40 +0000 [r96071]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Don't
	  use AST_C_DEFINE_CHECK for the two pthread things that may not
	  actually be definitions, they could be enums for example.

2008-01-02 22:29 +0000 [r96028]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: Add curly braces around a compound if
	  statement so that trunk will build properly

2008-01-02 21:51 +0000 [r96019]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: another
	  checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS
	  ioctl if it is present, but doesn't parse any supplied parameters
	  yet (this implementation is not very memory efficient as the
	  parameters and their values will be duplicated for each channel
	  that has the same settings, but we can worry about that later
	  once it is working)

2008-01-02 21:49 +0000 [r96018]  Russell Bryant <russell@digium.com>

	* main/libresample/include/libresample.h: Add doxygen documentation
	  to libresample.h while it's still fresh on my mind

2008-01-02 21:08 +0000 [r95994]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c,
	  apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo,
	  bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue
	  #11668, reported and patched by mvanbaak)

2008-01-02 20:26 +0000 [r95947]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4
	  lines Allocate a SIP refer structure when performing a transfer
	  using BYE with Also so that the transfer information is properly
	  stored. (AST-2008-001) (closes issue #11637) Reported by:
	  greyvoip ........

2008-01-02 20:23 +0000 [r95944-95945]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Since ',' is the standard argument separator in
	  trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of
	  AST_NONSTANDARD_APP_ARGS for determining member data.

	* include/asterisk/app.h: Fix a typo in a comment.
	  AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'.

2008-01-02 19:47 +0000 [r95893-95939]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: clean up hwgain CLI command and improve docs
	  for swgain CLI command

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate
	  the macro being checked for, but just check for its existence
	  finish implementation of check for Zaptel HWGAIN support add
	  check for Zaptel ECHOCANCEL_PARAMS support

	* codecs/Makefile, include/asterisk/libresample.h (added),
	  codecs/codec_resample.c: and now just to keep the libresample
	  party going... if the functions from libresample are going to be
	  in the main Asterisk binary, it makes sense for the header that
	  defines them to be available without any special CFLAGS and to
	  out-of-tree modules building against /usr/include/asterisk

	* channels/chan_zap.c: umm... this did not compile on x86-64, and
	  could not possibly have worked on any platform as it was passing
	  string pointers to a function expecting ints

2008-01-02 18:05 +0000 [r95891]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 95890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan
	  2008) | 9 lines A change to improve the accuracy of queue logging
	  in the case where a member does not answer during the specified
	  timeout period. Prior to this change, there was a small chance
	  that the member name recorded in this case would be blank. Also
	  prior to this change, if using the ringall strategy, if no one
	  answered the call during the specified timeout, the member name
	  listed in the queue log would randomly be one of the members that
	  was rung. (closes issue #11498, reported and tested by hloubser,
	  patched by me) ........

2008-01-02 17:38 +0000 [r95888]  Jason Parker <jparker@digium.com>

	* apps/app_osplookup.c: Update osplookup documentation to use
	  commas instead of pipes. Closes issue #11666, patch by Laureano.

2008-01-02 16:20 +0000 [r95864]  Russell Bryant <russell@digium.com>

	* main/Makefile, main/translate.c: For some odd reason, the last
	  set of libresample build changes from Kevin did not work for
	  everyone, but it did for some. This set of changes makes trunk
	  start again for those having problems. Instead of building
	  libresample as a static library, it just links the object files
	  in directly with the asterisk binary.

2008-01-02 14:53 +0000 [r95816-95841]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/Makefile: fix some long-time breakage that kept
	  chan_misdn from being embedded

	* channels/Makefile: use the proper technique for including
	  submodules so that embedding will work

	* CHANGES: note that chan_console requires portaudio v19

	* configure, configure.ac: actually check for a function present in
	  libiconv (don't know how this test could have worked before) and
	  don't do the check on Linux/GNU systems because libiconv is not
	  present there and attempting to link with '-liconv' always fails
	  (it's not necessary as the iconv functionality is always
	  available)

	* main/libresample/src/filterkit.h,
	  main/libresample/src/resample.c,
	  main/libresample/win/libresample.dsp, main/libresample/configure,
	  main/libresample/Makefile.in, res/Makefile,
	  main/libresample/configure.in, main/libresample/src,
	  main/libresample/tests/testresample.c,
	  main/libresample/win/libresample.vcproj,
	  main/libresample/tests/compareresample.c, main/libresample/tests,
	  codecs/codec_resample.c, res/res_resample.c (removed),
	  main/libresample/README.txt, main/libresample/src/resamplesubs.c,
	  main/libresample/tests/resample-sndfile.c,
	  main/libresample/src/configtemplate.h,
	  main/libresample/install-sh, main/Makefile, main/translate.c,
	  main/libresample/include, main/libresample/src/resample_defs.h,
	  codecs/Makefile, main/libresample/config.guess,
	  main/libresample/config.sub, main/libresample/win,
	  main/libresample/LICENSE.txt, main/libresample (added),
	  main/libresample/Makefile.asterisk, build_tools/strip_nonapi,
	  res/libresample (removed), main/libresample/src/filterkit.c,
	  main/libresample/include/libresample.h: go back to including
	  libresample in the main Asterisk binary, but this time including
	  a small hack to ensure that it does get linked in (and also
	  modify the strip_nonapi script to leave the resample_<foo>
	  symbols alone)

2008-01-02 11:34 +0000 [r95794]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Set stream flags to zero upon initialization.
	  When the XMPP over TLS/SSL connection resets for some reason, it
	  is wrongly believed as being secured, which makes the
	  re-connection process endlessly fail. This was reported by
	  mvanbaak in issue #11644.

2008-01-02 09:16 +0000 [r95771-95772]  Luigi Rizzo <rizzo@icir.org>

	* main/loader.c: some cleanup of this code while I am trying to
	  debug a problem with gdb dying while debugging asterisk. The
	  problem seems to be related with a race in the handling of
	  module_list, which in turn is triggeded by calling dlopen() on a
	  system which uses initializers to create locks.

	* include/asterisk/module.h: There are three instances of the
	  module definition macros, which make maintaining this file very
	  error prone. This commit merges the embedded and !embedded
	  versions, and fixes the C++ version. Eventually we should move to
	  a single version of the macro. Too bad C++ doesn't like the
	  C-style struct initializers .foo = some_value

2008-01-02 04:33 +0000 [r95697-95746]  Russell Bryant <russell@digium.com>

	* res/libresample/src/resample_defs.h,
	  res/libresample/src/resample.c: Don't make libresample print out
	  debugging output

	* main/translate.c: Make the translation table show slin16

	* apps/app_meetme.c: fix a spacing issue introduced in revision
	  95443.

	* main/Makefile, res/libresample/README.txt, res/Makefile,
	  res/libresample/install-sh, res/libresample/configure,
	  res/libresample/Makefile.in, res/libresample/include,
	  codecs/Makefile, res/libresample/configure.in,
	  res/libresample/src, res/libresample/config.guess,
	  main/libresample (removed), res/libresample/config.sub,
	  res/libresample/win, codecs/codec_resample.c,
	  res/libresample/LICENSE.txt, res/libresample (added),
	  res/libresample/Makefile.asterisk, res/libresample/tests,
	  res/res_resample.c (added): Instead of linking libresample into
	  the main Asterisk binary, build it as res_resample, and mark
	  codec_resample as dependent upon res_resample. This prevents the
	  linker from optimizing away libresample, and also makes it so the
	  libresample code isn't linked in to multiple places. (I have
	  another module in a branch that needs it, too.)

2008-01-01 23:55 +0000 [r95671-95673]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: call directly the cli command to
	  implement hangup.

	* channels/vcodecs.c: prevent a panic when destroying a channel
	  with no incoming video.

	* channels/console_video.c: remove a leftover sleep(1) used for
	  debugging

2008-01-01 23:09 +0000 [r95648]  Joshua Colp <jcolp@digium.com>

	* codecs/Makefile: Fix building of codec_resample on platforms
	  other then Cygwin. On everything else it actually gets built
	  after codec_resample, so you can't exactly link it in since it
	  doesn't exist.

2008-01-01 22:21 +0000 [r95624-95625]  Luigi Rizzo <rizzo@icir.org>

	* codecs/Makefile, codecs/codec_resample.c: make codec_resample
	  build on __CYGWIN__, and make it load on FreeBSD (and probably
	  other systems as well). Both need libresample.a to be specified
	  in the linking phase, and cygwin needs <float.h> as other BSD.
	  The checks for OS-specific headers should really be moved to some
	  common header though.

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  funcs/func_iconv.c, makeopts.in: implement "configure" checks for
	  libiconv, and add the iconv dependency for func_iconv. This fixes
	  some build issues on CYGWIN and FreeBSD and probably other
	  platforms where libiconv is not there by default

2007-12-31 23:44 +0000 [r95578]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 95577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec
	  2007) | 9 lines Avoiding a potentially bad locking situation.
	  ast_merge_contexts_and_delete writelocks the conlock, then calls
	  ast_hint_extension, which attempts to readlock the same lock.
	  Recursion with read-write locks is dangerous, so the inner lock
	  needs to be removed. I did this by copying the "guts" of
	  ast_hint_extension into ast_merge_contexts_and_delete (sans the
	  extra lock). (this change is inspired by the locking problems
	  seen in issue #11080, but I have no idea if this is the
	  problematic area experienced by the reporters of that issue)
	  ........

2007-12-31 22:41 +0000 [r95501-95550]  Russell Bryant <russell@digium.com>

	* codecs/codec_resample.c: Use float.h to fix the build on FreeBSD.
	  Also, add some other platforms as they are likely the same.

	* channels/chan_console.c: Update chan_console to natively use a 16
	  kHz sample rate. If it is talking to an 8 kHz endpoint, then
	  codec_resample will automatically be used to properly resample
	  the audio before sending it to/from chan_console.

	* main/libresample/src/filterkit.h, main/libresample/README.txt,
	  main/libresample/tests/resample-sndfile.c,
	  main/libresample/src/resamplesubs.c, main/Makefile,
	  main/libresample/install-sh,
	  main/libresample/src/configtemplate.h,
	  main/libresample/src/resample.c,
	  main/libresample/win/libresample.dsp, main/libresample/configure,
	  main/libresample/Makefile.in, main/libresample/include, CHANGES,
	  main/libresample/src/resample_defs.h,
	  main/libresample/configure.in, main/libresample/src,
	  main/libresample/config.guess, codecs/Makefile,
	  main/libresample/tests/testresample.c, codecs/slin_resample_ex.h
	  (added), main/libresample/config.sub, main/libresample/win,
	  main/libresample/win/libresample.vcproj,
	  main/libresample/LICENSE.txt, main/libresample (added),
	  main/libresample/Makefile.asterisk, main/libresample/tests,
	  main/libresample/tests/compareresample.c, codecs/codec_resample.c
	  (added), main/libresample/src/filterkit.c,
	  main/libresample/include/libresample.h: Merge changes from
	  team/russell/codec_resample This commit imports libresample for
	  use in Asterisk. It also adds a new codec module, codec_resample.
	  This module uses libresample to re-sample signed linear audio
	  between 8 kHz and 16 kHz. It also provides an alternative for
	  converting between 16 kHz G.722 and 8 kHz signed linear when
	  using G.722, which will likely be useful as some people have
	  complained about volume issues when the current codec_g722
	  converts to 8 kHz signed linear. But, to test this, you will have
	  to disable the g722-to-slin and g722-to-slin16 translators in
	  codec_g722.c.

2007-12-31 20:33 +0000 [r95490]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_env.c: Merged revisions 95470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007)
	  | 3 lines Allow the default "0" to be returned if the STAT fails
	  (Closes issue #11659) ........

2007-12-31 18:46 +0000 [r95443]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c: Fix a compiler warning (closes issue #11658,
	  reported and patched by eliel)

2007-12-31 16:13 +0000 [r95383-95412]  Russell Bryant <russell@digium.com>

	* configs/console.conf.sample (added), configs/modules.conf.sample,
	  channels/chan_console.c (added), CHANGES: Merge the main set of
	  changes from team/russell/chan_console. Add a new console channel
	  driver, chan_console, which is a console channel driver that uses
	  portaudio as a cross platform audio interface. It was written to
	  provide a console channel driver that works with Mac CoreAudio,
	  but it supports a number of other audio interfaces, as well,
	  including OSS and ALSA. It could one day be the single console
	  channel driver, but does not yet have as many features as
	  chan_oss.

	* include/asterisk/channel.h: fix a spelling error in a comment

	* include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets
	  you set a config variable to a string field. (from
	  team/russell/chan_console)

	* configure, include/asterisk/autoconfig.h.in: Regenerate configure
	  script to include check for portaudio.

	* build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add
	  configure script checking for portaudio.

2007-12-29 02:02 +0000 [r95262-95313]  Luigi Rizzo <rizzo@icir.org>

	* channels/vcodecs.c, channels/console_video.c, channels/Makefile,
	  channels/console_video.h, channels/vgrabbers.c (added): Move
	  grabbers definitions to a separate file, vgrabbers.c, so it is
	  easier to add more entries. This required moving struct grab_desc
	  to the common header, and adding an entry in the Makefile. On
	  passing, cleanup some comments and file headers (some are still
	  missing).

	* channels/console_gui.c, channels/console_video.c: virtualize the
	  interface for video grabbers, which should make it easier to add
	  support for more grabbers (V4L2, firewire, and so on).

	* channels/console_video.c: Add a few entries up to 1408x1152 in
	  the table of known video resolutions. This makes it very
	  convenient to enlarge images using the right-click on the video
	  window.

	* channels/vcodecs.c, channels/console_video.c: change the
	  interface of video encapsulation routines, they only need the
	  buffer and mtu as input.

	* channels/console_gui.c, channels/vcodecs.c,
	  channels/console_video.c, channels/console_video.h: various
	  rearrangements and renaming of console_video stuff

2007-12-28 18:39 +0000 [r95233]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: The diff for this change looks really bad, but
	  all I did here was decrease the indentation of most of the
	  queue_exec function by reversing the logic of an if statement.
	  This change makes the function comply better with the coding
	  guidelines. Since this change is purely a cosmetic change to the
	  code, I am only committing the change to trunk.

2007-12-28 18:26 +0000 [r95192]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) |
	  6 lines Remove duplicate increment of the header count in the
	  add_header() function. (closes issue #11648) Reported by: makoto
	  Patch provided by sergee, committed patch by me, inspired by
	  comments from putnopvut ........

2007-12-28 16:12 +0000 [r95167]  Mark Michelson <mmichelson@digium.com>

	* apps/app_amd.c, CHANGES: Some changes to app_amd. The channel
	  name is printed in verbose messages maximumWordLength option
	  added. Duration of words that do not meet the minimum word
	  duration will be logged The duration of pre-greeting silence will
	  be logged Only consider us in the greeting if we actually
	  detected a valid word duration. (closes issue #11650, reported
	  and patched by davevg)

2007-12-28 08:57 +0000 [r95139]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_video.c: fix a small bug in printing out
	  geometries - wrong input.

2007-12-28 00:17 +0000 [r95096]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 95095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec
	  2007) | 8 lines I found a bug while browsing the queue code and
	  managed to reproduce it in a small setup. If a queue uses the
	  ringall strategy, it was possible through unfortunate coincidence
	  for a single member at a given penalty level to make app_queue
	  think that all members at that penalty level were unavailable and
	  cause the members at the next penalty level to be rung. With this
	  patch, we will only move to the next penalty level if ALL the
	  members at a given penalty level are unreachable. ........

2007-12-27 23:32 +0000 [r95073]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove
	  more unnecessary casts for NULL. main/say.c is a big offender in
	  this respect.

2007-12-27 23:28 +0000 [r95070]  Jason Parker <jparker@digium.com>

	* doc/asterisk.8, main/asterisk.c: Fix -s socket option, and
	  document it as well. Closes issue #11645, patch by Laureano.

2007-12-27 23:13 +0000 [r95068-95069]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL
	  does not need to be cast to (char *)

	* channels/chan_oss.c: remove useless casts

2007-12-27 21:41 +0000 [r95025]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 95024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) |
	  9 lines Don't report a syntax error when an empty string is
	  passed to ast_get_group. Just return 0. (closes issue #11540)
	  Reported by: tzafrir Patches: group_empty.diff uploaded by
	  tzafrir (license 46) -- slightly changed by me ........

2007-12-27 20:11 +0000 [r94978]  Mark Michelson <mmichelson@digium.com>

	* /, main/io.c: Merged revisions 94977 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec
	  2007) | 3 lines Fixing a typo in a comment. ........

2007-12-27 17:34 +0000 [r94908-94934]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6
	  lines Include types.h in chan_h323 as without it it can not be
	  compiled on some operating systems like FreeBSD to name one.
	  (closes issue #11585) Reported by: sobomax Patches:
	  chan_h323.c.diff uploaded by sobomax (license 359) ........

	* /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4
	  lines Use ast_strlen_zero to see if our_contact is set or not on
	  the dialog. It is possible for it to be a pointer to NULL.
	  (closes issue #11557) Reported by: FuriousGeorge ........

2007-12-27 17:26 +0000 [r94904]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c: more
	  localization of gui stuff

2007-12-27 17:18 +0000 [r94903]  Mark Michelson <mmichelson@digium.com>

	* doc/manager_1_1.txt: Adding documentation for new manager actions
	  and events in app_queue

2007-12-27 16:51 +0000 [r94902]  Luigi Rizzo <rizzo@icir.org>

	* CHANGES: clarify the type of video support in chan_oss

2007-12-27 16:11 +0000 [r94830-94877]  Russell Bryant <russell@digium.com>

	* codecs/codec_g722.c: I went looking for where we downloaded the
	  g722 implementation and came across these two links. So, I'm
	  adding them so they are available for reference later.

	* /, main/translate.c, include/asterisk/translate.h: Merged
	  revisions 94828-94829 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) |
	  9 lines Change ast_translator_best_choice() to only pay attention
	  to audio formats. This fixes a problem where Asterisk claims that
	  a translation path can not be found for channels involving video.
	  (closes issue #11638) Reported by: cwhuang Tested by: cwhuang
	  Patch suggested by cwhuang, with some additional changes by me.
	  ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27
	  Dec 2007) | 2 lines Use the constant that I really meant to use
	  here ... ........

2007-12-27 09:13 +0000 [r94826-94827]  Olle Johansson <oej@edvina.net>

	* funcs/func_dialplan.c: This function checks more than just
	  contexts...

	* apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: -
	  This application needs better documentation and a RESULT code in
	  the dialplan.

2007-12-27 01:03 +0000 [r94825]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c, /: Merged revisions 94824 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007)
	  | 2 lines make this comment explain the situation in an even more
	  explicit fashion ........

2007-12-27 00:48 +0000 [r94819-94823]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: more steps to decouple the gui from the
	  rest of the code.

	* channels/console_gui.c, channels/console_video.c,
	  channels/console_video.h: Enable building the code even if SDL is
	  not present (similarly, SDL is also detected at runtime). Now we
	  should be able to stream video even without a rendering device
	  (useful for remote monitoring).

	* channels/console_gui.c, channels/console_video.c: more
	  localizations around sdl_setup

	* channels/console_gui.c: use fread instead of mmap to read in the
	  comment area from the keypad. fread is simpler and more portable,
	  and there is no performance gain in using mmap.

	* images/kpad2.jpg: update the region description with an empty
	  line at the beginning.

2007-12-26 22:38 +0000 [r94818]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, channels/chan_zap.c: Allow more spans
	  than 32. Also, rearrange compiler flags so the most often used
	  flags appear closer to the top. Reported by: tzafrir Patch by:
	  tzafrir,tilghman (Closes issue #11528)

2007-12-26 22:29 +0000 [r94817]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c: another bunch
	  of gui localizations

2007-12-26 22:14 +0000 [r94814]  Jason Parker <jparker@digium.com>

	* apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean
	  up the description and usage text a bit. Closes issue #11564,
	  patch by pnlarsson (with some extra cleanup by me).

2007-12-26 22:10 +0000 [r94810-94813]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c: more
	  localization of sdl stuff

	* channels/console_gui.c, channels/console_video.c,
	  channels/console_video.h: move more gui stuff into console_gui.c

2007-12-26 20:49 +0000 [r94809]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 94808 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007)
	  | 6 lines Workaround for what is probably a glibc bug (but we'll
	  see this crop up again and again, if we don't add the
	  workaround). Reported by: rolek Patch by: tilghman (Closes issue
	  #11601, closes issue #11426) ........

2007-12-26 20:02 +0000 [r94806]  Jason Parker <jparker@digium.com>

	* pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
	  apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c,
	  apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c,
	  pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c,
	  pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use
	  defined return values in load_module in more places. (closes
	  issue #11096) Patches: pbx_config.c.patch uploaded by moy
	  (license 222) pbx_dundi.c.patch uploaded by moy (license 222)
	  pbx_gtkconsole.c.patch uploaded by moy (license 222)
	  pbx_loopback.c.patch uploaded by moy (license 222)
	  pbx_realtime.c.patch uploaded by moy (license 222)
	  pbx_spool.c.patch uploaded by moy (license 222)
	  app_adsiprog.c.patch uploaded by moy (license 222)
	  app_alarmreceiver.c.patch uploaded by moy (license 222)
	  app_amd.c.patch uploaded by moy (license 222)
	  app_authenticate.c.patch uploaded by moy (license 222)
	  app_cdr.c.patch uploaded by moy (license 222)
	  app_zapateller.c.patch uploaded by moy (license 222)
	  app_zapbarge.c.patch uploaded by moy (license 222)
	  app_zapras.c.patch uploaded by moy (license 222)
	  app_zapscan.c.patch uploaded by moy (license 222)

2007-12-26 20:01 +0000 [r94805]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/vcodecs.c,
	  channels/console_video.c, channels/console_video.h: more
	  preparation for untangling of the various console_video stuff

2007-12-26 19:09 +0000 [r94796-94802]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: Merged revisions 94801 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) |
	  4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already
	  set before starting autoservice, remember it and ensure that the
	  channel has the same setting when autoservice gets stopped.
	  (pointed out by d1mas, patched up by me) ........

	* funcs/func_dialplan.c (added), CHANGES: Add a new dialplan
	  function, DIALPLAN_EXISTS(), which allows you to check for the
	  existence of a dialplan target. (closes issue #11579) Reported
	  by: irroot Patches: func_dialplan2.c uploaded by irroot (license
	  52) -- Additional changes by me.

	* main/autoservice.c, /: Merged revisions 94797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) |
	  4 lines When a channel is in autoservice, mark a flag on the
	  channel that says that we only care about the END of a digit.
	  That way, no magic digit emulation stuff will happen when all
	  we're doing is queueing up END frames. ........

	* main/channel.c: Leave a note for a minor bug that was pointed out
	  by d1mas

2007-12-26 18:05 +0000 [r94795]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Convert raw bits for callprogress bitfield
	  to use constants, for greater code clarity Reported by: dimas
	  Patch by: dimas (Closes issue #11280)

2007-12-26 17:26 +0000 [r94787-94794]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 94793 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) |
	  3 lines Don't try to send a parked call back to itself. (closes
	  issue #11622, reported by djrodman, patched by me) ........

	* Makefile, /: Merged revisions 94789 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) |
	  5 lines List include/asterisk/version.h as a .PHONY target
	  because we want the commands listed for this target to be
	  executed regardless of whether the file exists or not. This fixes
	  having the version not up to date when running from svn. (closes
	  issue #11619, reported by plack, fixed by me) ........

	* main/autoservice.c, /: Merged revisions 94790 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) |
	  5 lines Don't store DTMF BEGIN frames while a channel is in
	  autoservice. It's just going to make ast_read() do a lot of extra
	  work when the channel comes back out of autoservice. (closes
	  issue #11628, patched by me) ........

	* channels/chan_iax2.c: Fix a bug in peer handling that caused
	  multiple instances of a peer to end up in the peers container
	  after a reload. Somehow, this bug doesn't exist in 1.4 ...
	  (closes issue #11626) (reported by pnlarsson, additional info
	  from mvanbaak, fixed by me)

	* utils: update svn:ignore for astcanary

2007-12-26 15:58 +0000 [r94782]  Mark Michelson <mmichelson@digium.com>

	* configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding
	  support for storing the queue log entries in a realtime backend.
	  (closes issue #11625, reported and patched by sergee) Thank you
	  very much to sergee for adding this new feature!

2007-12-26 10:14 +0000 [r94774]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c (added), channels/vcodecs.c (added),
	  channels/console_video.c: Split console_video.c so that video
	  codecs and gui functions are in separate files (still #include'd
	  because of tangling in the data structures, but this is going to
	  be cleaned up). The video grabbing functions still need to be
	  moved to a separate file.

2007-12-25 04:10 +0000 [r94771-94773]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_pickupchan.c (added): Add pickup by channel (Closes
	  issue #11161)

	* channels/chan_zap.c, configs/zapata.conf.sample: Change the
	  abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF
	  character. Also, fix the documentation to match the code.

	* res/res_agi.c: Add channel thread ID to the information passed to
	  AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162)

2007-12-24 19:43 +0000 [r94764-94768]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 94767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007)
	  | 5 lines Race: we need to wait to queue a NewChannel event until
	  after the channel is inserted into the channel list. The reason
	  is because some manager users immediately queue requests from the
	  channel when they see that event and are confused when Asterisk
	  reports no such channel. (Closes issue #11632) ........

	* /: Merged revisions 94763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007)
	  | 5 lines Another bit of bad logic in realtime_peer Reported by:
	  dimas Patch by: dimas (Closes issue #11631) ........

2007-12-23 14:51 +0000 [r94713-94741]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_video.c, channels/console_video.h: support
	  sdl_videodriver to send output to x11/aalib/console

	* channels/console_video.c: move reading info from the keypad to a
	  separate function. Remove an unused keypad field and some
	  debugging messages. Adjust formatting on config file parsing

	* channels/console_video.c: make sure the minimum surface depth is
	  16bpp so we can create YUVoverlays. With this change we can do
	  setenv SDL_VIDEODRIVER aalib and output to an ascii window (which
	  is still in an X11 window). If you also do unsetenv DISPLAY then
	  the output goes into the main asterisk window, unfortunately it
	  interferes with the normal output so you don't see much. In any
	  case, i don't think we are very far away from having a working
	  xterm videophone!

	* channels/Makefile: avoid rebuilding dependent files if the
	  generated busy.h and ringtone.h do not change. Ths masks (but
	  does not solve) a but that i am seeing in doing a 'gmake install'
	  without donig a 'gmake all' first.

2007-12-23 01:38 +0000 [r94662]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007)
	  | 2 lines Argh... I suppose third time's the charm. ........

2007-12-22 22:44 +0000 [r94615-94638]  Luigi Rizzo <rizzo@icir.org>

	* configs/oss.conf.sample, channels/console_video.c: Change the
	  name of config file entries for keypad regions from
	  'keypad_entry' to 'region'. Fix the example file accordingly.
	  Also make some fixes in the code do reset entries on reload of
	  the keypad. The recently committed kpad2.jpg has the correct
	  names.

	* images/kpad2.jpg (added): add a sample keypad (with annotations)
	  for console video

	* channels/console_video.c, channels/Makefile, channels/chan_oss.c,
	  channels/console_video.h (added): Build console_video support by
	  linking in, as opposed to including, console_video.c This will
	  ease the task of splitting console_video.c into its components
	  (V4L and X11 grabbers, various video codecs and packetizers,
	  SDL), as well as ease future extensions (e.g. additional video
	  sources, codecs and rendering engines). For the time being
	  nothing changes for users: video support is off by default, and
	  requires -DHAVE_VIDEO_CONSOLE on the command line to be included
	  (if SDL and FFMPEG are available).

2007-12-21 21:19 +0000 [r94593]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Something I've been itching to do for a
	  while now. A minor optimization in app_voicemail. Since the
	  dtable in base_encode always gets populated with the same values
	  every time and never changes, make it static and const and only
	  initialize it once. Also, there's no reason to define
	  BASEMAXINLINE twice, so remove the redundant #define.

2007-12-21 20:50 +0000 [r94549-94551]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: We should only clear this value if we have
	  to

	* channels/chan_zap.c: Commit non TCP transport part of #11506.
	  Includes numerous additional parameters, as well as RLT support
	  for DMS type switches

2007-12-21 20:38 +0000 [r94542-94548]  Mark Michelson <mmichelson@digium.com>

	* res/res_config_sqlite.c: Store dates using local time instead of
	  UTC (closes issue #11610, reported and patched by
	  rbraun_performatique)

	* apps/app_queue.c: Fix a memory leak when reloading queue rules.

	* CHANGES: The one documentation source I forgot to update after
	  the merge of the queue-penalty branch was the CHANGES file. No
	  longer!

	* apps/app_voicemail.c: Lots of coding guidelines cleanup.

	* /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec
	  2007) | 8 lines Better quota support for using IMAP storage
	  voicemail (closes issue #11415, reported by jaroth) (closes issue
	  #11152, reported by selsky) Patch provided by jaroth ........

2007-12-21 20:12 +0000 [r94541]  Jason Parker <jparker@digium.com>

	* codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
	  codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c,
	  codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c:
	  codecs.conf really shouldn't be mandatory.. it never had been
	  before, so let's go back to being optional. A big "thank you" to
	  pnlarsson on IRC for allowing me access to his system to debug
	  this. Closes issue #11584.

2007-12-21 20:01 +0000 [r94477-94539]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec
	  2007) | 5 lines The mail_copy c-client function does not expect a
	  full imap mailbox string, just the name of the mailbox. (closes
	  issue #11419, reported and patched by jaroth, with additional
	  patchwork from me) ........

	* main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in
	  trunk

	* main/dial.c, /: Merged revisions 94468 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec
	  2007) | 6 lines Since we are freeing list elements within a list
	  traversal, we need to use the safe traversal and remove the item
	  from the list before freeing it. (closes issue 11612, reported by
	  dtyoo) ........

2007-12-21 16:12 +0000 [r94463-94465]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 94464 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec
	  2007) | 3 lines Removing a debug message I accidentally just
	  committed ........

	* /, main/say.c, apps/app_queue.c: Merged revisions 94420 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec
	  2007) | 5 lines Fixing Portuguese syntax for saying dates and
	  times. Also some coding guidelines cleanup. (closes issue #11599,
	  reported and patched by caio1982, coding guidelines cleanup by
	  me) ........

2007-12-21 15:14 +0000 [r94419]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 94418 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007)
	  | 2 lines Fix for restart-as-user problem reported via the -dev
	  list ........

2007-12-21 01:14 +0000 [r94345-94396]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Moved the update of the queue_ent's rule list
	  to just before we try to call queue members. This allows for the
	  change in penalty levels to be executed at the most logical time
	  frame.

	* configs/queues.conf.sample, doc/tex/channelvariables.tex,
	  apps/app_queue.c, configs/queuerules.conf.sample (added): Merging
	  the queue-penalty branch. In short, this allows one to
	  dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced
	  QUEUE_MIN_PENALTY during a call depending on the amount of time
	  passed. The purpose is to allow the call to open up to more (or
	  maybe just different) members without the caller's losing his
	  place in the queue. See configs/queuerules.conf.sample for an
	  example of how to set up queue rules and
	  configs/queues.conf.sample for how to associate a rule with a
	  queue. Along with the functional changes, new CLI and manager
	  commands exist to show the rules defined and there is an
	  additional CLI command to reload the queue rules. Future
	  enhancements that may be made: support for realtime queue rules
	  and support for dynamically adding a rule through the manager or
	  CLI. Also a manager command to reload the queue rules (I'll
	  probably write this myself very soon).

	* apps/app_voicemail.c: The changes to header inclusion in trunk
	  broke compilation of app_voicemail when using IMAP storage. The
	  reason is that c-client has its own definitions for LOG_WARNING
	  and LOG_DEBUG, so we need to be sure to include asterisk's
	  definitions last so that we use the proper values in
	  app_voicemail. (closes issue #11437, reported by blitzrage, patch
	  suggested by blitzrage)

2007-12-20 22:39 +0000 [r94320]  Russell Bryant <russell@digium.com>

	* configs/zapata.conf.sample: Add a bit more to the description of
	  the "mwimonitor" option.

2007-12-20 22:28 +0000 [r94319]  Steve Murphy <murf@digium.com>

	* build_tools/make_buildopts_h: closes issue #11287; thanks to
	  snuffy for this fix, which will surely make all solaris owners
	  shout praises to his name.

2007-12-20 20:25 +0000 [r94252-94257]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r94256 | russell | 2007-12-20 14:22:22 -0600
	  (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20
	  Dec 2007) | 5 lines Fix another potential seg fault ... (closes
	  issue #11606) Reported by: dimas ........ ................

	* channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) |
	  10 lines Fix a deadlock in d-channel handling in chan_zap. This
	  deadlock was introduced by the fix to ensure that channels are
	  properly locked when handling channel variables. There were
	  sections of this code where the channel pvt was locked before the
	  channel lock, when in fact it _must_ be the other way around.
	  (closes issue #11582) Reported by: bugi ........

2007-12-20 12:56 +0000 [r94168-94191]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_usbradio.c, include/asterisk/config.h,
	  channels/console_video.c, channels/chan_oss.c: add some macros to
	  simplify parsing the config file, see description in config.h .
	  They are a variant of the set of macros i used in chan_oss.c,
	  structured in a way to be more robust to the presence of spurious
	  ';' - basically, they define wrappers for 'do {' and '} while
	  (0)', plus some helper functions to deal with simple cases such
	  as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix
	  (CV_ as 'Config Variable') tries to be easy to remember and has
	  been chosen to not conflict with other existing macros in the
	  tree. For the time being, I have only updated the three source
	  files in the tree that used the old M_* macros. Hopefully, more
	  files will be converted. NOTE: I understand that inventing my own
	  dialect of C is generally wrong; however, the lack of adequate
	  support in the language encourages lazy programming practices
	  (such as ignoring errors, bounds, etc.) and this increases the
	  chance of vulnerability in the code, especially because we are
	  parsing user input here. Hopefully, these macros and the use of
	  ast_parse_arg (in config.h) should encourage the programmer to
	  write more robust code.

	* include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c,
	  utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify
	  http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that
	  paths and filename are writable by asterisk.c without causing
	  segfaults. This involves defining the variables as const char *,
	  and having them point to as static, writable buffer defined in
	  asterisk.c On passing, fix some errors in using these variables
	  in some files in utils/ , and in res/snmp/agent.c which was
	  redefining a variable without using paths.h (not applicable to
	  1.4)

2007-12-19 23:17 +0000 [r94123-94124]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 1. Unify the check for a penalty < 0 into the
	  set_member_penalty code. 2. Fix an error when checking the CLI
	  command for setting a member's penalty. 3. Fix a logging error if
	  the incorrect parameter was the queue name or interface. (closes
	  issue #11544, reported and patched by Laureano)

	* /, res/res_monitor.c: Merged revisions 94122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec
	  2007) | 6 lines Sox versions 13.0.0 and newer do not have
	  "soxmix" and instead use sox -m. res_monitor needs to use this if
	  the user does not have soxmix. (closes issue #11589, reported by
	  amessina, patch inspired by amessina but with a flourish from me)
	  ........

2007-12-19 22:51 +0000 [r94085]  Russell Bryant <russell@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 94077 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) |
	  4 lines Check for the existence of the soxmix application on the
	  target platform and have the result available in autoconfig.h.
	  (part of issue #11589) ........

2007-12-19 20:20 +0000 [r94052-94053]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Add 'voicemail reload' command. Reported
	  by: eliel Patch by: eliel (Closes issue #11365)

	* apps/app_waituntil.c (added): Add contributed WaitUntil app.
	  Original code by pprindeville, updated for trunk by tilghman.
	  (Closes issue #11487)

2007-12-19 19:29 +0000 [r94029]  Russell Bryant <russell@digium.com>

	* include/asterisk/time.h: Add a couple of new time API calls -
	  ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported
	  by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license
	  88)

2007-12-19 17:58 +0000 [r94002]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_video.c: Add instructions on how to generate
	  your own font.

2007-12-19 17:31 +0000 [r93956]  Joshua Colp <jcolp@digium.com>

	* /: Merged revisions 93955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2
	  lines Make the 1.4 builders happy, ensure var is NULL. ........

2007-12-19 17:13 +0000 [r93952]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007)
	  | 3 lines Avoid segfault in chan_iax when peer isn't defined
	  (Closes issue #11602) ........

2007-12-19 17:09 +0000 [r93925-93950]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c, include/asterisk/strings.h: Add a new API function,
	  written at least twice in app_voicemail.c and likely in other
	  places too. This is quite useful when placing mail/html stuff in
	  config files. /*! \brief Convert some C escape sequences
	  (\b\f\n\r\t) into the equivalent characters. \brief s The string
	  to be converted (will be modified). \return The converted string.
	  */ char *ast_unescape_c(char *s);

	* include/asterisk/config.h, main/config.c: add support for
	  PARSE_DOUBLE, and remove identifiers for types not supported
	  (INT16 and UINT16)

2007-12-19 09:20 +0000 [r93899]  Olle Johansson <oej@edvina.net>

	* CHANGES: Reorganize CHANGES a bit. The "misc" section grew too
	  large...

2007-12-19 08:57 +0000 [r93898]  Luigi Rizzo <rizzo@icir.org>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK()
	  and use it to check for NETSMP and GTK (GTK is not used thoug).
	  AST_EXT_TOOL_CHECK() could be used for checking curl status as
	  well, perhaps with a small addition because we currently seem to
	  require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE
	  entry in makeopts.in We don't have yet any macros for using
	  pkg-config to check for a specific package (right now there is
	  only gtk2+ in the category).

2007-12-19 08:57 +0000 [r93897]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the
	  ability to specify the To: header in an outbound INVITE by adding
	  an exclamation mark to the dial string. This patch also exists
	  for 1.4 in the fixtoheader-1.4 branch and has been in production
	  for quite some time.

2007-12-19 08:12 +0000 [r93875]  Luigi Rizzo <rizzo@icir.org>

	* res/snmp/agent.c: make netsmp build under AST_DEVMODE.
	  Description, included in the source, is below. I should note that
	  the PACKAGE_* macros that asterisk defines in autoconfig.h are
	  not used anywhere in the tree so they should just be removed. /*
	  * There is some collision collision between netsmp and asterisk
	  names, * causing build under AST_DEVMODE to fail. * * The
	  following PACKAGE_* macros are one place. * Also netsnmp has an
	  improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H
	  instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define
	  it to 0, however this will fail * when the proper check is
	  implemented. */ No

2007-12-19 07:01 +0000 [r93854]  Olle Johansson <oej@edvina.net>

	* CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for
	  starting remote Asterisk by naming the actual runtime socket
	  instead of pointing to configuration file with -C Reported by:
	  sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax
	  (license 359) doc changes by committer (closes issue #11598)

2007-12-19 00:09 +0000 [r93827]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_osplookup.c: add missing header file

2007-12-18 23:38 +0000 [r93804-93805]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Making the canary error message a little more
	  obvious.

	* utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a
	  canary process, for high priority mode (asterisk -p) to ensure
	  that if Asterisk goes into a busy loop, the machine will be
	  recoverable. We'd still need to do a restart to put Asterisk back
	  into high priority mode, but at least a reboot won't be required.
	  (Closes issue #11559)

2007-12-18 21:13 +0000 [r93741]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Move some warnings away to debug since some
	  devices send a packet with a silly string as a NAT keepalive
	  packet.

2007-12-18 18:39 +0000 [r93672]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  93668 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600
	  (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18
	  Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119)
	  ........ ................

2007-12-18 18:20 +0000 [r93666]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/paths.h: remove a leftover line with only a '#'
	  (wonder why the compiler does not complain!) and variables that
	  are only used in asterisk.c

2007-12-18 17:05 +0000 [r93626]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 93625 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec
	  2007) | 6 lines Rework deadlock avoidance used in ast_write,
	  since it meant that agent channels which were being monitored had
	  one audio file recorded and one empty audio file saved. (closes
	  issue #11529, reported by atis patched by me) ........

2007-12-18 10:24 +0000 [r93558-93603]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c,
	  utils/ael_main.c, utils/extconf.c, main/asterisk.c,
	  res/res_monitor.c, utils/conf2ael.c: make configuration variable
	  const so they are not accidentally modified. This requires
	  casting the strings in asterisk.c when writing to them, so we do
	  it through a macro to do it consistently.

	* channels/chan_unistim.c, res/res_crypto.c, main/astmm.c,
	  apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c,
	  main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c,
	  cdr/cdr_csv.c: remove unnecessary (char *) casts for
	  ast_config_AST_* variables. There are some left in the .flex
	  files, left to the maintainer...

	* build_tools/make_defaults_h, main/asterisk.c: Rename the macros
	  in defaults.h - they are not meant to be globally visible.
	  Document the fact that DEFAULT_TMP_DIR cannot be overridden from
	  the default configuration (this needs to be fixed, as you could
	  have a totally different spooldir configured at runtime, and yet
	  DEFAULT_TMP_DIR keeps the compile-time default). Remove two
	  unused entries for sounds and images.

	* Makefile.moddir_rules: make the code match documentation - now
	  you can specify multiple words in MODULE_PREFIX.

	* CREDITS: Name the people responsible for some recent
	  contributions to the tree.

	* Makefile: Two small changes: + document the difference between
	  "A=foo make ..." and "make A=foo ..." and suggest using
	  COPTS/LDOPTS if you need to use the second form to pass compiler
	  and loader flags; + define only in one place the environment used
	  to build stuff in menuselect/

2007-12-18 07:56 +0000 [r93557]  Olle Johansson <oej@edvina.net>

	* doc/CODING-GUIDELINES: A minor update, caused by a recent bug
	  report ;-)

2007-12-18 07:22 +0000 [r93536]  Luigi Rizzo <rizzo@icir.org>

	* doc/CODING-GUIDELINES: small documentation update (nothing
	  important).

2007-12-18 02:57 +0000 [r93514]  Joshua Colp <jcolp@digium.com>

	* channels/chan_unistim.c: You... will... build! I say so and
	  therefore you will.

2007-12-18 02:42 +0000 [r93493]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_unistim.c, include/asterisk/threadstorage.h: minor
	  cleanups

2007-12-17 23:10 +0000 [r93464]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_unistim.c: fix building under cygwin. At this point
	  WINARCH should go away.

2007-12-17 22:54 +0000 [r93405]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_unistim.c: remove some unnecessary includes

2007-12-17 22:50 +0000 [r93390]  Jason Parker <jparker@digium.com>

	* /, main/translate.c: Merged revisions 93381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4
	  lines What was I thinking when I wrote this masterpiece? -1 + 1 =
	  0.. who woulda thunk it?. ........

2007-12-17 22:38 +0000 [r93380]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: surprising as it may be, chan_oss compiles
	  correctly under cygwin as well, provided you look for soundcard.h
	  in the right place...

2007-12-17 22:29 +0000 [r93378]  Joshua Colp <jcolp@digium.com>

	* /, main/utils.c: Merged revisions 93377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7
	  lines Do not try to access information about a lock when printing
	  out a trylock attempt. It is possible for the lock that it
	  references to no longer be valid. This would have caused
	  segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
	  Reported by: callguy (closes issue #11100) Reported by: callguy
	  ........

2007-12-17 21:14 +0000 [r93337]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/time.h: Merged revisions 93336 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17
	  Dec 2007) | 6 lines Today is tomorrow's yesterday, and
	  yesterday's tomorrow is today, and tomorrow's tomorrow is the day
	  after tomorrow, so who cares if you recycle anyway? If this
	  confuses you, that's nothing compared to what this fixes. ;-)
	  ........

2007-12-17 21:12 +0000 [r93335]  Olle Johansson <oej@edvina.net>

	* channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c,
	  channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions
	  93182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8
	  lines Issue 11574: Add dependencies on res_monitor and
	  res_features. I wonder if Asterisk can run at all without
	  res_features. My guess is that there's propably a lot of more
	  modules and the core that depends on it. Reported by: caio1982
	  (closes issue #11574) ........

2007-12-17 20:42 +0000 [r93293-93297]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing some leftover debug messages from a
	  while back.

	* /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec
	  2007) | 6 lines We need to create the directory for a voicemail
	  user even if they are using IMAP storage since greetings are
	  stored in the filesystem. (closes issue #11388, reported by
	  spditner, patch by me inspired by a patch by spditner) ........

2007-12-17 18:07 +0000 [r93252]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6
	  lines If a call is received with a called number IE containing
	  nothing go to the 's' extension. (closes issue #9099) Reported
	  by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by
	  Corydon76 (license 14) ........

2007-12-17 17:16 +0000 [r93191-93224]  Kevin P. Fleming <kpfleming@digium.com>

	* utils: all created files need to be listed in the ignore property

	* channels/chan_unistim.c, build_tools/menuselect-deps.in,
	  configure, configure.ac, channels/Makefile, channels/chan_oss.c:
	  make the configure script detect that it is running on a Windows
	  platform, and report that information so that menuselect can use
	  it (all information that is used to decide whether to build
	  modules or not must be fed to menuselect so the user knows what
	  will be built and why... don't make module build decisions in the
	  makefiles, please)

	* Makefile: make using PRINT_DIR a little easier

2007-12-17 15:18 +0000 [r93187-93190]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix usage of rtptimeout. It can be used
	  without rtpkeepalive, and the value can not be accessed directly
	  in the SIP pvt structure. All RTP related timeouts have to be
	  retrieved using the ast_rtp_* function calls. (closes issue
	  #11562) Reported by: ibc

	* channels/chan_unistim.c: If no timezone is available use the
	  default message. (closes issue #11576) Reported by: junky

	* channels/chan_unistim.c: Make chan_unistim actually be able to
	  unload. When creating a thread that you want to pthread_join you
	  have to explicitly create it as joinable, and also if using
	  pthread_cancel you have to have a pthread_testcancel to see if it
	  has been called.

2007-12-17 07:27 +0000 [r93184-93185]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs, /, build_tools/make_version,
	  include/asterisk/autoconfig.h.in, configure.ac, apps,
	  Makefile.moddir_rules, res/Makefile, pbx/Makefile,
	  build_tools/prep_moduledeps (removed), channels/Makefile, cdr,
	  formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
	  configure, build_tools/embed_modules.xml, cdr/Makefile,
	  build_tools/prep_tarball, makeopts.in, formats/Makefile, res,
	  pbx, channels, funcs/Makefile: Merged revisions 93180 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007)
	  | 23 lines In
	  http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
	  rizzo brought up some issues related to the way that the metadata
	  required for menuselect and the rest of the build system is
	  extracted from the source files. Since I had a few hours to kill
	  on an airplane today, I decided to improve this situation... so
	  now the system caches the extracted metadata and uses it to build
	  the menuselect 'tree' as much as it can. The result of this is
	  that when a single source file is changed, only the metadata for
	  that file needs to be extracted again, and the rest is used from
	  the cache files. I also reduced the number of forked processes
	  required to do the metadata extraction; it was actually possible
	  to do most of what we needed in the Makefiles themselves without
	  using any shell scripts at all! On my laptop, these changes
	  resulted in an 80% decrease in the time required for the
	  'menuselect.makeopts' automatic check to occur after editing a
	  single source file. While doing this work I also cleaned up a few
	  minor things in the Makefiles, adding a check for 'awk' to the
	  configure script and changed all remaining places we use 'grep'
	  or 'awk' to use the ones found by the configure script, and
	  changed the 'prep_tarball' script to build the menuselect
	  metadata so that tarballs of Asterisk will include it and won't
	  require the user to wait while it is extracted after unpacking.
	  ........

2007-12-16 19:06 +0000 [r93173]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: menuselect.makeopts is not a .PHONY target

2007-12-16 13:38 +0000 [r93163-93167]  Olle Johansson <oej@edvina.net>

	* pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks,
	  dimas! (closes issue #11572) Reported by: dimas Patches:
	  dundilog-trunk.patch uploaded by dimas (license 88)

	* main/manager.c, CHANGES: Adding a new CLI command for "manager
	  reload", which is important now that you need to reload after
	  changes. Thanks YS. Reported by: ys Patches:
	  trunk93163_manager_reload.c.diff uploaded by ys (license 281)
	  (related to issue #11414)

	* main/manager.c, CHANGES: Change manager so that registered
	  accounts are stored in memory. This opens for a manager realtime
	  implementation. If you change accounts in manager.conf, you now
	  need to reload to activate the changes (deletions, additions).
	  This was not the case with 1.4. Reported by: ys Patches:
	  trunk93163_manager_reload.c.diff uploaded by ys (license 281)
	  (closes issue #11414)

	* CHANGES: Adding console_video to CHANGES. It's important that we
	  keep this file up to date, even with experimental stuff.

	* channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample,
	  channels/chan_sip.c, include/asterisk/rtp.h,
	  include/asterisk/netsock.h, channels/iax2-provision.c,
	  UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample,
	  CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c,
	  configs/h323.conf.sample, configs/iax.conf.sample,
	  channels/chan_skinny.c, configs/mgcp.conf.sample,
	  configs/unistim.conf.sample, channels/chan_h323.c,
	  configs/iaxprov.conf.sample, pbx/pbx_dundi.c,
	  configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements
	  to QoS/CoS handling by IgorG - Refer to the proper documentation
	  - Implement separate signalling/media QoS/CoS in many channels
	  using RTP - Improve warnings and verbose messages - Deprecate
	  some old settings Minor modifications by me, a big effort from
	  IgorG. Thanks! Reported by: IgorG Patches:
	  qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
	  Tested by: IgorG (closes issue #11145)

2007-12-16 10:34 +0000 [r93162]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: use a simpler idiom for 'cmp -s ...'

2007-12-16 09:37 +0000 [r93152-93161]  Olle Johansson <oej@edvina.net>

	* main/asterisk.c: Don't drop the first character randomly in long
	  listings in the CLI. Reported by: slavon Patches:
	  asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich
	  (license 176) Tested by: eliel (closes issue #9325)

	* configs/sip.conf.sample, CHANGES: Update documentation

	* channels/chan_sip.c, configs/sip.conf.sample: Make more timers
	  settable in SIP so that we can force timeout earlier on
	  non-responsive SIP servers. Thanks, jcmoore, for the patch!
	  Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt
	  uploaded by jcmoore (license 9) (closes issue #9771)

	* include/asterisk/file.h: Typo fixed earlier, that wasn't a typo
	  after all. Didn't a clever guy once say "Compile before you
	  commit" ? :-)

2007-12-15 08:10 +0000 [r93151]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h: fix a typo from revision 93138

2007-12-15 00:44 +0000 [r93138-93145]  Luigi Rizzo <rizzo@icir.org>

	* configs/oss.conf.sample: configuration options related to video
	  support.

	* channels/console_video.c (added): Bring in video console support
	  for chan_oss (and later chan_alsa too). This is disabled in the
	  default build, you need to explicitly enable it compiling with
	  make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do
	  a video call with chan_oss, using the webcam (or X11 grabbing) as
	  local source, and rendering the incoming stream on your screen.
	  Currently supported formats are h261, h263, h263+, h264, mpeg4
	  (all through the avcodec lib, part of ffmpeg). Incoming video is
	  on the left, outgoing video is on the right, while the center
	  displays a keypad (if configured so). Right clicking on the video
	  windows increases the size, center clicking reduces the size.
	  Dragging the mouse (with the left key) on the right window while
	  the X11 grabber is active moves the grab area. This is the result
	  of work by Sergio Fadda, Marta Carbone and myself, all properly
	  disclaimed to digium. Note, there is a lot of work left to do in
	  this module, including adding support for Video4LinuxV2 (I have
	  patches from Matteo Brancaleoni which should be integrated), and
	  making the GUI a lot more friendly than it is now (e.g.
	  supporting merging or switching among multiple sources, a text
	  window, and more).

	* channels/chan_oss.c: remove some redundant headers

	* include/asterisk/file.h: include mmap header if detected by
	  configure

2007-12-14 22:02 +0000 [r93094-93115]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Resolve a compiler warning

	* apps/app_voicemail.c: Change places where the name "INBOX" was
	  hardcoded to use the imapfolder setting from voicemail.conf
	  instead. This commit will help to get issue #11415 moving towards
	  commitment.

2007-12-14 21:09 +0000 [r93090]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h:
	  Solaris compat fixes Reported by: snuffy Patch by:
	  snuffy,tilghman (Closes issue #11315)

2007-12-14 19:31 +0000 [r93067]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: make something static

2007-12-14 19:27 +0000 [r93066]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_privacy.c, UPGRADE.txt, CHANGES,
	  configs/privacy.conf.sample (removed): Remove use of privacy.conf
	  by the Privacy app. Reported by: eliel Patch by: eliel (Closes
	  issue #11344)

2007-12-14 19:19 +0000 [r93042-93065]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to
	  increment the numbers used on the VERBOSITY_ATLEAST calls by 1.
	  Thanks to kpfleming for pointing this out.

	* include/asterisk/logger.h, main/pbx.c, main/manager.c,
	  funcs/func_timeout.c: Changed VERBOSITY_LEVEL to
	  VERBOSITY_ATLEAST to be more accurate.

	* include/asterisk/logger.h, main/pbx.c, main/manager.c,
	  funcs/func_timeout.c, main/logger.c: After reading Russell's
	  e-mail to the dev list stating that checking option_verbose is
	  not equivalent to the check done by ast_verb, I wrote a macro,
	  VERBOSITY_LEVEL, which does this check. I did a quick look in the
	  source and used this macro in some places where option_verbose
	  was used. I also converted some verbose messages in logger.c to
	  use ast_verb instead of ast_verbose.

2007-12-14 18:24 +0000 [r93041]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: gcc 4.1.3 wants a union used here.

2007-12-14 17:49 +0000 [r93001-93004]  Russell Bryant <russell@digium.com>

	* main/config.c: Print an error message if a #included file does
	  not exist

2007-12-14 17:29 +0000 [r92999]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Publish the AGI events to manager. Reported by:
	  moy Patch by: moy,tilghman (Closes issue #11337)

2007-12-14 15:59 +0000 [r92976]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_timeout.c: Reintroduce an optimization that was lost
	  when converting trunk to use ast_verb.

2007-12-14 15:49 +0000 [r92939]  Tilghman Lesher <tlesher@digium.com>

	* main/editline/sys.h: If malloc.h is included in a Solaris build,
	  the compilation breaks. Reported by: snuffy Patch by: snuffy
	  (Closes issue #11313)

2007-12-14 15:18 +0000 [r92938]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4
	  lines Up the length of the format on the SIP channel since it can
	  now be rather long. (closes issue #11552) Reported by:
	  francesco_r ........

2007-12-14 15:14 +0000 [r92936]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 92933 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007)
	  | 5 lines Change help documentation to match actual behavior
	  (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
	  (Closes issue #11548) ........

2007-12-14 15:08 +0000 [r92935]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14
	  Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state
	  setting and overlap_task thread starting. ........

2007-12-14 14:48 +0000 [r92913]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c,
	  res/res_features.c, apps/app_minivm.c, apps/app_amd.c,
	  res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
	  main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert
	  ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy
	  (Closes issue #11547)

2007-12-14 01:25 +0000 [r92876]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13
	  Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't
	  spam the CLI with messages about possible deadlocks. Instead just
	  print the intended single message every five seconds. (closes
	  issue 11537, reported and patched by dimas) ........

2007-12-13 23:10 +0000 [r92816-92855]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: When working with dates, use numeric form
	  whenever possible, as it's faster. Also, a bunch of coding
	  guidelines fixes.

	* channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007)
	  | 5 lines Properly initialize polarity statuses, so that they are
	  detected properly. Reported by: julianjm Patch by: julianjm
	  (Closes issue #10238) ........

2007-12-13 20:23 +0000 [r92811]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c,
	  apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the
	  new ast_module_user_* functions in a few documentation places.
	  (closes issue #11533) Reported by: IgorG Patches:
	  oldmacroclean.v1.diff uploaded by IgorG (license 20)

2007-12-13 20:14 +0000 [r92810]  Jason Parker <jparker@digium.com>

	* main/pbx.c, /: Merged revisions 92809 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1
	  line Make application help text a little more clear about the use
	  of extensions in a filename. ........

2007-12-13 20:12 +0000 [r92806-92808]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec
	  2007) | 3 lines Prevent another potential fd leak ........

	* /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec
	  2007) | 3 lines Prevent a possible fd leak. ........

2007-12-13 17:46 +0000 [r92779]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape
	  character, unless it really is an escape character.

2007-12-13 16:23 +0000 [r92758]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Remove remnants of a poorly merged commit.
	  (92697)

2007-12-13 15:40 +0000 [r92737]  Doug Bailey <dbailey@digium.com>

	* apps/app_voicemail.c: Tag voicemails with UTC time as opposed to
	  local time zone

2007-12-13 00:18 +0000 [r92697]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c, channels/chan_h323.c, main/config.c:
	  Merged revisions 92696 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600
	  (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file,
	  we previous continued on with what was already loaded. We do not
	  want to do this (see bug below for details). This makes it so
	  that if a [ is found without a ], the entire config will fail,
	  and nothing in it will be loaded. Issue 10690. ........

2007-12-12 23:44 +0000 [r92676]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Revert an "optimization" that I added in
	  revision 89887, as the user who reported issue #11449 has
	  demonstrated that it actually was a performance hit on his
	  machine. I think that it is possible that it could still be a
	  benefit on systems under higher load, especially SMP systems, but
	  I don't have enough time or interest to find out at the moment.
	  (closes issue #11449)

2007-12-12 21:22 +0000 [r92618]  Jason Parker <jparker@digium.com>

	* /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600
	  (Wed, 12 Dec 2007) | 4 lines Don't increment user count until
	  after name has been recorded (if enabled). Issue 11048, tested by
	  pep. ........

2007-12-12 20:05 +0000 [r92594]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, main/logger.c, main/utils.c,
	  apps/app_mixmonitor.c: Conversions of free to ast_free, where
	  applicable, and several other formatting fixes. Reported by:
	  eliel Patch by: eliel,tilghman (Closes issue #11209)

2007-12-12 19:50 +0000 [r92562]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Merged revisions 92556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) |
	  1 line resolve compiler warning ........

2007-12-12 17:51 +0000 [r92511-92526]  Mark Michelson <mmichelson@digium.com>

	* res/res_features.c: Same change to trunk as revision 92510. I'm
	  not sure why I merged this way, but I did.

2007-12-12 17:15 +0000 [r92476-92507]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Correctly handle possible memory allocation
	  failure Reported by: eliel Patch by: eliel (Closes issue #11512)

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 92463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007)
	  | 4 lines Test directly for the API that fixed AST-2007-026, to
	  ensure that older versions of PostgreSQL are no longer
	  acceptable. (Closes issue #11526) ........

2007-12-12 16:11 +0000 [r92444]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 92443 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec
	  2007) | 3 lines Removing an unused variable. ........

2007-12-11 22:20 +0000 [r92423]  Olle Johansson <oej@edvina.net>

	* include/asterisk/term.h, channels/misdn/isdn_msg_parser.c,
	  channels/ringtone.h, include/asterisk/ulaw.h,
	  include/jitterbuf.h, include/asterisk/manager.h,
	  include/asterisk/transcap.h, channels/misdn/isdn_lib.c,
	  channels/gentone.c, include/asterisk/zapata.h,
	  channels/misdn/isdn_lib.h, include/asterisk/doxyref.h,
	  channels/DialTone.h, channels/misdn/ie.c,
	  channels/misdn/chan_misdn_config.h, channels/iax2.h,
	  channels/misdn/portinfo.c, include/asterisk/udptl.h,
	  main/cygload.c, include/asterisk/translate.h: Doxygen updates,
	  formatting. misdn stuff needs a lot of doxygenification (Hello,
	  Qwell :-) )

2007-12-11 22:10 +0000 [r92422]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Trunk build would fail due to the nonexistence of
	  zaptel hwgain structures missing. Patched configure to check for
	  this stuff and put a #ifdef around the offending code in
	  chan_zap. Thanks to file for overseeing this.

2007-12-11 21:58 +0000 [r92421]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: We need to set the address we want to match
	  against before we actually do the match.. Closes issue #11518.

2007-12-11 21:46 +0000 [r92402]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c: Removing a pointless memset. The memory
	  was just calloc'd, so the memory is already zeroed out

2007-12-11 21:17 +0000 [r92401]  Jason Parker <jparker@digium.com>

	* apps/app_controlplayback.c: Add variable to show which key was
	  pressed to stop playback. Issue #11377, initial patch by johan.

2007-12-11 20:06 +0000 [r92364-92365]  Joshua Colp <jcolp@digium.com>

	* res/res_monitor.c: Only look to see if options are set if some
	  have been provided. (closes issue #11505) Reported by: Mike
	  Anikienko

	* main/global_datastores.c, /: Merged revisions 92363 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec
	  2007) | 6 lines Fix potential memory leak with the dialed
	  interfaces list if another memory allocation fails. (closes issue
	  #11507) Reported by: eliel Patches: global_datastores.c.patch
	  uploaded by eliel (license 64) ........

2007-12-11 17:44 +0000 [r92324]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 92323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec
	  2007) | 10 lines Fixing autofill to be more accurate.
	  Specifically, if calls ahead of the current caller were ringing
	  members (but not yet bridged) there could be available members
	  and waiting callers who would not get matched up. The member
	  availability checker was correctly determining the number of
	  available members in this scenario, but the queue itself did not
	  parallelly reflect this status on the pending calls. This commit
	  corrects the issue. (closes issue #11459, reported by
	  equissoftware, patched by me) ........

2007-12-11 16:29 +0000 [r92305]  Russell Bryant <russell@digium.com>

	* include/asterisk/unaligned.h, main/event.c: * In unaligned.h,
	  remove some unnecessary casts and mark the arg of the
	  get_unaligned functions as const * In event.c, use
	  get_unaligned_uint32() in a couple of places to fix issues on
	  architectures that don't allow unaligned access

2007-12-11 14:17 +0000 [r92267-92285]  Olle Johansson <oej@edvina.net>

	* include/asterisk/devicestate.h, include/asterisk/agi.h,
	  include/asterisk/astobj2.h, include/asterisk/extconf.h,
	  include/asterisk/io.h, include/asterisk/cdr.h,
	  include/asterisk/aes.h, include/asterisk/_private.h,
	  include/asterisk/localtime.h, include/asterisk/hashtab.h,
	  include/asterisk/callerid.h, include/asterisk/logger.h,
	  include/asterisk/doxyref.h, include/asterisk/app.h,
	  include/asterisk/adsi.h, include/asterisk/event.h,
	  include/asterisk/causes.h, include/asterisk/alaw.h,
	  include/asterisk/ast_expr.h, include/asterisk/dsp.h,
	  include/asterisk/mod_format.h, include/asterisk/ael_structs.h,
	  include/asterisk/astdb.h: A lot of doxygen updates

	* include/asterisk/frame.h: Doxygen updates

2007-12-10 20:18 +0000 [r92243]  Doug Bailey <dbailey@digium.com>

	* channels/chan_zap.c: Add CLI commands to dynamically set hw and
	  sw gains

2007-12-10 16:48 +0000 [r92205-92206]  Joshua Colp <jcolp@digium.com>

	* utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to
	  check_expr for platforms that need it. (closes issue #11484)
	  Reported by: snuffy

	* /, main/rtp.c: Merged revisions 92204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6
	  lines Add G729A as another possible payload name for G729. Some
	  devices use this instead of G729, which is perfectly normal since
	  the payload number itself is defined and can't be used by
	  anything else so the name doesn't matter that much. (closes issue
	  #11483) Reported by: revolution Patches: rtp.diff uploaded by
	  revolution (license 346) ........

2007-12-10 16:30 +0000 [r92203]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 92202 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec
	  2007) | 7 lines If there are no members in a queue, then the loop
	  where the datastore for detecting duplicate dialed numbers will
	  be skipped, meaning the datastore isn't created. This means that
	  when we try to free it, there's a crash. This stops that crash
	  from occurring. (closes issue #11499, reported by slavon, patched
	  by eliel) ........

2007-12-10 16:15 +0000 [r92199-92201]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Only send a SIGHUP if the pid is greater than -1,
	  otherwise all PIDs greater than -1 will get the SIGHUP... and
	  that is bad. (closes issue #11453) Reported by: alanmcmillan

2007-12-10 14:18 +0000 [r92140-92160]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Removing some LOG_DEBUG items

	* /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16
	  lines Avoid reinvite race situations with two Asterisks trying to
	  reinvite each other in 1.4 and trunk. This patch implements
	  support for the 491 error code that Asterisk 1.4 generates on
	  situations where we get an incoming INVITE and already has one in
	  progress. Thanks to mavetju for reporting and to Raj Jain for an
	  excellent explanation of the problem. Patch by myself. Tested
	  with 8 Asterisk servers connected to each other in a training
	  network. Closes issue #10481 ........

	* doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra
	  headers in the voicemail users listing in manager 1.1. Update
	  documentation too. (closes issue #11495) Reported by: caio1982
	  Patches: extra_vm_manager_info1.diff uploaded by caio1982
	  (license 22)

2007-12-10 09:00 +0000 [r91929-92122]  Luigi Rizzo <rizzo@icir.org>

	* build_tools/make_version, build_tools/make_version_h:
	  simplify/cleanup the scripts

	* utils/Makefile: remove relative paths and use ASTTOPDIR instead.
	  Give a default value to ASTTOPDIR if unset so we can at least do
	  a 'make clean' without too much trouble. The proper fix, however,
	  is to partition the top level Makefile in a 'setup' and a 'main'
	  part, in a way that the 'setup' part can be included from
	  subdirs' Makefiles and allow targets to be built without going
	  through the top level Makefile.

	* utils/clicompat.c: simplify this file

	* doc/CODING-GUIDELINES: add a bit of info on the build
	  infrastructure

	* Makefile: Fix the detection of modules installed from this build.
	  You can now add the path of local module subdirs from the command
	  line with make LOCAL_MOD_SUBDIRS= ....

	* codecs/Makefile, apps/Makefile, Makefile.moddir_rules,
	  cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile,
	  formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules
	  the common instructions used to generate loadable and embedded
	  module lists. Individual Makefiles now are a lot simpler,
	  possibly as simple as this: -include
	  $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
	  MODULE_PREFIX=cdr_ all: _all include
	  $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because
	  in a single directory we can combine various types of modules
	  (app_, cdr_, func_, ... ) by simply listing them in the
	  MODULE_PREFIX variable. The individual Makefiles can also create
	  list of modules to be excluded by listing them in the variablel
	  MODULE_EXCLUDE (see an example in channels/Makefile). With this
	  change it becomes trivial to integrate a directory with locally
	  created/modified sources into the main build.

	* Makefile, Makefile.moddir_rules: make the install target a bit
	  less noisy

	* Makefile: document usage of several exported variables

	* utils/Makefile: add hashtab.c to the list of files deleted

	* Makefile.moddir_rules: another place where ../ should have been
	  ASTTOPDIR

	* codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile,
	  pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile,
	  funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR
	  and not .. to reference the top level directory.

	* Makefile: Implement the outcome of a discussion on the -dev list
	  re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir
	  Cohen and Simon Perreault for extremely useful feedback: 1.
	  comment out the [directories] section the created asterisk.conf ;
	  you can set the correct defaults at build time using
	  INSTALL_PATH, so the repetition here is redundant and often
	  wrong. (The next step now is move asterisk.conf outside the
	  Makefile to asterisk.conf.sample, because there is little if
	  anything here that needs to be constructed at build/install
	  time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to
	  specify a path once if the two coincide. This should have no ill
	  side effects, because if you don't specify DESTDIR, you really
	  need INSTALL_PATH="" to set the correct defaults, and if you
	  specify DESTDIR the value is not overridden. The second part
	  required moving the 'export DESTDIR' right after the assignment
	  to prevent DESTDIR getting set by the export (this is documented
	  in the Makefile).o hopefully avoid the mistake)$ With this change
	  you can now do something like this from your source tree: make
	  INSTALL_PATH=/some/place install samples and then main/asterisk
	  -vdc which will pick up the correct config files and libraries
	  from /some/place - i.e. great for developers!

	* main/config.c: remove unused code, and simplify the logic for
	  #include/#exec (still a lot of cleanup needed here).

	* main/config.c: Implement comment_buffer and lline_buffer in terms
	  of the ast_str_*() API. I don't know if comment_buffers etc are
	  actually used at all...

	* main/config.c: unify some common code

	* main/config.c: normalize formatting

	* main/config.c: document a nice technique to exit from a block in
	  case of errors.

	* main/config.c: a little bit of documentation on how lines are
	  parsed.

	* utils/ael_main.c: normalize header order, and add a comment on
	  the need to clean up this file.

	* include/asterisk/network.h: some platforms (e.g. FreeBSD4) need
	  netinet/in.h to be included before arpa/inet.h

2007-12-07 23:32 +0000 [r91832-91891]  Jason Parker <jparker@digium.com>

	* /, main/dsp.c: Merged revisions 91890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600
	  (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the
	  input frame if we return a different frame in ast_dsp_process.
	  Issue 11273, pointed out by dimas, with a patch by eliel.
	  ........

	* pbx/pbx_lua.c, configs/extensions.lua.sample: Update
	  documentation for pbx_lua. Closes issue #11492, patch by
	  mnicholson.

2007-12-07 21:25 +0000 [r91784-91831]  Russell Bryant <russell@digium.com>

	* /, main/utils.c: Merged revisions 91830 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) |
	  5 lines Make the lock protecting each thread's list of locks it
	  currently holds recursive. I think that this will fix the
	  situation where some people have said that "core show locks"
	  locks up the CLI. (related to issue #11080) ........

	* /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07
	  Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code.
	  The ast_mutex_init() function had the mutex attribute object
	  marked as static. This means that multiple threads initializing
	  locks at the same time could step on each other and end up with
	  improperly initialized locks. (found when tracking down locking
	  issues related to issue #11080) ........

	* /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07
	  Dec 2007) | 6 lines I love fixing lock related errors in the lock
	  debugging code. That's about as ironic as it gets in Asterisk
	  programming land. Anyway, I spotted this bug while trying to
	  track down why systems are locking up and acting weird in issue
	  #11080. The mutex attribute object was marked as static in this
	  function when it should not have been. ........

	* apps/app_dial.c, /: Merged revisions 91783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) |
	  6 lines * Add channel locking around datastore operations that
	  expect the channel to be locked. * Document why we don't record
	  Local channels in the dialed interfaces list. * Remove the dialed
	  variable as it isn't needed. * Restructure some code for clarity
	  and coding guidelines stuff ........

2007-12-07 16:37 +0000 [r91782]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Fix a small typo in a comment. Closes issue
	  #11490

2007-12-07 16:28 +0000 [r91781]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 91780 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) |
	  7 lines * Add channel locking around datastore operations that
	  expect the channel to be locked. * Document why we don't record
	  Local channels in the dialed interfaces list. * Handle memory
	  allocation failure. * Remove the dialed variable, as it wasn't
	  actually needed. * Tweak some formatting to conform to coding
	  guidelines. ........

2007-12-07 16:11 +0000 [r91779]  Jason Parker <jparker@digium.com>

	* doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c,
	  include/asterisk/pbx.h, main/cli.c: Add count of total number of
	  calls processed by asterisk during it's lifetime. Add number of
	  total calls and current calls to SNMP. Closes issue #10057, patch
	  by jcmoore.

2007-12-07 16:11 +0000 [r91778]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: Merged revisions 91777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) |
	  6 lines * Add a bit more of a verbose comment as to why a hangup
	  frame needs to be queued up if autoservice gets a NULL return
	  from ast_read(). * Make the process of queueing the hangup frame
	  more efficient by putting the frame where it is going to end up
	  and avoiding some locking and extra memory allocations and
	  freeing. ........

2007-12-07 15:40 +0000 [r91738]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c, /: Merged revisions 91737 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec
	  2007) | 7 lines Hangups that happen during autoservice were not
	  processed appropriately. This is because a hangup actually causes
	  a NULL frame to be received, not a hangup frame. Queueing a
	  hangup if we receive a NULL frame during autoservice corrects
	  this problem (closes issue #11467, reported by jmls, patched by
	  me) ........

2007-12-07 02:52 +0000 [r91676-91700]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, /: Merged revisions 91693 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) |
	  2 lines Don't unlock the dialed_interfaces list until we're done
	  messing with the iterator. ........

	* apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) |
	  4 lines Allow dialing local channels from Queue() and Dial()
	  again. There was a slight flaw in the code to prevent call
	  forwards from looping that caused this problem. (related to issue
	  #11486) ........

	* /, apps/app_queue.c: Merged revisions 91675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) |
	  7 lines Fix in an issue in the call forwarding handling code that
	  was causing crashes on every call into a queue. I'm not entirely
	  sure about the logic in this part of the code, so I want to look
	  at it some more tomorrow. However, this makes it safe and keeps
	  it from crashing. (closes issue #11486, reported by adamg,
	  patched by me) ........

2007-12-07 00:58 +0000 [r91617-91638]  Tilghman Lesher <tlesher@digium.com>

	* /, main/rtp.c: Merged revisions 91637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007)
	  | 5 lines At the end of a call, when we're reporting, RTCP may
	  already be partially torn down, so check for NULL dereference
	  Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
	  ........

	* channels/chan_zap.c: Add a manager event for PRI events: this
	  will help manager users detect when a D-channel goes down

	* main/cdr.c: If duration or billsec are not yet calculated,
	  calculate them on demand.

2007-12-06 21:57 +0000 [r91598]  Jason Parker <jparker@digium.com>

	* cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3
	  cdr module.. Closes issue #11070, patch by seanbright.

2007-12-06 21:03 +0000 [r91579]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Handle allocation failure of the heard and
	  deleted arrays of the vm_state. (closes issue #11408, reported
	  and patched by jaroth)

2007-12-06 20:52 +0000 [r91561]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007)
	  | 3 lines Properly escape cdr->src and cdr->dst and ensure we use
	  thread-safe escaping (Fixes AST-2007-026) ........ r90736 |
	  tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines
	  If both dbhost and dbsock were not set, a NULL deref could result
	  Reported by: xrg Patch by: tilghman (Closes issue #11387)
	  ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
	  Dec 2007) | 5 lines Solaris requires the inclusion of
	  sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
	  snuffy,tilghman (Closes issue #11430) ........

2007-12-06 16:54 +0000 [r91472]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we clear these flags when libpri
	  is not installed

2007-12-06 16:51 +0000 [r91440-91458]  Joshua Colp <jcolp@digium.com>

	* main/udptl.c, /: Merged revisions 91450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6
	  lines Fix various in the udptl implementation. It could return
	  empty modem frames, have an incorrect sequence number on packets,
	  and display the wrong sequence number in the debug messages.
	  (closes issue #11228) Reported by: Cache Patches: udptl-4.patch
	  uploaded by dimas (license 88) ........

	* /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4
	  lines Add support for accepting and sending T.38 in the initial
	  INVITE. (closes issue #9402) Reported by: thdei ........

2007-12-06 15:56 +0000 [r91347-91438]  Olle Johansson <oej@edvina.net>

	* doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation
	  for the massive manager changes to manager version 1.1 -
	  hopefully a more consistent manager interface.

	* main/manager.c: - The Ping Action - Now use Response: success -
	  New header "Ping: pong" :-) - The Events action - Now use
	  Response: Success - The new status is reported as "Events: On" or
	  "Events: Off" - Report if manager is enabled in the reload event
	  Small cleanups... From moremanager

	* main/channel.c: Changes to manager events in channel.c - Newstate
	  event - Now has "CalleridNum" for numeric caller id, like
	  Newchannel - The event does not send "<unknown>" for unknown
	  caller IDs just an empty field - Newstate and Newchannel events -
	  these have changed headers "State" -> ChannelStateDesc Text based
	  channel state -> ChannelState Numeric channel state - The events
	  does not send "<unknown>" for unknown caller IDs just an empty
	  field - Newstate event - Now has "CalleridNum" for numeric caller
	  id, like Newchannel - The event does not send "<unknown>" for
	  unknown caller IDs just an empty field - Link and Unlink events -
	  The "Link" and "Unlink" bridge events in channel.c are now
	  renamed to "Bridge" - The link state is in the bridgestate:
	  header as "Link" or "Unlink" - For channel.c bridges,
	  "Bridgetype: core" is added. This opens up for bridge events in
	  rtp.c and channel drivers - The "Rename" manager event has a
	  renamed header, to use the same terminology for the current
	  channel as other events - Oldname -> Channel (Moremanager)

	* main/cdr.c: New manager event when a channel changes account
	  code. Maybe belongs in the new cdr category? ---moremanager---
	  Event: NewAccountCode Modules: cdr.c Purpose: To report a change
	  in account code for a live channel Example: Event: NewAccountCode
	  Privilege: call,all Channel: SIP/olle-01844600 Uniqueid:
	  1177530895.2 AccountCode: Stinas account 1234848484
	  OldAccountCode: Olles Account 12345

	* apps/app_dial.c: - Dial event - Event Dial has new headers, to
	  comply with other events - Source -> Channel Channel name
	  (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring
	  Dialstring in app data (moremanager)

	* apps/app_meetme.c: Adding small missing but important comma...

	* apps/app_meetme.c: A big oops...

	* apps/app_meetme.c: The MeetmeJoin now has caller ID name and
	  Caller ID number fields (like MeetMeLeave) (Moremanager)

	* channels/chan_zap.c: Update ZapShowChannels so that you can
	  specify one channel. Action ZapShowChannels Header changes -
	  Channel: -> ZapChannel For active channels, the Channel: and
	  Uniqueid: headers are added You can now add a "ZapChannel: "
	  argument to zapshowchannels actions to only get information about
	  one channel. From the moremanager branch

	* main/logger.c: Doxygen updates

	* include/asterisk/logger.h, /, main/logger.c, main/loader.c:
	  Merged revisions 91366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4
	  lines Make sure logger is reloaded at general reload in the cli.
	  (Discovered during Asterisk training in Portugal) ........

	* main/manager.c: Change description of new manager command

	* main/manager.c, CHANGES: Add manager command for showing all
	  current channels. Thanks, eliel, for writing the original patch.
	  Modified by me to follow other manager events and the new
	  "moremanager" style. (closes issue #11478) Reported by: eliel
	  Patches: manager.c.patch uploaded by eliel (license 64)

2007-12-06 04:37 +0000 [r91328]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Instead of iterating through the entire epoll
	  events array just look at the ones that will actually contain
	  data. (props to eliel on IRC for this)

2007-12-05 22:57 +0000 [r91291-91293]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec
	  2007) | 3 lines Reverting extra stuff I didn't mean to commit
	  ........

	* apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec
	  2007) | 11 lines The 'G' option for Dial() did not properly
	  handle the case where only a label was provided. This was due to
	  the fact that the answering channel did not have an extension
	  set, so ast_parseable_goto would fail. This fix eliminates the
	  call to ast_parseable_goto on the answering channel since it is a
	  wasteful call. The answering channel and the calling channel are
	  both directed to the same extension and context, just different
	  priorities, so we can just copy the values from the calling
	  channel to the answering channel and increment the answering
	  channel's priority. (closes issue #11382, reported by jon, patch
	  by me with correction by jon) ........

2007-12-05 21:46 +0000 [r91238]  Tilghman Lesher <tlesher@digium.com>

	* /, sounds/Makefile: Merged revisions 91237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007)
	  | 2 lines Upgrade to the latest version of extra sounds ........

2007-12-05 17:49 +0000 [r91193-91197]  Russell Bryant <russell@digium.com>

	* /, main/threadstorage.c: Merged revisions 91192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) |
	  10 lines Make the lock in the threadstorage debugging code
	  untracked to avoid a deadlock on thread destruction. (closes
	  issue #11207) Reported by: ys Patches: threadstorage.c.diff
	  uploaded by ys (license 281) Also fixes an open bug report:
	  (closes issue #11446) ........

	* apps/app_directory.c: Resolve compiler warnings.

2007-12-05 16:46 +0000 [r91172-91173]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, UPGRADE.txt, configs/manager.conf.sample,
	  CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change
	  cdr_manager to use a "CDR" level, rather than the (overcrowded)
	  "call" level. (Closes issue #11015)

	* CHANGES, apps/app_directory.c: Added multiple name listing.
	  (Closes issue #10413)

2007-12-05 16:14 +0000 [r91171]  Joshua Colp <jcolp@digium.com>

	* configs/http.conf.sample: Remove second prefix line. Only need it
	  documented once in the same file. (closes issue #11472) Reported
	  by: eserra Patches: http.conf.sample.diff uploaded by eserra
	  (license 45)

2007-12-05 13:09 +0000 [r91151-91152]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename
	  "username" to "defaultuser" to match with "defaultip". "Username"
	  still works, but is deprecated.

	* channels/chan_sip.c: Remove the cseqs from "sip show channel" and
	  make more place for the call ID.

2007-12-05 03:48 +0000 [r91133]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: revert part of my changes from earlier today
	  since this code is no longer dependent on libpri.h

2007-12-05 03:34 +0000 [r91029-91131]  Russell Bryant <russell@digium.com>

	* res/res_odbc.c: Use ast_free() instead of free(). (closes issue
	  #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded
	  by Laureano (license 265)

	* /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04
	  Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were
	  reported as happening on Mac systems. It turns out that the
	  problem was the Mac version of the ast_atomic_fetchadd_int()
	  function. The Mac atomic add function returns the _new_ value,
	  while this function is supposed to return the old value. So, the
	  crashes happened on unreferencing objects. If the reference count
	  was decreased to 1, ao2_ref() thought that it had been decreased
	  to zero, and called the destructor. However, there was still an
	  outstanding reference around. (closes issue #11176) (closes issue
	  #11289) ........

	* /, main/utils.c: Merged revisions 91074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) |
	  4 lines When DEBUG_THREADS is enabled, we only have the details
	  about who is holding a lock that we are waiting on for a mutex,
	  not rwlocks. This should fix the problem where people have
	  reported "core show locks" crashing sometimes. ........

	* channels/chan_zap.c: Fix mwimonitornotify on reload ... again.
	  This option was only read at startup so a reload would erase it
	  and not reset it. (pointed out by tzafrir)

	* utils/astman.c: Fix the build of astman. Any file that includes
	  any asterisk sub-headers needs to first include asterisk.h.
	  (closes issue #11394)

2007-12-04 22:44 +0000 [r91012]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Don't error when we don't have libpri
	  installed with libss7 support. Also, print the debug message
	  anyway if we can't find the right PRI

2007-12-04 22:07 +0000 [r91010-91011]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 90967 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) |
	  7 lines Make some changes to some additions I made recently for
	  doing channel autoservice when looking up extensions. This code
	  was added to handle the case where a dialplan switch was in use
	  that could block for a long time. However, the way that I added
	  it, it did this for all extension lookups. However, lookups in
	  the in-memory tree of extensions should _not_ take long enough to
	  matter. So, move the autoservice stuff to be only around
	  executing a switch. ........

	* channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I
	  guess I only added this line in my head. (thanks to tzafrir for
	  pointing it out)

2007-12-04 21:46 +0000 [r90993]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_usbradio.c: Coding guidelines fixups (Closes issue
	  #11412)

2007-12-04 21:23 +0000 [r90991]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c, CHANGES: Add manager action
	  'sipshowregistry'. Closes issue #11464, patch by eliel.

2007-12-04 19:08 +0000 [r90949]  Russell Bryant <russell@digium.com>

	* include/asterisk/callerid.h, channels/chan_zap.c,
	  main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support
	  for monitoring MWI on FXO lines. This introduces two new options
	  for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor
	  option enables MWI monitoring. When the MWI state on a line
	  changes, then the script specified by mwimonitornotify will be
	  executed for custom handling of the state change, similar to the
	  externnotify option of voicemail.conf. Also, when the MWI state
	  on an FXO line changes, an internal Asterisk event is generated
	  to indicate the new state of the associated mailbox. That may,
	  any module that cares about MWI information will get notified and
	  can handle it just as if app_voicemail had sent this
	  notification. (BE-253, original patch from markster, with some
	  minor modifications by me to add comments, documentation, and
	  internal event support)

2007-12-04 18:29 +0000 [r90930]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Kevin suggested doing the reverse of my
	  last commit, since imap_retrieve_file does not modify the
	  contents of the "mailbox" string. In other words, I'm changing
	  the imap_retrieve_file function to take a const char* as the
	  third argument so that I don't need to cast const char*'s as
	  char*'s to suppress compiler warnings.

2007-12-04 18:15 +0000 [r90929]  Jason Parker <jparker@digium.com>

	* Makefile: Add Makefile alias target 'pdf' which does the same
	  thing as asterisk.pdf. Issue 11452, reported by blitzrage.

2007-12-04 18:14 +0000 [r90928]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Suppress a compiler warning due to
	  discarding a "const" qualifier

2007-12-04 18:09 +0000 [r90927]  Jason Parker <jparker@digium.com>

	* main/global_datastores.c: Fix build, that some people aren't
	  seeing for some reason.

2007-12-04 17:51 +0000 [r90899]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Wrong locking style got merged from 1.4 to
	  trunk. My mistake.

2007-12-04 17:40 +0000 [r90880]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: fix build of this module when libpri and/or
	  libss7 are or are not present

2007-12-04 17:38 +0000 [r90879]  Jason Parker <jparker@digium.com>

	* main/channel.c, /: Merged revisions 90876 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600
	  (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after
	  allocating a timing fd, we need to make sure to close it. Issue
	  11454, patch by eliel. ........

2007-12-04 17:36 +0000 [r90878]  Russell Bryant <russell@digium.com>

	* main/Makefile: Fix a silly little typo :)

2007-12-04 17:35 +0000 [r90877]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but
	  blocked in trunk since this hadn't been merged yet.

2007-12-04 17:08 +0000 [r90873]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, main/global_datastores.c (added),
	  channels/chan_local.c, /, main/Makefile,
	  include/asterisk/channel.h, include/asterisk/global_datastores.h
	  (added), apps/app_queue.c: Merged revisions 90735 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03
	  Dec 2007) | 22 lines A big one... This is the merge of the
	  forward-loop branch. The main change here is that call-forwards
	  can no longer loop. This is accomplished by creating a datastore
	  on the calling channel which has a linked list of all devices
	  dialed. If a forward happens, then the local channel which is
	  created inherits the datastore. If, through this progression of
	  forwards and datastore inheritance, a device is attempted to be
	  dialed a second time, it will simply be skipped and a warning
	  message will be printed to the CLI. After the dialing has been
	  completed, the datastore is detached from the channel and
	  destroyed. This change also introduces some side effects to the
	  code which I shall enumerate here: 1. Datastore inheritance has
	  been backported from trunk into 1.4 2. A large chunk of code has
	  been removed from app_dial. This chunk is the section of code
	  which handles the call forward case after the channel has been
	  requested but before it has been called. This was removed because
	  call-forwarding still works fine without it, it makes the code
	  less error-prone should it need changing, and it made this set of
	  changes much less painful to just have the forwarding handled in
	  one place in each module. 3. Two new files, global_datastores.h
	  and .c have been added. These are necessary since the datastore
	  which is attached to the channel may be created and attached in
	  either app_dial or app_queue, so they need a common place to find
	  the datastore info. This approach was taken in case similar
	  datastores are needed in the future, there will be a common place
	  to add them. ........

2007-12-04 15:16 +0000 [r90852-90854]  Olle Johansson <oej@edvina.net>

	* apps/app_queue.c: (closes issue #11431) Reported by: Laureano
	  Patches: app_queue.c.patch uploaded by Laureano (license 265)

	* main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel
	  Patches: core.show.hint.patch uploaded by eliel (license 64)

	* CHANGES: (closes issue #11462) Reported by: eliel Patches:
	  CHANGES.patch uploaded by eliel (license 64)

2007-12-04 15:01 +0000 [r90851]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Pass the Asterisk version to AGI scripts as part
	  of the initial dump of info Reported by: acunningham Patch by:
	  acunningham (Closes issue #11398)

2007-12-04 11:50 +0000 [r90834]  Luigi Rizzo <rizzo@icir.org>

	* res/Makefile: fix build on cygwin

2007-12-03 23:52 +0000 [r90760]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
	  Dec 2007) | 5 lines Solaris requires the inclusion of
	  sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
	  snuffy,tilghman (Closes issue #11430) ........

2007-12-03 23:49 +0000 [r90746]  Steve Murphy <murf@digium.com>

	* main/hashtab.c: A small fix from snuffy

2007-12-03 23:48 +0000 [r90738]  Jason Parker <jparker@digium.com>

	* res/res_monitor.c: Add manager events for when a monitor is
	  started or stopped. Closes issue #10191, patch by dgradecak.

2007-12-03 23:29 +0000 [r90737]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03
	  Dec 2007) | 5 lines If both dbhost and dbsock were not set, a
	  NULL deref could result Reported by: xrg Patch by: tilghman
	  (Closes issue #11387) ........

2007-12-03 22:07 +0000 [r90697]  Jason Parker <jparker@digium.com>

	* /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600
	  (Mon, 03 Dec 2007) | 4 lines Make sure we always close the
	  conference fd if we have an open one. Issue 11383, reported by
	  markmhy, patch by eliel. ........

2007-12-03 21:24 +0000 [r90670]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Replacing some calls to free() with
	  ast_free(). (closes issue #11448, reported and patched by jaroth)

2007-12-03 21:03 +0000 [r90656]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands
	  for speech recognition. These mirror the dialplan applications
	  mostly but present the information in a nicer fashion. The SPEECH
	  RECOGNIZE command for example will return the results instead of
	  having to query the dialplan functions.

2007-12-03 21:00 +0000 [r90644]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec
	  2007) | 5 lines Changing some bad logic when calculating the
	  interdigit timeout. (closes issue #11402, reported and patched by
	  eferro) ........

2007-12-03 20:58 +0000 [r90631]  Jason Parker <jparker@digium.com>

	* /, res/res_features.c: Merged revisions 90607 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600
	  (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce
	  application. Issue #11436, reported by lytledd, patch by eliel.
	  ........

2007-12-03 20:30 +0000 [r90591]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Avoid an additional function call. Reported by:
	  eliel Patch by: eliel (Closes issue #11438)

2007-12-03 20:07 +0000 [r90550-90589]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 90588 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2
	  lines Do not create a smoother for G723.1 frames, they need to be
	  left alone to their native 20/24 byte size. ........

	* main/channel.c, /, include/asterisk/channel.h, .cleancount:
	  Merged revisions 90548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2
	  lines Preserve the indication currently playing on a channel when
	  a masquerade operation happens. (issue #BE-88) ........

2007-12-03 16:46 +0000 [r90528]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Updating sample queues.conf file to
	  show how multiple periodic announcements may be specified since
	  this was not documented previously (closes issue #11432, reported
	  and patched by Laureano)

2007-12-03 14:14 +0000 [r90508]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Remove the file descriptors from the main poll
	  channel when the channel is hung up during the dialing attempt,
	  and make sure a channel exists before trying to remove it at the
	  end. (closes issue #11441) Reported by: blitzrage

2007-12-02 18:20 +0000 [r90471]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 90470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) |
	  6 lines The other day when I went through making changes as a
	  result of the ao2_link() change, I added some code to set
	  pointers to NULL after they were unreferenced. This pointed out
	  that in this place, the object was unreferenced before the code
	  was done using it. So, move the unref down a little bit. (crash
	  reported by jmls on IRC) ........

2007-12-02 09:42 +0000 [r90433]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c, /: Merged revisions 90432 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007)
	  | 7 lines Clarify the return value on autoservice. Specifically,
	  if you started autoservice and autoservice was already on, it
	  would erroneously return an error. Reported by: adiemus Patch by:
	  dimas (Closes issue #11433) ........

2007-12-01 01:37 +0000 [r90410]  Jason Parker <jparker@digium.com>

	* res/res_adsi.c: Only reload if the config file has changed.
	  Closes issue #11281, patch by eliel.

2007-11-30 21:19 +0000 [r90388]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, include/asterisk/app.h,
	  include/asterisk/audiohook.h, res/res_features.c,
	  include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c,
	  configs/features.conf.sample: Adding support for the
	  "automixmonitor" dial and queue options. This works in much the
	  same way as the automonitor, except that instead of using the
	  monitor app, it uses the mixmonitor app. By providing an 'x' or
	  'X' as a dial or queue option, a DTMF sequence may be entered (as
	  defined in features.conf) to start the one-touch mixmonitor. This
	  patch also introduces some new API calls to the audiohooks code
	  for searching for an audiohook by type and for searching for a
	  running audiohook by type. Big thanks to joetester for writing
	  the initial patch, testing it and patiently waiting for it to be
	  committed. (closes issue #10185, reported and patched by
	  xmarksthespot)

2007-11-30 19:34 +0000 [r90311-90351]  Russell Bryant <russell@digium.com>

	* main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c,
	  channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged
	  revisions 90348 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) |
	  8 lines Change the behavior of ao2_link(). Previously, in
	  inherited a reference. Now, it automatically increases the
	  reference count to reflect the reference that is now held by the
	  container. This was done to be more consistent with ao2_unlink(),
	  which automatically releases the reference held by the container.
	  It also makes it so it is no longer possible for a pointer to be
	  invalid after ao2_link() returns. ........

	* /, include/asterisk/astobj2.h: Merged revisions 90310 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) |
	  2 lines Add some notes on the behavior of ao2_unlink() after a
	  discussion with Tilghman ........

2007-11-30 14:45 +0000 [r90270]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6
	  lines Fix locking issues under one legged replaces scenarios.
	  (closes issue #11420) Reported by: irroot Patches:
	  chan_sip_oneleg.patch uploaded by irroot (license 52) ........

2007-11-30 00:16 +0000 [r90164-90232]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov
	  2007) | 5 lines Clear the DTMF buffer if the call times out.
	  (closes issue #11418, reported and patched by eferro) ........

	* /, apps/app_queue.c: Merged revisions 90163 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov
	  2007) | 6 lines This patch handles the case where a queue member
	  with a negative penalty is added via the manager. If a negative
	  value is submitted for a member penalty, we set it to 0. (closes
	  issue #11411, reported and patched by Laureano) ........

2007-11-29 19:35 +0000 [r90156-90162]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29
	  Nov 2007) | 2 lines Properly escape input buffers (Fixes
	  AST-2007-025) ........

	* /, formats/format_wav.c, formats/format_pcm.c,
	  formats/format_ogg_vorbis.c, main/file.c,
	  include/asterisk/mod_format.h, formats/format_h263.c,
	  formats/format_h264.c, formats/format_wav_gsm.c,
	  formats/format_g726.c: Merged revisions 90155 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007)
	  | 5 lines Use of "private" as a field name in a header file
	  messes with C++ projects Reported by: chewbacca Patch by: casper
	  (Closes issue #11401) ........

	* include/asterisk/lock.h: Fix build of trunk

	* /, sounds/Makefile: Merged revisions 90154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007)
	  | 2 lines Upgrade the core sounds release version ........

2007-11-29 13:38 +0000 [r90149-90150]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/Makefile, utils/hashtest.c: let's try this again... *all*
	  compilation and linking in Asterisk should be done using the
	  standard compilation rules, not manually created ones. changing
	  hashtest.c to use these rules caused the compiler to notice a
	  large number of coding guidelines violations, so those are fixed
	  too.

	* main/manager.c: restore behavior from the 1.4 branch... manager
	  users created via users.conf should default to *all* permissions,
	  not none

2007-11-29 00:37 +0000 [r90139-90148]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/channel.h,
	  funcs/func_callerid.c: Merged revisions 90145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) |
	  5 lines This set of changes is to make some callerID handling
	  thread-safe. The ast_set_callerid() function needed to lock the
	  channel. Also, the handlers for the CALLERID() dialplan function
	  needed to lock the channel when reading or writing callerid
	  values directly on the channel structure. ........

	* include/asterisk/file.h, /, main/file.c: Merged revisions 90142
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) |
	  4 lines Merge a change from team/russell/chan_refcount ... This
	  makes ast_stopstream() thread-safe. ........

	* include/asterisk/audiohook.h: Merge another small doxygen change
	  from team/russell/chan_refcount to indicate that a channel
	  doesn't need to be locked before calling a certain function.

	* include/asterisk/channel.h: Merge some channel.h doxygen updates
	  from team/russell/chan_refcount This was mostly to note whether a
	  channel needed to be locked or not before calling these
	  functions. However, I added some other things, too.

2007-11-28 23:03 +0000 [r90102]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions
	  90101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6
	  lines Fix a few memory leaks. (closes issue #11405) Reported by:
	  eliel Patches: load_realtime.patch uploaded by eliel (license 64)
	  ........

2007-11-28 22:44 +0000 [r90100]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/users.conf.sample, main/manager.c, /: Merged revisions
	  90098 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007)
	  | 2 lines it is impossible to set permissions for manager
	  accounts created by users.conf (reported internally, patched by
	  me) ........

2007-11-28 22:32 +0000 [r90099]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: file says... compile before you commit!

2007-11-28 22:17 +0000 [r90060-90061]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Removing a pointless check of option_debug

	* main/pbx.c, /: Merged revisions 90059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov
	  2007) | 13 lines Removing some seemingly pointless code. This
	  sets a channel variable for every priority executed in the
	  dialplan if you have debug set to anything non-zero. This seems
	  pointless due to the fact that these channel variables are not
	  referenced anywhere else in the code and their names are esoteric
	  enough that they would not be practical to reference in the
	  dialplan. Plus the fact that this behavior isn't documented
	  anywhere means that the change is not likely to cause any
	  disruption. If anything, this may actually cause a slight
	  performance increase if running with debug on. The motivating
	  influence for this code change is the eventwhencalled option for
	  queues. If set to vars, all channel variables will be output to
	  the manager. These unnecessary channel variables make the output
	  a lot more difficult to deal with. ........

2007-11-28 20:33 +0000 [r90039]  Steve Murphy <murf@digium.com>

	* main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit
	  transparent

2007-11-28 20:27 +0000 [r90038]  Jason Parker <jparker@digium.com>

	* main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c:
	  Remove "old"-style CLI handler, since nothing uses it anymore.
	  Closes issue #11403, patch by eliel. This also completes the
	  janitor project.

2007-11-28 15:48 +0000 [r89981-89982]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: Hide CLI commands starting with _ from tab completion
	  as was done previously. (closes issue #11395) Reported by: eliel
	  Patches: cli.c.patch uploaded by eliel (license 64)

	* main/abstract_jb.c, res/res_agi.c: Fix a few log messages.
	  (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff
	  uploaded by IgorG (license 20)

2007-11-28 00:49 +0000 [r89947]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Merge some little changes from
	  team/russell/chan_refcount to help reduce the diff to trunk. This
	  just removes some checks on the return value of alloca(), as
	  behavior is undefined if it runs out of stack space, and we don't
	  check it anywhere else.

2007-11-28 00:47 +0000 [r89946]  Mark Michelson <mmichelson@digium.com>

	* configs/musiconhold.conf.sample, configs/extconfig.conf.sample,
	  res/res_musiconhold.c, CHANGES: Adding support for realtime music
	  on hold. The following are the main points: 1. When moh is
	  started, we search first in memory to find the class. If we do
	  not find it in memory, we search realtime instead. 2. When moh is
	  restarted (as in, it had been started on this particular channel,
	  stopped, and now we're starting it again), if using the "files"
	  mode, then realtime will always be rechecked. If you are using
	  other modes, however, we will simply reattach to the external
	  running process which was playing moh earlier in the call. This
	  is a necessary compromise so that we don't end up with too many
	  background processes. 3. musiconhold.conf has a general section
	  now. It has one option: cachertclasses. If set to yes, then moh
	  classes found in realtime will be added to the in-memory list.
	  This has the advantage of not requiring database lookups each
	  time moh is started, but it has the disadvantage of not truly
	  being realtime. I have tested this for functionality, and it
	  passes. I also tested this under valgrind and there are no memory
	  problems reported under typical use. Special thanks to Sergee for
	  implementing this feature and enduring my complaints on the
	  bugtracker! (closes issue #11196, reported and patched by sergee)

2007-11-28 00:24 +0000 [r89840-89915]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) |
	  4 lines - update documentation for some of the goto functions to
	  note that they handle locking the channel as needed - update
	  ast_explicit_goto() to lock the channel as needed ........

	* include/asterisk/channel.h: Document that the channel is not
	  locked when the send_digit_begin and end callbacks get called.

	* main/autoservice.c, /: Merged revisions 89886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) |
	  2 lines Don't do frame processing if ast_read() returned NULL.
	  ........

	* channels/chan_iax2.c: Merge changes from
	  team/russell/iax2_frame_queue This patch is an optimization for
	  chan_iax2. This module is now heavily multi-threaded. However,
	  there is still a good number of globally shared resources that
	  prevent things from happen asynchronously. One of those things
	  was the global IAX frame queue. This queue was used to hold
	  frames that have been deferred for transmitting by another
	  thread, and frames that may need to get retransmitted. I changed
	  the frame queue to be per-call, since almost all of the frame
	  queue handling only cares about frames specific to a call number.

	* /, apps/app_queue.c: Merged revisions 89844 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) |
	  3 lines Instead of depending on the return value of ast_true(),
	  explicitly set the eventwhencalled variable to 1. ........

	* main/pbx.c, /: Merged revisions 89839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) |
	  2 lines Don't start/stop autoservice in pbx_extension_helper()
	  unless a channel exists ........

2007-11-27 23:11 +0000 [r89838]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 89837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov
	  2007) | 12 lines Two changes with regards to the
	  'eventwhencalled' option of queues.conf 1) Due to some signed vs.
	  unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
	  did exactly the same thing. Thus the sign change of the ast_true
	  call. 2) The vars2manager function overwrote a \n for every
	  channel variable it parsed, resulting in bizarre output for the
	  channel variables. This patch remedies this. (related to issue
	  #11385, however I'm not sure if this will actually be enough to
	  close it) ........

2007-11-27 22:42 +0000 [r89835]  Russell Bryant <russell@digium.com>

	* channels/chan_misdn.c: Bring in a small change from
	  team/russell/chan_refcount This replaces tab completion code with
	  the use of a public function that does the same thing

2007-11-27 22:14 +0000 [r89792]  Steve Murphy <murf@digium.com>

	* main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the
	  conditional unlock of the contexts when the hash table is used
	  instead; also, used the ast_free_ptr as advised.

2007-11-27 22:05 +0000 [r89791]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) |
	  41 lines Merge changes from team/russell/autoservice_1.4 This set
	  of changes fixes an issue that was reported to me on IRC
	  yesterday. The user, d1mas, was using chan_zap for incoming calls
	  and was having DTMF recognition issues in some situations.
	  Specifically, he noticed that the problem occurred when using
	  DISA or WaitExten. He also noticed that when using Read, the
	  problem did not occur. His system also used DUNDi for dialplan
	  lookups. So, he theorized that if the DUNDi lookups blocked for
	  some period of time, that audio from the zap channel could get
	  lost. If the audio got lost, then it wouldn't be run through the
	  DTMF detector, and digits could get lost. He was correct, and the
	  following set of changes fixes the problem. However, the changes
	  go a little bit further than what was necessary to fix this exact
	  problem. 1) I updated pbx_extension_helper() to autoservice the
	  associated channel to handle cases where extension lookups may
	  take a long time. This would normally be a dialplan switch that
	  does some lookup over the network, such as the DUNDi or IAX2
	  switches. This ensures that even while a DUNDi lookup is
	  blocking, the channel will be continuously serviced. 2) I made a
	  change to the autoservice code. This is actually something that
	  has bothered me for a long time. When a channel is in
	  autoservice, _all_ frames get thrown away. However, some frames
	  really shouldn't be thrown away. The most notable examples are
	  signalling (CONTROL) frames, and DTMF. So, this patch queues up
	  important frames while a channel is in autoservice. When
	  autoservice is stopped on the channel, the queued up frames get
	  stuck back on the channel so that they can get processed instead
	  of thrown away. 3) I made another change to the autoservice code
	  to handle the case where autoservice is started on channels
	  recursively. Previously, you could call ast_autoservice_start()
	  multiple times on a channel, and it would stop the first time
	  ast_autoservice_stop() gets called. Now, it will ensure that
	  autoservice doesn't actually stop until the final call to
	  ast_autoservice_stop(). ........

2007-11-27 21:10 +0000 [r89769-89772]  Olle Johansson <oej@edvina.net>

	* main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A
	  few more "moremanager" fixes

	* include/asterisk.h, main/asterisk.c, main/loader.c: More
	  "moremanager" fixes. Manager commands to check module status.

	* include/asterisk/manager.h: More "moremanager" changes - doxygen
	  docs and changing manager version (finally) before making more
	  dramatic changes.

	* channels/chan_iax2.c: More additions from the "moremanager"
	  branch, this time for IAX2.

2007-11-27 20:21 +0000 [r89721]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/app.c: Merged revisions 89709 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007)
	  | 2 lines on second thought... revert all the other changes i've
	  made in app options parsing leaving only one: if an empty
	  argument is supplied for an option, set that argument pointer to
	  point to an empty string rather than NULL, so that the
	  application can do normal checks on it without worrying about it
	  being NULL ........

2007-11-27 20:17 +0000 [r89710]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: remove a duplicate manager event

2007-11-27 19:50 +0000 [r89706]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Manager events from the "moremanager"
	  branch

2007-11-27 19:47 +0000 [r89704]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/app.c: Merged revisions 89701 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007)
	  | 2 lines generate a warning when an application option that
	  requires an argument is ignored due to lack of an argument
	  ........

2007-11-27 19:45 +0000 [r89698-89702]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Starting to merge changes from the
	  "moremanager" branch. Documentation will follow.

	* /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The
	  following patch with updates for trunk. Works much better in
	  trunk. Also by accident fixed a bad typo by a previous committer,
	  which actually made video calls not work fully... Merged
	  revisions 89630 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12
	  lines If we get a codec offer using a well-known payload type,
	  but using it for another codec that we don't know, Asterisk did
	  not remove that codec from the list. With this patch, we remove
	  the codec from audio and video rtp objects and deny it ever
	  existed. Thanks to lasse for testing. (closes issue #11376)
	  Reported by: lasse Patches: bug11376.txt uploaded by oej (license
	  306) Tested by: lasse ........

2007-11-27 19:12 +0000 [r89683]  Jason Parker <jparker@digium.com>

	* include/asterisk/strings.h: Add an S_COR macro, which is similar
	  to the existing S_OR macro, except with an additional boolean
	  arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes:
	  S_COR(foo, bar, "baz")

2007-11-27 18:50 +0000 [r89682]  Steve Murphy <murf@digium.com>

	* res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11,
	  pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
	  pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
	  pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22,
	  res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex,
	  pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8:
	  made AEL 8-bit transparent; mainly the lexer was tossing chars
	  with the hi-order bit set. Not nice. Also, allow @ in extension
	  names, and a backslash, also.

2007-11-27 17:01 +0000 [r89637]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Ensure the value returned from ast_random is
	  between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348)
	  Reported by: sperreault

2007-11-27 16:13 +0000 [r89635]  Russell Bryant <russell@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 89634 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) |
	  3 lines Add a note to the sample voicemail config noting that
	  when using IMAP storage, only the first format specified will be
	  attached to the message. ........

2007-11-27 15:41 +0000 [r89632]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_env.c: Merged revisions 89631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007)
	  | 3 lines Default result of STAT should be "0" not "". Reported
	  via the -users mailing list, fixed by me. ........

2007-11-27 07:36 +0000 [r89625]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov
	  2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304)
	  Reported by: pj ........

2007-11-27 06:47 +0000 [r89623]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample,
	  include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1
	  line closes issue #11379; OK, this is an attempt to make both
	  sides happy. To the cdr.conf file, I added the option
	  'unanswered', which defaults to 'no'. In this mode, you will see
	  a cdr for a call, whether it was answered or not. The disposition
	  will be NO ANSWER or ANSWERED, as appropriate. The src is as
	  you'd expect, the destination channel will be one of the channels
	  from the Dial() call, usually the last in the list if more than
	  one chan was specified. With unanswered set to 'yes', you will
	  still see this cdr entry in both cases. But in the case where the
	  dial timed out, you will also see a cdr for each line attempted,
	  marked NO ANSWER, with no destination channel name. The new
	  option defaults to 'no', so you don't see the pesky extra cdr's
	  by default, and you will not see the irritating 'not posted'
	  messages. ........

2007-11-26 23:15 +0000 [r89617-89621]  Mark Michelson <mmichelson@digium.com>

	* pbx/ael/ael-test/ael-test19/extensions.ael,
	  pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt,
	  pbx/ael/ael-test/ael-test3/extensions.ael,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances
	  of "CALLERID(number)" to "CALLERID(num)" for consistency's sake
	  (closes issue #11381, reported and patched by jon)

	* /, apps/app_playback.c: Merged revisions 89618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov
	  2007) | 7 lines After issuing a "say load new", if a caller hangs
	  up during the middle of playback of a number, app_playback will
	  continue to try to play the remaining files. With this change, no
	  more files will be played back upon hangup. (closes issue #11345,
	  reported and patched by IgorG) ........

2007-11-26 22:52 +0000 [r89615]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Update the configure script check for
	  libpri to check for the newest function that was just added.
	  Cresl1n, please keep this in mind when making these changes to
	  libpri or libss7.

2007-11-26 21:23 +0000 [r89613]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Rename
	  "limitonpeer" to "counteronpeer" since the call-limit is
	  deprecated. Both still works in this version.

2007-11-26 21:14 +0000 [r89612]  Joshua Colp <jcolp@digium.com>

	* main/dial.c, /: Merged revisions 89610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2
	  lines Fix issues with async dialing with an application
	  executing. The application has to be terminated and control
	  returned to the thread before hanging things up. (issue #BE-252)
	  ........

2007-11-26 21:12 +0000 [r89606-89611]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting, doxygenification

	* channels/chan_sip.c: Formatting changes, cleaning up some code

	* include/asterisk/doxyref.h, channels/chan_sip.c: Start using
	  Doxygen groupings to group variables and defines.

	* apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark
	  "concise" as deprecated - Restructure other changes to
	  UPGRADE.txt and CHANGES We're still looking for scripts that
	  replace asterisk -rx "show shannels concise" by using the manager
	  interface, but still produces the same output. Anyone?

2007-11-26 18:11 +0000 [r89600-89602]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c, apps/app_queue.c: Perform some module use
	  counting audits. This is now done outside the scope of the
	  application/dialplan function so they do not need to worry about
	  it.

	* /, res/res_features.c: Merged revisions 89599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6
	  lines Add module counting removal for error conditions. (closes
	  issue #11333) Reported by: Laureano Patches:
	  res_features_v2.c.patch uploaded by Laureano (license 265)
	  ........

2007-11-26 17:49 +0000 [r89596]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 89594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) |
	  3 lines Add channel locking to a function that needed to be doing
	  it. This is just a little something I noticed while working on a
	  completely unrelated issue. ........

2007-11-26 17:46 +0000 [r89595]  Steve Murphy <murf@digium.com>

	* utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes
	  issue #11341; made changes to make utils again right with the
	  MTX_PROFILE world.

2007-11-26 17:38 +0000 [r89593]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6
	  lines Use ast_free to free memory, or else we shall implode if
	  MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys
	  Patches: pbx.pbx_config.c.diff uploaded by ys (license 281)
	  ........

2007-11-26 17:26 +0000 [r89591]  Steve Murphy <murf@digium.com>

	* main/hashtab.c: closes issue #11356; Many thanks to snuffy for
	  his code review and changes to cut down duplication. I tested
	  this against hashtest, and it passes. I reviewed the changes, and
	  they look reasonable. I had to remove a few const decls to make
	  things compile on my workstation,

2007-11-26 17:25 +0000 [r89590]  Russell Bryant <russell@digium.com>

	* Makefile: make sure we check to see if the configure script has
	  been executed on a new checkout or after a distclean

2007-11-26 17:23 +0000 [r89589]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov
	  2007) | 6 lines Close the audio file before sending it to the
	  post processing application. (closes issue #11357) Reported by:
	  reformed Patches: mixmonitor.patch uploaded by reformed (license
	  330) ........

2007-11-26 17:21 +0000 [r89588]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007)
	  | 2 lines when parsing application options that take arguments,
	  don't indicate that the option was supplied unless a
	  non-zero-length argument was found for it ........

2007-11-26 16:24 +0000 [r89583]  Steve Murphy <murf@digium.com>

	* main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to
	  pnlarsson for noting the spelling error in the cli commands.
	  Also, added some verbage about the new algorithm to CHANGES.

2007-11-26 16:20 +0000 [r89582]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Revert change for 11348 until it can be looked at
	  even more.

2007-11-26 15:50 +0000 [r89581]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov
	  2007) | 6 lines Revert vmu->email back to an empty string if it
	  was empty when imap_store_file was called. This prevents sending
	  a duplicate e-mail. (closes issue #11204, reported by spditner,
	  patched by me) ........

2007-11-26 15:36 +0000 [r89570-89578]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 89577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6
	  lines If channel allocation fails because the alert pipe could
	  not be created also free the scheduler context. (closes issue
	  #11355) Reported by: eliel Patches: main.channel.c.patch uploaded
	  by eliel (license 64) ........

	* main/utils.c: Make the behavior of using /dev/urandom for random
	  numbers the same as random(). (closes issue #11348) Reported by:
	  sperreault Patches: ast_random2.diff uploaded by sperreault
	  (license 252)

	* channels/chan_sip.c: Instead of printing out one codec in sip
	  show channels print out all of the native ones (this is for
	  video). (closes issue #11366) Reported by: ovi

	* /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4
	  lines When unloading app_meetme destroy any auto created contexts
	  created by SLA. (closes issue #11367) Reported by: eliel ........

	* apps/app_controlplayback.c: Don't crash if the 'o' option of
	  ControlPlayback is used without any value. (closes issue #11375)
	  Reported by: johan

2007-11-25 21:12 +0000 [r89564-89566]  Olle Johansson <oej@edvina.net>

	* channels/chan_usbradio.c: Formatting changes

	* main/channel.c, include/asterisk/channel.h: Try to get channel.h
	  and channel.c aligned in regards to ast_set_callerid as well as
	  change name of variables to follow the rest of the naming.

2007-11-25 17:50 +0000 [r89560-89561]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, res/res_config_odbc.c, /,
	  res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
	  89559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007)
	  | 14 lines We previously attempted to use the ESCAPE clause to
	  set the escape delimiter to a backslash. Unfortunately, this does
	  not universally work on all databases, since on databases which
	  natively use the backslash as a delimiter, the backslash itself
	  needs to be delimited, but on other databases that have no
	  delimiter, backslashing the backslash causes an error. So the
	  only solution that I can come up with is to create an option in
	  res_odbc that explicitly specifies whether or not backslash is a
	  native delimiter. If it is, we use it natively; if not, we use
	  the ESCAPE clause to make it one. Reported by: elguero Patch by:
	  tilghman (Closes issue #11364) ........

	* channels/chan_sip.c: Typo (someone needs to test compile before
	  committing his changes)

2007-11-25 12:18 +0000 [r89551-89557]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: More doxygen changes

	* channels/chan_sip.c: Housekeeping

	* channels/chan_sip.c: Formatting, doxygen updates

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: -
	  Deprecate "call-limit" in chan_sip. No other channel driver
	  enforces call-limits and we now have the groupcount system to
	  implement call-limits in the dialplan. You can use the "setvar"
	  option in realtime/sip.conf to set limits per device. - Implement
	  "callcounter" as a new option to enable the call counting we need
	  to report device status to queue, manager and SIP subscriptions.
	  The call counter setting is now enabled in the code by setting
	  the device call-limit to 999. When we remove the call limit, we
	  can simply enable this with a boolean setting.

	* channels/chan_sip.c, include/asterisk/channel.h: Housekeeping...
	  - Fix typo in chan_sip - Remove changes to caller ID structure,
	  moving it to branch (russellb)

2007-11-24 21:00 +0000 [r89547]  Steve Murphy <murf@digium.com>

	* main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c,
	  configs/extensions.conf.sample: closes issue #11363; where the
	  pattern _20x. buried in an included context, didn't match 2012;
	  There were a small set of problems to fix: 1. I needed NOT to
	  score patterns unless you are at the end of the data string. 2.
	  Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize
	  the patterns in the trie to caps. 3. When a pattern ends with dot
	  or exclamation, CANMATCH/MATCHMORE should always report this
	  pattern, no matter the length. With this commit, I also supplied
	  the wish of Luigi, where the user can select which pattern
	  matching algorithm to use, the old (legacy) pattern matcher, or
	  the new, trie based matcher. The OLD matcher is the default. A
	  new [general] section variable, extenpatternmatchnew, is added to
	  the extensions.conf, and the example config has it set to false.
	  If true, the new matcher is used. In all other respects, the
	  context/exten structs are the same; the tries and hashtabs are
	  formed, but in the new mode the tries are not used. A new CLI
	  command 'dialplan set extenpatternmatch true/false' is provided
	  to allow switching at run time. I beg users that are forced to
	  return to the old matcher to please report the reason in the bug
	  tracker. Measured the speed benefit of the new matcher against an
	  impossibly large context with 10,000 extensions: the new matcher
	  is 374 times faster.

2007-11-24 17:07 +0000 [r89546]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_adsi.c: Merged revisions 89545 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007)
	  | 5 lines Free some frames that would otherwise leak on error.
	  Reported by: Laureano Patch by: Laureano,tilghman (Closes issue
	  #11351) ........

2007-11-24 16:53 +0000 [r89544]  Steve Murphy <murf@digium.com>

	* main/app.c: Added <sys/file.h> include to allow trunk to compile.
	  Hope this doesn't louse thing up.

2007-11-24 13:57 +0000 [r89542-89543]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_h323.c: remove a DEBUG_THREADS message that
	  accesses private lock fields. If needed, the code to extract this
	  information should be implemented in some generic header or
	  library and the function called here. (closed bug #11362)

	* main/acl.c, main/http.c, main/app.c: remove some unnecessary
	  includes

2007-11-24 06:24 +0000 [r89535-89541]  Tilghman Lesher <tlesher@digium.com>

	* /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007)
	  | 9 lines Currently, zero-length voicemail messages cause a
	  hangup in VoicemailMain. This change fixes the problem, with a
	  multi-faceted approach. First, we do our best to avoid these
	  messages from being created in the first place, and second, if
	  that fails, we detect when the voicemail message is zero-length
	  and avoid exiting at that point. Reported by: dtyoo Patch by:
	  gkloepfer,tilghman (Closes issue #11083) ........

	* main/manager.c, /: Merged revisions 89536 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007)
	  | 10 lines Up until this point, the XML output of the manager has
	  been technically invalid, due to the repetition of certain
	  parameters in a single event. This caused various issues for XML
	  parsers, some of which refused to parse at all, given the
	  invalidity of the rendered XML. So this commit fixes the XML
	  output, ensuring that each entity parameter has a unique name,
	  thus ensuring valid XML. Reported by: msetim Patch by: tilghman
	  (Closes issue #10220) ........

	* res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23
	  Nov 2007) | 5 lines Use ESCAPE clause for the first parameter,
	  not just 2nd-Nth parameters. Reported by: apsaras Patch by:
	  tilghman (Closes issue #11353) ........

2007-11-23 15:54 +0000 [r89532-89533]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: put in the necessary hooks for video support
	  in the console. This is a NOP as far as the current code is
	  concerned, but there is already support in ./configure and the
	  Makefiles for the various libraries used by console_video.c (not
	  yet in the tree) so addition is trivial.

	* channels/chan_sip.c: set rtpmap video info according to what is
	  read from SDP; make the format explicit in a debug message; print
	  the audio instead of aggregated peer capability in a debugging
	  msg.

2007-11-23 09:40 +0000 [r89531]  Olle Johansson <oej@edvina.net>

	* include/asterisk/channel.h: Let's start with implementing the
	  base architecture for UTF8 caller ID's so we can handle multiple
	  formats properly. This is not carved in stone, but a proposal to
	  start with. We need to add support for transliterations as well
	  as UTF8 handling, propably with libiconv. Murf is looking into
	  that for the dialplan.

2007-11-23 09:03 +0000 [r89530]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/image.h, formats/format_jpeg.c: formatting
	  cleanup on the header, normalization of the assignment of
	  descriptor fields.

2007-11-23 02:37 +0000 [r89529]  Russell Bryant <russell@digium.com>

	* configs/agents.conf.sample, /: Merged revisions 89527 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) |
	  3 lines mvanbaak pointed out a spelling error in this sample
	  configuration file. While I was at it, I went ahead and tweaked
	  it a little bit more. ........

2007-11-22 07:10 +0000 [r89514-89526]  Luigi Rizzo <rizzo@icir.org>

	* doc/CODING-GUIDELINES: new info on the management of headers

	* apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c,
	  apps/app_milliwatt.c: more header removal

	* include/asterisk/channel.h: formatting cleanup

	* include/asterisk.h, apps/app_read.c, apps/app_record.c,
	  apps/app_echo.c, apps/app_readexten.c,
	  include/asterisk/channel.h, apps/app_system.c,
	  apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h,
	  apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c,
	  include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c,
	  apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c,
	  apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a
	  little bit the content of header files to reduce dependencies. In
	  this commit: - move the ast_register/unregister_app functions to
	  module.h to avoid the need to include pbx.h for the simpler apps;
	  - move the ast_group structure to channel.h to remove the
	  dependency of app.h on linkedlists.h Note, this is a long process
	  that I am doing in small steps. The main difficulty is that now
	  for each subsystem we have a single header (e.g. channel.h)
	  included by the subsystem provider (usually one file, e.g.
	  channel.c) and by its clients (dozens of them, e.g. we have some
	  70+ apps and 30+ functions). This requires the clients to include
	  all the extra headers required by the provider (eg. lock.h,
	  linkedlists.h, definitions of substructures...) even though many
	  of the clients would be just happy with opaque struct
	  declarations and function prototypes. The long term plan is to
	  eventually rectify this structure so that the compilation can
	  become faster, and also APIs are more stable.

	* funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c,
	  apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove
	  some useless includes

	* include/asterisk/audiohook.h, apps/app_dictate.c,
	  apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c,
	  apps/app_mixmonitor.c, apps/app_stack.c,
	  apps/app_controlplayback.c: more removal of redundant headers

	* apps/app_read.c, apps/app_echo.c, apps/app_record.c,
	  apps/app_userevent.c, apps/app_image.c, apps/app_system.c,
	  apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c,
	  apps/app_while.c: remove redundant headers

	* main/file.c, main/netsock.c: more removal of fcntl.h and other
	  system headers

	* codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
	  codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c,
	  res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c,
	  formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c,
	  apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c,
	  codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which
	  are either useless or done elsewhere

	* formats/format_sln.c, formats/format_wav.c,
	  formats/format_ogg_vorbis.c, include/asterisk/_private.h,
	  formats/format_wav_gsm.c, formats/format_ilbc.c,
	  include/asterisk/file.h, formats/format_vox.c,
	  formats/format_pcm.c, main/file.c, formats/format_h263.c,
	  formats/format_g723.c, formats/format_h264.c,
	  include/asterisk/frame.h, formats/format_jpeg.c,
	  formats/format_g726.c, formats/format_gsm.c,
	  formats/format_g729.c: implement the split of file.h and
	  mod_format.h

	* include/asterisk/mod_format.h (added): Add a specific header for
	  providers of file and format handling routines, moving here
	  structs and function declarations formerly in file.h

2007-11-21 23:54 +0000 [r89513]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c,
	  res/res_features.c, apps/app_queue.c, channels/chan_iax2.c:
	  closes issue #11285, where an unload of a module that creates a
	  dialplan context, causes a crash when you do a 'dialplan show' of
	  that context. This is because the registrar string is defined in
	  the module, and the stale pointer is traversed. The reporter
	  offered a patch that would always strdup the registrar string,
	  which is practical, but I preferred to destroy the created
	  contexts in each module where one is created. That seemed more
	  symmetric. There were only 6 place in asterisk where this is
	  done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial,
	  and app_queue. The two apps destroyed the context, but left the
	  contexts. All is fixed now and unloads should be dialplan
	  friendly.

2007-11-21 23:24 +0000 [r89511-89512]  Luigi Rizzo <rizzo@icir.org>

	* funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c,
	  channels/chan_local.c, apps/app_record.c, funcs/func_strings.c,
	  apps/app_sayunixtime.c, apps/app_test.c,
	  apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
	  apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c,
	  channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c,
	  pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c,
	  apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c,
	  channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c,
	  apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c,
	  funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c,
	  channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c,
	  channels/iax2-provision.c, funcs/func_cut.c,
	  apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
	  apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c,
	  apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
	  channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
	  pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
	  apps/app_voicemail.c, channels/chan_unistim.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c,
	  apps/app_readexten.c, funcs/func_vmcount.c,
	  channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c,
	  cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c,
	  channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c,
	  funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
	  apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c,
	  funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c,
	  apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
	  apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c,
	  apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
	  channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
	  channels/chan_jingle.c, channels/chan_usbradio.c,
	  apps/app_channelredirect.c, apps/app_flash.c,
	  apps/app_directed_pickup.c, funcs/func_blacklist.c,
	  channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c,
	  channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c,
	  apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c:
	  remove another set of redundant #include "asterisk/options.h"

	* main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c,
	  main/say.c, res/res_features.c, main/devicestate.c, main/utils.c,
	  res/res_musiconhold.c, res/res_jabber.c, main/indications.c,
	  main/enum.c, res/res_config_sqlite.c, main/config.c,
	  main/loader.c, main/term.c, main/cli.c, main/io.c,
	  main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c,
	  res/res_config_odbc.c, main/manager.c, res/res_agi.c,
	  main/http.c, main/logger.c, res/res_realtime.c, main/app.c,
	  main/image.c, main/dns.c, main/db.c, res/res_speech.c,
	  main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c,
	  main/translate.c, res/res_crypto.c, res/res_adsi.c,
	  main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c,
	  res/res_ael_share.c, res/res_monitor.c, main/rtp.c,
	  main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c,
	  main/adsistub.c, main/abstract_jb.c, main/file.c,
	  main/callerid.c, main/astmm.c, main/audiohook.c,
	  formats/format_g726.c, main/asterisk.c, res/res_odbc.c,
	  main/dsp.c: remove a bunch of useless #include "options.h"

2007-11-21 22:37 +0000 [r89509-89510]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Remove unneccessary explicit case for BRI

	* channels/chan_zap.c: Take some debug code out :-)

2007-11-21 22:20 +0000 [r89508]  Luigi Rizzo <rizzo@icir.org>

	* main/cygload.c: add a missing return

2007-11-21 22:07 +0000 [r89507]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add BRI support to chan_zap

2007-11-21 21:30 +0000 [r89506]  Luigi Rizzo <rizzo@icir.org>

	* utils/Makefile, configure, configure.ac: enable support for stack
	  backtrace for stuff built in utils/ (this was present in the main
	  tree but forgotten here).

2007-11-21 20:38 +0000 [r89505]  Steve Murphy <murf@digium.com>

	* main/pbx.c: closes issue #11290; the proposed patch was a good
	  guess, and would solve the bug to some extent, but was really
	  masking the real issue, that there were bad entries in the table.
	  This fix removes the condition that the hashtab updates be done
	  on exten removal only when the pattern_tree was present, which is
	  silly. The operations that apply to the pattern tree are instead
	  made conditional. Also, threw back in routines that kpfleming
	  deleted because of probs in the 64-bit world. Tested on both 32
	  and 64-bit machines (compile). Tested the reload problem with
	  over 20 reloads, and no problems. If you find more problems,
	  please reopen 11290.

2007-11-21 20:22 +0000 [r89504]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: Simplify comparison in parking fix

2007-11-21 19:28 +0000 [r89494-89496]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 89495 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov
	  2007) | 3 lines Fix a small error I made in my previous commit
	  ........

	* /, apps/app_queue.c: Merged revisions 89493 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov
	  2007) | 5 lines Changing an inaccurate debug message to be less
	  inaccurate. Under the circumstances, this message would always
	  report that there were 0 members available, even though that may
	  not be true. ........

2007-11-21 19:20 +0000 [r89492]  Terry Wilson <twilson@digium.com>

	* /, res/res_features.c: Merged revisions 89491 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) |
	  4 lines If a channel gets masqueraded in the middle of a park,
	  don't play the announcement to the masqueraded channel, and dial
	  back to the original channel on timeout. ........

2007-11-21 18:52 +0000 [r89490]  Russell Bryant <russell@digium.com>

	* main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove
	  the FAX_DETECT define and only do the calculations if fax
	  detection is enabled on the dsp. (closes issue #11331) Reported
	  by: dimas Patches: dsp.patch uploaded by dimas (license 88)

2007-11-21 18:38 +0000 [r89489]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set
	  READSTATUS as an indication of the result Also, some cleanup to
	  CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman
	  (Closes issue #11004)

2007-11-21 18:24 +0000 [r89488]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: fix a small gramatical error in a comment

2007-11-21 18:19 +0000 [r89487]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: There existed about a 1 in 4 billion chance that
	  reading from /dev/urandom would return LONG_MIN (1 in 9
	  quintillion if using 64-bit longs). Since there is no positive
	  equivalent of LONG_MIN, the result of labs() in this case is
	  unpredictable. This fixes that situation. (closes issue #11336,
	  reported and patched by sperreault)

2007-11-21 16:24 +0000 [r89484]  Russell Bryant <russell@digium.com>

	* channels/chan_unistim.c: Fix some code that was supposed to
	  ensure that a buffer was terminated, but was writing to the wrong
	  byte. Also, remove some non-thread safe test code. (closes issue
	  #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by
	  IgorG (license 20) - additional changes by me

2007-11-21 16:08 +0000 [r89483]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: I introduced a deadlock avoidance into 1.4, which I
	  attempted to port to trunk as well. Unfortunately, since trunk
	  uses read/write locks for the context lock, it means that I have
	  actually *introduced* a deadlock condition since they are not
	  recursive. Removing this change for now and will look into
	  introducing a different one.

2007-11-21 16:07 +0000 [r89480-89482]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c,
	  utils/conf2ael.c: move these forward declarations back to
	  asterisk.h where they belong... even though asterisk.h includes
	  compat.h, these declarations have nothing to do with the being
	  platform-compatible and are directly related to being part of
	  Asterisk

	* channels/chan_usbradio.c: get this to actually compile...

	* main/pbx.c: remove some debugging code that doesn't compile on
	  64-bit platforms

2007-11-21 15:17 +0000 [r89478-89479]  Steve Murphy <murf@digium.com>

	* res/res_features.c: OOOps! All the debug stuff I inserted was
	  accidentally committed. I hereby revert it.

	* main/hashtab.c, res/res_features.c: closes issue #11265; Thanks
	  to snuffy for his work on neatening up the code and removing
	  duplicated code.

2007-11-21 08:28 +0000 [r89475-89477]  Luigi Rizzo <rizzo@icir.org>

	* channels/gentone-ulaw.c (removed): remove this file, it is not
	  used anywhere.

	* main/astmm.c: add missing paths.h

	* configure, include/asterisk/autoconfig.h.in, configure.ac: add
	  check for video4linux

2007-11-21 01:09 +0000 [r89474]  Steve Murphy <murf@digium.com>

	* main/pbx.c: A free in add_pri was ultimately the source of the
	  grief we were having with parking. This set of changes fixes that
	  problem, and introduces some more error messages, and puts debugs
	  into ifdefs for what could be short-term usage. Txs to Terry W.
	  for his help, guidance, and especially patience.

2007-11-21 00:23 +0000 [r89472-89473]  Luigi Rizzo <rizzo@icir.org>

	* main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c,
	  main/minimime/mm.h, utils/check_expr.c: more header
	  removal/normalization

	* configure, include/asterisk/autoconfig.h.in, configure.ac: X11
	  checks (at least some - for other platforms with unusual X11
	  locations you might need to add more directories)

2007-11-21 00:21 +0000 [r89470]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, CHANGES: Merge changes from
	  team/russell/sla_trunk_moh ... * Added the ability to specify the
	  music on hold class used to play into the conference when there
	  is only one member and the M option is used. * Added the ability
	  to specify a music on hold class to play instead of ringing for
	  the SLATrunk application. (patched by me, and tested internally)

2007-11-21 00:20 +0000 [r89469]  Luigi Rizzo <rizzo@icir.org>

	* makeopts.in: complete support for X11

2007-11-20 23:29 +0000 [r89467-89468]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk
	  build again

	* main/say.c: Add support for new recorded character sounds Closes
	  issue #5208

2007-11-20 23:16 +0000 [r89465-89466]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
	  apps/app_dictate.c, apps/app_test.c, apps/app_ices.c,
	  apps/app_followme.c, channels/chan_iax2.c, main/config.c,
	  main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c,
	  main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h,
	  res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c,
	  main/app.c, main/image.c, apps/app_directory.c, main/db.c,
	  cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c,
	  include/asterisk/utils.h, include/asterisk.h, main/pbx.c,
	  channels/chan_sip.c, res/res_crypto.c,
	  include/asterisk/channel.h, res/res_monitor.c,
	  include/asterisk/paths.h, main/file.c, apps/app_sms.c,
	  include/asterisk/ael_structs.h, pbx/pbx_config.c,
	  apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
	  move asterisk/paths.h outside asterisk.h and into those files who
	  really need it.

	* main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c,
	  main/threadstorage.c, main/devicestate.c,
	  include/asterisk/_private.h (added), main/astobj2.c,
	  main/loader.c, main/term.c, main/cli.c, main/channel.c,
	  main/manager.c, main/logger.c, build_tools/strip_nonapi,
	  main/event.c, main/asterisk.c, main/db.c: move internal function
	  declarations to include/asterisk/_private.h

2007-11-20 19:29 +0000 [r89464]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: i got a little carried away with commas
	  ...

2007-11-20 19:28 +0000 [r89463]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/module.h, build_tools/make_buildopts_h,
	  main/loader.c: switch compile-time option checking to string
	  storage mode in this branch too

2007-11-20 19:11 +0000 [r89460]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: fix the zaptel configure script check

2007-11-20 18:20 +0000 [r89459]  Luigi Rizzo <rizzo@icir.org>

	* acinclude.m4: the 'version' is now $7 not $6 (wait a bit before
	  regenerating configure, i have more changes)

2007-11-20 17:59 +0000 [r89458]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 89457 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov
	  2007) | 9 lines According to comments in main/pbx.c, it is
	  essential that if we are going to lock the conlock as well as the
	  hints lock, it must be locked in that respective order. In order
	  to prevent a potential deadlock, we need to lock the conlock
	  prior to locking the hints lock in ast_hint_state_changed (see
	  the call stack example on issue #11323 for how this can happen).
	  (closes issue #11323, reported by eelcob, suggestion for patch by
	  eelcob, patch by me) ........

2007-11-20 17:11 +0000 [r89454-89455]  Luigi Rizzo <rizzo@icir.org>

	* makeopts.in: prepare to support console_video

	* apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile,
	  channels/Makefile: Fix building of modules under cygwin. After
	  this commit we can actually load modules under windows, and we
	  can start debugging more interesting problems related to the load
	  order and functionality of modules.

2007-11-20 16:11 +0000 [r89453]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample: Changed occurrences of "busy-level" to
	  "busylevel" in sip.conf.sample in light of commit 89441. Thanks
	  to pj for pointing out the need for this (closes issue #11307,
	  reported by pj)

2007-11-20 15:39 +0000 [r89452]  Luigi Rizzo <rizzo@icir.org>

	* configure, configure.ac, acinclude.m4: add an argument for extra
	  headers to AC_EXT_LIB_CHECK, and on passing simplify the code.
	  Too bad that every time we need to regenerate configure...

2007-11-20 15:30 +0000 [r89451]  Steve Murphy <murf@digium.com>

	* /, doc/tex/queues-with-callback-members.tex: Merged revisions
	  89450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1
	  line closes issue #11324; break statements missing in switch
	  cases. ........

2007-11-20 15:00 +0000 [r89449]  Joshua Colp <jcolp@digium.com>

	* main/translate.c: Minor documentation tweak and if an incorrect
	  parameter is given to core show translation return the usage
	  information. (closes issue #11316) Reported by: eliel Patches:
	  translate.c.patch uploaded by eliel (license 64)

2007-11-20 14:54 +0000 [r89448]  Luigi Rizzo <rizzo@icir.org>

	* configure, acinclude.m4: comment a bit the code in acinclude.m4
	  There is still a lot of code to clean up there, but hopefully
	  this should clarify what goes on in there.

2007-11-20 14:49 +0000 [r89447]  Joshua Colp <jcolp@digium.com>

	* channels/h323/ast_h323.cxx: Include the compatibility header file
	  in ast_h323.cxx for compatibility reasons. (closes issue #11311)
	  Reported by: falves11

2007-11-20 14:44 +0000 [r89444-89446]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fix sip show history. Closes issue #11312

	* channels/chan_sip.c: Change terminology a bit for CLI commands
	  handling SIP channels/calls/dialogs/whatever. Closes issue #11312

2007-11-20 07:42 +0000 [r89443]  Luigi Rizzo <rizzo@icir.org>

	* Makefile, main/Makefile, Makefile.moddir_rules: initial makefile
	  changes to build loadable modules under cygwin (not complete yet
	  - still need to sort out dependecies on res_*)

2007-11-20 00:17 +0000 [r89442]  Steve Murphy <murf@digium.com>

	* main/pbx.c: Get rid of some debug messages in pbx.c

2007-11-19 23:24 +0000 [r89441]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, CHANGES: Changed the "busy-level" option in
	  sip.conf to "busylevel" to be more parallel with the SIPPEER()
	  argument of the same name. The deprecation procedure is not being
	  used here since this is a trunk-only option. (closes issue
	  #11307, reported by pj, patched by me)

2007-11-19 23:03 +0000 [r89439-89440]  Russell Bryant <russell@digium.com>

	* include/asterisk/module.h: Be a bit more pedantic about the type
	  for holding the md5 sum for the build options. Also, doxygenify
	  the comment.

	* funcs/func_sysinfo.c: Make the SYSINFO documentation reflect
	  which options were compiled in

2007-11-19 22:55 +0000 [r89438]  Steve Murphy <murf@digium.com>

	* main/pbx.c: These changes were made in response to
	  niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in
	  two different contexts, included in the same context. In that
	  particular case, we were behaving the same as 1.4, but after
	  experimenting, I quickly found that if 20 and 201 were in the
	  same extension, 1.4 would return 201, and this code returns 20.
	  These changes now enable the current code to replicate the
	  behavior of 1.4 in respect to MATCHMORE in cases like this.

2007-11-19 21:18 +0000 [r89430-89433]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c:
	  another few errno.h removals

	* pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
	  apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c,
	  pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
	  apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c,
	  apps/app_milliwatt.c: more errno.h removal

	* funcs/func_sysinfo.c: remove unnecessary headers

	* funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary
	  includes.

2007-11-19 20:13 +0000 [r89429]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Change delimiter of SIPPEER to be comma
	  (instead of pipe) and further deprecate the old ':' delimiter
	  Reported by: pj Patch by: tilghman Closes issue #11305

2007-11-19 19:51 +0000 [r89424-89428]  Luigi Rizzo <rizzo@icir.org>

	* codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c,
	  codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c,
	  codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c,
	  codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless
	  includes from codecs

	* formats/format_ilbc.c, formats/format_sln.c,
	  formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
	  formats/format_ogg_vorbis.c, formats/format_g723.c,
	  formats/format_h263.c, formats/format_h264.c,
	  formats/format_wav_gsm.c, formats/format_g726.c,
	  formats/format_jpeg.c, formats/format_gsm.c,
	  formats/format_g729.c: format handlers don't need network, lock,
	  channel and scheduler headers

	* include/asterisk.h, include/asterisk/compat.h,
	  include/asterisk/lock.h, utils/extconf.c,
	  include/asterisk/abstract_jb.h: move the declaration of struct
	  ast_channel ast_frame and ast_module to compat.h so it is always
	  available - hopefully this will let us reduce the number of
	  inclusions of channel.h and frame.h

	* main/udptl.c, main/autoservice.c, funcs/func_rand.c,
	  cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c,
	  main/threadstorage.c, main/say.c, funcs/func_env.c,
	  funcs/func_strings.c, main/devicestate.c,
	  cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c,
	  main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c,
	  main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c,
	  main/sched.c, main/pbx.c, funcs/func_timeout.c,
	  funcs/func_math.c, funcs/func_cut.c, main/chanvars.c,
	  main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c,
	  funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c,
	  funcs/func_iconv.c, main/alaw.c, main/asterisk.c,
	  funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c,
	  main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c,
	  funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c,
	  funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c,
	  main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c,
	  funcs/func_groupcount.c, main/manager.c, main/tdd.c,
	  funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c,
	  main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c,
	  funcs/func_db.c, main/dnsmgr.c, main/translate.c,
	  main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c,
	  cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c,
	  funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c,
	  main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c,
	  funcs/func_callerid.c, main/dsp.c: another bunch of include
	  removals (errno.h and asterisk/logger.h)

	* channels/chan_local.c, apps/app_record.c,
	  apps/app_alarmreceiver.c, apps/app_chanisavail.c,
	  apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c,
	  channels/chan_skinny.c, formats/format_pcm.c,
	  apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c,
	  codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c,
	  codecs/codec_g726.c, formats/format_ogg_vorbis.c,
	  apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c,
	  apps/app_speech_utils.c, apps/app_sendtext.c,
	  formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c,
	  apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c,
	  res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c,
	  apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c,
	  channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c,
	  res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c,
	  apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c,
	  apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  apps/app_adsiprog.c, codecs/codec_lpc10.c,
	  res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c,
	  codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c,
	  channels/chan_jingle.c, apps/app_channelredirect.c,
	  apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c,
	  formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c,
	  apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c,
	  apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c,
	  formats/format_wav_gsm.c, res/res_smdi.c,
	  include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c,
	  channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
	  formats/format_jpeg.c, formats/format_gsm.c,
	  apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
	  apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c,
	  channels/chan_agent.c, apps/app_disa.c,
	  channels/iax2-provision.c, res/res_ael_share.c,
	  apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c,
	  channels/chan_misdn.c, apps/app_waitforring.c,
	  apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c,
	  apps/app_zapateller.c, res/res_indications.c,
	  codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c,
	  apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c,
	  codecs/codec_zap.c, res/res_config_sqlite.c,
	  channels/misdn_config.c, apps/app_controlplayback.c,
	  formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c,
	  main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c,
	  res/res_clioriginate.c, apps/app_while.c, include/asterisk.h,
	  apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c,
	  res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
	  res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c,
	  apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c,
	  apps/app_flash.c, apps/app_directed_pickup.c,
	  channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
	  apps/app_senddtmf.c, formats/format_g729.c: include "logger.h"
	  and errno.h from asterisk.h - usage shows that they were included
	  almost everywhere. Remove some of the instances.

2007-11-19 17:18 +0000 [r89422]  Steve Murphy <murf@digium.com>

	* main/pbx.c: a correction to code involved in an extension removal

2007-11-19 16:29 +0000 [r89421]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan
	  function for retrieval of system information

2007-11-19 15:55 +0000 [r89417-89420]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 89419 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6
	  lines Print out the correct filename (features.conf) in the log
	  message when parkpos options are incorrect. (closes issue #11295)
	  Reported by: Laureano Patches: res_features.c.patch uploaded by
	  Laureano (license 265) ........

	* /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov
	  2007) | 4 lines Clarify documentation a bit, include that a frame
	  has to pass through the core in order for the Local channel
	  optimization to happen. (closes issue #11246) Reported by: jon
	  ........

2007-11-19 14:36 +0000 [r89412]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/logger.h: revert inclusion of options.h

2007-11-19 14:03 +0000 [r89410]  Joshua Colp <jcolp@digium.com>

	* apps/app_playback.c: Change warning messages (which are really
	  debug messages) into debug messages. (closes issue #11288)
	  Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded
	  by IgorG (license 20)

2007-11-19 09:16 +0000 [r89404-89407]  Olle Johansson <oej@edvina.net>

	* CHANGES: Update CHANGES

	* channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan
	  function. With this, you can control the peer in the dialplan, so
	  you avoid placing outbound calls when the device has reached
	  busy-level. Reported by pj. Closes bug #11180

	* main/acl.c: Add some debugging to the routines that finds our
	  local IP address. Related to bug #9225

	* channels/chan_sip.c: Make some notes about a problem I found with
	  the OPTIONs handler while working with the bug tracker. Since we
	  don't authenticate devices (peers/users) on OPTIONS we don't have
	  the proper context set for the user/peer. However, we might not
	  want to process an authentication for every OPTIONS, so we could
	  have a config option for this, "optionsforceok" to always answer
	  200 OK on the request and not check device or destination, nor
	  add a SDP. If Asterisk sends the OPTIONs request, it doesn't care
	  about the reply. Some devices use OPTIONs to discover
	  capabilities, since we should answer like an INVITE from the
	  device and we need to support that properly too, which we don't
	  today. So much to do :-)

2007-11-18 21:50 +0000 [r89394-89399]  Joshua Colp <jcolp@digium.com>

	* build_tools/make_buildopts_h: Add OSX into the logic that uses
	  md5 instead of md5sum.

	* include/asterisk/compat.h: Use the easy way that rizzo mentioned,
	  only include malloc.h on the Windows platform.

	* include/asterisk/compat.h: Revert last commit, apparently
	  buildbot lied to me.

	* include/asterisk/compat.h: Change how we handle alloca to conform
	  with how it is suggested in the autoconf manual for
	  AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms
	  should be broken by this.

	* configure, configure.ac: Change autoconf logic a bit so it says
	  what it is looking for in two instances where it didn't.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h, include/asterisk/network.h: Use autoconf
	  logic to determine the presence of
	  PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and
	  PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h
	  in "

2007-11-17 21:47 +0000 [r89393]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add SS7 Generic address support (#11156)

2007-11-17 19:29 +0000 [r89389-89392]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/compat.h: if alloca.h is not present, try
	  malloc.h

	* agi/Makefile: temporarily disable this target in mingw

	* Makefile: will i ever get precedences for windows right ? in the
	  meantime, use a variable to ease enabling/disabling print
	  subdirectories.

	* Makefile: reformulate dependencies in a more correct way

2007-11-17 17:46 +0000 [r89388]  Steve Murphy <murf@digium.com>

	* main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make
	  it so it will compile. Hope I did the right thing. And some
	  additions to removal of extens to take care of hashtab pointers
	  in all cases.

2007-11-17 17:27 +0000 [r89363-89387]  Luigi Rizzo <rizzo@icir.org>

	* Makefile.moddir_rules, Makefile.rules: as discussed some time ago
	  on the -dev list, create embedde object with a .eo suffix even if
	  they are coming from .cc sources. This simplifies the handling in
	  the build scripts.

	* include/asterisk/network.h: prefer socket.h over other variants
	  (winsock etc.)

	* channels/chan_local.c, main/translate.c,
	  channels/chan_features.c, main/http.c, main/config.c: trim more
	  redundant headers

	* main/acl.c: remove unnecessary includes

	* main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c,
	  main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by
	  previous mistake

	* Makefile: wrong variable, wrong order -> broken build.

	* include/asterisk/acl.h, include/asterisk/utils.h,
	  include/asterisk/autoconfig.h.in, include/asterisk/rtp.h,
	  configure.ac, main/acl.c, include/asterisk/netsock.h,
	  main/utils.c, include/asterisk/manager.h, main/netsock.c,
	  main/manager.c, res/res_agi.c, pbx/pbx_dundi.c,
	  include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
	  main/asterisk.c: start using asterisk/network.h for network
	  related headers. Also remove some unnecessary includes.

	* include/asterisk/network.h (added): wrapper for all generic
	  network headers that have different names and locations on the
	  various systems.

	* main/cygload.c: main is called main not amain!

	* main/Makefile: conditional targets for building the windows
	  version

	* Makefile: support cygwin targets

	* Makefile.moddir_rules: and this is the last one to have asterisk
	  compile (not run yet) natively under cygwin.

	* apps/app_sms.c: another cygwin compatibility fix. This one must
	  be handled in a better way in configure, also for other
	  architectures

	* utils/Makefile, main/Makefile, utils/extconf.c: more
	  cygwin/mingw32 compatibility fixes

	* include/asterisk/channel.h: use autoconf results to conditionally
	  compile timersub

	* include/asterisk/lock.h: compatibility fixes for cygwin

	* include/asterisk/compat.h: some version of flex produce code that
	  wants __STDC_VERSION__ defined, but the compiler does not always
	  define it.

	* Makefile: these linker flags apply to both cygwin and mingw32

	* utils/hashtest2.c: add a return NULL to a function that is
	  expected to return a value so compilers that don't understand
	  that this code is NOTREACHED will not complain (the fault is not
	  much on the compiler but on the declaration of pthread_exit on
	  certain platforms) s/certain platform/cygwin/ if you are really
	  curious

	* main/loader.c: define RTLD_LOCAL for platforms that don't have
	  it. This is only to complete the build, clearly the linker
	  behaviour will be completely different and likely to cause
	  trouble in those cases.

	* channels/Makefile: filter out modules that do not compile under
	  windows (this should be handled with the dependencies generated
	  by configure and menuselect, but will be fixed later)

	* main/utils.c: netdb.h is used for gethostbyname, and it was not
	  included in some platforms.

	* main/cygload.c (added): Loader for cygwin where asterisk is
	  really a big dll (something like this is already in 1.2)

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  timersub is a macro not a function, so write the check in a way
	  that detects both formats.

2007-11-17 06:34 +0000 [r89359-89362]  Russell Bryant <russell@digium.com>

	* pbx/pbx_lua.c: fix the build of pbx_lua

	* configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, configure.ac, include/asterisk/io.h,
	  include/asterisk/channel.h: Update the configure script check for
	  sys/poll.h to also provide the result in
	  include/asterisk/autoconfig.h. Also, move the conditional include
	  of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h
	  instead of the two headers it existed in before.

	* build_tools/make_buildopts_h: actually let this compile, oops :(

	* build_tools/make_buildopts_h: Use the fix suggested by Tilghman
	  on the -dev to make cutting up the BUILDSUM friendly to non-bash
	  shells. I think this should work for BSD/mingw as well, but did
	  not yet remove the switch statement.

2007-11-17 04:19 +0000 [r89348-89358]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: linker flags for mingw32

	* configure, include/asterisk/autoconfig.h.in, configure.ac: add
	  detection for timersub() and winsock.h/winsock2.h

	* include/asterisk/endian.h: provide definitions for
	  __LITTLE_ENDIAN and __BIG_ENDIAN if not present.

	* main/Makefile, include/asterisk/io.h, include/asterisk/channel.h:
	  use poll as detected by configure

	* configure, configure.ac, makeopts.in: use autoconf to check for
	  the existence of sys/poll.h

	* build_tools/make_buildopts_h: this script is run on the build
	  system, not on the host.

	* Makefile.moddir_rules: compatibility fix for mingw32

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4, makeopts.in: acinclude.m4: add a function to help
	  checking sdl-config, gtk-config and the like (this could be used
	  for gtk and gtk2 as well) Other files: add tests for sdl,
	  sdl_image and avcodec and regenerate configure and
	  autoconfig.h.in

	* include/asterisk/autoconfig.h.in, configure.ac: add check for the
	  presence of glob

	* channels/chan_jingle.c, channels/chan_unistim.c,
	  funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c,
	  utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
	  apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate
	  #include lines

	* main/udptl.c, funcs/func_module.c, res/res_features.c,
	  funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c,
	  channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c,
	  pbx/pbx_loopback.c, funcs/func_blacklist.c,
	  channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c,
	  main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
	  apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c:
	  remove a bunch of duplicate includes Reproduce with grep -r
	  #include . | grep -v .svn | grep -v Binary | sort | uniq -c |
	  sort -nr

2007-11-16 23:44 +0000 [r89347]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: Fix broken parking dial-back

2007-11-16 23:33 +0000 [r89346]  Steve Murphy <murf@digium.com>

	* main/pbx.c: My goodness, haven't handled an extension deletion.
	  Add code to ast_context_remove_extension2() to remove an
	  extension from the trie. Done by marking it deleted. The
	  scoreboard won't update for it any more. Also, a couple of calls
	  to insert hashtab had a spurious ->exten, which was removed.

2007-11-16 23:28 +0000 [r89341-89345]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/paths.h, include/asterisk.h: paths are already
	  in include/asterisk/paths.h so don't duplicate them in
	  include/asterisk.h

	* include/asterisk/utils.h, include/asterisk/lock.h: whitespace
	  only change - adjust indentation and add some comments on the
	  content of these two files. utils.h (which is included in over
	  150 files) contains a lot of unrelated functions which require
	  the inclusion of a large number of other headers. At some point
	  we should partition its content in a better way.

2007-11-16 21:23 +0000 [r89333-89338]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/logger.h: logger.h does not need options.h

	* include/asterisk/utils.h, channels/chan_sip.c,
	  include/asterisk/astobj.h, include/asterisk/compat.h,
	  include/asterisk/channel.h, include/asterisk/strings.h,
	  utils/extconf.c, include/asterisk/frame.h,
	  include/asterisk/stringfields.h, include/asterisk/endian.h:
	  remove redundant #include "asterisk/compat.h", but make sure that
	  asterisk/compiler.h is included everywhere

	* main/acl.c, main/asterisk.c: remove duplicate headers. Properly
	  check for netdb.h (there is actually tens of places to fix)

	* Makefile.rules: put back default optimization to -O6 (previously
	  changed by mistake)

	* main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c,
	  apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c,
	  channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c,
	  apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c,
	  include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c,
	  main/pbx.c, funcs/func_timeout.c, formats/format_wav.c,
	  apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c,
	  apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c,
	  main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c,
	  main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h,
	  apps/app_voicemail.c, formats/format_sln.c,
	  apps/app_authenticate.c, apps/app_readexten.c,
	  apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules,
	  apps/app_setcallerid.c, include/asterisk/astmm.h,
	  res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c,
	  apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c,
	  main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c,
	  funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
	  res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c,
	  apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c,
	  main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c,
	  apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
	  main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c,
	  cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c,
	  formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c,
	  funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c,
	  main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c,
	  formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c,
	  main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c,
	  channels/iax2-provision.c, res/res_monitor.c, main/netsock.c,
	  apps/app_waitforring.c, main/fixedjitterbuf.c,
	  include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c,
	  channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c,
	  apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c,
	  res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c,
	  res/res_config_sqlite.c, main/fskmodem.c,
	  channels/misdn_config.c, apps/app_controlplayback.c,
	  cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c,
	  channels/chan_phone.c, funcs/func_enum.c, main/dial.c,
	  main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c,
	  main/logger.c, main/image.c, apps/app_ivrdemo.c,
	  res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c,
	  channels/chan_zap.c, main/slinfactory.c, res/res_convert.c,
	  pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c,
	  channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
	  channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c,
	  include/asterisk/strings.h, apps/app_senddtmf.c,
	  funcs/func_callerid.c, include/asterisk/time.h,
	  channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c,
	  apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c,
	  pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c,
	  main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c,
	  apps/app_morsecode.c, formats/format_ogg_vorbis.c,
	  channels/iax2-parser.c, apps/app_speech_utils.c,
	  include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c,
	  funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c,
	  utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c,
	  formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c,
	  channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
	  res/res_jabber.c, funcs/func_uri.c, main/io.c,
	  include/asterisk/abstract_jb.h, main/channel.c,
	  apps/app_minivm.c, res/res_realtime.c, main/dns.c,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
	  codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c,
	  res/res_adsi.c, include/asterisk/plc.h,
	  apps/app_channelredirect.c, formats/format_vox.c,
	  main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c,
	  funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c,
	  cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c,
	  apps/app_sayunixtime.c, apps/app_test.c, main/config.c,
	  main/loader.c, main/term.c, main/cli.c, res/res_smdi.c,
	  include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c,
	  pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
	  include/asterisk/utils.h, apps/app_privacy.c,
	  codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c,
	  funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c,
	  apps/app_transfer.c, include/asterisk/manager.h,
	  apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c,
	  channels/chan_misdn.c, funcs/func_curl.c,
	  channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c,
	  res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c,
	  include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c,
	  funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c,
	  include/asterisk/file.h, main/tdd.c, funcs/func_volume.c,
	  res/res_agi.c, main/app.c, apps/app_parkandannounce.c,
	  cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c,
	  res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
	  main/translate.c, include/asterisk/config.h, main/jitterbuf.c,
	  main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c,
	  funcs/func_extstate.c, apps/app_directed_pickup.c,
	  main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c,
	  main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c,
	  formats/format_g729.c: Start untangling header inclusion in a way
	  that does not affect build times - tested, there is no
	  measureable difference before and after this commit. In this
	  change: use asterisk/compat.h to include a small set of system
	  headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h,
	  stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the
	  inclusion is conditional on HAVE_FOO_H as determined by autoconf.
	  Normally, source files should not include any of the above system
	  headers, and instead use either "asterisk.h" or
	  "asterisk/compat.h" which does it better. For the time being I
	  have left alone second-level directories (main/db1-ast, etc.).

2007-11-16 19:51 +0000 [r89331-89332]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Fixing a problem pointed out by Qwell

	* main/manager.c: Added some locks that should have been around
	  astman_send_error, at least according to the comments. (closes
	  issue #11258, reported and patched by eliel)

2007-11-16 19:26 +0000 [r89329-89330]  Steve Murphy <murf@digium.com>

	* main/pbx.c: This corrects a hashtab removal, given a bad argument

	* main/pbx.c, res/res_features.c: This fixes a problem with pattern
	  ranges; and corrects a situation in res_features, where an
	  extension would be created with the name Zap/51, as an example.
	  THe / is bad because it would tend to mean that the 51 is to be
	  cid matched.

2007-11-16 18:48 +0000 [r89328]  Luigi Rizzo <rizzo@icir.org>

	* build_tools/make_buildopts_h: both md5sum and variable
	  substitutions such as ${BUILDSUM:0:8} are not available in
	  FreeBSD. For the time being, put in a workaround so we can build
	  the system, and wait for the result of the discussion on whether
	  we can store the md5 as a string rather than 4 ints (if so, we
	  won't need more complex tricks with awk or sed for splitting the
	  md5). 1.4 will be fixed when we decide the issue.

2007-11-16 17:11 +0000 [r89327]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Adding confirmation playback when
	  forwarding voicemail messages. This will attempt to play the
	  name(s) of the person(s) to whom you are forwarding the message
	  prior to prompting for prepending. If no name is found, the
	  extension is read back verbatim. (closes issue #9046, reported
	  and patched by jaroth)

2007-11-16 16:56 +0000 [r89326]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/module.h, build_tools/make_buildopts_h,
	  main/loader.c: Merged revisions 89325 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007)
	  | 4 lines To help combat problems where people build external
	  modules (asterisk-addons or others) and then change the build
	  options of the Asterisk build in a way that makes the
	  incompatible without warning, this commit introduces an MD5
	  signature of the important build-time options and includes that
	  signature into modules when they are built. When the loader loads
	  one of these modules and notices the problem, it will emit a
	  warning to console and refuse to initialize the module, as doing
	  so could cause the system to be unstable or even crash. If you
	  upgrade to this version of Asterisk, you must rebuild *all* of
	  your modules that came from other sources before trying to run
	  this version. If you are using Digium's G.729 binary codec
	  module, you will need v33 or newer. ........

2007-11-16 15:44 +0000 [r89324]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 89323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov
	  2007) | 5 lines Make realtime queues accessible from the
	  QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
	  patched by atis, with small modifications from me) ........

2007-11-16 10:07 +0000 [r89322]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/config.h, main/config.c: add a small new
	  function to retrieve variables from a config once we have a
	  pointer to the category.

2007-11-16 10:06 +0000 [r89321]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed #10631, about one way audio. thanks
	  IgorG again.

2007-11-16 09:51 +0000 [r89320]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: move the inner part of config file parsing
	  to a separate function, so it can be reused in the implementation
	  of cli commands when they have a similar syntax.

2007-11-16 08:54 +0000 [r89319]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed compilation of chan_misdn, #11269,
	  thanks IgorG.

2007-11-15 23:50 +0000 [r89299-89312]  Tilghman Lesher <tlesher@digium.com>

	* main/utils.c, include/asterisk/stringfields.h: If we're going to
	  be passing a negative value for the size of a stringfield, in
	  order to indicate something, then using an UNSIGNED parameter is
	  bad, mmmmmkay?

	* Makefile, /: Merged revisions 89302 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007)
	  | 2 lines Start Asterisk in Debian at a more reasonable time
	  (since zaptel is at level 20) ........

	* /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15
	  Nov 2007) | 2 lines Fix an uninitialized memory read found by
	  valgrind ........

	* apps/app_zapscan.c: Fix trunk breakage due to chan->lock being
	  renamed.

	* /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007)
	  | 5 lines Yet another memory corruption issue. Reported by: atis
	  Patch by: tilghman Fixes issue #10923 ........

2007-11-15 17:27 +0000 [r89297]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) |
	  8 lines Update the SLAStation application to account for the case
	  where the SLA thread has a call out to the station, but the user
	  has pressed a line button to answer the call instead of picking
	  up the handset. If they do, the phone sends out a new INVITE. So,
	  the SLAStation app must check to see if it is picking up a
	  ringing trunk, and ensure that the other stations stop ringing.
	  (reported internally, patched by me, tested by mogorman) ........

2007-11-15 16:50 +0000 [r89294-89295]  Steve Murphy <murf@digium.com>

	* main/pbx.c: Get rid of a previously missed ast_log call for
	  debug, no longer nec.

	* main/pbx.c: Perhaps I went overboard on initializing things. I
	  can remove unnecc. stuff later. A few bug fixes. Killing small
	  bugs on the way to killing bigger ones. Removed locking on
	  hashtabs; there's plenty of locks already being taken. A small
	  bug in the root_tree hashtab compare func.

2007-11-15 16:20 +0000 [r89293]  Luigi Rizzo <rizzo@icir.org>

	* main/channel.c, apps/app_channelredirect.c, main/manager.c,
	  res/res_features.c, apps/app_softhangup.c,
	  include/asterisk/channel.h, include/asterisk/lock.h,
	  apps/app_senddtmf.c: access channel locks through
	  ast_channel_lock/unlock/trylock and not through ast_mutex
	  primitives. To detect all occurrences, I have renamed the lock
	  field in struct ast_channel so it is clear that it shouldn't be
	  used directly. There are some uses in res/res_features.c (see
	  details of the diff) that are error prone as they try and lock
	  two channels without caring about the order (or without
	  explaining why it is safe).

2007-11-15 15:39 +0000 [r89290-89291]  Joshua Colp <jcolp@digium.com>

	* UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been
	  used, not 'increasing'.

	* channels/chan_sip.c, channels/chan_h323.c,
	  channels/misdn_config.c: And file said... let trunk build again!
	  Accomplished by some more constification, and marking a function
	  in chan_sip as purposely unused until it is fixed up.

2007-11-15 14:58 +0000 [r89287-89289]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 89288 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov
	  2007) | 3 lines Undoing previous commit since I realize it was
	  wrong ........

	* main/manager.c, /: Merged revisions 89286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov
	  2007) | 4 lines Adding a missing mutex unlock. (closes issue
	  11256, reported and patched by ys) ........

2007-11-15 12:21 +0000 [r89278-89285]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Always relying on the responses when
	  crossing NAT's are not a good solution, it breaks communication.
	  Rizzo - you need to implement a configuration option for this
	  code. It's good, but maybe should be off by default.

	* /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6
	  lines Don't send re-invites during pending INVITE transactions.
	  Patch by one47 - thanks! Closes issue #9305 ........

	* /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5
	  lines Improve support for multipart messages. Code by gasparz,
	  changes by me (mostly formatting). Thanks, gasparz! Closes issue
	  #10947 ........

	* channels/chan_sip.c: Exit early instead of deciding to exit after
	  processing the message.

	* channels/chan_sip.c, configs/sip.conf.sample: Add support for
	  application/dtmf SIP INFO dtmf handling. Yep, another way of
	  handling DTMF in SIP. Totally undocumented, but implemented in
	  enough devices so we have to support it. Code by sergee, small
	  changes by oej. Closes issue #11049

2007-11-15 01:42 +0000 [r89277]  Steve Murphy <murf@digium.com>

	* main/pbx.c: Had trouble playing with parking; spent a long time
	  trying to reason out MATCHMORE mode. made these updates and xfers
	  on zaptel lines seem to work ok now

2007-11-15 00:01 +0000 [r89273-89276]  Tilghman Lesher <tlesher@digium.com>

	* /, main/app.c: Merged revisions 89275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007)
	  | 5 lines When a recording ends with '#', we are improperly
	  trimming an extra 200ms from the recording. Reported by: sim
	  Patch by: tilghman Closes issue #11247 ........

	* main/channel.c: Typo

	* main/channel.c: Add callerid to the Hangup manager event.
	  Reported by: outtolunc Patch by: outtolunc Closes issue #11248

2007-11-14 18:05 +0000 [r89271-89272]  Steve Murphy <murf@digium.com>

	* main/pbx.c: Rescaled the weights of the patterns to give
	  something more independent of pattern length; and make . less
	  likely to win. Question: which should win for 14102241145--
	  _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will
	  win.

	* main/pbx.c: A further problem highlighted by 11233 has been
	  resolved; a certain combination of patterns in a certain order,
	  led to a malformed trie, due to a ptr not being initialized in
	  the loop. Also, some tree printing prettifications.

2007-11-14 15:13 +0000 [r89269-89270]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c,
	  res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more
	  typo in config.c; and missed conversions due to the constifying
	  of ast_variable_new parameters

	* main/config.c: Typo

2007-11-14 13:18 +0000 [r89268]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/acl.h, channels/chan_sip.c,
	  include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c,
	  main/acl.c, pbx/dundi-parser.c, apps/app_queue.c,
	  channels/chan_iax2.c, main/enum.c, channels/chan_oss.c,
	  apps/app_playback.c, main/config.c, pbx/dundi-parser.h,
	  include/asterisk/abstract_jb.h, main/manager.c,
	  channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c,
	  main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c,
	  apps/app_voicemail.c: make the 'name' and 'value' fields in
	  ast_variable const char * This prevents modifying the strings in
	  the stored variables, and catched a few instances where this was
	  actually done. Given the differences between trunk and 1.4 (and
	  the fact that this is effectively an API change) it is better to
	  fix 1.4 independently. These are chan_sip.c::sip_register()
	  chan_skinny.c:: near line 2847 config.c:: near line 1774
	  logger.c::make_components() res_adsi.c:: near line 1049 I may
	  have missed some instances for modules that do not build here.

2007-11-14 03:22 +0000 [r89263-89266]  Russell Bryant <russell@digium.com>

	* main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding
	  guidelines issues ... - handle memory allocation failures - add
	  an ast_ prefix to a publicly exported function - put curly braces
	  in the right places - add a bunch of spaces where they should be
	  be used

	* res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple
	  places - return errors from load_module / unload_module

	* apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the
	  syntax highlighting in my editor happy

	* apps/app_queue.c: Instead of reserving 800 bytes for periodic
	  announcements, use an array of ast_str pointers and only alloate
	  space for the strings as needed.

2007-11-14 01:16 +0000 [r89262]  Joshua Colp <jcolp@digium.com>

	* main/srv.c, /: Merged revisions 89260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4
	  lines Return the proper value when the srv_callback function
	  executes properly. (closes issue #11240) Reported by: jtodd
	  ........

2007-11-14 01:15 +0000 [r89261]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Convert most of the strings in the call_queue
	  struct to use stringfields.

2007-11-14 00:54 +0000 [r89259]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, main/pbx.c: use simpler technique for removing
	  known entries from lists

2007-11-14 00:33 +0000 [r89258]  Russell Bryant <russell@digium.com>

	* main/image.c: - Simplify removing an item from a list - move a
	  verbose message to after the item is added to the list - make use
	  of the ARRAY_LEN macro in one spot

2007-11-13 23:43 +0000 [r89256-89257]  Steve Murphy <murf@digium.com>

	* main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't
	  covered the case of a context with no patterns. (blush)

	* main/pbx.c: closes issue #11233 -- where some fine points in the
	  algorithm to build the tree needed to be corrected. Many thanks
	  for the test case, jtodd

2007-11-13 21:01 +0000 [r89250-89253]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: This fixes a build error on my mac. It
	  also works on my linux box. Let me know if it breaks any other
	  platform ...

	* res/res_features.c: Fix a typo pointed out by outtolunc, thanks
	  :)

	* channels/chan_sip.c: - Convert initialization of a struct to C99
	  style instead of GNU style - Fix a minor spelling error in a
	  comment

	* res/res_features.c, CHANGES: Update the ParkedCall application to
	  grab the first available parked call if no parked extension is
	  provided as an argument. (closes issue #10803) Reported by:
	  outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by
	  outtolunc (license 237) - modified by me to work a bit
	  differently ...

2007-11-13 19:48 +0000 [r89249]  Jason Parker <jparker@digium.com>

	* /, res/res_features.c: Merged revisions 89248 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600
	  (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064.
	  It is documented behavior that if a parking extension already
	  exists while using PARKINGEXTEN, dialplan execution will
	  continue. If blind transferring to a Park with PARKINGEXTEN, you
	  must keep this in mind, and handle the failure yourself. Issue
	  11237, reported by jon. ........

2007-11-13 17:41 +0000 [r89247]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007)
	  | 2 lines If we set a value for qualify, we should actually pay
	  attention to it, instead of overriding the value ........

2007-11-13 16:03 +0000 [r89242]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13
	  Nov 2007) | 5 lines Reverting commit made in revision 89205 since
	  it is unnecessary. Thanks to Kevin for pointing this out ........

2007-11-13 14:03 +0000 [r89240]  Tilghman Lesher <tlesher@digium.com>

	* /, main/utils.c: Merged revisions 89239 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007)
	  | 4 lines Debugging is running into the 16-lock limit. Increase
	  to avoid. (This define is only effective when debugging is turned
	  on, so there's no effect for most installations.) ........

2007-11-13 01:19 +0000 [r89206-89207]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: There is the potential to copy
	  uninitialized memory into the mixmonitor->post_process string.
	  This fix prevents that.

	* /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12
	  Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only
	  1 argument is given, then the args.options and args.post_process
	  strings are uninitialized and could contain garbage. This change
	  handles this situation properly by only using arguments that we
	  have parsed. ........

2007-11-13 00:19 +0000 [r89202-89203]  Jason Parker <jparker@digium.com>

	* Makefile: oops, somebody left out the directory here...

	* channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c,
	  include/asterisk/config.h, res/res_convert.c, res/res_crypto.c,
	  pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h,
	  res/res_config_sqlite.c, res/res_monitor.c,
	  include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen
	  fixes. Also fix a common typo I kept seeing (arguement) in
	  various files. Closes issue #11222, patch by snuffy (with
	  arguement > argument by me).

2007-11-12 23:33 +0000 [r89196-89201]  Steve Murphy <murf@digium.com>

	* utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake
	  of the mtx_prof stuff.

	* include/asterisk/hashtab.h: Thanks to snuffy for this doxygen
	  update to hashtab.h; closes issue #11223

	* main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work,
	  who brought up that these fixes might need to be made.

2007-11-12 20:48 +0000 [r89195]  Jason Parker <jparker@digium.com>

	* main/pbx.c, /: Merged revisions 89194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1
	  line Fix a typo pointed out by De_Mon on #asterisk-dev ........

2007-11-12 20:16 +0000 [r89190]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported
	  by: eliel Patches: utils.Makefile.patch uploaded by eliel
	  (modified by me) (license 64)

2007-11-12 18:44 +0000 [r89186]  Steve Murphy <murf@digium.com>

	* main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
	  funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c,
	  apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in
	  asterisk-dev by Brian Capouch, I determined I too agressive in
	  not initializing arrays passed to pbx_substitute_variables_xxxx;
	  I reviewed the code (again) and hopefully found every possible
	  spot where substitute_variables is called conditionally, and made
	  sure the char array involved was set to a null string.

2007-11-12 17:44 +0000 [r89185]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /, channels/chan_sip.c: Merged revisions 89184
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007)
	  | 5 lines Fix two cases of memory corruption caused by background
	  threads. Reported by: atis Patch by: tilghman Fixes issue #10923
	  ........

2007-11-12 13:36 +0000 [r89178-89179]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged
	  revisions 89173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) |
	  1 line if we're NT and no number was dialed and overlapdial is
	  set, we wait for the ISDN timeout instead of starting our own
	  timer. added a comment for the misdn.conf.sample for the
	  overlapdial config option. ........

	* channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  Merged revisions 89172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) |
	  1 line added restart all interfaces Restart_Indicator, to
	  automatically send a RESTART after the L2 of a PTP Port comes up.
	  Also fixed some places where we have send a RELEASE without need
	  for it. ........

2007-11-12 13:26 +0000 [r89177]  Joshua Colp <jcolp@digium.com>

	* channels/chan_unistim.c, utils/hashtest.c: Fix building on
	  FreeBSD by including/not including some headers. (closes issue
	  #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys
	  (license 281)

2007-11-12 13:22 +0000 [r89174-89176]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 89171 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) |
	  1 line fixed a state/event issue with overlapdial=yes when no
	  extension matched. removed the general sending of a
	  RELEASE_COMPLETE when we receive a RELEASE, this is done by
	  mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with
	  mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to
	  upgrade to at least mISDNuser-1.1.6 (when using the NT mode at
	  all) ........

	* /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12
	  Nov 2007) | 1 line fixed the support for CW and therefore for the
	  reject_cause option. ........

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
	  revisions 89169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) |
	  1 line aded ntkeepcalls option, to avoid droÃpping calls when the
	  L2 goes down on a PTP link. There are some pbx which do turn off
	  the L1 for a very short while and restart it immediately.
	  normally T310 should be started and after 10 seconds or so the
	  calls should be dropped, this is a simple fix wihtout this timer.
	  ........

2007-11-09 18:57 +0000 [r89130-89132]  Jason Parker <jparker@digium.com>

	* configs/usbradio.conf.sample (added): Add usbradio.conf.sample
	  from branches/1.4/configs - r84162. It was mistakenly deleted in
	  1.4 without ever being merged to trunk. Reported by eliel on
	  #asterisk-dev.

	* cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf
	  (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a
	  few potential deadlocks in cdr_sqlite3_custom. (also rename
	  sample config to .sample) Closes issue #11208, patch by Laureano.

2007-11-09 16:00 +0000 [r89129]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c
	  (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c,
	  include/asterisk/hashtab.h (added), main/config.c: This is the
	  perhaps the biggest, boldest, most daring change I've ever
	  committed to trunk. Forgive me in advance any disruption this may
	  cause, and please, report any problems via the bugtracker. The
	  upside is that this can speed up large dialplans by 20 times (or
	  more). Context, extension, and priority matching are all fairly
	  constant-time searches. I introduce here my hashtables
	  (hashtabs), and a regression for them. I would have used the
	  ast_obj2 tables, but mine are resizeable, and don't need the
	  object destruction capability. The hashtab stuff is well tested
	  and stable. I introduce a data structure, a trie, for extension
	  pattern matching, in which knowledge of all patterns is
	  accumulated, and all matches can be found via a single traversal
	  of the tree. This is per-context. The trie is formed on the first
	  lookup attempt, and stored in the context for future lookups.
	  Destruction routines are in place for hashtabs and the pattern
	  match trie. You can see the contents of the pattern match trie by
	  using the 'dialplan show' cli command when 'core set debug' has
	  been done to put it in debug mode. The pattern tree traversal
	  only traverses those parts of the tree that are interesting. It
	  uses a scoreboard sort of approach to find the best match. The
	  speed of the traversal is more a function of the length of the
	  pattern than the number of patterns in the tree. The tree also
	  contains the CID matching patterns. See the source code comments
	  for details on how everything works. I believe the approach
	  general enough that any issues that might come up involving fine
	  points in the pattern matching algorithm, can be solved by just
	  tweaking things. We shall see. The current pattern matcher is
	  fairly involved, and replicating every nuance of it is difficult.
	  If you find and report problems, I will try to resolve than as
	  quickly as I can. The trie and hashtabs are added to the existing
	  context and exten structs, and none of the old machinery has been
	  removed for the sake of the multitude of functions that use them.
	  In the future, we can (maybe) weed out the linked lists and save
	  some space.

2007-11-08 23:53 +0000 [r89124-89126]  Jason Parker <jparker@digium.com>

	* /, main/say.c: Merged revisions 89125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600
	  (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here..
	  Issue 11203, fix described by vma. ........

	* pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be
	  used to check whether it is time to hangup a channel. Closes
	  issue #11202, patch by mnicholson

2007-11-08 22:33 +0000 [r89122-89123]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: app_voicemail failed to build when
	  compiling with IMAP_STORAGE Now it does not.

	* main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one
	  argument. Thanks to snuffy for pointing this out on IRC

2007-11-08 21:27 +0000 [r89121]  Joshua Colp <jcolp@digium.com>

	* funcs/func_env.c: Make func_env build again.

2007-11-08 21:01 +0000 [r89120]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov
	  2007) | 7 lines Rework of the commit I made yesterday to use the
	  already built-in ast_uri_decode function as opposed to my
	  home-rolled one. Also added comments. Thanks to oej for pointing
	  me in the right direction ........

2007-11-08 20:39 +0000 [r89118]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_features.c: convert this code to a more efficient
	  idiom

2007-11-08 18:49 +0000 [r89116-89117]  Jason Parker <jparker@digium.com>

	* res/res_smdi.c: Change a warning to a notice. Issue #11195, patch
	  by eliel

	* /, configs/cdr_adaptive_odbc.conf.sample,
	  configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600
	  (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using
	  sample configuration files. Issue 11195, patch by eliel. ........

2007-11-08 17:32 +0000 [r89113-89114]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan
	  function and deprecate ReadFile.

	* channels/chan_features.c: Fix missed conversion to linkedlists
	  macro change

2007-11-08 16:51 +0000 [r89112]  Mark Michelson <mmichelson@digium.com>

	* /: Blocking changes from previous 1.4 commit

2007-11-08 09:21 +0000 [r89108-89110]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored
	  anyways).

	* main/audiohook.c: use %d and cast to int instead of %zd for
	  size_t object, this helps portability.

	* channels/chan_unistim.c: initialize a variable to silence
	  compiler. The type of warnings emitted depends on the
	  optimization level, at the lower levels the compiler doesn't
	  always understand what the programmer has in mind. In this case I
	  could not understand it either.

2007-11-08 05:36 +0000 [r89106-89107]  Kevin P. Fleming <kpfleming@digium.com>

	* main/srv.c, /: Merged revisions 89105 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007)
	  | 2 lines fix a glaring bug in the new SRV record handling that
	  would cause incorrect weight sorting ........

	* main/autoservice.c, main/frame.c, apps/app_meetme.c,
	  res/res_features.c, funcs/func_strings.c, main/devicestate.c,
	  res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
	  codecs/codec_zap.c, res/res_jabber.c, main/indications.c,
	  main/astobj2.c, main/config.c, main/loader.c, main/cli.c,
	  main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c,
	  main/logger.c, main/app.c, main/image.c, res/res_speech.c,
	  main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c,
	  channels/chan_agent.c, utils/astman.c, apps/app_queue.c,
	  channels/iax2-parser.c, main/srv.c,
	  include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c,
	  main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c,
	  apps/app_voicemail.c: improve linked-list macros in two ways: -
	  the *_CURRENT macros no longer need the list head pointer
	  argument - add AST_LIST_MOVE_CURRENT to encapsulate the
	  remove/add operation when moving entries between lists

2007-11-08 05:00 +0000 [r89104]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007)
	  | 2 lines Typo ........

2007-11-08 02:28 +0000 [r89096-89102]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4
	  lines Do not add a sip: to the beginning of the To URI unless
	  needed. (closes issue #10756) Reported by: goestelecom ........

	* /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6
	  lines Improve the devicestate logic for multiple devices. If any
	  are available then the extension is considered available. (closes
	  issue #10164) Reported by: nic_bellamy Patches:
	  sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
	  ........

	* /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8
	  lines Add support for allowing one outgoing transaction. This
	  means if a response comes back out of order chan_sip will still
	  handle it. I dream of a chan_sip with real transaction support.
	  (closes issue #10946) Reported by: flefoll (closes issue #10915)
	  Reported by: ramonpeek (closes issue #9567) Reported by:
	  atca_pres ........

	* /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4
	  lines If callerid is configured in sip.conf use that for checking
	  the presence of an extension in the dialplan. (closes issue
	  #11185) Reported by: spditner ........

2007-11-07 23:47 +0000 [r89094]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: Merged revisions 89093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007)
	  | 7 lines The member refcount must be incremented, to avoid using
	  it after deallocation. A huge thanks go to lvl- for patiently
	  providing the necessary valgrind output that was necessary to
	  finding this problem of memory corruption. Reported by: lvl-
	  Patch by: tilghman Closes issue #11174 ........

2007-11-07 23:18 +0000 [r89091-89092]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: If imapfolder has been specified in
	  voicemail.conf, we should not connect to INBOX... ever. It may
	  not exist. (closes issue #11151, reported by selsky, patched by
	  me)

	* /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov
	  2007) | 6 lines This patch makes it possible for SIP phones to
	  dial extensions defined with '#' characters in extensions.conf
	  AND maintain their escaped characters when forming URI's (closes
	  issue #10681, reported by cahen, patched by me, code review by
	  file) ........

2007-11-07 22:09 +0000 [r89089]  Steve Murphy <murf@digium.com>

	* /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1
	  line In response to 10578, I just ran 1.4 thru valgrind; some of
	  the config leakage I've already fixed, but it doesn't hurt to
	  double check. I found and fixed leaks in res_jabber, cdr_tds,
	  pbx_ael. Nothing major, tho. ........

2007-11-07 17:45 +0000 [r89086]  Joshua Colp <jcolp@digium.com>

	* channels/h323/ast_h323.cxx: Minor change so chan_h323 builds
	  again.

2007-11-07 13:12 +0000 [r89082-89084]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: remove enter/exit comments when handling subdirectory.
	  If we really want them we can remove the --no-print-directory

	* main/loader.c: remove a debugging message which i forgot in.

	* Makefile: match changes in menuselect's Makefile

2007-11-07 04:21 +0000 [r89077-89081]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_playback.c: Suppress erroneous warnings on load.
	  Reported by: eliel Patch by: eliel Closes issue #11177

	* /, configs/extensions.ael.sample: Merged revisions 89079 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007)
	  | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch
	  by: eliel Closes issue #11178 ........

	* channels/chan_zap.c, configs/zapata.conf.sample: Provide the
	  ability to directly manipulate the TON/NPI bits in the
	  dialstring. Reported by: thetatag Patch by:
	  thetatag/stevens/tilghman Closes issue #5331

	* contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy,
	  which provides FastAGI functionality for EAGI, while also
	  buffering the audio stream. Reported by: devil_slayer Patch by:
	  devil_slayer Closes issue #8921

2007-11-07 00:16 +0000 [r89076]  Russell Bryant <russell@digium.com>

	* main/astmm.c: Fix another CLI command so it doesn't run the real
	  code when called for initialization.

2007-11-07 00:04 +0000 [r89075]  Mark Michelson <mmichelson@digium.com>

	* doc/tex/imapstorage.tex: Adding documentation regarding
	  imapfolder, imapgreetings, and greetingsfolder options in
	  voicemail.conf (closes issue #11133, reported by selsky, patched
	  by blitzrage)

2007-11-07 00:00 +0000 [r89073-89074]  Russell Bryant <russell@digium.com>

	* include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the
	  channel name as a prefix to the "agi debug" output. This makes
	  AGI debugging on busy systems much easier. (closes issue #10730)
	  Reported by: junky Patches: agi_debug_chan.diff uploaded by junky
	  (license 177) 20070923_10730.diff uploaded by mvanbaak (license
	  7)

	* apps/app_meetme.c, CHANGES: Added the ability to do "meetme
	  concise" with the "meetme" CLI command. This extends the concise
	  capabilities of this CLI command to include listing all
	  conferences, instead of an addition to the other sub commands for
	  the "meetme" command. (closes issue #11078) Reported by: jthomas
	  Patches: meetme-concise.patch uploaded by jthomas (license 293)

2007-11-06 23:08 +0000 [r89072]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Fix up some PBX logic that became broken. The code
	  would exit prematurely when it should have been collecting more
	  digits. (closes issue #11175) Reported by: pj

2007-11-06 22:51 +0000 [r89071]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_jingle.c, channels/chan_phone.c,
	  codecs/codec_g722.c, main/frame.c, channels/chan_sip.c,
	  channels/chan_skinny.c, main/translate.c, channels/chan_h323.c,
	  main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h,
	  main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h:
	  Commit some cleanups to the format type code. - Remove the
	  AST_FORMAT_MAX_* types, as these are consuming 3 out of our
	  available 32 bits. - Add a native slin16 type, so that 16kHz
	  codecs can translate without losing resolution. (This doesn't
	  affect anything immediately, until another codec has wb support.)

2007-11-06 22:36 +0000 [r89070]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
	  queue strategy wrandom (closes issue #10942, reported and patched
	  by julianjm, documentation changes by me)

2007-11-06 22:15 +0000 [r89069]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added
	  the S() and L() options to the MeetMe application. These are
	  pretty much identical to the S() and L() options to Dial(). They
	  let you set timeouts for the conference, as well as have warning
	  sounds played to let the caller know how much time is left, and
	  when it is running out. (closes issue #8030) Reported by: areski
	  Patches: meetme_timeout_timelimit_v2.patch uploaded by areski
	  (license 29)

2007-11-06 22:05 +0000 [r89068]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added CLI and manager commands for changing a
	  queue member's penalty (closes issue #9374, reported and
	  initially patched by wuwu, intermediate patch by eliel, and final
	  patch by me)

2007-11-06 22:01 +0000 [r89067]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add some more locking as well as API update
	  for libss7 for new transport types

2007-11-06 21:08 +0000 [r89062]  Steve Murphy <murf@digium.com>

	* /, main/config.c: Merged revisions 89036 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1
	  line closes issue #8786 - where the [catname](!) and
	  [catname](othercat1,othercat2,...) notation gets dropped across a
	  ConfigUpdate (or any other thing that would cause a config file
	  to be written). While I was at it, I also cleaned up some of the
	  destroy routines to free up comments, which was not being done.
	  Made sure the new struct I introduced is also cleaned up properly
	  at destruction time. My code handles multiple template
	  inclusions. Many thanks to ssokol for his patch, which, while not
	  literally used in the final merge, served as a foundation for the
	  fix. ........

2007-11-06 20:55 +0000 [r89057]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Remove native bridging check for DTMF based
	  transfers. Thanks to the last batch of RTP changes it is no
	  longer required for the media stream to go through Asterisk if
	  DTMF is going over signalling. It will simply reinvite back as
	  needed. (closes issue #11172) Reported by: ibc

2007-11-06 20:32 +0000 [r89055]  Mark Michelson <mmichelson@digium.com>

	* res/res_features.c: Instead of trying to callback a local channel
	  on a failed attended transfer, call the device that made the
	  transfer instead. This makes for much smoother calling back when
	  queues are involved. (closes issue #11155, reported by IPetrov)
	  Tremendous thanks to Russell for pulling me out of my block I was
	  having on this one

2007-11-06 20:22 +0000 [r89052-89054]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06
	  Nov 2007) | 3 lines Fix init_classes() so that classes that
	  actually do have files loaded aren't treated as empty, and
	  immediately destroyed ... ........

	* main/astmm.c: Fix the memory show allocations CLI command so that
	  it doesn't spew out all of the current memory allocations when
	  you start Asterisk, when the command's handler gets called for
	  initialization.

2007-11-06 19:40 +0000 [r89051]  Steve Murphy <murf@digium.com>

	* main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in
	  OSX for a problem blitzrage found

2007-11-06 19:23 +0000 [r89050]  Olle Johansson <oej@edvina.net>

	* main/fskmodem.c: Formatting. Illegaly using some spare spaces
	  from Russell's space-bucket.

2007-11-06 19:16 +0000 [r89049]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06
	  Nov 2007) | 2 lines We went to the trouble of creating a method
	  of tracking failed trylocks, then never turned it on (oops).
	  ........

2007-11-06 19:10 +0000 [r89048]  Olle Johansson <oej@edvina.net>

	* main/tdd.c, include/asterisk/tdd.h: Additional TDD changes
	  (preparing for SIP changes - adding TDD support to SIP)

2007-11-06 19:10 +0000 [r89047]  Jason Parker <jparker@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4
	  lines Correctly set the total number of channels from a zaptel
	  transcoder board. SPD-49, patch by Matthew Nicholson. ........

2007-11-06 19:04 +0000 [r89044]  Mark Michelson <mmichelson@digium.com>

	* apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c,
	  apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c,
	  apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c,
	  apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c,
	  apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c,
	  apps/app_softhangup.c, apps/app_waitforsilence.c,
	  channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c,
	  apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c,
	  apps/app_forkcdr.c, apps/app_waitforring.c,
	  apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c,
	  res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
	  apps/app_stack.c: "show application <foo>" changes for clarity.
	  (closes issue #11171, reported and patched by blitzrage) Many
	  thanks!

2007-11-06 19:04 +0000 [r89043]  Olle Johansson <oej@edvina.net>

	* /, main/tdd.c: Merged revisions 89042 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2
	  lines Bug fixes to tdd support in zaptel. ........ (Small changes
	  for trunk)

2007-11-06 18:44 +0000 [r89041]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, include/asterisk/jabber.h,
	  channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle
	  to use TLS connections again. Closes issue #9972

2007-11-06 18:23 +0000 [r89038]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06
	  Nov 2007) | 11 lines If someone were to delete the files used by
	  an existing MOH class, and then issue a reload, further use of
	  that class could result in a crash due to dividing by zero. This
	  set of changes fixes up some places to prevent this from
	  happening. (closes issue #10948) Reported by: jcomellas Patches:
	  res_musiconhold_division_by_zero.patch uploaded by jcomellas
	  (license 282) Additional changes added by me. ........

2007-11-06 17:10 +0000 [r89034]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4
	  lines Make it so that if a peer is determined to be unreachable
	  using qualify their devicestate will report back unavailable.
	  (closes issue #11006) Reported by: pj ........

2007-11-06 17:05 +0000 [r89031]  Luigi Rizzo <rizzo@icir.org>

	* main/loader.c: Fix embedding of modules on FreeBSD: the
	  constructor for the list of modules was run after the
	  constructors for the embedded modules (which appended entries to
	  the list). As a result, the list appeared empty when it was time
	  to use it. On linux the order of execution of constructor was
	  evidently different (it may depend on the ordering of modules in
	  the ELF file). This is only a workaround - there may be other
	  situations where the execution of constructors causes problems,
	  so if we manage to find a more general solution this workaround
	  can go away.

2007-11-06 16:29 +0000 [r88974-88995]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, /, configs/zapata.conf.sample: Merged
	  revisions 88994 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6
	  lines Fix improbable but possible memory leaks in chan_zap.
	  (closes issue #11166) Reported by: eliel Patches:
	  chan_zap.c.patch uploaded by eliel (license 64) ........

	* channels/chan_agent.c: Update chan_agent documentation. Change a
	  | to , as that is now the required way. (closes issue #11167)
	  Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel
	  (license 64)

2007-11-06 15:01 +0000 [r88973]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_unistim.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Set up detection
	  of IP_PKTINFO in autoconf for chan_unistim

2007-11-06 14:17 +0000 [r88932-88937]  Russell Bryant <russell@digium.com>

	* channels/chan_unistim.c: convert uses of LOG_DEBUG to use
	  ast_debug()

	* channels/chan_unistim.c, configs/unistim.conf.sample: Add
	  jitterbuffer support to chan_unistim. (closes issue #11168)
	  Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by
	  IgorG (license 20)

	* main/pbx.c, /, channels/busy.h, channels/ringtone.h,
	  include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) |
	  12 lines After seeing crashes related to channel variables, I
	  went looking around at the ways that channel variables are
	  handled. In general, they were not handled in a thread-safe way.
	  The channel _must_ be locked when reading or writing from/to the
	  channel variable list. What I have done to improve this situation
	  is to make pbx_builtin_setvar_helper() and friends lock the
	  channel when doing their thing. Asterisk API calls almost all
	  lock the channel for you as necessary, but this family of
	  functions did not. (closes issue #10923, reported by atis)
	  (closes issue #11159, reported by 850t) ........

	* /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06
	  Nov 2007) | 8 lines Remove some checks to see if locks are
	  initialized from the non-DEBUG_THREADS versions of the lock
	  routines. These are incorrect for a number of reasons: - It
	  breaks the build on mac. - If there is a problem with locks not
	  getting initialized, then the proper fix is to find that place
	  and fix the code so that it does get initialized. - If additional
	  debug code is needed to help find the problem areas, then this
	  type of things should _only_ be put in the DEBUG_THREADS
	  wrappers. ........

2007-11-06 08:17 +0000 [r88898-88913]  Luigi Rizzo <rizzo@icir.org>

	* channels/Makefile: explain that the host environment must be used
	  to build gentone; Remove unset variables, they would be
	  misleading.

	* Makefile: don't export variables that can be retrieved from
	  makeopts in child subdirs

2007-11-06 02:53 +0000 [r88863]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05
	  Nov 2007) | 2 lines update comment to match the state of the code
	  ........

2007-11-05 23:31 +0000 [r88827]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 88826 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov
	  2007) | 6 lines Reworked deadlock avoidance in __ast_read.
	  Restored audio to callback agents. (closes issue #11071, reported
	  by callguy, patched by me, tested by callguy and Ted Brown)
	  ........

2007-11-05 21:36 +0000 [r88770]  Luigi Rizzo <rizzo@icir.org>

	* Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level
	  Makefile. This too is used only in one place.

2007-11-05 21:35 +0000 [r88769]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) |
	  8 lines When traversing the list of channel variables here in
	  transmit_invite(), the asterisk channel must be locked, as this
	  data may change at any time. (I have seen numerous reports of
	  crashes related to the handling of channel variables. There are a
	  couple of issues on the bug tracker related to it, but it has
	  also been noted on IRC and mailing lists. So, I am finding and
	  fixing some places where channel variables are handled
	  improperly.) ........

2007-11-05 21:27 +0000 [r88767]  Luigi Rizzo <rizzo@icir.org>

	* Makefile, main/Makefile: Move the last instance of AST_LIBS to
	  the only place it is used, namely main/Makefile . I am unclear
	  where decisions on the build environment (CFLAGS, LDFLAGS, LIBS
	  and so on) should be made - right now they are split here and
	  there. As a first step in cleaning up this situation, i am trying
	  to at least collect all instances of each variable in one place.

2007-11-05 21:23 +0000 [r88766]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) |
	  2 lines Fix up some indentation. ........

2007-11-05 20:50 +0000 [r88764]  Luigi Rizzo <rizzo@icir.org>

	* Makefile.moddir_rules: comment out an unused variable. Remove it
	  in a few days if no problems arise.

2007-11-05 20:44 +0000 [r88710-88740]  Russell Bryant <russell@digium.com>

	* main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) |
	  7 lines Merge changes from
	  asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
	  record support in Asterisk was broken. There was no guarantee on
	  what record Asterisk would choose to actually use. This set of
	  changes improves the situation by ensuring that Asterisk will
	  choose the highest priority record. ........

	* main/channel.c, /: Merged revisions 88709 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) |
	  20 lines Merge the last bit of changes from
	  asterisk/team/russell/readq-1.4 The issue here is that the
	  channel frame readq handling got broken when the code was
	  converted to use the linked list macros. It caused corruption of
	  the list head and tail pointers. So, I fixed up the usage of the
	  linked list macros and in passing, simplified the code. I also
	  documented what the code is doing, as it was a bit difficult to
	  figure out at first. This bug showed itself with crashes showing
	  messed up head/tail pointers for the readq. However, there are a
	  couple of crashes that aren't quite as obvious, but I think may
	  be related. So, if your bug gets closed by this commit, but you
	  still have a problem, please reopen or create a new bug report.
	  (closes issue #10936) (closes issue #10595) (closes issue #10368)
	  (closes issue #11084) (closes issue #10040) (closes issue #10840)
	  ........

2007-11-05 19:22 +0000 [r88675]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: Cleanup the installation of samples, avoiding
	  repetitions. I am preserving the behaviour on *.adsi files, i.e.
	  overwrite anything there without making a backup. However I am
	  not sure that this is the intended behaviour.

2007-11-05 18:52 +0000 [r88673]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7
	  lines If a SIP channel is put on hold multiple times do not keep
	  incrementing the onHold value. (closes issue #11085) Reported by:
	  francesco_r Tested by: blitzrage (closes issue #10474) Reported
	  by: acennami ........

2007-11-05 18:22 +0000 [r88653]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Change wording to that suggested by MasterYoda

2007-11-05 18:00 +0000 [r88652]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: simplify (hopefully) the printing of $(MAKE) in aligned
	  output.

2007-11-05 17:52 +0000 [r88651]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 88624 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) |
	  5 lines Fix up datastore handling in ast_do_masquerade(). The
	  code is intended to move any channel datastores from the old
	  channel to the new one. However, it did not use the linked list
	  macros properly to accomplish the task. The existing code would
	  only work if there was only a single datastore on the old
	  channel. ........

2007-11-05 17:44 +0000 [r88587-88615]  Luigi Rizzo <rizzo@icir.org>

	* Makefile: print messages when entering/leaving a directory so we
	  know where we are (sometimes it is obvious, sometimes it is not).

	* Makefile.moddir_rules: merge two rules with the same right hand;
	  document a bit what is done here.

2007-11-05 17:21 +0000 [r88586]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600
	  (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config
	  structure on configuration failure. Issue 11163, patch by eliel.
	  ........

2007-11-05 17:00 +0000 [r88584]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile.rules: use a variable name that actually indicates what
	  it is for

2007-11-05 16:41 +0000 [r88553]  Luigi Rizzo <rizzo@icir.org>

	* Makefile.rules: Put extra compiler flags into a variable so they
	  are not repeated too many times. On passing, add some comments
	  and fix indentation a bit. On passing, i suspect that the
	  following pattern is wrong %.eoo: %.o but in case it will be
	  fixed in a later commit.

2007-11-05 16:30 +0000 [r88540]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 88539 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007)
	  | 4 lines Don't check used pooled connections for connection
	  status, as it will cause issues for prepared queries. Reported
	  by: Nick Gorham (via -dev list) Patch by: tilghman ........

2007-11-05 15:15 +0000 [r88525]  Luigi Rizzo <rizzo@icir.org>

	* main/db.c: remove a cygwin-specific function remap that does not
	  work.

2007-11-05 13:11 +0000 [r88510]  Joshua Colp <jcolp@digium.com>

	* channels/chan_unistim.c: Fix memory leaks and deadlocks in
	  chan_unistim. (closes issue #11158) Reported by: eliel Patches:
	  chan_unistim.c.patch uploaded by eliel (license 64)

2007-11-04 22:42 +0000 [r88454-88490]  Luigi Rizzo <rizzo@icir.org>

	* /: block merging of not-applicable patch

	* main/channel.c, main/pbx.c, apps/app_meetme.c,
	  channels/chan_sip.c, res/res_features.c, main/utils.c,
	  channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify
	  the implementation and the API for stringfields; details and
	  examples are in include/asterisk/stringfields.h. Not applicable
	  to older branches except for 1.4 which will receive a fix for the
	  routines that free memory pools.

2007-11-03 14:19 +0000 [r88437]  Tilghman Lesher <tlesher@digium.com>

	* main/term.c: Revert commit #86119. Some users intentionally do
	  not want colorized terminals, so this was a misfeature.

2007-11-03 04:55 +0000 [r88422]  James Golovich <james@gnuinter.net>

	* main/db.c: Set CLI command to the correct name. Rev 85460
	  introduced two 'database show' commands when this one should have
	  been 'database showkey'

2007-11-02 22:36 +0000 [r88368-88409]  Russell Bryant <russell@digium.com>

	* channels/chan_unistim.c: fix some issues with crashing on unload,
	  when it didn't completely load cleanly

	* channels/chan_unistim.c: Convert the CLI commands to the new
	  format

	* pbx/pbx_lua.c: propagate the DECLINE return value back to the
	  loader

	* pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not
	  present.

	* main/cli.c: Show the channel unique ID in the "show channel
	  concise" output (closes issue #11148, requested by falves11,
	  patched by me)

	* channels/chan_unistim.c (added), CREDITS,
	  configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt
	  (added): Merge the code from asterisk/team/group/chan_unistim:
	  This introduces a new channel driver, chan_unistim, that supports
	  the Unistim VoIP protocol for Nortel phones. The following models
	  have been confirmed to work: i2002, i2004 and i2050. (closes
	  issue #8864) Reported by: c_hans Patches: chan_unistim.patch
	  uploaded by c (license 304) ustm_no_conf.diff uploaded by junky
	  (license 177) Tested by: c_hans, dbowerman, math, junky, loloski

2007-11-02 20:51 +0000 [r88329-88367]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4
	  lines Make subscribecontext behave as advertised. It will now
	  look for the presence of a hint in the given context (be it
	  subscribecontext or context). (closes issue #10702) Reported by:
	  slavon ........

	* /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6
	  lines If an INFO request within a dialog is received with a
	  content length of 0 simply send back a 200 OK. It is valid to do
	  this and the remote side is probably using it to make sure the
	  signalling is still alive. (closes issue #5747) Reported by:
	  chandi Patches: infofix-81430-1.patch uploaded by IgorG (license
	  20) ........

2007-11-02 20:13 +0000 [r88327]  Russell Bryant <russell@digium.com>

	* doc/tex/Makefile: Fix replacing the version number when it has a
	  '/' in it, like SVN-group-chan_unistim-r88326M-/trunk

2007-11-02 17:34 +0000 [r88287]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90

2007-11-02 16:54 +0000 [r88284]  Jason Parker <jparker@digium.com>

	* /, main/say.c: Merged revisions 88283 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500
	  (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a
	  language to ast_fileexists, otherwise it may fail for anything
	  besides en Issue 11147, fix discovered by both citats and myself
	  (independently), with input from Corydon76 ........

2007-11-02 16:26 +0000 [r88209-88267]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Add a few bytes on LUA

	* main/pbx.c, utils/build-extensions-conf.lua (added),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c
	  (added), configs/extensions.lua.sample (added),
	  include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of
	  doing extensions Reported by: mnicholson Patch by: mnicholson
	  Closes issue #11140

	* main/config.c: Don't re-cache the filename, but check to see if
	  it already exists Reported by: jamesgolovich Patch by:
	  jamesgolovich Closes issue #11144

	* /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02
	  Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy
	  Patch by: ys Closes issue #11143 ........

	* main/pbx.c: 'h' extension doesn't execute past first priority
	  Reported by: dimas Patch by: dimas Closes bug #11146

2007-11-02 03:09 +0000 [r88197]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_odbc.c: Restore building under 64-bit platforms.

2007-11-01 23:26 +0000 [r88184]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in:
	  Remove traces of gnutls, since we no longer use/need it.

2007-11-01 23:26 +0000 [r88182-88183]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Modify WaitExten to include an optional dialtone
	  Closes issue #10783

	* UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc
	  managed connections Closes issue #10614

2007-11-01 22:26 +0000 [r88166]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c,
	  funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c,
	  apps/app_exec.c, apps/app_queue.c, apps/app_playback.c,
	  res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c,
	  apps/app_minivm.c, res/res_agi.c, main/logger.c,
	  pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c,
	  utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c,
	  apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c,
	  cdr/cdr_manager.c: This commits the performance mods that give
	  the priority processing engine in the pbx, a 25-30% speed boost.
	  The two updates used, are, first, to merge the
	  ast_exists_extension() and the ast_spawn_extension() where they
	  are called sequentially in a loop in the code, into a slightly
	  upgraded version of ast_spawn_extension(), with a few extra args;
	  and, second, I modified the substitute_variables_helper_full, so
	  it zeroes out the byte after the evaluated string instead of
	  demanding you pre-zero the buffer; I also went thru the code and
	  removed the code that zeroed this buffer before every call to the
	  substitute_variables_helper_full. The first fix provides about a
	  9% speedup, and the second the rest. These figures come from the
	  'PIPS' benchmark I describe in blogs, conf. reports, etc.

2007-11-01 22:19 +0000 [r88164-88165]  Jason Parker <jparker@digium.com>

	* /: Crap, accidentally copied the props. Thanks for pointing this
	  out mvanbaak. The odds are quite high that this will break
	  automerge on every team branch.

	* /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber
	  to use openssl rather than gnutls. Closes issue #9972, patch by
	  phsultan. Copied from branch at
	  http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/

2007-11-01 17:25 +0000 [r88117]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01
	  Nov 2007) | 2 lines Add some notes on using valgrind ........

2007-11-01 16:22 +0000 [r88079]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4
	  lines Make sure we set the poll fds to NULL after free()ing it.
	  Part of issue 11017, patch by tzafrir. ........

2007-11-01 15:56 +0000 [r88062-88077]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free()
	  to ast_free(). (No functional differences.) (closes issue #11138)
	  Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel
	  (license 64) chan_sip.c.patch uploaded by eliel (license 64)

	* utils/Makefile: Remove another copied source file on "make
	  clean". (closes issue #11137) Reported by: IgorG Patches:
	  addonclean-87971-1.patch uploaded by IgorG (license 20)

2007-11-01 13:30 +0000 [r88027]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2
	  lines Fix up commit for my Zap channel with spies in Meetme fix.
	  (thanks Tony Mountifield!) ........

2007-11-01 06:12 +0000 [r88007-88010]  Tilghman Lesher <tlesher@digium.com>

	* main/utils.c: Conditionally free lock_info->thread_name to avoid
	  a useless warning Reported by: snuffy Patch by: snuffy Closes
	  issue #11125

	* apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to
	  pair calls of ast_malloc and ast_calloc Reported by: eliel Patch
	  by: eliel Closes issue #11135

	* cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed
	  by: tilghman Closes issue #11136

2007-11-01 01:55 +0000 [r87953-87971]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4
	  lines If a Zap channel contains a spy or a spy is added take it
	  out of the conference in kernel space and make it go through
	  Asterisk so the spy gets audio from both sides. (closes issue
	  #10060) Reported by: mparker ........

	* main/pbx.c: Drop any more references to type in the Exception
	  dialplan function. (closes issue #11134) Reported by: blitzrage
	  Patches: exception_patch.txt uploaded by blitzrage (license 10)

2007-10-31 21:23 +0000 [r87889-87909]  Jason Parker <jparker@digium.com>

	* /, res/res_jabber.c: Merged revisions 87908 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500
	  (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated
	  memory before returning. Issue 11131, patch by eliel. ........

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 87906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11130) (closes issue #11132) ........ r87906 | qwell |
	  2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try
	  to allocate memory that we're just going to re-allocate later
	  anyways. Issues 11130 and 11132, patch by eliel. ........

	* formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c,
	  formats/format_wav_gsm.c, res/res_musiconhold.c,
	  codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c,
	  formats/format_pcm.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_jpeg.c,
	  formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c,
	  codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
	  formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c,
	  res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c,
	  formats/format_ogg_vorbis.c, res/res_ael_share.c,
	  formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c,
	  res/res_indications.c, codecs/codec_ilbc.c,
	  formats/format_g726.c, formats/format_g729.c: More changes to
	  change return values from load_module functions. (issue #11096)
	  Patches: codec_adpcm.c.patch uploaded by moy (license 222)
	  codec_alaw.c.patch uploaded by moy (license 222)
	  codec_a_mu.c.patch uploaded by moy (license 222)
	  codec_g722.c.patch uploaded by moy (license 222)
	  codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch
	  uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy
	  (license 222) codec_lpc10.c.patch uploaded by moy (license 222)
	  codec_speex.c.patch uploaded by moy (license 222)
	  codec_ulaw.c.patch uploaded by moy (license 222)
	  codec_zap.c.patch uploaded by moy (license 222)
	  format_g723.c.patch uploaded by moy (license 222)
	  format_g726.c.patch uploaded by moy (license 222)
	  format_g729.c.patch uploaded by moy (license 222)
	  format_gsm.c.patch uploaded by moy (license 222)
	  format_h263.c.patch uploaded by moy (license 222)
	  format_h264.c.patch uploaded by moy (license 222)
	  format_ilbc.c.patch uploaded by moy (license 222)
	  format_jpeg.c.patch uploaded by moy (license 222)
	  format_ogg_vorbis.c.patch uploaded by moy (license 222)
	  format_pcm.c.patch uploaded by moy (license 222)
	  format_sln.c.patch uploaded by moy (license 222)
	  format_vox.c.patch uploaded by moy (license 222)
	  format_wav.c.patch uploaded by moy (license 222)
	  format_wav_gsm.c.patch uploaded by moy (license 222)
	  res_adsi.c.patch uploaded by eliel (license 64)
	  res_ael_share.c.patch uploaded by eliel (license 64)
	  res_clioriginate.c.patch uploaded by eliel (license 64)
	  res_convert.c.patch uploaded by eliel (license 64)
	  res_indications.c.patch uploaded by eliel (license 64)
	  res_musiconhold.c.patch uploaded by eliel (license 64)
	  res_smdi.c.patch uploaded by eliel (license 64)
	  res_speech.c.patch uploaded by eliel (license 64)

2007-10-31 18:53 +0000 [r87888]  Steve Murphy <murf@digium.com>

	* /: Merged revisions 87849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1
	  line closes issue #11108 -- where the 'dialplan save' cli command
	  saves a file where the semicolon is not escaped. Fixed this; User
	  also wanted comments to be preserved across dialplan save, but
	  this is impossible at this point in time, because comments are
	  not stored in the dialplan. They are 'compiled' out of
	  extensions.conf. The only way to preserve those comments is to
	  use the config file reader/writer that the GUI uses to allow
	  online user edits. extensions.conf is first and foremost, a
	  config file, and is read in by the normal config-file reading
	  routines. Then, it is processed into a dialplan (context/exten
	  structs). (in the case of trunk, tho, no mods needed to be made
	  -- works OK there -- just make sure you use ',' to sep app args!)
	  ........

2007-10-31 18:09 +0000 [r87854]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, /: Merged revisions 87852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007)
	  | 2 lines Create samples for ALL of the available options in
	  asterisk.conf ........

2007-10-31 18:03 +0000 [r87833-87851]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Add volume adjustment in.

	* apps/app_mixmonitor.c: Restore operation of the option that only
	  writes when the channel is bridged.

	* apps/app_chanspy.c: Add volume adjustment to spy audiohook in
	  app_chanspy.

2007-10-31 16:13 +0000 [r87817]  Tilghman Lesher <tlesher@digium.com>

	* CREDITS: Formatting cleanups, remove obsolete contributions
	  (modules no longer in Asterisk), and obfuscate email addresses
	  enough to stop most spam harvesters.

2007-10-31 16:07 +0000 [r87815]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/channel.h: Remove old whisper remnants from
	  channel.h

2007-10-31 15:46 +0000 [r87811]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Optimize pbx_substitute_variables

2007-10-31 04:20 +0000 [r87776]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /: Merged revisions 87775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1
	  line Included some verbage in the check_includes func, to inform
	  the user that included contexts that have no match in the AEL,
	  might be OK, as AEL cannot check in the extensions.conf or the
	  in-memory contexts, as they may not be there at the time of the
	  check. ........

2007-10-30 23:08 +0000 [r87724-87740]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30
	  Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD
	  Reported by: ys Fixed by: ys Closes issue #11116 ........

	* apps/app_exec.c: If no '?' is found in the arguments, don't
	  attempt to continue. Reported by: blitzrage Fixed by: tilghman
	  Closes issue #11111

2007-10-30 21:22 +0000 [r87687]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) |
	  11 lines Merge the changes from team/russell/iax2_poke_fix and
	  iax2-poke-fix-trunk There was a race condition related to the
	  handling of POKEing peers. Essentially, a reference to a peer is
	  held by the scheduler when there are pending callbacks, but the
	  reference count didn't reflect it. So, it was possible for a peer
	  to hit a reference count of zero and have its destructor begin to
	  be called at the same time that the scheduler thread ran a POKE
	  related callback. If that happened, a crash would likely occur.
	  (closes issue #11082, closes issue #11094) ........

2007-10-30 20:30 +0000 [r87626-87651]  Jason Parker <jparker@digium.com>

	* /, channels/Makefile: Merged revisions 87650 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1
	  line Only try to clean out h323/ if the h323/Makefile exists.
	  ........

	* main/pbx.c: Update documentation to give an example of how to use
	  the return status of RaiseException Closes issue #11117, patch by
	  blitzrage (yay blitzrage)

2007-10-30 17:07 +0000 [r87573-87608]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: The priority gets incremented after raising an
	  exception, so the priority should be set to 0

	* main/pbx.c: Jumped the gun a bit in the RaiseException app. It
	  would always return -1 since it checked for the existence of
	  something that will never exist.

2007-10-30 16:15 +0000 [r87572]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 87571 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4
	  lines Add two more checks before printing out a warning message
	  about bridging. If either channel has hungup of course the bridge
	  will have failed. (closes issue #10009) Reported by: dimas
	  ........

2007-10-30 15:47 +0000 [r87568]  Jason Parker <jparker@digium.com>

	* /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500
	  (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris.
	  Issue 11113, patch by snuffy. ........

2007-10-29 22:44 +0000 [r87462-87498]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to
	  fix #10995, I found all kinds of cruft in this Makefile. It
	  should all be gone now, and as a side effect hashtest2 now builds
	  with --enable-dev-mode enabled without a host of errors

	* agi/Makefile, utils/Makefile, codecs/g722/Makefile,
	  main/editline/Makefile.in, Makefile.moddir_rules,
	  codecs/ilbc/Makefile, codecs/lpc10/Makefile,
	  main/db1-ast/Makefile: clean up assembler and preprocessor files
	  if they are here too

	* utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs,
	  codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main,
	  codecs/gsm, main/minimime, pbx, res, channels: ignore
	  preprocessor and assembler files if they are present

	* Makefile, /: Merged revisions 87460 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007)
	  | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is
	  already there (used when debugging preprocessor issues) because
	  the compiler will whine about each compile command ........

2007-10-29 21:34 +0000 [r87397-87428]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: If a caller is listen-only, then don't bother
	  with doing talker detection. (closes issue #10911, reported by
	  junky, patched by me)

	* /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) |
	  5 lines Add some more details to the output of "core show locks".
	  When a thread is waiting for a lock, this will now show the
	  details about who currently has it locked. (inspired by issue
	  #11100) ........

2007-10-29 20:13 +0000 [r87395]  Mark Michelson <mmichelson@digium.com>

	* UPGRADE.txt, apps/app_queue.c: Adding the more flexible
	  QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function.
	  A deprecation notice will be issued the first time
	  QUEUE_MEMBER_COUNT is used.

2007-10-29 20:02 +0000 [r87394]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Drop the RTCP Read too short message to debug. There
	  are some phones out there that send a sort of keep alive packet
	  in the RTCP that trigger this every 5 seconds.

2007-10-29 19:56 +0000 [r87393]  Jason Parker <jparker@digium.com>

	* apps/app_record.c: Make sure we set flags to a 0 value before
	  trying to use it. Pointed out by seanbright while I was debugging
	  issue 11109.

2007-10-29 19:47 +0000 [r87392]  Russell Bryant <russell@digium.com>

	* /, main/astmm.c: Merged revisions 87373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) |
	  5 lines Remove a lock that doesn't make any sense. The regions
	  lock needs to be held when traversing the list of allocated
	  chunks so that they can be printed out to the CLI. (Thanks to
	  eliel on #asterisk-dev for pointing this out!) ........

2007-10-29 17:22 +0000 [r87343]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6
	  lines Fix issue where if both sides of the dialog cancelled the
	  dialog at the same time chan_sip could kepe retransmitting a
	  response for no reason. (closes issue #9566) Reported by:
	  atca_pres Patches: bug9566.patch uploaded by oej ........

2007-10-29 16:38 +0000 [r87295-87327]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes
	  issue #11105) Reported by: eliel Patches: app_voicemail.c.patch
	  uploaded by eliel (license 64)

	* channels/chan_misdn.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Add autoconf
	  checks for extra suppserv definitions that are not present in
	  releases yet. chan_misdn should now build against the latest
	  release. (closes issue #11103) Reported by: IgorG

	* /, main/utils.c: Merged revisions 87294 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6
	  lines Fix issue with ast_unescape_semicolon going into an endless
	  loop. (closes issue #10550) Reported by: ramonpeek Patches:
	  unescape-85177-1.patch uploaded by IgorG (license 20) ........

2007-10-28 14:16 +0000 [r87263-87264]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_dialgroup.c (added): Add a simple dialgroup function.
	  By taking one of the simpler uses of Queue away from Queue, we
	  simplify the lives of people who do not need all the bells and
	  whistles. Also, this is part of the functions that people need to
	  reimplement Queue in the dialplan, as a set of logic, rather than
	  as a single app with hundreds of options.

	* /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c,
	  funcs/func_realtime.c: Merged revisions 87262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007)
	  | 7 lines Add autoservice to several more functions which might
	  delay in their responses. Also, make sure that func_odbc
	  functions have a channel on which to set variables. Reported by
	  russell Fixed by tilghman Closes issue #11099 ........

2007-10-27 15:41 +0000 [r87233-87247]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Update the configure script for the last
	  libss7 API change

	* funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists
	  before attempting to start or stop channel autoservice in
	  func_lock and func_shell.

2007-10-27 00:48 +0000 [r87231-87232]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add Circuit Group Queury message code

	* channels/chan_zap.c: Make sure we turn on the DSP when we answer
	  the call

2007-10-26 22:21 +0000 [r87217]  Mark Michelson <mmichelson@digium.com>

	* CHANGES: Forgot to update CHANGES when I committed the linear
	  queue strategy. Thank you Russell, for pointing this out!

2007-10-26 21:37 +0000 [r87202]  Jason Parker <jparker@digium.com>

	* channels/chan_local.c, channels/chan_zap.c,
	  channels/chan_agent.c, channels/chan_features.c,
	  res/res_crypto.c, res/res_realtime.c, res/res_monitor.c:
	  Correctly use defined return values in (some) load_module
	  functions. (issue #11096) Patches: chan_agent.c.patch uploaded by
	  eliel (license 64) chan_local.c.patch uploaded by eliel (license
	  64) chan_features.c.patch uploaded by eliel (license 64)
	  chan_zap.c.patch uploaded by eliel (license 64)
	  res_monitor.c.patch uploaded by eliel (license 64)
	  res_realtime.c.patch uploaded by eliel (license 64)
	  res_crypto.c.patch uploaded by eliel (license 64)

2007-10-26 17:39 +0000 [r87187]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c,
	  res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c,
	  res/ael/ael.tab.h, utils/ael_main.c,
	  pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex,
	  utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged
	  revisions 87168 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1
	  line closes issue #11086 where a user complains that references
	  to following contexts report a problem; The problem was REALLy
	  that he was referring to empty contexts, which were being
	  ignored. Reporter stated that empty contexts should be OK. I
	  checked it out against extensions.conf, and sure enough, empty
	  contexts ARE ok. So, I removed the restriction from AEL. This,
	  though, highlighted a problem with multiple contexts of the same
	  name. This should be OK, also. So, I added the extend keyword to
	  AEL, and it can preceed the 'context' keyword (mixed with
	  'abstract', if nec.). This will turn off the warnings in AEL if
	  the same context name is used 2 or more times. Also, I now call
	  ast_context_find_or_create for contexts now, instead of just
	  ast_context_create; I did this because pbx_config does this. The
	  'extend' keyword thus becomes a statement of intent. AEL can now
	  duplicate the behavior of pbx_config, ........

2007-10-26 15:19 +0000 [r87153-87154]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Added queue
	  strategy "linear". This strategy is useful for those who always
	  wish for their phones to be rung in a specific order. (closes
	  issue #7279, reported and initially patched by diLLec, patch
	  reworked by me)

	* configs/queues.conf.sample: Remove information about the
	  roundrobin strategy from trunk's queues.conf.sample since it no
	  longer exists

2007-10-26 14:00 +0000 [r87103-87121]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007)
	  | 7 lines The addition of autoservice to func_curl additionally
	  made func_curl dependent on the existence of a channel, with no
	  real reason. This should make func_curl once again work without a
	  channel. Reported by jmls. Fixed by tilghman. Closes issue #11090
	  ........

	* include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c,
	  main/app.c: Use the same delimited character as the FILTER
	  function in FIELDQTY and CUT.

2007-10-25 23:11 +0000 [r87070]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, /, include/asterisk/linkedlists.h: Merged
	  revisions 87069 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007)
	  | 2 lines appending one list to another should leave the first
	  list empty, and not require the user to do that ........

2007-10-25 18:59 +0000 [r87040]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Add support for a muted user to request to
	  talk. The '2' option in the user menu will adjust this status if
	  a user is muted. The talk request status will be reflected in the
	  CLI commands as well as the manager interface. (closes issue
	  #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded
	  by imesper (license 275)

2007-10-25 16:21 +0000 [r87024]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c:
	  closes issue #11045 - each file needs to define
	  ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the
	  compiler flags; the problem was that the fixes were getting made
	  to the generated .c file, and erased the next time someone
	  regenerated that file from the corresponding .y or .flex file.
	  Moral of story: keep your eyes open and make mods to the .y (or
	  flex input file) and re-run bison (or flex) as the Makefile
	  directs for that file, and then check in both. Also,
	  res_config_sqlite was kinda missed, and has the same issue.

2007-10-24 21:26 +0000 [r86985]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Adding the general
	  option "shared_lastcall" to queues so that a member's wrapuptime
	  may be used across multiple queues. (closes issue #9777, reported
	  and patched by eliel)

2007-10-24 20:59 +0000 [r86983]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500
	  (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname
	  configuration option. Simplify code slightly in the process.
	  Issue 11079, reported by ddv2005 ........

2007-10-24 13:21 +0000 [r86900-86967]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2,
	  pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c,
	  pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue
	  #11005, where #include uses the current dir instead of the config
	  dir (/etc/asterisk) for relative path includes for AEL

	* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1
	  line closes issue #11037 -- unable to specify app:spec in hint
	  arguments ........

	* /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1
	  line closes issue #11052 -- where nothing after the ? will allow
	  un-initialized variable values to corrupt and crash asterisk on
	  64-bit platforms ........

	* /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1
	  line This should get rid of a really, really irritating warning
	  generated by some 64-bit platforms from libc, where free(0) is
	  frowned upon ........

	* /, main/Makefile: Merged revisions 86881 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1
	  line this update to Makefile corrects how ast_expr2f.c should be
	  generated ........

2007-10-22 21:37 +0000 [r86835-86839]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22
	  Oct 2007) | 9 lines If lock tracking is not enabled, then we can
	  not attempt to log any mutex failures. If so, we could end up in
	  infinite recursion. The only lock that is affected by this is a
	  mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue
	  #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
	  (license 281) ........

	* apps/app_playback.c: Convert some spaces to tabs and make it so
	  the CLI command is only registered once instead of 3 times.
	  (closes issue #11053) Reported by: seanbright Patches:
	  app_playback.patch uploaded by seanbright (license 71)

2007-10-22 20:05 +0000 [r86820]  Jason Parker <jparker@digium.com>

	* main/udptl.c, channels/chan_local.c, main/frame.c,
	  res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
	  main/astobj2.c, main/config.c, main/cli.c,
	  channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
	  channels/chan_alsa.c, main/db.c, main/pbx.c,
	  channels/chan_agent.c, channels/iax2-provision.c,
	  apps/app_playback.c, channels/chan_misdn.c,
	  channels/chan_features.c, res/res_indications.c,
	  pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
	  res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
	  main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
	  res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
	  main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
	  res/res_agi.c, apps/app_minivm.c, main/logger.c,
	  res/res_realtime.c, main/image.c, apps/app_rpt.c,
	  channels/chan_mgcp.c, res/res_clioriginate.c,
	  res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
	  channels/chan_sip.c, res/res_limit.c, main/translate.c,
	  res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
	  apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
	  channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
	  channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
	  funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to
	  AST_CLI_DEFINE, since the former didn't make much sense

2007-10-22 17:40 +0000 [r86790]  Tilghman Lesher <tlesher@digium.com>

	* /, main/astmm.c: Merged revisions 86787 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007)
	  | 2 lines Minor FreeBSD build fix ........

2007-10-22 16:36 +0000 [r86755-86757]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4
	  lines After reading online I have confirmed that Record-Route
	  headers should be copied to 1xx responses as well. (closes issue
	  #10113) Reported by: makoto ........

	* /, apps/app_controlplayback.c: Merged revisions 86754 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4
	  lines Make sure res is a positive value before performing the
	  check to determine whether the user stopped it or not. (closes
	  issue #11023) Reported by: cfc ........

2007-10-22 15:57 +0000 [r86734-86751]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 86750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) |
	  8 lines Don't leak a frame in the case that an END frame is
	  received and the time since the BEGIN is less than that of the
	  defined minimum DTMF duration. (closes issue #11051) Reported by:
	  casper Patches: channel.c.86664.diff uploaded by casper (license
	  55) ........

	* channels/chan_zap.c: There is a really fun game that you can play
	  before committing code, and it's called "make". :)

	* /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22
	  Oct 2007) | 4 lines Update the static mutex initializer to
	  include the initialization of the internal mutex used to protect
	  the lock debugging data. (closes issue #11044, patch suggested by
	  Ivan) ........

2007-10-22 14:59 +0000 [r86697]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: resetinterval
	  defaulting to something other than 'never' doesn't seem to
	  accomplish any good and causes problems for plenty of people...

2007-10-22 14:58 +0000 [r86696]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct
	  2007) | 5 lines Account for the fact that sometimes headers may
	  be terminated with \r\n instead of just \n (closes issue #11043,
	  reported by yehavi) ........

2007-10-22 14:56 +0000 [r86695]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c: merging patches that don't compile is bad...
	  mmkay?

2007-10-22 14:28 +0000 [r86631-86664]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 86663 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6
	  lines Move log message to before the frame it references is
	  freed. (closes issue #11050) Reported by: slavon Patches:
	  channel.c.86662.diff uploaded by casper (license 55) ........

	* /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6
	  lines Fix tab completion for dundi show peer. (closes issue
	  #11041) Reported by: jsmith Patches:
	  asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
	  (license 176) ........

	* /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct
	  2007) | 6 lines Fixes for building under OpenSolaris. (closes
	  issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
	  uploaded by snuffy (license 35) ........

2007-10-22 10:18 +0000 [r86616-86617]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 86598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) |
	  1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if
	  the dialplan does not match after an overlap call. Also added
	  out_cause=1 ........

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  started to add some basic support for supplementary services like
	  CallForwarding and so forth

2007-10-21 22:52 +0000 [r86585]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged
	  revisions 85532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) |
	  8 lines Properly handle the case where read() may return the text
	  for more than one CLI command at once for a remote console.
	  (closes issue #10888) Reported by: jamesgolovich Patches:
	  asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
	  176) ........

2007-10-20 19:56 +0000 [r86572]  Matthew Fredrickson <creslin@digium.com>

	* configs/zapata.conf.sample: Improved comments and organization
	  for zapata.conf (#10904)

2007-10-19 18:46 +0000 [r86549]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add better support for blocking and
	  unblocking of CICs (#10965)

2007-10-19 18:29 +0000 [r86534-86536]  Jason Parker <jparker@digium.com>

	* main/udptl.c, channels/chan_local.c, main/frame.c,
	  res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
	  main/astobj2.c, main/config.c, main/cli.c,
	  channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
	  channels/chan_alsa.c, main/db.c, main/pbx.c,
	  channels/chan_agent.c, channels/iax2-provision.c,
	  apps/app_playback.c, channels/chan_misdn.c,
	  channels/chan_features.c, res/res_indications.c,
	  pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
	  res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
	  main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
	  res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
	  main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
	  res/res_agi.c, apps/app_minivm.c, main/logger.c,
	  res/res_realtime.c, main/image.c, apps/app_rpt.c,
	  channels/chan_mgcp.c, res/res_clioriginate.c,
	  res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
	  channels/chan_sip.c, res/res_limit.c, main/translate.c,
	  res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
	  apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
	  channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
	  channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
	  funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue
	  #11039, as suggested by seanbright.

	* channels/chan_usbradio.c, res/res_config_pgsql.c,
	  channels/chan_misdn.c, channels/chan_h323.c,
	  res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c,
	  res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes
	  to NEW_CLI. Also fixes a few cli messages and some minor
	  formatting. (closes issue #11001) Reported by: seanbright
	  Patches: newcli.1.patch uploaded by seanbright (license 71)
	  newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch
	  uploaded by seanbright (license 71) newcli.5.patch uploaded by
	  seanbright (license 71) newcli.6.patch uploaded by seanbright
	  (license 71) newcli.7.patch uploaded by seanbright (license 71)

2007-10-19 16:40 +0000 [r86470-86503]  Joshua Colp <jcolp@digium.com>

	* /, main/app.c: Merged revisions 86502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4
	  lines When returning a DTMF digit from ast_control_streamfile
	  cast it as a char so that 0 does not overlap with the success
	  return code. (closes issue #11023) Reported by: cfc ........

	* /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6
	  lines Fix two issues with domains and transfers. If a port was
	  given in the hostname it was treated as part of the hostname. If
	  domains were configured but external domains were not enabled all
	  transfers would be considered remote. (closes issue #11027)
	  Reported by: ramonpeek Patches: 11027-1.diff uploaded by
	  ramonpeek (license 266) ........

	* /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4
	  lines Set port number in received as information for
	  registrations as well. (closes issue #11028) Reported by: brad-x
	  ........

2007-10-19 01:56 +0000 [r86439]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Fixed a buffer size issue.

2007-10-18 22:03 +0000 [r86407-86408]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 86405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500
	  (Thu, 18 Oct 2007) | 4 lines Add documentation for options in
	  asterisk.conf Issue 11029, patch by eserra ........

2007-10-18 18:40 +0000 [r86350]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: Fixing a segfault from tab-completing a "zap
	  restart" CLI command. (patch made by seanbright, pointed out in
	  #asterisk-dev on IRC)

2007-10-18 18:06 +0000 [r86331]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/channel.h: Merged revisions
	  86330 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) |
	  10 lines The channel needs to stay locked while running timer
	  callbacks, as they access and modify channel data that may change
	  elsewhere. I went through every timer callback in the source tree
	  to make sure that none of them did any additional locking that
	  could introduce deadlocks, and all is well. (closes issue #10765)
	  Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff
	  uploaded by Ivan (license 229) ........

2007-10-18 17:40 +0000 [r86298-86329]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 86328 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct
	  2007) | 5 lines If a non-existent file is specified to be played
	  either as a periodic announcement or as a hold/position
	  announcement, the caller would be kicked out of the queue. No
	  longer does this happen. ........

	* apps/app_queue.c: Changed some spaces to tabs

2007-10-18 15:57 +0000 [r86297]  Russell Bryant <russell@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) |
	  3 lines Execute the RELEASE operation on transcoder channels in
	  the destroy callback. (patch from jsloan) ........

2007-10-18 07:23 +0000 [r86277-86278]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue
	  #10784

	* res/res_musiconhold.c: On reload, re-read the files in the
	  specified moh directory (closes issue #10536)

2007-10-18 04:41 +0000 [r86238]  Russell Bryant <russell@digium.com>

	* /, main/utils.c: Merged revisions 86237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) |
	  9 lines Revert a change that I made for issue #10979 which, as
	  has been pointed out to me in issue #11018, doesn't really make
	  sense. There is no reason to have the base64 decode function
	  force a '\0' terminated buffer, when the result is almost always
	  binary, anyway. In fact, this caused some breakage, as some code
	  in res_crypto passed in a buffer exactly the right size to get
	  its binary result, which got stomped on by this patch. (closes
	  issue #11018, reported by dimas) ........

2007-10-17 21:41 +0000 [r86208]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 86202 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct
	  2007) | 6 lines Changing the strategy field of the call_queue
	  struct to be signed instead of unsigned, since the code attempts
	  to set the strategy to -1 if you specify a bogus strategy. While
	  this isn't a huge issue in 1.4, it could be a problem for someone
	  who, say, tries to use the roundrobin strategy in trunk (despite
	  all the deprecation warnings in 1.4). ........

2007-10-17 21:16 +0000 [r86195-86197]  Tilghman Lesher <tlesher@digium.com>

	* main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify
	  some preprocessor logic by using #elif

	* CHANGES, configs/meetme.conf.sample: Document the changes made
	  earlier today to meetme

2007-10-17 20:06 +0000 [r86180-86182]  Steve Murphy <murf@digium.com>

	* utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and
	  then, I noticed the clicompat stuff.

	* utils/check_expr.c: more stub routines to allow linkage in
	  stand-alone environment, with thread debugs turned on

	* utils/hashtest2.c: more stub routines to allow linkage in
	  stand-alone environment, with thread debugs turned on

2007-10-17 18:01 +0000 [r86150]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) |
	  8 lines If Asterisk is in the middle of shutting down, respond to
	  OPTIONS with 503 Unavailable. (closes issue #10994) Reported by:
	  eserra Patches: sip-options-503.patch uploaded by eserra (license
	  45) ........

2007-10-17 17:06 +0000 [r86119]  Tilghman Lesher <tlesher@digium.com>

	* main/term.c: Support color on certain platforms, even when
	  started at boot (before TERM is set) Closes issue #9048

2007-10-17 17:00 +0000 [r86118]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4
	  lines Whoops, forgot to remove the original sip_scheddestroy.
	  (closes issue #11010) Reported by: vadim ........

2007-10-17 16:09 +0000 [r86104]  Jason Parker <jparker@digium.com>

	* channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow
	  chan_usbradio to compile again. Closes issue #11014, patch by
	  seanbright.

2007-10-17 15:39 +0000 [r86079]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 86066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007)
	  | 3 lines When runuser/rungroup is specified, a remote console
	  could only be attained by root (Closes issue #9999) ........

2007-10-17 15:30 +0000 [r86067]  Joshua Colp <jcolp@digium.com>

	* channels/chan_usbradio.c: Change dependency for chan_usbradio to
	  asound. Let's keep everything uniform. (closes issue #11013)
	  Reported by: seanbright

2007-10-17 15:13 +0000 [r86065]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: Enhancements to realtime (closes issue #9609)

2007-10-17 15:09 +0000 [r86064]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4
	  lines Don't schedule dialog destruction if a MESSAGE is received
	  using an existing dialog. (closes issue #11010) Reported by:
	  vadim ........

2007-10-16 23:36 +0000 [r86029-86033]  Mark Michelson <mmichelson@digium.com>

	* /, configs/queues.conf.sample: Merged revisions 86032 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct
	  2007) | 3 lines Since monitor-join is deprecated now, remove the
	  example from the sample queues.conf file ........

	* apps/app_queue.c: Removed the monitor-join option. If one wishes
	  to mix audio, they should instead use monitor-type=mixmonitor.
	  (related to issue #10885)

2007-10-16 22:36 +0000 [r85995-85998]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16
	  Oct 2007) | 1 line really picky formatting tweak ... ........

	* /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16
	  Oct 2007) | 16 lines Some locking errors exposed the fact that
	  the lock debugging code itself was not thread safe. How ironic!
	  Anyway, these changes ensure that the code that is accessing the
	  lock debugging data is thread-safe. Many thanks to Ivan for
	  finding and fixing the core issue here, and also thanks to those
	  that tested the patch and provided test results. (closes issue
	  #10571) (closes issue #10886) (closes issue #10875) (might close
	  some others, as well ...) Patches: (from issue #10571)
	  ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
	  229) - a few small changes by me ........

2007-10-16 21:51 +0000 [r85959-85992]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing the build.

	* apps/app_read.c: Fixing app_read so that if a timeout of less
	  than 1 ms is specified, assume that 1 ms is desired. (closes
	  issue #11000, reported and patched by michael-fig, with a warning
	  line added by me)

	* /, apps/app_queue.c: Merged revisions 85958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct
	  2007) | 5 lines Trying to remove a non-dynamic queue member via
	  dynamic means can lead to some interesting (read nasty)
	  situations. This patch clears up the issue by making only dynamic
	  queue members removable via dynamic methods. ........

2007-10-16 20:55 +0000 [r85957]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Don't hangup the call for SS7 if we get an
	  alarm

2007-10-16 20:32 +0000 [r85944]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't
	  improperly memset() over an ast_str. This was leftover from
	  before it got changed to use ast_str. (closes issue #11003,
	  reported by pj) (closes issue #10770, reported by yehavi)
	  (patched by me)

2007-10-16 19:47 +0000 [r85943]  Tilghman Lesher <tlesher@digium.com>

	* /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16
	  Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z
	  gnu extension to strftime) Reported and fixed by jcmoore Closes
	  issue #11002 ........

2007-10-16 19:12 +0000 [r85897]  Russell Bryant <russell@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) |
	  2 lines Remove a pointless lock. ........

2007-10-16 16:40 +0000 [r85853-85883]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix IMAP compilation error. (closes issue
	  #10986, reported and patched by snuffy)

	* /: Blocking changes from previous commit

2007-10-16 15:15 +0000 [r85819-85851]  Joshua Colp <jcolp@digium.com>

	* /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4
	  lines Check to make sure a value has been given to the VMCOUNT
	  dialplan function. (closes issue #10996) Reported by: marsosa
	  ........

	* main/threadstorage.c: Permit building under DEBUG_THREADLOCALS.
	  Thanks snuff.

	* /, main/threadstorage.c: Merged revisions 85818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6
	  lines Fix memory allocation issue in threadstorage. (closes issue
	  #10995) Reported by: snuffy Patches: new-patch.diff uploaded by
	  snuffy (license 35) ........

2007-10-16 10:38 +0000 [r85777-85787]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help
	  output

	* channels/chan_jingle.c: Added two CLI functions, taken from
	  chan_gtalk : - jingle reload ; - jingle show channels.

	* channels/chan_jingle.c: Make an audio path under the following
	  call configuration : SIP Phone 1 --- [chan_sip]Asterisk
	  1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP
	  Phone 2 Modifications : - set bridge type to partial ; - process
	  media candidates from the remote peer properly. Now we have
	  Jingle audio, at least between two Asterisk Jingle clients.

2007-10-15 23:20 +0000 [r85764]  Jason Parker <jparker@digium.com>

	* configs/dundi.conf.sample, channels/chan_sip.c,
	  channels/chan_h323.c, main/acl.c, UPGRADE.txt,
	  channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c,
	  channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new
	  tos config format. Remove old unused defines for old style.
	  Closes issue 10860, patch by IgorG.

2007-10-15 21:11 +0000 [r85718-85721]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 85720 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) |
	  3 lines Ensure that no pending state changes are leaked when the
	  device state change thread gets stopped on module unload.
	  ........

	* /, main/say.c: Merged revisions 85686 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) |
	  7 lines Add a small fix for the tw version of saying dates.
	  (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
	  uploaded by sharkey (license 172) ........

2007-10-15 20:16 +0000 [r85685]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 85684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500
	  (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config'
	  target. Do not try to run chkconfig or similar if using DESTDIR.
	  Issue 10938, patch by cabal95. ........

2007-10-15 20:09 +0000 [r85648-85683]  Russell Bryant <russell@digium.com>

	* doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the
	  channel var docs

	* res/res_features.c, CHANGES: Added support for reading the
	  TOUCH_MONITOR_PREFIX channel variable. It allows you to configure
	  a prefix for auto-monitor recordings. (closes issue #6353)
	  Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded
	  by ivanfm (original patch) - updated patch:
	  6353-touch_monitor_prefix.diff uploaded by qwell (license 4)

	* /, main/utils.c: Merged revisions 85649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) |
	  2 lines Be pedantic about handling memory allocation failure.
	  ........

	* /, main/utils.c: Merged revisions 85647 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) |
	  5 lines The loop in the handler for the "core show locks" could
	  potentially block for some amount of time. Be a little bit more
	  careful and prepare all of the output in an intermediary buffer
	  while holding a global resource. Then, after releasing it, send
	  the output to ast_cli(). ........

2007-10-15 17:51 +0000 [r85633]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c: Document my changes from Friday

2007-10-15 16:59 +0000 [r85605]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) |
	  6 lines Make the default for the srvlookup option to be yes. It
	  doesn't really make sense for it to default to off. The default
	  configuration file has it on, and proper RFC behavior, as
	  indicated by a comment in the code, is for it to be on. So, let's
	  have it on by default to make lives easier. (closes issue #10954,
	  suggested by jtodd) ........

2007-10-15 16:41 +0000 [r85578]  Joshua Colp <jcolp@digium.com>

	* /, configs/features.conf.sample: Merged revisions 85571 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4
	  lines Document that DTMF based features only work when two
	  channels are bridged together. (closes issue #10773) Reported by:
	  pbayley ........

2007-10-15 16:36 +0000 [r85562]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/strings.h: Merged revisions 85561 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) |
	  4 lines Make a few changes so that characters in the upper half
	  of the ISO-8859-1 character set don't get stripped when reading
	  configuration. (closes issue #10982, dandre) ........

2007-10-15 16:23 +0000 [r85560]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 85559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4
	  lines Bring both DTMF begin and end frames up through to the core
	  for DTMF feature handling. (closes issue #10826) Reported by:
	  dimas ........

2007-10-15 15:55 +0000 [r85557-85558]  Russell Bryant <russell@digium.com>

	* pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c.
	  This also makes the code a bit safer by removing various
	  assumptions about sizes. (No vulnerabilities, though) (closes
	  issue #10977) Reported by: dimas Patches: dundiparser.patch
	  uploaded by dimas (license 88)

	* /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) |
	  9 lines Ensure the buffer passed to ast_canmatch_extension() is
	  properly initialized so that it is null terminated. (issue
	  #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by
	  dimas (license 88) - small mods by me ........

2007-10-15 15:26 +0000 [r85555]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c: Allow RTP structure registration

2007-10-15 15:07 +0000 [r85553-85554]  Joshua Colp <jcolp@digium.com>

	* main/frame.c: Add packetization data for G.722. (closes issue
	  #10900) Reported by: andrew Patches: frame.diff uploaded by
	  andrew (license 240)

	* /, main/rtp.c: Merged revisions 85552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4
	  lines If Monitor or a spy was added to a P2P or native bridged
	  channel bring the channel back to the generic bridging core so
	  the monitor or spy operations work. (closes issue #10943)
	  Reported by: julianjm ........

2007-10-15 13:51 +0000 [r85551]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Allocate more space for the base64 output we
	  need to generate. Closes issue #10913, reported by tootai, who
	  graciously granted us access to his Asterisk server, thanks!
	  Daniel, feel free to reopen the bug in case you can reproduce
	  this on 1.4.

2007-10-15 13:44 +0000 [r85539-85550]  Russell Bryant <russell@digium.com>

	* main/cli.c: Move the CLI commands that were in builtins[] into
	  the cli_cli[] array of CLI commands and remove the cli_iterator
	  struct. This gets tab completion working again. (closes issue
	  #10970) Reported by: jamesgolovich Patches:
	  asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license
	  176)

	* doc/tex/jitterbuffer.tex, doc/tex/extensions.tex,
	  doc/tex/channelvariables.tex, doc/tex/ael.tex,
	  doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex,
	  doc/tex/dundi.tex, doc/tex/security.tex,
	  doc/tex/configuration.tex, doc/tex/ajam.tex,
	  doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex,
	  doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex,
	  doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c,
	  doc/tex/localchannel.tex, doc/tex/cdrdriver.tex,
	  doc/tex/queuelog.tex: Another major doc directory update from
	  IgorG. This patch includes - Many uses of the astlisting
	  environment around verbatim text to ensure that it gets properly
	  formatted and doesn't run off the page. - Update some things that
	  have been deprecated. - Add escaping as needed - and more ...
	  (closes issue #10978) Reported by: IgorG Patches:
	  texdoc-85542-1.patch uploaded by IgorG (license 20)

	* /, main/asterisk.c: Merged revisions 85545 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) |
	  7 lines Make sure remote consoles unmute themselves again after
	  reconnecting. (closes issue #10847) Reported by: atis Patches:
	  console_unmute_on_reconnect.patch uploaded by atis (license 242)
	  ........

	* /, main/utils.c: Merged revisions 85543 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) |
	  8 lines Make sure that the base64 decoder returns a terminated
	  string. (closes issue #10979) Reported by: ys Patches:
	  util.c.diff uploaded by ys (license 281) - small mods by me
	  ........

	* configure, configure.ac: Change the configure script to check for
	  a function that was recently added to libss7.

	* /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) |
	  7 lines Don't create the context for users in users.conf until we
	  know at least one user exists. (closes issue #10971) Reported by:
	  dimas Patches: pbxconfig.patch uploaded by dimas (license 88)
	  ........

	* doc/tex/backtrace.tex (added): When merging the last
	  documentation update, I forgot to "svn add" a file. Here it is.
	  (closes issue #10962)

2007-10-13 08:38 +0000 [r85535]  James Golovich <james@gnuinter.net>

	* main/cli.c: Fix compiling cli.c due to differences with new cli
	  system (closes issue 0010966)

2007-10-13 05:53 +0000 [r85534]  Russell Bryant <russell@digium.com>

	* include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged
	  revisions 85533 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) |
	  12 lines Fix an issue with console verbosity when running
	  asterisk -rx to execute a command and retrieve its output. The
	  issue was that there was no way for the main Asterisk process to
	  know that the remote console was connecting in the -rx mode. The
	  way that James has fixed this is to have all remote consoles
	  muted by default. Then, regular remote consoles automatically
	  execute a CLI command to unmute themselves when they first start
	  up. (closes issue #10847) Reported by: atis Patches:
	  asterisk-consolemute.diff.txt uploaded by jamesgolovich (license
	  176) ........

2007-10-12 20:06 +0000 [r85527]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Allow for the
	  position announcement to be turned off if desired. (closes issue
	  #8515, reported by bruno_rocha, initial patch by bruno_rocha,
	  final patch by qwell)

2007-10-12 19:41 +0000 [r85525-85526]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to
	  finish the last of the charge_number patch up #10916

	* channels/chan_zap.c: Add support for receive charge number in
	  dialplan #10916

2007-10-12 18:37 +0000 [r85522-85524]  Tilghman Lesher <tlesher@digium.com>

	* doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions
	  85523 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007)
	  | 2 lines Change Digium address ........

	* funcs/func_strings.c: Enable ranges, hexadecimal, octal, and
	  special backslashed characters for the FILTER function

2007-10-12 15:50 +0000 [r85516-85519]  Russell Bryant <russell@digium.com>

	* doc/tex/odbcstorage.tex, doc/tex/extensions.tex,
	  doc/tex/channelvariables.tex, doc/tex/ael.tex,
	  doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex,
	  doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex,
	  doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex,
	  doc/tex/localchannel.tex, doc/tex/ices.tex,
	  doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory
	  improvements, including: - Added development section
	  (backtrace.tex) - Correct filesystem path formating - Replace all
	  "|" argument separator to "," - Endless count of spaces at the
	  end of line - Using astlisting to make listings do not take so
	  much place - Take back ASTRISKVERSION on first page - Make
	  localchannel.tex readable by inserting extra end of lines (closes
	  issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch
	  uploaded by IgorG (license 20)

	* res/res_smdi.c, /: Merged revisions 85517 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) |
	  3 lines Fix a spelling error in a log message. SMDI, not SDMI.
	  (closes issue #10959) ........

	* /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) |
	  7 lines Fix the potential use of an uninitialized buffer in a log
	  message. (closes issue #10958) Reported by: dimas Patches:
	  realtime.patch uploaded by dimas (license 88) ........

2007-10-11 22:42 +0000 [r85474-85499]  Matthew Fredrickson <creslin@digium.com>

	* apps/app_dial.c: Make sure we propogate ANI2 to the outbound
	  channel

	* funcs/func_callerid.c: See if I can fix this borked ANI2 code I
	  added

	* channels/chan_zap.c: Make sure we set the ANI2 field for PRI

	* funcs/func_callerid.c: Add ANI2 support to func_callerid

	* channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916

	* channels/chan_zap.c: Add CCR test support #10916

2007-10-11 19:03 +0000 [r85460]  Russell Bryant <russell@digium.com>

	* main/udptl.c, main/threadstorage.c, res/res_limit.c,
	  main/translate.c, res/res_crypto.c, res/res_convert.c,
	  channels/iax2-provision.c, channels/chan_gtalk.c,
	  channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c,
	  main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c,
	  pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c,
	  pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c,
	  main/asterisk.c, main/db.c, channels/chan_mgcp.c,
	  res/res_clioriginate.c: Merge a ton of NEW_CLI conversions.
	  Thanks to everyone that helped out! :) (closes issue #10724)
	  Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel
	  (license 64) chan_oss.c.patch uploaded by eliel (license 64)
	  chan_mgcp.c.patch2 uploaded by eliel (license 64)
	  pbx_config.c.patch uploaded by seanbright (license 71)
	  iax2-provision.c.patch uploaded by eliel (license 64)
	  chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch
	  uploaded by seanbright (license 71) file.c.patch uploaded by
	  seanbright (license 71) image.c.patch uploaded by seanbright
	  (license 71) cli.c.patch uploaded by moy (license 222)
	  astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch
	  uploaded by moy (license 222) res_limit.c.patch uploaded by
	  seanbright (license 71) res_convert.c.patch uploaded by
	  seanbright (license 71) res_crypto.c.patch uploaded by seanbright
	  (license 71) app_osplookup.c.patch uploaded by seanbright
	  (license 71) app_rpt.c.patch uploaded by seanbright (license 71)
	  app_mixmonitor.c.patch uploaded by seanbright (license 71)
	  channel.c.patch uploaded by seanbright (license 71)
	  translate.c.patch uploaded by seanbright (license 71)
	  udptl.c.patch uploaded by seanbright (license 71)
	  threadstorage.c.patch uploaded by seanbright (license 71)
	  db.c.patch uploaded by seanbright (license 71) cdr.c.patch
	  uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy
	  (license 222) app_osplookup-rev83558.patch uploaded by moy
	  (license 222) res_clioriginate.c.patch uploaded by moy (license
	  222)

2007-10-11 17:17 +0000 [r85431-85444]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Let's hard code this until I fix it

	* channels/chan_zap.c: Make sure we are clean to build without
	  libpri

2007-10-11 04:40 +0000 [r85357]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 85356 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007)
	  | 2 lines A dollar sign by itself, not indicating a start of a
	  variable or expression prematurely ends substitution (closes
	  issue #10939) ........

2007-10-10 16:01 +0000 [r85317]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, /: Merged revisions 85316 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10
	  Oct 2007) | 6 lines I introduced a new member to the
	  ast_filestream struct in 1.4.12, but put it in the middle of the
	  struct, instead of at the end. One of the Debian folks, paravoid,
	  pointed out that this breaks binary compatability with modules
	  compiled against older headers. So, I'm moving the new member to
	  the end of the struct to resolve the situation. ........

2007-10-10 14:43 +0000 [r85281]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4
	  lines If devicestate is passed a port number strip it out.
	  (closes issue #10930) Reported by: ibc ........

2007-10-10 14:38 +0000 [r85279]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct
	  2007) | 5 lines A bunch of changes from sprintf to snprintf. See
	  security advisory AST-2002-022 ........

2007-10-10 14:30 +0000 [r85234-85278]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6
	  lines Add support for handling a 182 Queued response. (closes
	  issue #10924) Reported by: ramonpeek Patches: queued-182.diff
	  uploaded by ramonpeek (license 266) ........

	* /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6
	  lines Close voicemail message description file if duration did
	  not meet the minimum, or else we will eventually run out of file
	  descriptors. (closes issue #10918) Reported by: brak2718 Patches:
	  vm1.4.12.1.patch uploaded by brak2718 (license 279) ........

	* main/logger.c: Process outstanding log messages before shutting
	  down the logger thread. (closes issue #10933) Reported by:
	  sperreault

2007-10-10 06:48 +0000 [r85197]  Luigi Rizzo <rizzo@icir.org>

	* bootstrap.sh: Adapt the autotools names to different versions of
	  FreeBSD (and open the way to better adaptation for other
	  platforms as well).

2007-10-10 06:41 +0000 [r85196]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10
	  Oct 2007) | 2 lines use a macro instead of an inline function, so
	  that backtraces will report the caller of ast_frame_free()
	  properly ........

2007-10-09 22:35 +0000 [r85177]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Patch to add one-touch parking for queues.
	  (closes issue #10869, reported and patched by bluecrow76)

2007-10-09 22:21 +0000 [r85140-85176]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged
	  revisions 85158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007)
	  | 5 lines This commit fixes the following issues: - Deadlock in
	  ast_write (issue #10406) - Deadlock in ast_read (issue #10406) -
	  Possible mutex initialization error in lock.h (issue #10571)
	  ........

	* apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c,
	  apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c,
	  utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove
	  redundant includes (patch by snuffy) (Closes issue #10922)

2007-10-09 15:12 +0000 [r85097-85098]  Russell Bryant <russell@digium.com>

	* CHANGES: Note jitterbuffer support for chan_local in CHANGES

	* channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer
	  support for chan_local. To enable it, you use the 'j' option in
	  the Dial command. The 'j' option _must_ be used in conjunction
	  with the 'n' option. This feature will allow you to use the
	  existing jitterbuffer implementation to put a jitterbuffer on
	  incoming SIP calls connecting to Asterisk applications by putting
	  a local channel in the middle.

2007-10-09 14:31 +0000 [r84991-85094]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4
	  lines Don't perform a reinvite if a transfer is in progress.
	  (issue #10915) Reported by: ramonpeek ........

	* /, main/rtp.c: Merged revisions 85057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4
	  lines Only update codec information if the channel has a
	  technology private structure. (issue #10915) Reported by:
	  ramonpeek ........

	* res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c,
	  main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it
	  compiles when MTX Profiling is enabled. (closes issue #10898)
	  Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by
	  qwell (license 4)

	* /, main/rtp.c: Merged revisions 85023 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4
	  lines Update codec information as well as address when doing hold
	  reinvites. (issue #10868) Reported by: mavince ........

	* main/channel.c, /: Merged revisions 84990 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4
	  lines Don't keep trying to native bridge if either of the
	  channels are involved in a masquerade operation to be done.
	  (closes issue #10696) Reported by: tbelder ........

2007-10-08 03:29 +0000 [r84958]  Russell Bryant <russell@digium.com>

	* /, Makefile.rules: Merged revisions 84957 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) |
	  6 lines Enable file dependency tracking for _all_ builds, and not
	  just for builds with dev-mode enabled. I have seen enough
	  problems caused by this that I don't think it's worth keeping. I
	  want to continue to encourage anybody that is interested to
	  continue to run Asterisk from svn. Furthermore, I do not want
	  their systems to break when we change a structure definition in a
	  header file. :) ........

2007-10-07 16:28 +0000 [r84891-84939]  Philippe Sultan <philippe.sultan@gmail.com>

	* configs/jabber.conf.sample, include/asterisk/jabber.h,
	  res/res_jabber.c: Make the status and priority configurable.
	  Closes issue #10785, patch by Luke-Jr, thanks!

	* /, res/res_jabber.c: Merged revisions 84902 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007)
	  | 5 lines Presence packets from a client who's connected with our
	  Jabber ID are valid, therefore, those clients must be considered
	  as buddies. The resource string helps us make the distinction
	  between clients. Closes issue #10707, reported by yusufmotiwala.
	  ........

	* res/res_jabber.c: Fix indentation

	* /, res/res_jabber.c: Merged revisions 84890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007)
	  | 5 lines Prevent Asterisk from crashing when receiving a
	  presence packet without resource from a buddy that is known to
	  have a resource list. Revert a change I previously made, where
	  Asterisk could point to a freed memory location. ........

2007-10-05 19:48 +0000 [r84852]  Tilghman Lesher <tlesher@digium.com>

	* /, main/db.c: Merged revisions 84851 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007)
	  | 2 lines Log exactly why we can't open the database, if we fail
	  (closes issue #10887) ........

2007-10-05 18:57 +0000 [r84819]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 84818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4
	  lines Update the remembered RTP peer information when putting an
	  endpoint on hold or taking it off hold so that the RTP stack does
	  not initiate a needless reinvite. (closes issue #10868) Reported
	  by: mavince ........

2007-10-05 16:49 +0000 [r84743-84784]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) |
	  4 lines Do deadlock avoidance in a couple more places. You can't
	  lock two channels at the same time without doing extra work to
	  make sure it succeeds. (closes issue #10895, patch by me)
	  ........

	* main/manager.c, /: Merged revisions 84742 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) |
	  3 lines Fix a copy/paste error in the description of UpdateConfig
	  that was pointed out by JerJer on #asterisk-dev ........

2007-10-04 22:58 +0000 [r84693-84726]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix
	  some bad logic in the counting of statistics for QueueSummary
	  manager event. Variables were not being reset for each additional
	  queue, so cumulative totals were reported on each successive
	  queue. 2) Add a longest hold time stat to QueueSummary manager
	  event.

	* /, apps/app_queue.c: Merged revisions 84692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct
	  2007) | 5 lines Don't allocate space for queue members unless
	  it's needed. You end up deleting dynamic members on a reload. Not
	  good. closes issue (#10879, reported by dazza76, patched by me)
	  ........

2007-10-04 21:38 +0000 [r84691]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007)
	  | 2 lines callers of sig2str already add the word 'signalling' in
	  the appropriate place, so don't duplicate it ........

2007-10-04 16:56 +0000 [r84671]  Tilghman Lesher <tlesher@digium.com>

	* res/res_jabber.c: Update to current coding standards, also
	  changing the argument delimiter to ',' (Closes issue #10876)

2007-10-04 14:54 +0000 [r84613-84638]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 84637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4
	  lines Create a duplicate of the channel's member name as the tab
	  completion stuff will free it. (closes issue #10884) Reported by:
	  adamg ........

	* main/pbx.c: Don't register the exception function with module
	  information. Since it is in the core there is none and it will
	  explode.

2007-10-03 23:05 +0000 [r84580-84582]  Tilghman Lesher <tlesher@digium.com>

	* /, main/rtp.c: Merged revisions 84581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007)
	  | 2 lines When an RFC 2833 event is sent that we don't recognize,
	  ignore it, don't queue a NULL digit (closes issue #10877)
	  ........

	* main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h:
	  Create a universal exception handling extension, "e" (closes
	  issue #9785)

2007-10-03 18:23 +0000 [r84512-84545]  Steve Murphy <murf@digium.com>

	* /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the
	  CUT in AEL

	* res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1
	  line closes issue #10834 ; where a null input to a switch
	  statement results in a hangup; since switch is implemented with
	  extensions, and the default case is implemented with a '.', and
	  the '.' matches 1 or more remaining characters, the case where 0
	  characters exist isn't matched, and the extension isn't matched,
	  and the goto fails, and a hangup occurs. Now, when a default case
	  is generated, it also generates a single fixed extension that
	  will match a null input. That extension just does a goto to the
	  default extension for that switch. I played with an alternate
	  solution, where I just tack an extra char onto all the patterns
	  and the goto, but not the default case's pattern. Then even a
	  null input will still have at least one char in it. But it made
	  me nervous, having that extra char in , even if that's a pretty
	  secret and low-level issue. ........

2007-10-02 20:07 +0000 [r84475]  Russell Bryant <russell@digium.com>

	* Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) |
	  5 lines * Don't build the menuselect-tree for the tarball, as it
	  requires running the configure script first * Change the Makefile
	  to note that menuselect-tree depends on the configure script.
	  ........

2007-10-02 19:02 +0000 [r84432-84440]  Jason Parker <jparker@digium.com>

	* /, res/res_features.c: Merged revisions 84410 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500
	  (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel
	  before return on failure. Issue 10821, patch by Ivan ........

2007-10-02 18:12 +0000 [r84405]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Add MSet for people who prefer the old, deprecated
	  syntax of Set (Closes issue #10549)

2007-10-02 14:13 +0000 [r84371]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) |
	  6 lines Use snprintf instead of sprintf in one place. There is no
	  vulnerability here due to various buffer sizes around the code,
	  but I still didn't like seeing a non length-limited copy of data
	  coming off of the wire into a stack buffer, as this would be a
	  problem in the future if buffer sizes elsewhere got changed or
	  size limitations removed ... ........

2007-10-02 13:58 +0000 [r84368]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Don't swap channel priority if using epoll as polling
	  should/will only happen off the first channel. (closes issue
	  #10867) Reported by: phsultan

2007-10-01 23:33 +0000 [r84327-84331]  Steve Murphy <murf@digium.com>

	* utils/check_expr.c: OK. THis a DEBUG_THREADS situation.

	* utils/check_expr.c: picky gcc versions... sigh.

	* utils/check_expr.c: This mod will allow check_expr to compile in
	  the presence of DEBUG_THREAD situations. At least, it does for
	  me. And it's less expensive than several other approaches I
	  tried.

	* res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y,
	  pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1
	  line closes issue #10777 -- by returning a null for the parse
	  tree when there's really nothing there, and making sure we don't
	  try to do checking on a null tree. ........

2007-10-01 21:54 +0000 [r84300]  Jason Parker <jparker@digium.com>

	* Makefile, /, Makefile.rules, channels/Makefile: Merged revisions
	  84291 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6
	  lines Add dist-clean support for subdirs. Change h323 to only
	  remove the Makefile on a dist-clean, rather than a clean. This
	  fixes a bug I found with trying to run make after a make clean
	  ........

2007-10-01 21:31 +0000 [r84275]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged
	  revisions 84274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007)
	  | 1 line moved get_base_channel() code from action_redirect to
	  ast_channel_masquerade() for issue 7706 and BE-160 ........

2007-10-01 21:15 +0000 [r84207-84272]  Russell Bryant <russell@digium.com>

	* /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) |
	  4 lines Fulfull a feature request from Qwell on the "core show
	  locks" output. It will now note the lock type for each lock that
	  a thread holds. (mutex, rdlock, or wrlock) ........

	* /, res/res_agi.c: Merged revisions 84236 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) |
	  5 lines Add another sanity check in the AGI read loop. We really
	  don't care about EAGAIN unless we didn't read an entire line. If
	  there is a newline at the end if the read buffer, break, because
	  we got the whole thing. (reported and patched by bmd) ........

	* /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01
	  Oct 2007) | 2 lines Show rwlocks in the "core show locks" output.
	  Before, it only showed mutexes. ........

2007-10-01 15:57 +0000 [r84176]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Check to make sure a structure pointer is
	  non-NULL before touching it... crashing is bad, mmmk? (closes
	  issue #10831) Reported by: eliel Patches: chan_sip.c.patch
	  uploaded by eliel (license 64)

2007-10-01 15:34 +0000 [r84167-84174]  Russell Bryant <russell@digium.com>

	* main/say.c: Change simple uses of snprintf to ast_copy_string.
	  This was provided by mvanbaak as a part of issue #10843, but this
	  part didn't apply because of a patch I applied right beforehand.

	* channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c,
	  apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
	  main/say.c, apps/app_minivm.c, pbx/dundi-parser.c,
	  channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c,
	  main/rtp.c, channels/chan_mgcp.c: Corydon posted this janitor
	  project to the bug tracker and mvanbaak provided a patch for it.
	  It replaces a bunch of simple calls to snprintf with
	  ast_copy_string (closes issue #10843) Reported by: Corydon76
	  Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7)

	* main/say.c: Simplify code by using the -= and %= operators.
	  (closes issue #10848) Reported by: opticron Patches: saymod.diff
	  uploaded by opticron (license 267)

	* codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The
	  trunk version of this patch also includes a couple more small
	  clean fixes from IgorG. Merged revisions 84170 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) |
	  3 lines Remove another file in "make clean". (closes issue
	  #10814, paravoid) ........

	* main/cli.c: Don't set the full command string until after
	  verifying that there is not another CLI command with the same
	  command text registered. This prevents a crash if someone
	  accidentally calls ast_cli_register() on the same CLI command
	  data twice. This also fixes a small bug where the helpers list
	  would get unlocked without being locked if building the full
	  command failed. (closes issue #10858, reported by jamesgolovich,
	  patched by me)

	* configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new
	  option for files-based music on hold to ensure that the sort
	  order of the files is alphabetical. (closes issue #10855)
	  Reported by: jamesgolovich Patches:
	  asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license
	  176)

	* apps/app_dial.c, /: Merged revisions 84166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) |
	  2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........

2007-10-01 14:21 +0000 [r84159-84165]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Add MP4 to part of the SDP code. (closes
	  issue #10820) Reported by: ruikubo Patches: chan_sip.patch
	  uploaded by ruikubo (license 250)

	* main/dnsmgr.c: Don't register the dnsmgr refresh CLI command
	  twice. (closes issue #10856) Reported by: jamesgolovich Patches:
	  asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license
	  176)

	* /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct
	  2007) | 6 lines Fix randomness. save_pos was being set to 0
	  initially instead of -1, causing it to jump to position 0 when
	  moh started. (closes issue #10859) Reported by: jamesgolovich
	  Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
	  (license 176) ........

	* apps/app_dial.c, /: Merged revisions 84158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4
	  lines Only attempt early bridging if the options given to Dial()
	  permit it. (closes issue #10861) Reported by: peekyb ........

2007-09-30 20:06 +0000 [r84143-84147]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/module.h: Merged revisions 84146 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30
	  Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++
	  modules. The load and reload parameters were in the wrong place.
	  (closes issue #10846, alebm) ........

	* funcs/func_lock.c: * The documentation for the LOCK() function
	  says that it will block for up to 3 seconds while waiting on a
	  lock when other locks are currently held to avoid deadlocks.
	  Change the code to reflect this. * Since trying to grab a lock
	  may block for some time, put the channel in autoservice so that
	  audio is still read from the channel and that any active
	  generators on the channel don't pause.

2007-09-29 23:47 +0000 [r84134-84137]  Steve Murphy <murf@digium.com>

	* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1
	  line This issue sort of closes 10786; All config files support
	  #include with globbing (you know, *,[chars],?,{list,list},etc),
	  so I've updated the AEL system to support this also. ........

	* pbx/ael/ael-test/ael-ntest22/t2 (added),
	  pbx/ael/ael-test/ael-ntest22/t3 (added),
	  pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
	  pbx/ael/ael-test/ael-ntest22 (added),
	  pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
	  (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
	  pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
	  pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
	  pbx/ael/ael-test/ael-ntest22/qq.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL
	  affected a small number of tests. Added a regression test for
	  glob'd includes

2007-09-29 18:21 +0000 [r84130]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_manager.c: Set enablecdr at the end of re-reading the
	  config file (Closes issue #10852)

2007-09-29 00:19 +0000 [r84115]  Matthew Fredrickson <creslin@digium.com>

	* main/translate.c: Let's use process time instead of wall clock
	  time for show translation

2007-09-28 14:35 +0000 [r84050-84080]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Autoconf requires version 2.60, not
	  2.59, to process (Closes issue #10842)

	* /, main/say.c: Merged revisions 84078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007)
	  | 2 lines Correct pronunciations of numbers for .nl (Closes issue
	  #10837) ........

	* main/channel.c, /: Merged revisions 84049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007)
	  | 3 lines Avoid a deadlock with ALL of the locks in the
	  masquerade function, not just the pairs of channels. (Closes
	  issue #10406) ........

2007-09-27 23:18 +0000 [r84019]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/manager.c, /, channels/chan_agent.c,
	  include/asterisk/channel.h: Merged revisions 84018 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27
	  Sep 2007) | 1 line if an Agent is redirected, the base channel
	  should actually be redirected. This was causing multiple issues,
	  especially issue 7706 and BE-160 ........

2007-09-27 00:08 +0000 [r83978-83986]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007)
	  | 2 lines avoid the weird usage of assert() in the ALSA header
	  files that gcc 4.2 wants to complain about ........

	* res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const
	  pointer warnings

2007-09-27 00:02 +0000 [r83911-83977]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) |
	  1 line remove a todo item that has been completed ........

	* /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) |
	  2 lines I changed my mind ... I think this should be a
	  LOG_NOTICE. ........

	* /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) |
	  5 lines Add a log message that was requested by the masses in the
	  developer tutorial session at Astricon. chan_sip did not output
	  any message when a call was rejected because the extension was
	  not found. This adds a verbose message (at verbose level 3) to
	  note when this happens. ........

	* /: Merged revisions 83910 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) |
	  3 lines Fix building chan_misdn under dev-mode. (please run the
	  configure script with --enable-dev-mode so this doesn't happen
	  again ...) ........

2007-09-26 18:43 +0000 [r83880]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007)
	  | 2 lines Remove unused 4k of memory on the program stack (closes
	  issue #10827) ........

2007-09-26 06:53 +0000 [r83849-83864]  Russell Bryant <russell@digium.com>

	* include/asterisk/event.h: fix a typo in a comment

	* include/asterisk/file.h: Change function documentation to use
	  doxygen tags. (Really, I just needed to make some minor change in
	  trunk to test something with automerge ...)

2007-09-25 23:14 +0000 [r83834]  Matthew Fredrickson <creslin@digium.com>

	* doc/ss7.txt: Fix typo in readme

2007-09-25 21:06 +0000 [r83819]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h: Don't note that functions are
	  deprecated in favor of themselves. This was found by showing a
	  very poor example doxygen function in a presentation this
	  morning. :)

2007-09-25 16:34 +0000 [r83804]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Added a CLI command that shows our buddy list,
	  as suggested by Daniel McKeehan, thanks!

2007-09-25 14:18 +0000 [r83774]  Tilghman Lesher <tlesher@digium.com>

	* /, main/app.c: Merged revisions 83773 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007)
	  | 2 lines jmls pointed out that unsetting the group and setting
	  the group to the blank string aren't quite the same. ........

2007-09-25 13:41 +0000 [r83758]  Joshua Colp <jcolp@digium.com>

	* res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a
	  value without freeing the previous result is bad, mmmk?

2007-09-25 09:07 +0000 [r83743]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, include/asterisk/jingle.h: Comply with
	  latest XEP-0166, XEP-0167, XEP-0176. No real Jingle
	  implementation being available, testing was made using two
	  Asterisk servers relaying SIP calls over their Jingle channels:
	  SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] ---
	  [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was
	  possible to test the code in both ways, and make the Jingle
	  channel comply with the latest specifications. No sound available
	  yet. Main modifications include : - modified the
	  'jingle_candidate' structure and the 'jingle_create_candidates'
	  function according to XEP-0176 ; - modified the 'jingle_action'
	  function in order to properly terminate a Jingle session, in
	  conformance with XEP-0166 ; - modified username format used in
	  STUN requests ; - actually make the bindaddr configuration field
	  useable. Todo : - set audio paths up (no native bridging) ; -
	  make the CLI gtalk functions available to jingle ; - clean up the
	  storage space used in strings.

2007-09-25 08:09 +0000 [r83741]  Russell Bryant <russell@digium.com>

	* utils/Makefile, utils: Add some files to the utils directory
	  svn:ignore and Makefile clean target (closes issue #10808,
	  reported by mvanbaak)

2007-09-24 22:06 +0000 [r83696-83726]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, main/asterisk.c: Permit custom locations for astdb and
	  the keys directory (though default to the current locations)
	  (Closes issue #10267)

	* /, build_tools/make_defaults_h: Merged revisions 83695 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007)
	  | 4 lines In the source, keys are relative to the datadir, not
	  varlib (which is the same in most cases, but it's good to be
	  accurate). Closes issue #10811 ........

2007-09-24 17:10 +0000 [r83671]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's
	  patch for configurable SDP origin-field username and session
	  field, closes issue# 10795

2007-09-24 17:00 +0000 [r83656]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: interface_exists_global was never returning 1.
	  Most likely an error from my merge on Friday. (closes issue
	  #10817, reported and patched by snar, patch simplified by me)

2007-09-24 16:42 +0000 [r83654-83655]  Tilghman Lesher <tlesher@digium.com>

	* /, main/app.c: Merged revisions 83637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007)
	  | 3 lines Making change to group splitting, as discussed on the
	  -dev list. The main effect of this will be to permit
	  Set(GROUP([cat])=), i.e. unsetting a group. ........

2007-09-22 19:54 +0000 [r83575-83590]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /: Merged revisions 83589 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1
	  line This closes issue #10788 -- The exact same fixes are made
	  here for the first arg in the for(arg1; arg2; arg3) {} statement,
	  as were done for the 3rd arg. It can now be an assignment that
	  will embedded in a Set() app, or a macro call, or an app call.
	  ........

	* res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1
	  line This closes issue #10788 -- the 3rd arg in the for statement
	  is now wrapped in Set() only if there's an '=' in that string.
	  Otherwise, if it begins with '&', then a Macro call is generated;
	  otherwise it is made into an app call. A bit more accomodating,
	  keeps the new guys happy, and the guys with ael-1 code should be
	  happy, too ........

2007-09-22 17:37 +0000 [r83574]  Matthew Fredrickson <creslin@digium.com>

	* doc/ss7.txt: Fix potential point of confusion

2007-09-22 14:45 +0000 [r83517-83545]  Tilghman Lesher <tlesher@digium.com>

	* utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix
	  build of check_expr and hashtest2 when DEBUG_THREADLOCAL is
	  defined

	* main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload
	  manager commands, which supplement the need to have Command
	  privilege. (closes issue #10736)

	* configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
	  main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for
	  FreeBSD... testing for every conceivable math function now

2007-09-21 19:55 +0000 [r83500]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix compilation errors in CLI command
	  updates to SS7 CLI commands

2007-09-21 19:54 +0000 [r83499]  Matthew Fredrickson <creslin@digium.com>

	* doc/ss7.txt (added): Add an SS7 readme for setup and use of
	  libss7 and asterisk

2007-09-21 18:41 +0000 [r83484]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Fix some areas where we were still using '|'
	  for an argument delimiter (closes issue #10793)

2007-09-21 18:27 +0000 [r83483]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Update app_queue to use commas as application
	  argument separators. (closes issue #10793, snar)

2007-09-21 17:36 +0000 [r83466]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_manager.c: Fix cdr_manager, such that if the config file
	  is created past load, it'll start logging (and conversely, if the
	  config file is destroyed or deactivated, the logging is
	  disabled). Reported by Juggie via IRC, fix by me.

2007-09-21 14:40 +0000 [r83433]  Russell Bryant <russell@digium.com>

	* res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c,
	  main/db1-ast/hash/hash.c, include/asterisk/channel.h,
	  channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c,
	  main/cdr.c, main/channel.c, channels/chan_misdn.c,
	  main/ast_expr2f.c, main/file.c, include/asterisk/sched.h,
	  channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c,
	  main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged
	  revisions 83432 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) |
	  4 lines gcc 4.2 has a new set of warnings dealing with cosnt
	  pointers. This set of changes gets all of Asterisk (minus
	  chan_alsa for now) to compile with gcc 4.2. (closes issue #10774,
	  patch from qwell) ........

2007-09-21 14:25 +0000 [r83431]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
	  main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the
	  presence of trunc and round, and make the ISOC99 detection a
	  little more sane (closes issue #10776)

2007-09-20 23:14 +0000 [r83381]  Jason Parker <jparker@digium.com>

	* apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More
	  NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch
	  uploaded by moy (license 222) app_minivm.c.patch uploaded by
	  eliel (license 64) astmm.c.patch uploaded by eliel (license 64)

2007-09-20 21:37 +0000 [r83350-83351]  Mark Michelson <mmichelson@digium.com>

	* /: Oops. Getting rid of svnmerge-integrated and automerge stuff

	* /, apps/app_queue.c: Merging changes from queue_refcount_trunk
	  into trunk. Refcounted queues now in place.

2007-09-20 21:17 +0000 [r83293-83349]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.c: Merged revisions 83348 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) |
	  4 lines When daemonizing, don't change working directory to "/".
	  It makes it not be able to do a core dump when not running as
	  uid=root. (closes issue #10766, xrg) ........

	* /, contrib/scripts/safe_asterisk: Merged revisions 83316 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) |
	  3 lines Change safe_asterisk to explicitly ask for /bin/bash, as
	  it uses bashisms. (closes issue #10772, reported by culrich)
	  ........

	* main/dsp.c: trivial formatting change

	* main/asterisk.c: trivial formatting change

	* main/app.c: minor spelling fixes in a comment

	* main/app.c: minor grammar fix

	* channels/chan_sip.c: fix spelling in a comment

	* main/asterisk.c: trivial formatting change

2007-09-20 19:05 +0000 [r83251-83278]  Jason Parker <jparker@digium.com>

	* doc/modules.txt: Fix a trivial typo, to test our new commit bot

	* /, apps/app_disa.c: Merged revisions 83246 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8
	  lines If # is pressed after dialing an extension in DISA, stop
	  trying to collect more digits. (closes issue #10754) Reported by:
	  atis Patches: app_disa.c.branch.patch uploaded by atis (license
	  242) app_disa.c.trunk.patch uploaded by atis (license 242)
	  ........

2007-09-20 16:28 +0000 [r83234]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7
	  lines Make sure the minimum T1 timer value is obeyed in all
	  cases. (closes issue #10768) Reported by: flefoll Patches:
	  chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license
	  244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll
	  (license 244) ........

2007-09-20 16:27 +0000 [r83233]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Don't start the event processing thread until
	  after forking. (reported by Simon on the -dev list, thanks!)

2007-09-20 16:19 +0000 [r83229-83231]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7
	  lines Fix a minor spelling error. (closes issue #10769) Reported
	  by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded
	  by flefoll (license 244) chan_sip.c.br14.83070.inita-patch
	  uploaded by flefoll (license 244) ........

	* pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks
	  in pbx_dundi, cdr_pgsql, and the configuration file parser.

2007-09-19 23:16 +0000 [r83213]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c,
	  apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724)
	  Patches: chan_zap.c.patch uploaded by moy (license 222)
	  app_queue.c.patch uploaded by eliel (license 64)
	  app_voicemail.c.patch uploaded by eliel (license 64)
	  app_meetme.c.patch uploaded by eliel (license 64)

2007-09-19 20:06 +0000 [r83182-83183]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme
	  for me today?)

	* res/res_adsi.c: Clean up code in res_adsi.

2007-09-19 19:54 +0000 [r83176-83181]  Russell Bryant <russell@digium.com>

	* funcs/func_shell.c: put the channel in autoservice when executing
	  func_shell

	* /, apps/app_system.c: Merged revisions 83179 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) |
	  5 lines The System() and TrySystem() applications can take a
	  substantial amount of time to execute while not servicing the
	  channel. So, put the channel in autoservice while the command is
	  being executed. (closes issue #10726, reported by mnicholson)
	  ........

	* funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) |
	  4 lines Using curl can take a substantial amount of time, so the
	  channel should be autoserviced while waiting for it to complete.
	  (closes issue #10725, reported by mnicholson) ........

	* /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) |
	  8 lines When handling a reload of chan_iax2, don't use an
	  ao2_callback() to POKE all peers. Instead, use an iterator. By
	  using an iterator, the peers container is not locked while the
	  POKE is being done. It can cause a deadlock if the peers
	  container is locked because poking a peer will try to lock pvt
	  structs, while there is a lot of other code that will hold a pvt
	  lock when trying to go lock the peers container. (reported to me
	  directly by Loic Didelot. Thank you for the debug info!) ........

2007-09-19 17:22 +0000 [r83155-83157]  Joshua Colp <jcolp@digium.com>

	* apps/app_db.c: Fix indentation in app_db.

	* apps/app_authenticate.c: Clean up code in app_authenticate.

	* apps/app_adsiprog.c: Clean up code in app_adsiprog.

2007-09-19 15:11 +0000 [r83126]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 83121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) |
	  4 lines Fix up another potential race condition. Do the loop
	  decrementing use count on events with the eventq protected from
	  being changed. (reported on IRC by Ivan) ........

2007-09-19 15:08 +0000 [r83105-83114]  Joshua Colp <jcolp@digium.com>

	* apps/app_disa.c: DISA only needs to know about the end of DTMF,
	  not the beginning/duration.

	* apps/app_disa.c: Clean up app_disa code a bit.

2007-09-19 13:55 +0000 [r83076]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c: Replace Google namespace occurrences with
	  Jingle. The former namespace is handled by chan_gtalk.

2007-09-19 13:49 +0000 [r83073-83075]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 83074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6
	  lines Protect the CDR record from modification by pbx_exec so
	  that the application data contains the Queue data. (closes issue
	  #10761) Reported by: snar Patches: app-queue-mixmonitor.patch
	  uploaded by snar (license 245) ........

	* main/manager.c: Extend manager show connected with additional
	  information. (closes issue #10757) Reported by: outtolunc
	  Patches: manager.c.sessionstart.diff uploaded by outtolunc
	  (license 237)

2007-09-19 13:29 +0000 [r83072]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c: Remove namespaces in payload-type tags.

2007-09-19 13:21 +0000 [r83071]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6
	  lines (closes issue #10760) Reported by: dimas Patches:
	  chan_sip.patch uploaded by dimas (license 88) Read in
	  subscribecontext option in general to be the default. ........

2007-09-19 12:23 +0000 [r83055]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, include/asterisk/jingle.h: Transmit
	  proper invitation, thus conforming to XEP-0166 (Jingle general
	  specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176
	  (Jingle ICE Transport).

2007-09-19 09:48 +0000 [r83025]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
	  channels/misdn_config.c: Merged revisions 83023-83024 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) |
	  1 line added 'astdtmf' option to allow configuring the asterisk
	  dtmf detector instead of the mISDN_dsp ones. also added the patch
	  from irroot #10190, so that dtmf tones detected by the asterisk
	  detector are passed outofband to asterisk, to make any use of
	  dtmf tones at all. ........ r83024 | crichter | 2007-09-19
	  11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which
	  violates the coding guidelines. ........

2007-09-19 00:21 +0000 [r82993]  Russell Bryant <russell@digium.com>

	* /, apps/app_flash.c: Merged revisions 82992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) |
	  4 lines Change the description of app_flash to note how it can be
	  a useful tool instead of just saying that it is generally a
	  worthless feature. (Thanks to Jim Van Meggelen for pointing it
	  out and providing the proposed text) ........

2007-09-18 23:42 +0000 [r82962]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 82961 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2
	  lines Initialize a variable to NULL to make the world happy.
	  ........

2007-09-18 22:46 +0000 [r82931]  Russell Bryant <russell@digium.com>

	* include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) |
	  11 lines Add a new patch to handle interrupting the fgets() call
	  when using FastAGI. This version of the patch maintains the
	  original behavior of the code when not using FastAGI. (closes
	  issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch
	  uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch
	  uploaded by juggie (license 24) Slight mods by me Tested by:
	  juggie, festr ........

2007-09-18 22:43 +0000 [r82871-82930]  Jason Parker <jparker@digium.com>

	* main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c,
	  channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
	  res/res_musiconhold.c, res/res_jabber.c, main/manager.c,
	  res/res_agi.c, channels/chan_features.c, main/logger.c,
	  main/http.c, channels/chan_alsa.c, res/res_realtime.c,
	  res/res_odbc.c: (issue #10724) Reported by: eliel Patches:
	  res_features.c.patch uploaded by eliel (license 64)
	  res_agi.c.patch uploaded by seanbright (license 71)
	  res_musiconhold.c.patch uploaded by seanbright (license 71)
	  pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded
	  by moy (license 222) frame.c.patch uploaded by moy (license 222)
	  manager.c.patch uploaded by moy (license 222) http.c.patch
	  uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy
	  (license 222) res_realtime.c.patch uploaded by eliel (license 64)
	  res_odbc.c.patch uploaded by seanbright (license 71)
	  res_jabber.c.patch uploaded by eliel (license 64)
	  chan_local.c.patch uploaded by eliel (license 64)
	  chan_agent.c.patch uploaded by eliel (license 64)
	  chan_alsa.c.patch uploaded by eliel (license 64)
	  chan_features.c.patch uploaded by eliel (license 64)
	  chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch
	  (includes all of the above patches) uploaded by seanbright
	  (license 71) Convert many CLI commands to the NEW_CLI format.

	* configs/voicemail.conf.sample, apps/app_voicemail.c: (closes
	  issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff
	  uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by
	  qwell (license 4) Tested by: callguy, ruffle Add an option to
	  disable the automatic moving of "heard" messages to the Old
	  folder.

2007-09-18 20:59 +0000 [r82868]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 82867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) |
	  10 lines Fix a memory leak that can occur on systems under higher
	  load. The issue is that when events are appended to the master
	  event queue, they use the number of active sessions as a use
	  count so it will know when all active sessions at the time the
	  event happened have consumed it. However, the handling of the
	  number of sessions was not properly synchronized, so the use
	  count was not always correct, causing an event to disappear
	  early, or get stuck in the event queue for forever. (closes issue
	  #9238, reported by bweschke, patch from Ivan, modified by me)
	  ........

2007-09-18 20:10 +0000 [r82866]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 82865 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep
	  2007) | 4 lines Moving the logic for handling an empty membername
	  to the create_member function so that there is a common place
	  where this occurs instead of being spread out to several
	  different places. ........

2007-09-18 19:06 +0000 [r82835]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_queue.c: Merged revisions 82834 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007)
	  | 2 lines there is no need for conditional logic to select
	  ->interface or ->membername, snince ->membername will always be
	  populated ........

2007-09-18 16:34 +0000 [r82803]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) |
	  4 lines When copying the contents from the wildcard peer, do a
	  deep copy instead of shallow copy so that it doesn't crash when
	  beging destroyed. (closes issue #10546, patch by me) ........

2007-09-18 16:16 +0000 [r82800]  Jason Parker <jparker@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: (closes issue
	  #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch
	  uploaded by snar (license 245) queues.conf.patch uploaded by snar
	  (license 245) Add an updatecdr option to queues.conf, so that if
	  a "member name" is specified, the cdr record will be updated with
	  that, rather than the channel.

2007-09-18 16:14 +0000 [r82776-82793]  Russell Bryant <russell@digium.com>

	* include/asterisk/threadstorage.h: Make sure that libpthread
	  doesn't try to call free() directly when MALLOC_DEBUG is enabled.
	  If it does, Asterisk will crash as the address isn't the real
	  beginning of the allocation.

	* channels/chan_zap.c: Don't use ast_channel_lock_both() here, it
	  only exists in one of my branches. This is theoretically a
	  potential deadlock, but it's the way it was before so I'm going
	  to leave it this way for now.

2007-09-18 15:29 +0000 [r82752]  Jason Parker <jparker@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500
	  (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains
	  option in SIP sample config. Issue 10753 ........

2007-09-17 22:59 +0000 [r82728]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c,
	  channels/chan_agent.c, channels/chan_alsa.c,
	  channels/chan_iax2.c, channels/chan_mgcp.c: convert various
	  places that access the channel lock directly to use the channel
	  lock wrappers

2007-09-17 21:52 +0000 [r82710-82712]  Jason Parker <jparker@digium.com>

	* cdr/cdr_sqlite3_custom.c: Don't try to continue loading
	  cdr_sqlite3_custom on a module load failure (such as the config
	  not existing) Closes issue #10749, patch by seanbright.

	* configs/http.conf.sample: Fix the sample redirect to point to a
	  valid file in the Asterisk GUI. Closes issue #10748, patch by
	  bkruse

2007-09-17 20:24 +0000 [r82595-82679]  Russell Bryant <russell@digium.com>

	* doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support
	  for #include, var_metric, and cat_metric in res_config_sqlite
	  (closes issue #10738, rbraun_proformatique)

	* /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged
	  revisions 82676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) |
	  4 lines Put a memset in ast_localtime() instead of a couple
	  places in app_voicemail to prevent the problem everywhere instead
	  of just a couple of places. (related to issue #10746) ........

	* /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) |
	  6 lines Initialize some memory to fix crashes when leaving
	  voicemail. This problem was fixed by running Asterisk under
	  valgrind. (closes issue #10746, reported by arcivanov, patched by
	  me) *** IMPORTANT NOTE: We need to check to see if this same bug
	  exists elsewhere. ........

	* apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h,
	  apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
	  res/res_features.c, apps/app_queue.c, channels/chan_iax2.c,
	  pbx/pbx_config.c: Make the MALLOC_DEBUG output for free() useful
	  again. After changing calls to free to be ast_free, astmm said
	  all calls to free were coming from utils.h

	* /, res/res_features.c: Merged revisions 82594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) |
	  5 lines Handle the case where there are multiple dynamic features
	  with the same digit mapping, but won't always match the activated
	  on/by access controls. In that case, the code needs to keep
	  trying features for a match. (reported by Atis on the
	  asterisk-dev list, patched by me) ........

2007-09-17 16:44 +0000 [r82593]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17
	  Sep 2007) | 2 lines fix a couple of places where a logical member
	  name (if specified) was not used, but instead the direct
	  interface was listed ........ r82592 | kpfleming | 2007-09-17
	  11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that
	  wasn't supposed to be committed... doh! ........

2007-09-17 14:58 +0000 [r82568]  Doug Bailey <dbailey@digium.com>

	* main/http.c: Fix memory leak introduced when POST support was
	  added.

2007-09-17 02:20 +0000 [r82516-82546]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10715) Reported by:
	  the-chopper Don't bother hanging up the new channel if it does
	  not exist yet.

	* main/pbx.c, /: Merged revisions 82514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4
	  lines (closes issue #10734) Reported by: asgaroth Instead of
	  passing a NULL pointer into snprintf pass "". It makes Solaris
	  much happier. ........

2007-09-16 15:32 +0000 [r82496]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Option maxmessage should be maxsecs
	  per-folder, too (closes issue #10729)

2007-09-14 21:30 +0000 [r82457]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 82444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1
	  line closes issue #10668; thanks to arkadia for his patch; had to
	  leave out the bit about ending the previous cdr in the fork; it
	  would destroy current implementations. ........

2007-09-14 21:21 +0000 [r82454]  Russell Bryant <russell@digium.com>

	* /, configs/zapata.conf.sample: Merged revisions 82435 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) |
	  3 lines Add a note to help clarify the value set with the
	  echocancel option. (inspired by Malcolm's blog post on
	  blogs.digium.com about HPEC) ........

2007-09-14 19:49 +0000 [r82401]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Add support
	  in chan_skinny for sending RTP directly to the endpoints. Closes
	  issue #9154, patch by DEA

2007-09-14 18:37 +0000 [r82397-82400]  Mark Michelson <mmichelson@digium.com>

	* /: Blocking revision 82398

	* /, apps/app_queue.c: Merged revisions 82396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep
	  2007) | 5 lines Adding member name field to manager events where
	  they were missing before (closes issue #10721, reported by snar)
	  ........

2007-09-14 17:51 +0000 [r82395]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5
	  lines If a channel does not have an owner, do not try to set a
	  channel variable. This will end up making the channel variable
	  global, which is not right. Closes issue #10720, patch by
	  flefoll. ........

2007-09-14 17:29 +0000 [r82393]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute
	  method to res_odbc (closes issue #10722)

2007-09-14 16:02 +0000 [r82386-82391]  Russell Bryant <russell@digium.com>

	* channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed),
	  channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c,
	  channels/xpmr/xpmr_coef.h: use the standard license header for
	  the xpmr files

	* channels/chan_usbradio.c (added), channels/xpmr (added): Add
	  chan_usbradio to trunk

	* /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  Merged revisions 82385 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) |
	  3 lines Add checking for libusb here, so nobody has to deal with
	  conflicts in the chan_usbradio-1.4 branch every time the
	  configure script gets changed ........

2007-09-14 14:44 +0000 [r82377]  Mark Michelson <mmichelson@digium.com>

	* doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14
	  Sep 2007) | 5 lines Fixing a typo in the coding guidelines
	  (closes issue #10717, reported and patched by leedm777) ........

2007-09-14 13:02 +0000 [r82373]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c: Fix DTMF following what has been done in
	  issue #9401. Thanks irroot.

2007-09-13 23:12 +0000 [r82359]  Jason Parker <jparker@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4
	  lines Fix a small typo. retrytime > waittime ........

2007-09-13 21:53 +0000 [r82347-82352]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changed "in" to "queue" in "queue
	  {pause|unpause} member" command to be more clear. Also added
	  check to be sure that sixth argument is the word "reason" if full
	  command is given

	* CHANGES, apps/app_queue.c: Added the ability to pause and unpause
	  members via the CLI

	* /, apps/app_queue.c: Merged revisions 82346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep
	  2007) | 4 lines Preemptively fixing a possible segfault. It is
	  possible that queuename is NULL (meaning pause ALL queues), so
	  use q->name instead. ........

2007-09-13 20:13 +0000 [r82345]  Jason Parker <jparker@digium.com>

	* /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9
	  lines Fix a crash that could occur in cdr_csv when mutliple
	  threads tried to close the same file. Do we actually need the
	  locking here? What happens if you open the same file twice, and
	  two threads try to write to it at the same time? Is fputs() going
	  to write out the entire line at once? I suspect that it could be
	  possible for the second fopen to run during the first fputs, so
	  the position could be in the middle of the previously written
	  line... Issue 10347, initial patch by explidous (but I removed
	  all of the paranoia stuff..) ........

2007-09-13 19:16 +0000 [r82338-82341]  Russell Bryant <russell@digium.com>

	* /, main/astobj2.c: Merged revisions 82339 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) |
	  1 line resolve a warning when not building under dev mode
	  ........

	* include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged
	  revisions 82337 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) |
	  4 lines Only compile in tracking astobj2 statistics if dev-mode
	  is enabled. Also, when dev mode is enabled, register the CLI
	  command that can be used to run the astobj2 test and print out
	  statistics. ........

2007-09-13 18:13 +0000 [r82336]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: Merged revisions 82335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700
	  (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13
	  Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license
	  exceptions ........ ................

2007-09-13 16:58 +0000 [r82329]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
	  setvar support to chan_zap. Just like you can in chan_sip and
	  chan_iax2 you can now use it with zaptel channels. (done while in
	  Montreal at the Asterisk bootcamp!)

2007-09-13 16:27 +0000 [r82327]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 82326 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep
	  2007) | 7 lines Added logic to handle the unlikely case that
	  someone has two queues with the same name. Asterisk will log a
	  warning message letting the user know that one was already
	  defined with that name and is it skipping all further instances.
	  This also will work for realtime queues but in order for that to
	  happen, the user would have to trigger a perfectly timed reload
	  as a realtime queue is being looked up, which is highly unlikely
	  (but taken care of nonetheless). ........

2007-09-13 15:26 +0000 [r82321]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxyref.h, doc/res_config_sqlite.txt,
	  res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various
	  code and documentation cleanups for res_config_sqlite (closes
	  issue #10711, rbraun_proformatique)

2007-09-13 15:25 +0000 [r82312-82320]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c: Modify rule filters to match with the
	  Jingle namespace constant

	* include/asterisk/jingle.h: Assign namespace properly

	* channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle
	  and Jingle DTMF namespaces. As both specifications are in the
	  Experimental status, the namespaces specified therein shall be of
	  the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See
	  the Namespace issuance section in XEP-0053 :
	  http://www.xmpp.org/extensions/xep-0053.html#namespaces

	* channels/chan_jingle.c: Reflect Jingle DTMF specification changes

2007-09-13 13:34 +0000 [r82311]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix a missing unref of a member struct. This
	  was pointed out by Marta. Thanks! This function in 1.4 didn't
	  have the problem.

2007-09-13 11:54 +0000 [r82310]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13
	  Sep 2007) | 4 lines Closes issue #9401, reported and patched by
	  irrot, with slight modifications by me. Handle DTMF sent by
	  Asterisk properly. ........

2007-09-12 21:57 +0000 [r82297]  Russell Bryant <russell@digium.com>

	* /, res/res_agi.c: Merged revisions 82296 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) |
	  3 lines Fix a check of the wrong pointer, as pointed out by an
	  XXX comment left in the code. The problem was harmless, however.
	  ........

2007-09-12 21:55 +0000 [r82294]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: After some discussions, we decided that the
	  return values here were a bit messy. This also fixes a bug on
	  reload, where peers may not have reregistered properly.

2007-09-12 21:32 +0000 [r82290-82292]  Tilghman Lesher <tlesher@digium.com>

	* /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12
	  Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files
	  ........

	* main/stdtime/private.h, /, main/stdtime/tzfile.h,
	  funcs/func_strings.c, apps/app_sms.c,
	  include/asterisk/localtime.h, main/stdtime/localtime.c: Merged
	  revisions 82285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007)
	  | 4 lines Working on issue #10531 exposed a rather nasty 64-bit
	  issue on ast_mktime, so we updated the localtime.c file from
	  source. Next we'll have to write ast_strptime to match. ........

2007-09-12 21:17 +0000 [r82289]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem
	  because unless get_member_status returned QUEUE_NORMAL, a NULL
	  member would be unreferenced. While this didn't cause any crashes
	  or anything terrible, it still is incorrect

2007-09-12 20:50 +0000 [r82288]  Steve Murphy <murf@digium.com>

	* main/config.c: This fix closes issue #10642 -- it's not perfect,
	  but should retain most blank lines in config files, via
	  read/write cycles.

2007-09-12 20:47 +0000 [r82287]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007)
	  | 1 line remove a race condition for the creation of
	  recordthread's, and fix a small memory leak. This closes issue#
	  10636 ........

2007-09-12 16:24 +0000 [r82283]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build
	  warnings (closes issue #10698, reported and patched by snuffy)

2007-09-12 15:53 +0000 [r82279-82282]  Russell Bryant <russell@digium.com>

	* utils/hashtest2.c: Change the traversal to use ao2_callback()
	  instead of an ao2_iterator. Using ao2_callback() is a much more
	  efficient way of performing an operation on every item in the
	  container. This change makes hashtest2 run in about 25% of the
	  time it ran before on my system. In general, I would say that it
	  makes the most sense to use an ao2_iterator if the operation
	  being performed is going to take a long time and you don't want
	  to keep the container locked while you work with each object.
	  Otherwise, the use of ao2_callback is preferred.

	* /, main/asterisk.c: Merged revisions 82280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) |
	  4 lines Clean up the output of "asterisk -h". This tweaks the
	  wording and wraps lines at 80 characters. (closes issue #10699,
	  seanbright) ........

	* /, res/res_agi.c: Merged revisions 82278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) |
	  3 lines revert patch from issue #10553, as someone not using
	  fastagi reported that this broke their system. ........

2007-09-12 14:31 +0000 [r82275-82277]  Mark Michelson <mmichelson@digium.com>

	* /: Blocking changes from revision 82276

	* /, apps/app_queue.c: Merged revisions 82274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep
	  2007) | 6 lines We should only initialize a realtime queue when
	  it is allocated, not every time we access it. This prevents the
	  members ao2_container from being reallocated every time the queue
	  is accessed. I also removed a debug message I had accidentally
	  left in on a previous commit. ........

2007-09-11 23:07 +0000 [r82273]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Fix to make sure we don't hangup a call when
	  getting a RLC without sending REL. Found making sure we are Q.784
	  (the SS7 test specification) compliant

2007-09-11 22:38 +0000 [r82269-82270]  Russell Bryant <russell@digium.com>

	* main/config.c: remove unused functions that made this file not
	  build under dev mode

	* /, apps/app_queue.c: Merged revisions 82267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) |
	  3 lines Fix incorrect uses of ao2_find(). Every one of these
	  calls was reading bogus memory ... ........

2007-09-11 22:37 +0000 [r82268]  Steve Murphy <murf@digium.com>

	* utils/Makefile, main/config.c: This solves an unreported solaris
	  compile problem (missing -lnsl -lsocket).

2007-09-11 21:43 +0000 [r82266]  Joshua Colp <jcolp@digium.com>

	* /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged
	  revisions 82265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4
	  lines (closes issue #10679) Reported by: andrew Build under dev
	  mode when K6OPTS is enabled. ........

2007-09-11 20:50 +0000 [r82264]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 82263 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) |
	  5 lines Fix another missing unref of member objects. This one was
	  pointed out by Marta. When building the outgoing list in
	  try_calling(), a member reference is stored in each outgoing
	  entry. However, when this list got destroyed, the reference was
	  not released. ........

2007-09-11 20:49 +0000 [r82262]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 82261 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1
	  line this change should fix issue # 10659 -- what I worry about
	  is how many other bug reports it may generate. Hopefully, we can
	  please the/a majority. Hopefully. We shall see. Calls not marked
	  ANSWERED and with only one channel name will not be posted. This
	  should eliminate the double CDR's. ........

2007-09-11 18:37 +0000 [r82257-82258]  Joshua Colp <jcolp@digium.com>

	* configs/sip.conf.sample: Lil' bit more documentation to keep
	  folks happy.

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes
	  issue #9433) Reported by: junky Patches: register_trying.diff.txt
	  uploaded by jcmoore Disable sending 100 Trying on REGISTER
	  attempts and make it an option. This has been signed off by oej.

2007-09-11 17:16 +0000 [r82256]  Steve Murphy <murf@digium.com>

	* utils/Makefile: fixing up the pthread stuff for hashtest2

2007-09-11 16:15 +0000 [r82254]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged
	  revisions 82249 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) |
	  1 line fixed a hold/retrieve issue. ........

2007-09-11 16:12 +0000 [r82253]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 82252 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep
	  2007) | 6 lines All instances of ao2_iterators which were just
	  named 'i' have been renamed to 'mem_iter' so that when refcounted
	  queues are merged into trunk, there will be little confusion
	  regarding iterator names, especially when a queue and member
	  iterator are used in the same function. ........

2007-09-11 16:05 +0000 [r82251]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) |
	  4 lines The sample dundi.conf claims support for a wildcard peer
	  entry - [*], but the code did not support it. This patch makes it
	  work. (closes issue #10546, patch by dds, with some changes by
	  me) ........

2007-09-11 15:34 +0000 [r82248]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c: (closes issue #10666) Reported by: arkadia Patches:
	  cdr_lockorder.patch uploaded by arkadia (license 233) Optimize
	  CDR stuff a bit.

2007-09-11 15:31 +0000 [r82246-82247]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Remove an unused variable. I have no idea why this
	  was marked with the unused attribute instead of just removing it.
	  :)

	* /, res/res_agi.c: Merged revisions 82245 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) |
	  9 lines (closes issue #10553) Reported by: juggie Patches:
	  res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
	  juggie When using fastagi, fgets() can return before a full line
	  is read. Add explicit handling for the case where it gets
	  interrupted. ........

2007-09-11 14:58 +0000 [r82242-82244]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6
	  lines (closes issue #10577) Reported by: jamesgolovich Patches:
	  asterisk-dundifree.diff.txt uploaded by jamesgolovich (license
	  176) Don't leak memory when unloading DUNDi. ........

	* apps/app_meetme.c: (closes issue #10560) Reported by: ruffle
	  Patches: rb uploaded by ruffle (license 201) Show whether the
	  conference is locked or not on the CLI.

2007-09-11 14:35 +0000 [r82237-82241]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 82240 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) |
	  2 lines Add a couple more missing unrefs of queue member objects
	  ........

	* /, apps/app_queue.c: Merged revisions 82238 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) |
	  2 lines Add a missing unref of a queue member in an error
	  handling block ........

	* /, apps/app_queue.c: Merged revisions 82236 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) |
	  2 lines Document why membercount can not simply be replaced by
	  ao2_container_count() ........

2007-09-11 13:46 +0000 [r82231-82235]  Joshua Colp <jcolp@digium.com>

	* utils/Makefile: Include string compatibility file in hashtest2.

	* utils/hashtest2.c: Include compat.h to hopefully make it
	  compatible with FreeBSD.

	* utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h
	  exists before including it.

	* main/manager.c: (closes issue #10695) Reported by: junky Patches:
	  count_showconn.diff uploaded by junky (license 177) Provide a
	  count of connected users to manager.

	* main/minimime/minimime.c, main/minimime/tests/create.c,
	  main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes
	  issue #10692) Reported by: snuffy Patches: minivm.diff uploaded
	  by snuffy (license 35) Instead of using err (which is not
	  available under Solaris) use fdprintf with stderr.

2007-09-10 20:03 +0000 [r82200]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to
	  have manager-style output, instead of CLI-style output (closes
	  issue #8254)

2007-09-10 19:10 +0000 [r82185]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing a problem where NULL channels would
	  cause a crash when calling indisposed queue members (i.e. paused,
	  wrapup time not completed, etc.)

2007-09-10 18:32 +0000 [r82178]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: Merged revisions 82155 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007)
	  | 2 lines Convert struct member to use refcounts (closes issue
	  #10199) ........

2007-09-10 17:39 +0000 [r82154]  Jason Parker <jparker@digium.com>

	* main/db.c: Add a counter to the 'database deltree' CLI command.
	  Note: this is slightly different than the initial patch, because
	  I felt that using res <= 0 would be a change in behavior. Closes
	  issue #10687, patch by junky

2007-09-10 16:59 +0000 [r82140]  Steve Murphy <murf@digium.com>

	* utils/Makefile, utils/hashtest2.c (added): Committing my test for
	  astobj2, hashtest2.c, along with makefile changes in utils.

2007-09-10 16:24 +0000 [r82125]  Jason Parker <jparker@digium.com>

	* main/db.c: Add counter to 'database show' CLI command. (also a
	  minor whitespace change that I found along the way) Closes issue
	  #10683, patch by junky

2007-09-10 16:19 +0000 [r82124]  Steve Murphy <murf@digium.com>

	* main/astobj2.c: Changes applied from marta's team/marta/astobj2
	  branch to solve a race condition

2007-09-10 15:05 +0000 [r82092]  Mark Michelson <mmichelson@digium.com>

	* /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10
	  Sep 2007) | 5 lines Removing non-existent options from misdn
	  configuration sample. (closes issue #10678, reported and patched
	  by IgorG) ........

2007-09-10 14:26 +0000 [r82062-82077]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10688) Reported by: casper
	  Patches: chan_sip.c.82076.diff uploaded by casper (license 55)
	  Remove double check for zombie flag and optimize things a bit.

	* res/res_agi.c: (closes issue #10684) Reported by: junky Patches:
	  debug.diff uploaded by junky (license 177) Fix issue with debug
	  always showing up.

	* apps/app_meetme.c: (closes issue #10686) Reported by: junky
	  Patches: meet.diff uploaded by junky (license 177) Change NOTICE
	  message to DEBUG.

2007-09-09 02:45 +0000 [r82029]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08
	  Sep 2007) | 2 lines Fix inline compiles on really old compilers
	  (who uses gcc 2.7 anymore, really?) (closes issue #10675)
	  ........

2007-09-08 19:01 +0000 [r81998-81999]  Russell Bryant <russell@digium.com>

	* include/asterisk/slinfactory.h: Add doxygen documentation for
	  slinfactory_destroy(), mainly just noting that it doesn't free
	  the slinfactory itself. (This isn't related to a bug, i'm just
	  looking over random code)

	* /, main/asterisk.c: Merged revisions 81997 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) |
	  2 lines Fix a small memory leak. ast_unregister_atexit() did not
	  free the entry it removed. ........

2007-09-08 16:37 +0000 [r81984]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Make Callerid more consistent in IMAP mail
	  headers (closes issue #10056, reported and patched by jaroth,
	  with small modification by me)

2007-09-08 13:45 +0000 [r81953]  Russell Bryant <russell@digium.com>

	* /, .cleancount: Merged revisions 81952 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) |
	  11 lines (closes issue #10672) Bump the cleancount so that a
	  "make clean" will be forced. This is needed because my fix in
	  revision 81599 made a change to a data structure in file.h, and
	  since file dependency tracking is only on with dev-mode enabled,
	  file format modules that don't get rebuilt may crash, as is the
	  case with this issue. This makes me wonder - how much faster does
	  the code build without the file dependency tracking enabled? If
	  it doesn't make much of a difference, then it may be worth just
	  keeping it on all of the time, or perhaps just not in release
	  tarballs, so that this type of issue is avoided. ........

2007-09-07 19:53 +0000 [r81910-81924]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 81923 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500
	  (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable
	  to be used as the mixmonitor filename. This moves the setting of
	  the MEMBERINTERFACE variable to before mixmonitor. Issue 10671,
	  patch by sim. ........

	* apps/app_queue.c: Add an optional reason parameter to
	  PauseQueueMember/UnpauseQueueMember applications and manager
	  events. Issue 8738, patch by rgollent

2007-09-07 15:29 +0000 [r81891]  Mark Michelson <mmichelson@digium.com>

	* /, configs/queues.conf.sample: Merged revisions 81886 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep
	  2007) | 3 lines Moving the explanation for joinempty to a more
	  appropriate place ........

2007-09-07 12:32 +0000 [r81858-81873]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Don't check for epoll support when cross
	  compiling.

	* main/channel.c, main/audiohook.c: Fix memory issue that crept up
	  with Russell's testing. It is *not* proper to free the frame we
	  get in ast_write.

2007-09-06 22:32 +0000 [r81839-81849]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix the build ... oops

	* /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) |
	  16 lines (closes issue #9724, closes issue #10374) Reported by:
	  kenw Patches: 9724.txt uploaded by russell (license 2) Tested by:
	  kenw, russell Resolve a deadlock that occurs when doing a SIP
	  transfer to parking. I come across this type of deadlock fairly
	  often it seems. It is very important to mind the boundary between
	  the channel driver and the core in respect to the channel lock
	  and the channel-pvt lock. Channel drivers lock to lock the pvt
	  and then the channel once it calls into the core, while the core
	  will do it in the opposite order. The way this is avoided is by
	  having channel drivers either release their pvt lock while
	  calling into the core, or such as in this case, unlocking the pvt
	  just long enough to acquire the channel lock. ........

2007-09-06 22:06 +0000 [r81827]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 81826 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1
	  line We added COPTS for ASTCFLAGS additions, but not LDOPTS for
	  ASTLDFLAGS. This adds LDOPTS ........

2007-09-06 21:01 +0000 [r81814]  Joshua Colp <jcolp@digium.com>

	* channels/iax2-parser.c: Initialize iax_frames variable to NULL,
	  keeps valgrind happy.

2007-09-06 20:54 +0000 [r81783-81813]  Russell Bryant <russell@digium.com>

	* CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE()
	  function that can retrieve the state of an extension that has a
	  hint. (closes issue #10635, adamgundy)

	* CHANGES: s/DEVSTATE/DEVICE_STATE/

	* funcs/func_devstate.c: Rename the DEVSTATE() function to
	  DEVICE_STATE() to better conform to how other functions are
	  named. (inspired by issue #10635)

	* CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function
	  from my sandbox branch into trunk. This function will let you
	  retrieve the list of devices or name associated with a hint.
	  (inspired by issue #10635)

2007-09-06 20:16 +0000 [r81782]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c, CHANGES: (closes issue #10377) Reported
	  by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak
	  (license 7) Add skinny show device, skinny show line, and skinny
	  show settings CLI commands.

2007-09-06 20:05 +0000 [r81781]  Russell Bryant <russell@digium.com>

	* configs/extensions.conf.sample: Fix the syntax of declaring a
	  hint with a name to be compatible with trunk

2007-09-06 20:00 +0000 [r81779]  Jason Parker <jparker@digium.com>

	* /, include/asterisk/astobj2.h: Merged revisions 81778 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2
	  lines This should fix a build issue that people building against
	  uClibc were seeing with the addition of astobj2 ........

2007-09-06 19:43 +0000 [r81777]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7
	  lines (closes issue #10122) Reported by: stevefeinstein Patches:
	  meetme-unmute-manager.diff uploaded by qwell (license 4) Tested
	  by: stevefeinstein After looking over the code I agree with
	  Qwell. Setting the file descriptor to conference each time just
	  causes a fight back and forth. ........

2007-09-06 17:00 +0000 [r81745]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged
	  revisions 81743 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007)
	  | 1 line Various string length fixes. Removed an unused variable
	  in aji_client structure (context) ........

2007-09-06 16:57 +0000 [r81744]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/safe_asterisk: Incorporate the ability to log
	  output of safe_asterisk to syslog (closes issue #9882)

2007-09-06 16:38 +0000 [r81742]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Patch on 10575. Add support for unequipped
	  CIC (UCIC) message as well as improve some of our CIC flags in
	  chan_zap

2007-09-06 16:31 +0000 [r81730]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep
	  2007) | 6 lines Fixes an issue where valid DTMF had to be pressed
	  twice to exit a queue if a member's phone was ringing. (closes
	  issue #10655, reported by strider2k, patched by me) ........

2007-09-06 15:43 +0000 [r81712]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/astobj2.h, main/astobj2.c: various changes to
	  the documentation, and redefinition of ao2_hash_fn and
	  ao2_callback_fn typedefs, in preparation to more cleanup of the
	  _search_flags Please do not merge this change to 1.4 yet - there
	  are no functional changes anyways.

2007-09-06 15:21 +0000 [r81683]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_features.c: Merged revisions 81682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep
	  2007) | 5 lines Fixes a memory leak (closes issue #10658,
	  reported and patched by Ivan) ........

2007-09-06 14:24 +0000 [r81651]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 81650 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007)
	  | 3 lines According to both RFC 3920 - section 9.1.2 - and
	  Google's XMPP server complaint, if set, the 'from' attribute must
	  be set to the user's full JID. ........

2007-09-05 21:59 +0000 [r81632]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Not having this epoll specific code in
	  wait_for_answer was causing app_queue to infinitely loop. This
	  makes it so it doesn't. Thanks to file for pointing out where the
	  problem was and showing a similar function in app_dial as an
	  example of how to fix it.

2007-09-05 21:45 +0000 [r81631]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05
	  Sep 2007) | 2 lines Solaris x86 compatibility fix ........

2007-09-05 20:58 +0000 [r81601]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller

2007-09-05 20:58 +0000 [r81600]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, /, main/say.c, res/res_features.c,
	  main/file.c, include/asterisk/channel.h: Merged revisions 81599
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) |
	  11 lines Fix an issue that can occur when you do an attended
	  transfer to parking. If you complete the transfer before the
	  announcement of the parking spot finishes, then the channel being
	  parked will hear the remainder of the announcement. These changes
	  make it so that will not happen anymore. Basically, res_features
	  sets a flag on the channel is playing the announcement to so that
	  the file streaming core knows that it needs to watch out for a
	  channel masquerade, and if it occurs, to abort the announcement.
	  (closes BE-182) ........

2007-09-05 16:48 +0000 [r81568]  Tilghman Lesher <tlesher@digium.com>

	* utils: Add two more generated files (requested by mvanbaak via
	  irc)

2007-09-05 16:31 +0000 [r81560]  Jason Parker <jparker@digium.com>

	* include/asterisk/devicestate.h, res/res_config_odbc.c,
	  channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c,
	  res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c,
	  include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c,
	  res/res_config_sqlite.c, include/asterisk/sha1.h,
	  include/asterisk/stringfields.h, include/asterisk/features.h:
	  Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy

2007-09-05 15:32 +0000 [r81526-81535]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Weird. When I merged my changes from 1.4, they
	  merged into the wrong function. This should fix the build for
	  trunk.

	* /, apps/app_queue.c: Merged revisions 81525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep
	  2007) | 4 lines Fixing the build... ........

2007-09-05 15:16 +0000 [r81524]  Jason Parker <jparker@digium.com>

	* channels/chan_phone.c, /: Merged revisions 81523 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500
	  (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL
	  channel tech. Also changed load_module function to use defines
	  rather than numbers for return values. Issue 10651, patch by
	  rbraun_proformatique, with additions by me. ........

2007-09-05 15:04 +0000 [r81522]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81520 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep
	  2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only
	  count members who are logged in and available. (related to issue
	  #10652, reported by wuwu) ........

2007-09-05 14:47 +0000 [r81519]  Steve Murphy <murf@digium.com>

	* include/asterisk/config.h, main/config.c: this set of changes
	  fixes issue # 10643 by keeping track of the last object defined
	  in a file, and attaching any accumulated comments to that object
	  (category header or variable declaration). The file_save routine
	  also had to be upgraded to output these trailing comments.
	  Config.h was modified to include the trailing comment list on
	  categories and variables.

2007-09-05 13:13 +0000 [r81459-81493]  Joshua Colp <jcolp@digium.com>

	* main/editline/sys.h: Finish up commit from revision 81452 by
	  removing last remnants of strlcat/strlcpy checks.

2007-09-04 20:59 +0000 [r81454-81456]  Jason Parker <jparker@digium.com>

	* /, apps/app_followme.c: Merged revisions 81455 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500
	  (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file,
	  we can just check whether it exists. Issue 10634, patch by me,
	  testing by pabelanger, sanity checked by bweschke ........

	* /, configs/followme.conf.sample: Merged revisions 81453 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500
	  (Tue, 04 Sep 2007) | 4 lines Change default followme config file
	  to point to the correct files. Issue 10644, patch by pabelanger
	  ........

2007-09-04 19:51 +0000 [r81445-81452]  Russell Bryant <russell@digium.com>

	* main/editline/configure, main/editline/configure.in: Don't check
	  for and include strlcpy and strlcat in editline. We also include
	  them directly in Asterisk. For platforms that need them (like my
	  mac), you will get a linker error due to the functions being
	  included twice.

	* /, include/asterisk/astobj2.h, channels/chan_iax2.c,
	  main/astobj2.c: Merged revisions 81448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) |
	  4 lines Remove the typedefs on ao2_container and ao2_iterator.
	  This is simply because we don't typedef objects anywhere else in
	  Asterisk, so we might as well make this follow the same
	  convention. ........

	* include/asterisk/logger.h: logger.h depends on options.h, so go
	  ahead and include it

2007-09-04 16:41 +0000 [r81443]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007)
	  | 2 lines there is no point in sending 401 Unauthorized to a UAS
	  that sent us a properly-formatted Authentication header with the
	  expected username and nonce but an incorrect response (which
	  indicates the shared secret does not match)... instead, let's
	  send 403 Forbidden so that the UAS doesn't retry with the same
	  authentication credentials repeatedly ........

2007-09-04 14:28 +0000 [r81436-81441]  Joshua Colp <jcolp@digium.com>

	* configs/extensions.ael.sample: (closes issue #10633) Reported by:
	  pabelanger Patches: extensions.ael.sample.patch uploaded by
	  pabelanger (license 224) Update extensions.ael.sample with
	  voicemail and | changes.

	* /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6
	  lines (closes issue #10632) Reported by: jamesgolovich Patches:
	  asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich
	  (license 176) Fix memory leak when unloading chan_iax2. The
	  firmware files were not being freed. ........

	* main/channel.c, /: Merged revisions 81437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4
	  lines (closes issue #10476) Reported by: mdu113 Only look for the
	  end of a digit when waiting for a digit. This in turn disables
	  emulation in the core. ........

	* /, main/dns.c: Merged revisions 81435 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7
	  lines (closes issue #10610) Reported by: john Patches:
	  dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
	  Don't return a match if no SRV record actually exists. ........

2007-09-03 18:59 +0000 [r81434]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) |
	  5 lines Remove a couple of calls to ast_string_field_free_pools()
	  on peers in error handling blocks in the code for building peers.
	  The peer object destructor does this and doing it twice will
	  cause a crash. (closes issue #10625, reported by and patched by
	  pnlarsson) ........

2007-09-03 18:01 +0000 [r81430-81432]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c: Once we get past the file checks, we're loading,
	  so clear the FILEUNCHANGED flag (fixes #include) (closes issue
	  #10629)

	* /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007)
	  | 2 lines The IF() function was not allowing true values that had
	  embedded colons (closes issue #10613) ........

	* main/config.c: We shouldn't use a filename blindly without
	  checking to make sure it's unused first

2007-09-01 06:03 +0000 [r81427]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
	  81426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep
	  2007) | 4 lines Making match_by_addr into ao2_match_by_addr and
	  making it available everywhere since it could be a handy callback
	  to have ........

2007-08-31 21:29 +0000 [r81419]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/astobj2.h: Merged revisions 81418 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) |
	  2 lines Remove references to a debugging parameter that does not
	  exist ........

2007-08-31 19:50 +0000 [r81417]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81416 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug
	  2007) | 6 lines Fixed broken behavior of a reload on realtime
	  queues. Prior to this patch, if a reload was issued and a
	  realtime queue had callers waiting in it, then the queue would be
	  removed from the queue list, but it would not actually be freed
	  (in fact, a debug message warning about a memory leak would come
	  up). With this patch, reloads do not touch realtime queues at
	  all. ........

2007-08-31 18:46 +0000 [r81413]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, /: Merged revisions 81412 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500
	  (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line
	  with the existing alpha order. Issue 10621, initial patch by
	  junky ........

2007-08-31 17:43 +0000 [r81411]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31
	  Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command
	  available. Closes issue 10548, reported by keepitcool. ........

2007-08-31 15:58 +0000 [r81408]  Kevin P. Fleming <kpfleming@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007)
	  | 2 lines add missing "transcoder show" (and deprecated "show
	  transcoder") CLI commands that were in 1.2 but never added to 1.4
	  ........

2007-08-31 15:54 +0000 [r81402-81407]  Joshua Colp <jcolp@digium.com>

	* /, res/res_speech.c: Merged revisions 81406 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2
	  lines Make it the engine's responsible to check for the presence
	  of results. ........

	* /, res/res_features.c: Merged revisions 81403 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4
	  lines (closes issue #10618) Reported by: dimas Don't pass through
	  the stopped sounds frame.... just drop it. ........

	* /, res/res_features.c: Merged revisions 81401 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4
	  lines (closes issue #10009) Reported by: dimas Don't output a
	  bridge failed warning message if it failed because one of the
	  channels was part of the masquerade process. That is perfectly
	  normal. ........

2007-08-30 23:52 +0000 [r81400]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Add new queryable fields from zaptel to 'zap
	  show status'

2007-08-30 22:08 +0000 [r81398]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug
	  2007) | 7 lines Removing an extraneous (and possibly misleading)
	  log message. Firstly, if the announce file isn't found, the
	  streaming functions will report it. Secondly, not all non-zero
	  returns from play_file mean that the announce file wasn't found.
	  Positive return values simply mean that a digit was pressed (most
	  likely to skip through the announcement). (closes issue #10612,
	  reported and patched by dimas) ........

2007-08-30 21:25 +0000 [r81394-81396]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6
	  lines (closes issue #10514) Reported by: casper Patches:
	  chan_sip.c.80129.diff uploaded by casper (license 55) Remove
	  needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it
	  to ever be that value. ........

	* channels/chan_sip.c: (closes issue #10565) Reported by: tootai
	  Make sure the external IP address has the standard SIP port set
	  for when the user does not specify the port in the externip
	  setting.

2007-08-30 21:16 +0000 [r81393]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 81392 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1
	  line via issue 10599, where 'CDR already initialized' messages
	  are being generated. Since all channels will have an init'd CDR
	  attached at creation time, this message is now particularly
	  useless. Removed. ........

2007-08-30 20:55 +0000 [r81391]  Joshua Colp <jcolp@digium.com>

	* apps/app_minivm.c: (closes issue #10336) Reported by: junky
	  Patches: minivm_output2.diff uploaded by junky (license 177)
	  Change console output of minivm show stats to be more simple for
	  external parsing.

2007-08-30 20:31 +0000 [r81389-81390]  Tilghman Lesher <tlesher@digium.com>

	* main/sched.c: A schedule id of 0 is not possible and is used to
	  flag that we want to add a new item

	* apps/app_readexten.c: Change wording as requested by Kevin

2007-08-30 18:52 +0000 [r81388]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Added note to sample queues.conf file
	  to line up with most recent change regarding setinterfacevar.
	  MEMBERREALTIME indicates whether a member is realtime.

2007-08-30 17:51 +0000 [r81387]  Tilghman Lesher <tlesher@digium.com>

	* main/logger.c: Always force reread of the config when we're
	  rotating the log file (closes issue #10598)

2007-08-30 15:40 +0000 [r81384]  Russell Bryant <russell@digium.com>

	* /, channels/h323/ast_h323.cxx: Merged revisions 81383 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) |
	  3 lines Add missing checks for the PTRACING define. (closes issue
	  #10559, paravoid) ........

2007-08-30 15:36 +0000 [r81382]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug
	  2007) | 3 lines Changed some manager event messages to reflect
	  whether a queue member is a realtime member or not ........

2007-08-30 15:34 +0000 [r81380]  Russell Bryant <russell@digium.com>

	* configs/modem.conf.sample (removed), /, configs/enum.conf.sample,
	  configs/extensions.ael.sample: Merged revisions 81379 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) |
	  3 lines Fix a typo, update a reload command, and remove an unused
	  configuration file. (closes issue #10606, casper) ........

2007-08-30 15:24 +0000 [r81378]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN
	  function (closes issue #10082)

2007-08-30 14:54 +0000 [r81376]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30
	  Aug 2007) | 1 line Fixed some warnings. ........

2007-08-30 14:42 +0000 [r81370-81372]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls
	  Patches: pbx.diff uploaded by jmls (license 141) Add REASON
	  dialplan variable for when an originated call fails and the
	  failed extension is executed.

	* /, res/res_features.c: Merged revisions 81369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4
	  lines (issue #10599) Reported by: dimas Handle the -1 control
	  subclass during feature dialing (it indicates to stop sounds).
	  ........

2007-08-30 08:50 +0000 [r81368]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 81367 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) |
	  11 lines Fixed a severe issue where a misdn_read would lock the
	  channel, but read would not return because it blocks. later
	  chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER
	  which could result in a deadlock situation. misdn_read will now
	  not block forever anymore, and we don't queue the ANSWER frame at
	  all when we already was called with misdn_answer -> answer would
	  be called twice. Also we don't explicitly send a RELEASE_COMPLETE
	  on receiption of a RELEASE anymore, because mISDN does that for
	  us, this resulted in a problem on some switches, which would
	  block our port after some calls for a short while. ........

2007-08-29 22:05 +0000 [r81365]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added the MEMBERREALTIME variable when using
	  setinterfacevar in queues.conf

2007-08-29 21:55 +0000 [r81364]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/event.h: Make the event header file work under
	  C++.

2007-08-29 21:30 +0000 [r81363]  Steve Murphy <murf@digium.com>

	* main/config.c: init newer so compile won't complain.

2007-08-29 21:25 +0000 [r81362]  Russell Bryant <russell@digium.com>

	* main/config.c: make trunk build again. murf will have to review
	  this to see if it was the right fix, as it is related to his last
	  change.

2007-08-29 20:55 +0000 [r81361]  Steve Murphy <murf@digium.com>

	* res/res_config_pgsql.c, channels/chan_sip.c,
	  include/asterisk/config.h, channels/chan_iax2.c,
	  channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c,
	  main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c,
	  main/manager.c, channels/chan_skinny.c, apps/app_minivm.c,
	  main/http.c, utils/extconf.c, apps/app_directory.c,
	  apps/app_parkandannounce.c, apps/app_voicemail.c: This code was
	  in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I
	  didn't add it to 1.4 yet, because it's not entirely clear to me
	  if this is a bug fix or an enhancement. A lot of files were
	  affected by small changes like ast_variable_new getting an added
	  arg, for the file name the var was defined in; ast_category_new
	  gets added args of filename and lineno; ast_category and
	  ast_variable structures now record file and lineno for each
	  entry; a list of all #include and #execs in a config file (or any
	  of its inclusions are now kept in the ast_config struct; at save
	  time, each entry is put back into its proper file of origin, in
	  order. #include and #exec directives are folded in properly.
	  Headers indicating that the file was generated, are generated
	  also for each included file. Some changes to main/manager.c to
	  take care of file renaming, via the UpdateConfig command.
	  Multiple inclusions of the same file are handled by exploding
	  these into multiple include files, uniquely named. There's
	  probably more, but I can't remember it right now.

2007-08-29 19:41 +0000 [r81353-81356]  Russell Bryant <russell@digium.com>

	* main/event.c: Try to clarify the rules on changing ast_event and
	  ast_event_ie

	* main/event.c: Fix parenthesis from my last commit

	* main/event.c: Change pointer aritmetic on void * to char *

	* main/event.c: there is not actually code that sends these over
	  the network in trunk yet

2007-08-29 16:39 +0000 [r81350]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug
	  2007) | 12 lines This patch, in essence, will correctly pause a
	  realtime queue member and reflect those changes in the realtime
	  engine. (issue #10424, reported by irroot, patch by me) This
	  patch creates a new function called update_realtime_member_field,
	  which is a generic function which will allow any one field of a
	  realtime queue member to be updated. This patch only uses this
	  function to update the paused status of a queue member, but it
	  lays the foundation for persisting the state of a realtime member
	  the same way that static members' state is maintained when using
	  the persistentmembers setting ........

2007-08-29 16:25 +0000 [r81348]  Joshua Colp <jcolp@digium.com>

	* main/event.c: Return ast_event_get_ie_raw to using an iterator
	  and fix logic in ast_event_iterator_next.

2007-08-29 16:09 +0000 [r81347]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug
	  2007) | 3 lines Changed some tabs to spaces ........

2007-08-29 16:07 +0000 [r81344-81345]  Joshua Colp <jcolp@digium.com>

	* main/event.c: This concludes bringing trunk back to a working
	  state.

	* include/asterisk/event.h, main/event.c: To keep others happy...
	  revert part of my additions so trunk works.

2007-08-29 15:59 +0000 [r81343]  Russell Bryant <russell@digium.com>

	* /, main/Makefile: Merged revisions 81342 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) |
	  3 lines If chan_h323 is not being built, don't use g++ to do the
	  final link of Asterisk. (in response to a question on the
	  asterisk-dev list) ........

2007-08-29 15:57 +0000 [r81341]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81340 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug
	  2007) | 8 lines This fix creates a more accurate way of detecting
	  whether realtime members were deleted. (closes issue 10541,
	  reported by Alric, patched by me) The REALLY nice things about
	  this patch is that queue members now have a "realtime" field
	  which will be true if the member is a realtime member. This means
	  we can check this value prior to certain processing if it should
	  ONLY be done for realtime members. ........

2007-08-29 15:21 +0000 [r81335]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Changed one too many variable settings in
	  issue #9315 (closes issue #10592)

2007-08-29 15:19 +0000 [r81334]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/event.h, include/asterisk/event_defs.h,
	  main/event.c: Add API calls for iterating through an event. This
	  should allow events to have multiple information elements (while
	  there was nothing preventing it before you could not actually
	  access any except the first one).

2007-08-29 14:19 +0000 [r81333]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue
	  #10591, reported and patched by junky, with small modification by
	  me)

2007-08-29 14:16 +0000 [r81326-81332]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4
	  lines (closes issue #9690) Reported by: mattv Make rtp timeouts
	  work even if two RTP streams are directly bridged in the RTP
	  stack. ........

	* include/asterisk/utils.h: Add inline function for signed linear
	  subtraction.

2007-08-28 21:39 +0000 [r81292]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) |
	  3 lines Change the message about receiving a mini-frame before
	  the first full voice frame to a DEBUG message. ........

2007-08-28 21:35 +0000 [r81290]  Joshua Colp <jcolp@digium.com>

	* main/logger.c: Add some read/write locking magic to make logger
	  reload operate again.

2007-08-28 20:03 +0000 [r81277]  Tilghman Lesher <tlesher@digium.com>

	* main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support
	  better rotation of log files to be more like system logging
	  (closes issue #10398)

2007-08-28 19:12 +0000 [r81227-81264]  Russell Bryant <russell@digium.com>

	* include/asterisk/audiohook.h: Change the audiohook lock and
	  unlock wrappers to macros instead of inline functions. As inline
	  functions, the lock debug information will show that these are
	  always locked in audiohooks.h instead of the file where the lock
	  was actually acquired.

	* funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking
	  around the uses of datastore_add and _find. There are still more
	  places in the tree that I have not yet changed if someone wants
	  to go through and find the places they are used without the
	  channel locked.

	* main/channel.c, funcs/func_volume.c, include/asterisk/channel.h:
	  * Constify the uid field of channel datastores * Convert some
	  spaces to tabs in func_volume * Add a note in channel.h making it
	  clear that none of the datastore API calls lock the channel they
	  are given, so the channel should be locked before calling the
	  functions that take a channel argument.

	* include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c,
	  doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by:
	  nic_bellamy Patches:
	  2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by
	  nic_bellamy (license 213) Add support for configurable file
	  locking methods. The default is "lockfile", which is the old
	  behavior. There is an additional option, "flock", which is
	  intended for use in situations where the lockfile method will not
	  work, such as with SMB/CIFS mounts.

	* /, configs/indications.conf.sample: Merged revisions 81226 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) |
	  2 lines Add Russian tones. (closes issue #7953, hanabana)
	  ........

2007-08-28 14:37 +0000 [r81210]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10579) Reported by: ornati
	  Make sure the called channel during the attended transfer process
	  becomes associated with the calling channel so that the
	  ast_waitfor_* call works properly under epoll.

2007-08-28 14:12 +0000 [r81121-81190]  Mark Michelson <mmichelson@digium.com>

	* /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28
	  Aug 2007) | 5 lines Fixes a forwarding problem when using
	  res_config_mysql (closes issue #10573, reported by chrisvaughan,
	  patch suggested by chrisvaughan as well) ........

	* /, apps/app_queue.c: Merged revisions 81158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug
	  2007) | 5 lines Resolve a potential deadlock. In this case, a
	  single queue is locked, then the queue list. In changethread(),
	  the queue list is locked, and then each individual queue is
	  locked. Under the right circumstances, this could deadlock. As
	  such, I have unlocked the individual queue before locking the
	  queue list, and then locked the queue back after the queue list
	  is unlocked. ........

	* /, channels/chan_agent.c: Merged revisions 81120 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27
	  Aug 2007) | 7 lines DTMF begin frames should be ignored so that
	  when an agent acks a call with the '#' key, he doesn't cause a
	  queue's announce file to be interrupted. Also went ahead and did
	  the same for the '*' key and for ending a call. (closes issue
	  #10528, reported by deskhack, patched by me) ........

2007-08-27 20:55 +0000 [r81118]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_directed_pickup.c: Enhance Pickup to do native
	  pickupgroup pickup when no arguments are specified (closes issue
	  #10404)

2007-08-27 17:44 +0000 [r81043-81098]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh
	  well ... it's harmless. Merged revisions 81065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) |
	  1 line explicity define a variable as a boolean ........

	* /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) |
	  3 lines Add a \todo to note that this module leaks most of the
	  memory it allocates on unload and should be fixed (when I'm not
	  in the middle of something else ...). ........

	* /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27
	  Aug 2007) | 11 lines (closes issue #10419) Reported by:
	  mustardman Patches: asterisk-mohposition.diff.txt uploaded by
	  jamesgolovich (license 176) This patch fixes a few problems with
	  music on hold. * Fix issues with starting at the beginning of a
	  file when it shouldn't. * Fix the inuse counter to be decremented
	  even if the class had not been set to be deleted when not in use
	  anymore * Don't arbitrarily limit the number of MOH files to 255
	  ........

2007-08-27 15:03 +0000 [r81013]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6
	  lines (closes issue #10561) Reported by: jesselang Patches:
	  chan_sip-ChannelReload-20080825.patch uploaded by jesselang
	  (license 202) Remove an extra \r\n to make the ChannelReload
	  event conform with every other event. ........

2007-08-27 14:56 +0000 [r81011]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 81010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug
	  2007) | 3 lines Found a case where the queue's membercount is
	  off. It does not take into account dynamic members on a reload.
	  ........

2007-08-27 13:35 +0000 [r80962-80991]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove places that say if no language is
	  specified it will default to english... since on some setups this
	  is untrue.

	* /, main/rtp.c: Merged revisions 80974 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4
	  lines (closes issue #10562) Reported by: idkpmiller Correct
	  jitter value output in the CLI to be as expected. ........

	* configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG
	  Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20)
	  Fix up sip.conf sample configuration.

2007-08-26 18:12 +0000 [r80933]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) |
	  3 lines Remove an extra signal_condition() for the scheduler
	  thread. (closes issue #10564, patch from casper) ........

2007-08-25 17:55 +0000 [r80821-80898]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) |
	  7 lines Fix some issues with the handling of the scheduler in
	  chan_iax2. Most of the places that scheduled items to be executed
	  by the scheduler thread did not signal the scheduler thread to
	  wake up so that it could recalculate the time until the next
	  action. These changes will make the scheduler thread more
	  responsive and ensure that actions get executed as close to when
	  intended as possible instead of it being possible for very long
	  delays. ........

	* pbx/pbx_dundi.c: localize a variable and remove a duplicate error
	  message

	* apps/app_queue.c: use ast_strlen_zero

	* /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) |
	  5 lines If dnsmgr is in use, and no DNS servers are available
	  when Asterisk first starts, then don't give up on poking peers.
	  Allow the poke to get rescheduled so that it will work once the
	  dnsmgr is able to resolve the host. (closes issue #10521, patch
	  by jamesgolovich) ........

	* /, main/dsp.c: Merged revisions 80820 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) |
	  7 lines Improve the debouncing logic in the DTMF detector to fix
	  some reliability issues. Previously, this code used a shift
	  register of hits and non-hits. However, if the start of the digit
	  isn't clean, it is possible for the leading edge detector to miss
	  the digit. These changes replace the flawed shift register logic
	  and also does the debouncing on the trailing edge as well.
	  (closes issue #10535, many thanks to softins for the patch)
	  ........

2007-08-24 20:21 +0000 [r80819]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Merged revisions 80818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007)
	  | 3 lines A minor correction to the available logic of autofill.
	  If a queue member is paused, they're not really "available" so
	  don't count them as such. Somewhat related to issue #10155
	  ........

2007-08-24 19:50 +0000 [r80817]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Fix documentation for Set (closes issue #10549)

2007-08-24 19:03 +0000 [r80790]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 80789 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1
	  line From a complaint by jmls, I realize that the message in
	  cdr_disposition is unnecessary. To get failure disposition, just
	  return -1; no use having more than one case do that. ........

2007-08-24 18:05 +0000 [r80778]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add VMWI chan_zap support #9909

2007-08-24 15:53 +0000 [r80751]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug
	  2007) | 3 lines Fix a possible crash in IMAP voicemail. ........

2007-08-24 15:42 +0000 [r80748]  Steve Murphy <murf@digium.com>

	* utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well

2007-08-24 15:29 +0000 [r80725]  Russell Bryant <russell@digium.com>

	* /, utils/ael_main.c: Merged revisions 80722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) |
	  3 lines Tweak the formatting of this MODULEINFO block. I think
	  this would have caused a "*" to get in the menuselect-tree file.
	  ........

2007-08-24 14:55 +0000 [r80690-80718]  Steve Murphy <murf@digium.com>

	* /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80717 | murf | 2007-08-24 08:48:49 -0600 (Fri, 24 Aug 2007) | 1
	  line This change addresses JerJer's complaint that aelparse
	  builds and installs even if pbx_ael is unchecked in the
	  menuselect stuff. ........

2007-08-24 11:49 +0000 [r80662]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: Merged revisions 80661 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24
	  Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are
	  answered too early, which results in CDRs wrongly stating that a
	  call was ANSWERED when the calling party cancelled a call before
	  before being established. We must not answer the call upon
	  reception of a 'transport-accept' iq packet, but this packet
	  still needs to be acknowledged, otherwise the remote peer would
	  close the call (like in #8970). ........

2007-08-23 23:37 +0000 [r80649]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
	  res/ael/ael.y, res/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-test7: an unreported crash I debugged,
	  looked like it was backing up way too far after hitting the
	  syntax error. An inspection of the code revealed that error
	  tokens in lists were not rearranged when the rules were
	  rearranged as part of a code neatening-up process. By moving the
	  error tokens to where they should be, I also reduced the number
	  of shift/reduce conflicts to 3 instead of 8. This introduces
	  subtle differences in error messages, so the regressions had to
	  be updated.

2007-08-23 21:34 +0000 [r80510-80616]  Russell Bryant <russell@digium.com>

	* apps/app_while.c: Use the comma separator in app_while. reported
	  by blitzrage on irc, patched by me

	* /, res/res_features.c, include/asterisk/features.h: Merged
	  revisions 80573 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) |
	  5 lines When executing a dynamic feature, don't look it up a
	  second time by digit pattern after we already looked it up by
	  name. This causes broken behavior if there is more than one
	  feature defined with the same digit pattern. (closes issue
	  #10539, reported by bungalow, patch by me) ........

	* /, funcs/func_timeout.c: Merged revisions 80547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80547 | russell | 2007-08-23 14:29:44 -0500 (Thu, 23 Aug 2007) |
	  3 lines Revert very broken fix for issue #10540 ... none of these
	  values take ms so I don't know what I was thinking ........

	* /, funcs/func_timeout.c: Merged revisions 80539 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80539 | russell | 2007-08-23 14:21:53 -0500 (Thu, 23 Aug 2007) |
	  4 lines Fix func_timeout to take values in floating point so 1.5
	  actually means 1.5 seconds instead of being rounded. (closes
	  issue #10540, reported by spendergrass, patch by me) ........

	* doc/asterisk-mib.txt, res/snmp/agent.c: Fix a typo in the
	  Asterisk MIB and fix astNumChanBridged so it acts as a counter
	  again (closes issue #10118, patch by jeffg)

2007-08-23 17:18 +0000 [r80508]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 80501 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80501 | kpfleming | 2007-08-23 12:08:25 -0500 (Thu, 23 Aug 2007)
	  | 2 lines report the actual channel number that was unregistered,
	  instead of assuming that the interface list consists of channels
	  1 through <x> with no gaps in the sequence ........

2007-08-23 17:04 +0000 [r80470-80500]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 80499 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80499 | russell | 2007-08-23 12:02:50 -0500 (Thu, 23 Aug 2007) |
	  3 lines Fix some code where it was possible for a reference to a
	  peer to not get released when it should. Thank you to Marta
	  Carbone for pointing this out! ........

	* /, res/res_agi.c: Merged revisions 80469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80469 | russell | 2007-08-23 10:49:28 -0500 (Thu, 23 Aug 2007) |
	  2 lines Revert res_agi fix that didn't quite work until we get it
	  right ... ........

2007-08-23 15:48 +0000 [r80453-80468]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If no default language has been specified
	  print out that it will default to english when using sip show
	  peer or sip show user.

	* main/minimime/mm.h: Return trunk to a working state by including
	  compat.h in minimime.

2007-08-22 23:26 +0000 [r80428-80429]  Jason Parker <jparker@digium.com>

	* main/minimime/mm_util.c, main/minimime/mm_codecs.c,
	  main/minimime/mm_mem.h, main/minimime/mm_base64.c,
	  main/minimime/mm.h: Convert minimime to use the proper uint*_t
	  types, rather than u_int*_t

	* apps/app_minivm.c: Cast calls to getpid. This was done in 1.4
	  already, this one was just new

2007-08-22 22:54 +0000 [r80361-80427]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/astobj2.h: Merged revisions 80426 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) |
	  6 lines Add some more documentation on iterating ao2 containers.
	  The documentation implies that is possible to miss an object or
	  see an object twice while iterating. After looking through the
	  code and talking with mmichelson, I have documented the exact
	  conditions under which this can happen (which are rare and
	  harmless in most cases). ........

	* /, main/astobj2.c: Merged revisions 80424 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) |
	  10 lines When converting this code to use the list macros, I
	  changed it so objects are added to the head of a bucket instead
	  of the tail. However, while looking over code with mmichelson, we
	  noticed that the algorithm used in ao2_iterator_next requires
	  that items are added to the tail. This wouldn't have caused any
	  huge problem, but it wasn't correct. It meant that if an object
	  was added to a container while you were iterating it, and it was
	  added to the same bucket that the current element is in, then the
	  new object would be returned by ao2_iterator_next, and any other
	  objects in the bucket would be bypassed in the traversal.
	  ........

	* channels/chan_iax2.c: allow peers and users to go into a hash
	  table

	* /, channels/chan_sip.c: Merged revisions 80390 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80390 | russell | 2007-08-22 16:00:44 -0500 (Wed, 22 Aug 2007) |
	  3 lines Don't crash when using realtime in chan_sip without an
	  insecure setting in the database. (closes issue #10348, reported
	  by link55, fixed by me) ........

	* channels/chan_iax2.c: Unsubscribe from MWI events in the peer
	  destructor

	* /, main/Makefile, include/asterisk/astobj2.h (added),
	  include/asterisk/strings.h, channels/chan_iax2.c, main/astobj2.c
	  (added): Merged revisions 80362 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) |
	  34 lines Merge changes from team/russell/iax_refcount. This set
	  of changes fixes problems with the handling of iax2_user and
	  iax2_peer objects. It was very possible for a thread to still
	  hold a reference to one of these objects while a reload operation
	  tries to delete them. The fix here is to ensure that all
	  references to these objects are tracked so that they can't go
	  away while still in use. To accomplish this, I used the astobj2
	  reference counted object model. This code has been in one of
	  Luigi Rizzo's branches for a long time and was primarily
	  developed by one of his students, Marta Carbone. I wanted to go
	  ahead and bring this in to 1.4 because there are other problems
	  similar to the ones fixed by these changes, so we might as well
	  go ahead and use the new astobj if we're going to go through all
	  of the work necessary to fix the problems. As a nice side benefit
	  of these changes, peer and user handling got more efficient.
	  Using astobj2 lets us not hold the container lock for peers or
	  users nearly as long while iterating. Also, by changing a define
	  at the top of chan_iax2.c, the objects will be distributed in a
	  hash table, drastically increasing lookup speed in these
	  containers, which will have a very big impact on systems that
	  have a large number of users or peers. The use of the hash table
	  will be made the default in trunk. It is not the default in 1.4
	  because it changes the behavior slightly. Previously, since peers
	  and users were stored in memory in the same order they were
	  specified in the configuration file, you could influence peer and
	  user matching order based on the order they are specified in the
	  configuration. The hash table does not guarantee any order in the
	  container, so this behavior will be going away. It just means
	  that you have to be a little more careful ensuring that peers and
	  users are matched explicitly and not forcing chan_iax2 to have to
	  guess which user is the right one based on secret, host, and
	  access list settings, instead of simply using the username. If
	  you have any questions, feel free to ask on the asterisk-dev
	  list. ........

	* /, res/res_agi.c: Merged revisions 80360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) |
	  5 lines Juggie in #asterisk-dev was reporting problems where
	  fgets would return without reading the whole line when using
	  fastagi. When this happens, errno was set to EINTR or EAGAIN.
	  This patch accounts for the possibility and lets fgets continue
	  in that case. ........

2007-08-22 18:54 +0000 [r80303-80331]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/mkpkgconfig, /, build_tools/make_build_h,
	  build_tools/strip_nonapi, build_tools/prep_moduledeps,
	  build_tools/make_buildopts_h: Merged revisions 80330 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r80330 | qwell | 2007-08-22 13:53:18 -0500 (Wed, 22 Aug
	  2007) | 7 lines Fix a few build issues in Solaris (and likely
	  others). Use GREP and ID variables from autoconf. Reported to me
	  in #asterisk-dev I forgot who reported this - sorry. :( ........

	* Makefile, /: Merged revisions 80304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80304 | qwell | 2007-08-22 13:25:34 -0500 (Wed, 22 Aug 2007) | 2
	  lines Change a syntax that the GNU make in Solaris dislikes.
	  ........

	* /, build_tools/make_version: Merged revisions 80302 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r80302 | qwell | 2007-08-22 13:06:00 -0500 (Wed, 22 Aug
	  2007) | 3 lines Fix a bashism (we explicitly request /bin/sh).
	  Remove some oddly placed quotes I found in passing. ........

2007-08-22 16:27 +0000 [r80258-80262]  Russell Bryant <russell@digium.com>

	* utils/check_expr.c: Ensure that the object code for
	  ast_atomic_fetchadd_int() gets included in the check_expr binary
	  when building with LOW_MEMORY defined. (reported by Brian Capouch
	  on the asterisk-dev list, patch by me)

	* Makefile, /: Merged revisions 80257 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) |
	  4 lines Honor the contents of the COPTS variable as custom target
	  CFLAGS. Apparently this is what openwrt does. (reported by Brian
	  Capouch on the asterisk-dev list, patch by me) ........

2007-08-22 16:16 +0000 [r80256]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 80255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4
	  lines (closes issue #10526) Reported by: sinistermidget Revert
	  commit from issue #10355 and return timestamp skew to 640.
	  ........

2007-08-22 14:17 +0000 [r80241-80242]  Steve Murphy <murf@digium.com>

	* /: blocking 80167

	* /, main/alaw.c: Merged revisions 80166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1
	  line This patch solves problem 1 in 8126; it should not slow down
	  the alaw codec, but should prevent signal degradation via
	  multiple trips thru the codec. Fossil estimates the twice thru
	  this codec will prevent fax from working. 4-6 times thru would
	  result hearable, noticeable, voice degradation. ........

2007-08-21 21:58 +0000 [r80226]  Russell Bryant <russell@digium.com>

	* funcs/func_odbc.c: use ast_atomic_fetchadd_int for incrementing
	  resultcount

2007-08-21 20:55 +0000 [r80217]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c: As per 10472, mvanbaak thought the generated code
	  would look better this way.

2007-08-21 18:49 +0000 [r80184]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 80183 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) |
	  7 lines Don't record SIP dialog history if it's not turned on.
	  Also, put an upper limit on how many history entires will be
	  stored for each SIP dialog. It is currently set to 50, but can be
	  increased if deemed necessary. (closes issue #10421, closes issue
	  #10418, patches suggested by jmoldenhauer, patches updated by me)
	  (Security implications documented in AST-2007-020) ........

2007-08-21 15:51 +0000 [r80157]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c: Minor tweak. Don't manipulate volume of the
	  audio in the buffer if no audio is actually there.

2007-08-21 15:23 +0000 [r80133]  Russell Bryant <russell@digium.com>

	* /, channels/chan_mgcp.c: Merged revisions 80132 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) |
	  3 lines Don't try to dereference the owner channel when it may
	  not exist (issue #10507, maxper) ........

2007-08-21 15:04 +0000 [r80131]  Jason Parker <jparker@digium.com>

	* /, configs/cdr.conf.sample: Merged revisions 80130 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug
	  2007) | 7 lines (closes issue #10510) Reported by: casper
	  Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
	  errors in sample cdr config file. ........

2007-08-20 22:53 +0000 [r80113]  Steve Murphy <murf@digium.com>

	* build_tools/cflags.xml, main/ulaw.c, codecs/slin_ulaw_ex.h,
	  codecs/ulaw_slin_ex.h, include/asterisk/alaw.h, main/translate.c,
	  include/asterisk/ulaw.h, main/alaw.c: This change set fixes bug
	  8126 in trunk. It is implemented via compile time options,
	  activated via the menuselect stuff, which defaults to the old
	  way. non-zero sample data added. Translate tables expressed in
	  microseconds instead of milliseconds, with 5-digit data now
	  instead of 3, giving 2 more digits of precision.

2007-08-20 17:37 +0000 [r80075]  Steve Murphy <murf@digium.com>

	* include/asterisk/lock.h, utils/extconf.c: Stephn Davies reports
	  that this will help make things work on 64-bit machines

2007-08-20 16:18 +0000 [r80050]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 80049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80049 | mmichelson | 2007-08-20 11:17:43 -0500 (Mon, 20 Aug
	  2007) | 4 lines Found a pointless ternary if. member->dynamic was
	  set to 1 and has no opportunity to change between then and this
	  line, so "dynamic" will ALWAYS be output. ........

2007-08-20 16:12 +0000 [r80048]  Jason Parker <jparker@digium.com>

	* /, configs/extensions.conf.sample: Merged revisions 80047 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7
	  lines (closes issue #10499) Reported by: casper Patches:
	  extensions.conf.sample.diff uploaded by casper (license 55)
	  Update CLI examples in extensions.conf.sample to reflect command
	  changes. ........

2007-08-20 15:53 +0000 [r80046]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Remove remnants of last commit so trunk
	  builds again.

2007-08-20 15:37 +0000 [r80045]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 80044 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r80044 | mmichelson | 2007-08-20 10:34:43 -0500 (Mon, 20 Aug
	  2007) | 5 lines Ukrainian language voicemail support. (closes
	  issue #10458, reported and patched by Oleh) ........

2007-08-20 15:27 +0000 [r80037]  Steve Murphy <murf@digium.com>

	* utils/pval.c (removed): pval.c should not be in svn, in the utils
	  dir

2007-08-20 15:10 +0000 [r80023-80033]  Joshua Colp <jcolp@digium.com>

	* utils/pval.c: Bring pval.c in utils up to date with pval.c in
	  res/ael.

	* channels/chan_zap.c: Fix random segfault issue when loading
	  chan_zap. Trying to access a configuration structure that has
	  already been destroyed is bad, mmmk?

2007-08-20 02:46 +0000 [r79999]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 79998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79998 | tilghman | 2007-08-19 21:42:49 -0500 (Sun, 19 Aug 2007)
	  | 2 lines Missing curly braces. Oops. (Reported by snuffy via
	  IRC) ........

2007-08-20 00:54 +0000 [r79988-79990]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: (closes issue #10495) Reported by:
	  stevedavies Make sure context pointer is valid or else chan_iax2
	  will go kaboom.

	* utils/Makefile: (closes issue #10496) Reported by: caio1982 Fix
	  building on OSX.

	* channels/chan_h323.c: Fix building of trunk. I'm doing work on a
	  Sunday night just to avoid watching Snakes on a Plane which my
	  roommate is watching.

2007-08-19 14:17 +0000 [r79980]  Tilghman Lesher <tlesher@digium.com>

	* utils/Makefile: Add strcompat dependency for check_expr (needed
	  for platforms that don't have strndup)

2007-08-18 23:58 +0000 [r79972]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Actually check the return value of
	  epoll_create to make sure it works.

2007-08-18 14:34 +0000 [r79940-79949]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 79947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79947 | tilghman | 2007-08-18 09:30:44 -0500 (Sat, 18 Aug 2007)
	  | 3 lines Don't allocate vmu for messagecount when we could just
	  use the stack instead (closes issue #10490) Also, remove a
	  useless (and leaky) SQLAllocHandle (closes issue #10480) ........

	* channels/chan_zap.c, channels/chan_sip.c, channels/chan_h323.c,
	  channels/chan_iax2.c: We weren't properly encapsulating the mtime
	  ignores of config files (closes issue #10488)

2007-08-17 21:19 +0000 [r79915]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: I broke the build. Now I'm fixing it.

2007-08-17 21:04 +0000 [r79913]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 79912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) |
	  4 lines Avoid a crash in the handling of DTMF based Caller ID. It
	  is valid for ast_read to return NULL in the case that the channel
	  has been hung up. (crash reported by anonymouz666 on IRC in
	  #asterisk-dev) ........

2007-08-17 19:16 +0000 [r79907]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 79906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug
	  2007) | 6 lines Patch allows for more seamless transition from
	  file storage voicemail to ODBC storage voicemail. If a retrieval
	  of a greeting from the database fails, but the file is found on
	  the file system, then we go ahead an insert the greeting into the
	  database. The result of this is that people who switch from file
	  storage to ODBC storage do not need to rerecord their voicemail
	  greetings. ........

2007-08-17 19:13 +0000 [r79903-79905]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
	  Merged revisions 79904 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10430) ........ r79904 | qwell | 2007-08-17 14:12:19 -0500
	  (Fri, 17 Aug 2007) | 11 lines Don't send a semicolon over the
	  wire in sip notify messages. Caused by fix for issue 9938. I
	  basically took the code that existed before 9938 was fixed, and
	  copied it into a new function - ast_unescape_semicolon There
	  should be very few places this will be needed (pbx_config does
	  NOT need this (see issue 9938 for details)) Issue 10430, patch by
	  me, with help/ideas from murf (thanks murf). ........

	* channels/chan_local.c, /: Merged revisions 79902 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500
	  (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name
	  and number. Issue 10485, reported by and fix explained by
	  paradise. ........

2007-08-17 16:39 +0000 [r79901]  Tilghman Lesher <tlesher@digium.com>

	* configs/logger.conf.sample: Documentation for %q in logger.conf,
	  as suggested by jtodd (closes issue #10475)

2007-08-17 16:04 +0000 [r79888-79894]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Fix Dial arguments in res_features. Closes
	  issue #10484, patch by lunn.

	* pbx/pbx_dundi.c: Correct the argument separator for a Dial
	  statement in pbx_dundi. Closes issue #10483, patch by lunn

2007-08-17 14:41 +0000 [r79885]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c: Change this flag... might not otherwise unlock in
	  an OOM situation

2007-08-17 14:14 +0000 [r79861-79862]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Make use of ast_sched_replace() in some
	  places in chan_iax2

	* channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: This
	  commit adds a scheduler API call, ast_sched_replace that can be
	  used in place of a very common construct. I also used it in a
	  number of places in chan_sip. if (id > -1) ast_sched_del(sched,
	  id); id = ast_sched_add(sched, ...); changes to:
	  ast_sched_replace(id, sched, ...);

2007-08-17 13:45 +0000 [r79859-79860]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c, res/res_config_sqlite.c: store and destroy
	  implementations for sqlite (closes issue #10446) and odbc (closes
	  issue #10447)

	* res/res_config_pgsql.c, funcs/func_lock.c: store and destroy
	  implementations for realtime pgsql (closes issue #10372)

2007-08-17 13:39 +0000 [r79858]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 79857 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) |
	  5 lines Fix some crashes in chan_sip. This patch changes various
	  places that add items to the scheduler to ensure that they don't
	  overwrite the ID of a previously scheduled item. If there is one,
	  it should be removed. (closes issue #10391, closes issue #10256,
	  probably others, patch by me) ........

2007-08-17 08:29 +0000 [r79841]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 79833 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17
	  Aug 2007) | 1 line sometimes we don't need to signal dtmf tones
	  to asterisk, we just want them to go through as inband. Otherwise
	  they might be generated by the other channel partner and then
	  there is a double tone. ........

2007-08-17 01:19 +0000 [r79824]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Fix building of chan_zap under development
	  mode without libpri and libss7 installed.

2007-08-16 23:31 +0000 [r79813]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_lock.c: Revise dialplan locks to permit multiple locks
	  per channel, but with deadlock avoidance

2007-08-16 22:33 +0000 [r79764-79794]  Russell Bryant <russell@digium.com>

	* /: Merged revisions 79792 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79792 | russell | 2007-08-16 17:32:33 -0500 (Thu, 16 Aug 2007) |
	  4 lines Fix a little race condition that could cause a crash if
	  two channels had MOH stopped at the same time that were using a
	  class that had been marked for deletion when its use count hits
	  zero. ........

	* /, res/res_musiconhold.c: Merged revisions 79778 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16
	  Aug 2007) | 14 lines This patch fixes a bug where reloading the
	  module with "module reload" did not delete classes from memory
	  that were no longer in the config. This patch fixes that problem
	  as well as another one. Previously, if you reloaded MOH using the
	  "moh reload" CLI command, which behaved differently than "module
	  reload ...", MOH had to be stopped on every channel and started
	  again immediately. However, there was no way to tell what class
	  was being used, so they would all fall back to the default class.
	  (closes issue #10139) Reported by: blitzrage Patches:
	  asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
	  (license 176) Tested by: jamesgolovich ........

	* /, channels/chan_iax2.c: Merged revisions 79756 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79756 | russell | 2007-08-16 16:29:24 -0500 (Thu, 16 Aug 2007) |
	  11 lines Fix more deadlocks in chan_iax2 that were introduced by
	  making frame handling and scheduling multi-threaded.
	  Unfortunately, we have to do some expensive deadlock avoidance
	  when queueing frames on to the ast_channel owner of the IAX2 pvt
	  struct. This was already handled for regular frames, but
	  ast_queue_hangup and ast_queue_control were still used directly.
	  Making these changes introduced even more places where the IAX2
	  pvt struct can disappear in the context of a function holding its
	  lock due to calling a function that has to unlock/lock it to
	  avoid deadlocks. I went through and fixed all of these places to
	  account for this possibility. (issue #10362, patch by me)
	  ........

2007-08-16 21:28 +0000 [r79755]  Joshua Colp <jcolp@digium.com>

	* /: Fix properties on trunk again.

2007-08-16 21:21 +0000 [r79749]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 79748 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r79748 | mmichelson | 2007-08-16 16:16:40 -0500 (Thu, 16
	  Aug 2007) | 8 lines Fixes a problem where agents would get stuck
	  busy due to their wrapuptime being longer than the queue's
	  wrapuptime and ringinuse=no for the queue. (closes issue #10215,
	  reported by Doug, repaired by me) Special thanks to fkasumovic
	  for pointing out the source of the problem and to bweschke for
	  helping to come up with a solution! ........

2007-08-16 21:09 +0000 [r79747]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, cdr/cdr_sqlite3_custom.c, /, res/res_features.c,
	  codecs/codec_adpcm.c, apps/app_alarmreceiver.c,
	  cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/config.c,
	  main/loader.c, res/res_smdi.c, channels/chan_skinny.c,
	  main/http.c, apps/app_amd.c, channels/chan_alsa.c,
	  cdr/cdr_odbc.c, cdr/cdr_manager.c, codecs/codec_g722.c,
	  apps/app_privacy.c, codecs/codec_speex.c, channels/chan_agent.c,
	  codecs/codec_g726.c, channels/iax2-provision.c,
	  apps/app_playback.c, channels/iax2-provision.h,
	  channels/chan_misdn.c, res/res_indications.c, pbx/pbx_config.c,
	  main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
	  channels/chan_vpb.cc, res/res_snmp.c, apps/app_meetme.c,
	  codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c,
	  cdr/cdr_pgsql.c, apps/app_followme.c, res/res_jabber.c,
	  cdr/cdr_radius.c, codecs/codec_zap.c, res/res_config_sqlite.c,
	  main/enum.c, channels/misdn_config.c, cdr/cdr_csv.c, main/cdr.c,
	  channels/chan_phone.c, res/res_config_odbc.c, main/manager.c,
	  apps/app_osplookup.c, funcs/func_odbc.c, apps/app_minivm.c,
	  main/logger.c, apps/app_directory.c, apps/app_rpt.c,
	  cdr/cdr_custom.c, channels/chan_mgcp.c, codecs/codec_lpc10.c,
	  res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
	  channels/chan_sip.c, apps/app_festival.c, codecs/codec_alaw.c,
	  res/res_adsi.c, include/asterisk/config.h, apps/app_queue.c,
	  channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c,
	  channels/chan_jingle.c, channels/misdn/chan_misdn_config.h,
	  channels/chan_h323.c, pbx/pbx_dundi.c, codecs/codec_ulaw.c: Don't
	  reload a configuration file if nothing has changed.

2007-08-16 19:40 +0000 [r79736]  Steve Murphy <murf@digium.com>

	* utils/pval.c, utils/conf2ael.c: Many thanks to mvanbaak for his
	  update to translate hints; I added the -d option for local
	  testing purposes. This is from bug 10472

2007-08-16 18:23 +0000 [r79724-79725]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_iax2.c: added counter for iax2 show registry CLI
	  output, closes issue 10461, thanks junky

	* apps/app_voicemail.c: added counter for voicemail show users,
	  issue 10462, thanks junky

2007-08-16 17:34 +0000 [r79714-79719]  Steve Murphy <murf@digium.com>

	* utils/conf2ael.c: mvanbaak asks: why did you include that twice?
	  Answer: dunno. removed redundant include

	* utils/extconf.c, utils/conf2ael.c: svn did me dirty for some
	  reason. Left 5 files out of the commit; Tilghman copied them in
	  from the branch, but I had made changes to these. Here they are.

2007-08-16 15:59 +0000 [r79691]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 79690 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79690 | mmichelson | 2007-08-16 10:58:34 -0500 (Thu, 16 Aug
	  2007) | 5 lines base_encode is not trying to open a log file, so
	  we should not call it a log file in the warning. (related to
	  issue #10452, reported by bcnit) ........

2007-08-16 15:29 +0000 [r79687-79688]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: (closes issue #10467) Reported by: lunn Patches:
	  pbx_dundi.diff uploaded by lunn (license 179) Don't print a
	  warning saying an ethernet interface was found when it indeed
	  was.

	* utils/conf2ael.c: Make conf2ael build on 64-bit systems.

2007-08-16 09:45 +0000 [r79666]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: Merged revisions 79665 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007)
	  | 21 lines A fix for two critical problems detected while working
	  with Daniel McKeehan in issue #10184. Upon priority change, the
	  resource list is not NULL terminated when moving an item to the
	  end of the list. This makes Asterisk endlessy loop whenever it
	  needs to read the list. Jids with different resource and priority
	  values, like in Gmail's and GoogleTalk's jabber clients put that
	  problem in evidence. Upon reception of a 'from' attribute with an
	  empty resource string, Asterisk crashes when trying to access the
	  found->cap pointer if the resource list for the given buddy is
	  not empty. This situation is perfectly valid and must be handled.
	  The Gizmoproject's jabber client put that problem in evidence.
	  Also added a few comments in the code as well as a handle for the
	  capabilities from Gmail's jabber client, which are stored in a
	  caps:c tag rather than the usual c tag. Closes issue #10184.
	  ........

2007-08-16 09:22 +0000 [r79660]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/ie.c: Merged revisions 79642 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79642 | crichter | 2007-08-16 10:21:21 +0200 (Do, 16 Aug 2007) |
	  1 line 0x80 + protocol is wrong for USERUSER when we want to send
	  IA5 Chars. ........

2007-08-16 06:52 +0000 [r79638]  Olle Johansson <oej@edvina.net>

	* CHANGES: Doc change

2007-08-15 22:53 +0000 [r79634]  Jason Parker <jparker@digium.com>

	* res/res_musiconhold.c: Modify the names of functions/variables in
	  res_musiconhold to be useful. Closes issue #10464, patch by
	  caio1982

2007-08-15 21:25 +0000 [r79623]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/pval.h (added), utils/pval.c (added),
	  include/asterisk/extconf.h (added), utils/extconf.c (added),
	  utils/conf2ael.c (added): Missing from murf's last trunk commit,
	  which was why trunk won't compile

2007-08-15 19:34 +0000 [r79611]  Joshua Colp <jcolp@digium.com>

	* /: Remove properties that appeared from Steve's last branch
	  merge. Automerge has already run so everyone's branches based off
	  of trunk are probably toast by now.

2007-08-15 19:21 +0000 [r79595]  Steve Murphy <murf@digium.com>

	* /, pbx/ael/ael.y (removed), pbx/ael/ael-test/ref.ael-test11,
	  res/Makefile, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test16,
	  pbx/ael/ael-test/ref.ael-test19, include/asterisk/ast_expr.h,
	  pbx/ael/ael_lex.c (removed), pbx/pbx_ael.c, pbx/ael/ael.flex
	  (removed), res/ael (added), main/pbx.c, UPGRADE.txt,
	  res/res_ael_share.c (added), pbx/Makefile, CHANGES,
	  utils/Makefile, pbx/ael/ael-test/ref.ael-ntest10,
	  pbx/ael/ael.tab.c (removed), pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ref.ael-test4, include/asterisk/ael_structs.h,
	  pbx/ael/ael.tab.h (removed), pbx/ael/ael-test/ref.ael-test5,
	  utils/ael_main.c, include/asterisk/pbx.h,
	  pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7,
	  utils/check_expr.c: This commit closes bug 7605, and half-closes
	  7638. The AEL code has been redistributed/repartitioned to allow
	  code re-use both inside and outside of Asterisk. This commit
	  introduces the utils/conf2ael program, and an external
	  config-file reader, for both normal config files, and for
	  extensions.conf (context, exten, prio); It provides an API for
	  programs outside of asterisk to use to play with the dialplan and
	  config files.

2007-08-15 14:42 +0000 [r79558]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 79553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6
	  lines (closes issue #10440) Reported by: irroot (closes issue
	  #10454) Reported by: flo_turc Increase maximum timestamp skew to
	  120. 20 was apparently far too low. ........

2007-08-15 14:27 +0000 [r79529]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 79527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug
	  2007) | 5 lines Fixed an error in the Russian language voicemail
	  intro. (issue #10458, reported and patched by Oleh) ........

2007-08-15 14:20 +0000 [r79524]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 79523 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6
	  lines (closes issue #10456) Reported by: irroot Patches:
	  sip_timeout.patch uploaded by irroot (license 52) Change
	  hardcoded timer value to defined value. I'm doing this in 1.4 as
	  well so if it needs to be changed in the future this place would
	  not have been forgotten. ........

2007-08-15 11:27 +0000 [r79507]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: Merged revisions 78936 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) |
	  1 line fixed a bug with the useruser information element. We send
	  them now also in the disconnect message. ........

2007-08-14 18:50 +0000 [r79437-79471]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 79470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) |
	  2 lines Fix another spot where an iax2_peer would be leaked if
	  realtime was in use. ........

	* /, channels/chan_iax2.c: Merged revisions 79436 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79436 | russell | 2007-08-14 12:31:39 -0500 (Tue, 14 Aug 2007) |
	  3 lines Fix some memory leaks throughout chan_iax2 related to the
	  use of realtime. I found these while working on iax2_peer object
	  reference tracking. ........

2007-08-14 15:30 +0000 [r79403]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 79397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4
	  lines (closes issue #10415) Reported by: atis Revert fix for
	  #10327 as it causes more issues then it solves. ........

2007-08-14 14:32 +0000 [r79392]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest17, /,
	  pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ael-test5/extensions.ael,
	  pbx/ael/ael-test/ael-test6/extensions.ael,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ael-vtest21/extensions.ael,
	  pbx/ael/ael-test/ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test2, pbx/pbx_ael.c,
	  pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4,
	  utils/ael_main.c, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 79255 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79255 | murf | 2007-08-13 11:49:54 -0600 (Mon, 13 Aug 2007) | 1
	  line This patch fixes bug 10411. I added a new regression test,
	  some regression test cleanups ........

2007-08-14 14:17 +0000 [r79379]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: (closes issue #10427) Reported by: pj Two of the
	  three places ast_waitfor_nandfds could branch off to did not
	  clear outfd and exception. If the calling function did not clear
	  these there was a chance they could get a false positive on
	  testing to see whether they were set.

2007-08-14 13:46 +0000 [r79378]  Steve Murphy <murf@digium.com>

	* main/channel.c, channels/chan_zap.c: Don't ask me why, but
	  waitfordigit will immediately return a 1 on my system, unless the
	  outfd is initialized to -1 before calling the nandfds func

2007-08-13 21:59 +0000 [r79335]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/speech.h, res/res_speech.c,
	  apps/app_speech_utils.c: Merged revisions 79334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2
	  lines Instead of accepting a single DTMF character accept a full
	  string. ........

2007-08-13 21:44 +0000 [r79333]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Only use the sanitysql if it's not zero-len

2007-08-13 20:40 +0000 [r79273-79306]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 79301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79301 | russell | 2007-08-13 15:37:50 -0500 (Mon, 13 Aug 2007) |
	  3 lines Don't call find_peer in registry_authrequest with the pvt
	  lock held to avoid a deadlock. ........

	* /, channels/chan_iax2.c: Merged revisions 79276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79276 | russell | 2007-08-13 15:18:30 -0500 (Mon, 13 Aug 2007) |
	  4 lines Release the pvt lock before calling find_peer in
	  register_verify to avoid a deadlock. Also, remove some
	  unnecessary locking in auth_fail that was only done recursively.
	  ........

	* /, channels/chan_iax2.c: Merged revisions 79274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79274 | russell | 2007-08-13 15:02:57 -0500 (Mon, 13 Aug 2007) |
	  3 lines Don't call find_peer within update_registry with a pvt
	  lock held. This can cause a deadlock as the code will eventually
	  call find_callno. ........

	* /, channels/chan_iax2.c: Merged revisions 79272 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79272 | russell | 2007-08-13 14:27:39 -0500 (Mon, 13 Aug 2007) |
	  9 lines I am fighting deadlocks in chan_iax2. I have tracked them
	  down to a single core issue. You can not call find_callno() while
	  holding a pvt lock as this function has to lock another (every)
	  other pvt lock. Doing so can lead to a classic deadlock. So, I am
	  tracking down all of the code paths where this can happen and
	  fixing them. The fix I committed earlier today was along the same
	  theme. This patch fixes some code down the path of
	  authenticate_reply. ........

2007-08-13 15:39 +0000 [r79238]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, apps/app_queue.c: Allow non-realtime queues to have
	  realtime members (issue #10424, reported and patched by irroot)

2007-08-13 15:32 +0000 [r79222]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 79214 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79214 | russell | 2007-08-13 10:28:13 -0500 (Mon, 13 Aug 2007) |
	  4 lines Fix a potential deadlock in socket_process.
	  check_provisioning can eventually call find_callno. You can't
	  hold a pvt lock while calling find_callno because it goes through
	  and locks every single one looking for a match. ........

2007-08-13 14:55 +0000 [r79208]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/speech.h, res/res_speech.c,
	  apps/app_speech_utils.c: Merged revisions 79207 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2
	  lines Add an API call to allow the engine to know that DTMF was
	  received. ........

2007-08-13 14:23 +0000 [r79176]  Russell Bryant <russell@digium.com>

	* main/channel.c, include/asterisk/channel.h: constify the return
	  value of reason2str

2007-08-13 14:22 +0000 [r79175]  Joshua Colp <jcolp@digium.com>

	* channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_zap.c, /, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
	  channels/chan_mgcp.c: Merged revisions 79174 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4
	  lines (closes issue #10437) Reported by: haklin Don't set the
	  callerid name and number a second time on a newly created
	  channel. ast_channel_alloc itself already sets it and setting it
	  twice would cause a memory leak. ........

2007-08-11 05:28 +0000 [r79147]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 79142 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79142 | tilghman | 2007-08-11 00:23:04 -0500 (Sat, 11 Aug 2007)
	  | 2 lines Ensure the connection gets marked as used at allocation
	  time (closes issue #10429, report and fix by mnicholson) ........

2007-08-10 21:29 +0000 [r79109]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Use localized softkey labels. Add some
	  information about localization "codes".

2007-08-10 21:03 +0000 [r79100]  Steve Murphy <murf@digium.com>

	* main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h:
	  Merged revisions 79099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1
	  line From a user complaint on #asterisk, I have forced pbx_spool
	  to explain what reason codes mean, when they are logged ........

2007-08-10 20:48 +0000 [r79098]  Russell Bryant <russell@digium.com>

	* funcs/func_devstate.c: Store custom device states in astdb so
	  that they will persist a restart. As a side benefit, this
	  simplifies the code a bit, too.

2007-08-10 18:37 +0000 [r79074]  Joshua Colp <jcolp@digium.com>

	* main/dial.c: Bring up to date with poll changes.

2007-08-10 18:35 +0000 [r79045-79068]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 79049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79049 | murf | 2007-08-10 12:25:51 -0600 (Fri, 10 Aug 2007) | 1
	  line Re bug behavior mentioned in #asterisk, made this tweak to
	  code, to prevent hundreds of log messages from being generated
	  ........

	* /: oops. forgot to commit the prop change on .

	* main/cdr.c: Merged revisions 79044 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r79044 | murf | 2007-08-10 11:43:49 -0600 (Fri, 10 Aug 2007) | 1
	  line This will help debug; from a question asked on #asterisk
	  ........

2007-08-10 16:24 +0000 [r79005-79027]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h, apps/app_meetme.c,
	  res/res_features.c, main/devicestate.c, main/event.c,
	  funcs/func_devstate.c: Merge a set of device state improvements
	  from team/russell/events. The way a device state change
	  propagates is kind of silly, in my opinion. A device state
	  provider calls a function that indicates that the state of a
	  device has changed. Then, another thread goes back and calls a
	  callback for the device state provider to find out what the new
	  state is before it can go send it off to whoever cares. I have
	  changed it so that you can include the state that the device has
	  changed to in the first function call from the device state
	  provider. This removes the need to have to call the callback,
	  which locks up critical containers to go find out what the state
	  changed to. This change set changes the "simple" device state
	  providers to use the new method. This includes parking, meetme,
	  and SLA. I have also mostly converted chan_agent in my branch,
	  but still have some more things to think through before
	  presenting the plan for converting channel drivers to ensure all
	  of the right events get generated ...

	* /, include/asterisk/lock.h: Merged revisions 78995 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10
	  Aug 2007) | 4 lines The last set of changes that I made to "core
	  show locks" made it not able to track mutexes unless they were
	  declared using AST_MUTEX_DEFINE_STATIC. Locks initialized with
	  ast_mutex_init() were not tracked. It should work now. ........

2007-08-10 14:17 +0000 [r78952-78956]  Joshua Colp <jcolp@digium.com>

	* /, main/file.c: Merged revisions 78955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2
	  lines Don't bother having the core pass through or emulate begin
	  DTMF frames when in an ast_waitstream. It only cares about the
	  end of DTMF. ........

	* /, configs/queues.conf.sample: Merged revisions 78951 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78951 | file | 2007-08-10 10:49:19 -0300 (Fri, 10 Aug 2007) | 4
	  lines (closes issue #10422) Reported by: bhowell Add note to
	  sample configuration about module load order and how it can cause
	  perfectly good queue members to be marked as invalid. ........

2007-08-09 23:49 +0000 [r78908]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 78907 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78907 | mmichelson | 2007-08-09 18:47:00 -0500 (Thu, 09 Aug
	  2007) | 4 lines Improved a bit of logic regarding comma-separated
	  mailboxes in has_voicemail. Also added some braces to some
	  compound if statements since unbraced if statements scare me in
	  general. ........

2007-08-09 23:32 +0000 [r78906]  Steve Murphy <murf@digium.com>

	* Makefile, /: Merged revisions 78891 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78891 | murf | 2007-08-09 17:10:46 -0600 (Thu, 09 Aug 2007) | 1
	  line This fixes bug 10416; thanks to mvanbaak for the pretty
	  output ........

2007-08-09 22:19 +0000 [r78861-78862]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 78859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78859 | mmichelson | 2007-08-09 16:51:17 -0500 (Thu, 09 Aug
	  2007) | 9 lines Quite a few changes regarding IMAP storage. 1.
	  instead of using inboxcount as the core message counting
	  function, we use messagecount instead. This makes it possible to
	  count messages in folders besides just INBOX and Old. 2.
	  inboxcount and hasvoicemail now use messagecount as their means
	  of determining return values. 3. Added a copy_message function
	  for IMAP storage. Unfortunately I don't have the means to test
	  it, but it seems like a pretty straightforward function. 4.
	  Removed a #ifndef IMAP_STORAGE and matching #endif from
	  leave_voicemail for a couple of reasons. One, we want to support
	  copying mail to multiple IMAP boxes, and two, IMAP was broken
	  because a STORE macro had been moved into this section of code.
	  ........

2007-08-09 20:07 +0000 [r78829]  Russell Bryant <russell@digium.com>

	* apps/app_minivm.c: Don't use strncpy for moving a chunk of memory
	  to another that is overlapping. This was found by running
	  Asterisk under valgrind.

2007-08-09 19:35 +0000 [r78718-78824]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: When looking up a mailbox, use the default
	  context if not specified as something else

	* channels/chan_sip.c: Restore the ability to have multiple
	  mailboxes listed for the mailbox option in sip.conf. chan_sip now
	  maintains separate internal MWI subscriptions for each one.

	* /, apps/app_voicemail.c: Merged revisions 78778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78778 | russell | 2007-08-09 12:58:31 -0500 (Thu, 09 Aug 2007) |
	  1 line add a comment to indicate that inboxcount for ODBC_STORAGE
	  needs to be fixed to support multiple mailboxes ........

	* /, apps/app_voicemail.c: Merged revisions 78749 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78749 | russell | 2007-08-09 12:24:40 -0500 (Thu, 09 Aug 2007) |
	  9 lines Fix subscriptions to multiple mailboxes for ODBC_STORAGE.
	  Also, leave a comment for this to be fixed for IMAP_STORAGE, as
	  well. I left IMAP alone since I know MarkM was working on this
	  code right now for another reason. This is broken even worse in
	  trunk, but for a different reason. The fact that the mailbox
	  option supported multiple mailboxes is completely not obvious
	  from the code in the channel drivers. Anyway, I will fix that in
	  another commit ... ........

	* channels/chan_zap.c, channels/chan_sip.c,
	  include/asterisk/event_defs.h, channels/chan_iax2.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c: Fix a problem that I
	  had introduced into MWI handling. I had ignored the mailbox
	  context. Now, all related MWI event dealings pay attention to the
	  context as well.

	* /, apps/app_meetme.c: Merged revisions 78717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) |
	  7 lines Fix a problem with the combination of the 'F' option to
	  pass DTMF through a conference and options that use DTMF to
	  activate various features. The problem was that the BEGIN frame
	  would be passed through, but the END frame would get intercepted
	  to activate a feature. Then, the other conference members would
	  hear DTMF for forever, which they didn't seem to like very much.
	  (closes issue #10400, reported by stevefeinstein, fixed by me)
	  ........

2007-08-08 22:05 +0000 [r78649-78686]  Joshua Colp <jcolp@digium.com>

	* configure: Regenerate configure script. This actually just
	  updated the revision number... since my last merge changed it to
	  an older number, while it was in fact generated from a much newer
	  revision.

	* channels/chan_skinny.c: Minor fix for building under dev mode
	  when byteswapping macro header files are not available.

	* apps/app_dial.c, channels/chan_zap.c, channels/chan_sip.c,
	  include/asterisk/autoconfig.h.in, channels/chan_agent.c,
	  configure.ac, include/asterisk/channel.h, channels/chan_gtalk.c,
	  channels/chan_oss.c, main/rtp.c, main/channel.c,
	  channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_skinny.c, configure,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c:
	  Add support for using epoll instead of poll. This should increase
	  scalability and is done in such a way that we should be able to
	  add support for other poll() replacements.

	* channels/chan_zap.c: HAVEL_SS7 should be HAVE_SS7. Reported by
	  kwallace.

	* main/channel.c, include/asterisk/audiohook.h (added),
	  funcs/func_volume.c (added), main/Makefile, main/slinfactory.c,
	  include/asterisk/chanspy.h (removed), include/asterisk/channel.h,
	  main/audiohook.c (added), apps/app_chanspy.c,
	  apps/app_mixmonitor.c, include/asterisk/slinfactory.h: Merge
	  audiohooks branch into trunk. This is a new API for developers to
	  listen and manipulate the audio going through a channel.

2007-08-08 19:30 +0000 [r78648]  Jason Parker <jparker@digium.com>

	* /, doc/jabber.txt: Merged revisions 78646 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78646 | qwell | 2007-08-08 14:29:42 -0500 (Wed, 08 Aug 2007) | 2
	  lines Fix mogs email address. ........

2007-08-08 19:03 +0000 [r78637]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Correct spelling. s/threaads/threads/

2007-08-08 18:34 +0000 [r78590-78635]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 78575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78575 | mmichelson | 2007-08-08 09:26:36 -0500 (Wed, 08 Aug
	  2007) | 4 lines Changing a bit of logic so that someone will
	  NEVER exit the queue on timeout unless they have enabled the 'n'
	  option. This commit relates to issue #10320. Thanks to
	  jfitzgibbon for detailing the idea behind this code change.
	  ........

2007-08-08 13:52 +0000 [r78570]  Joshua Colp <jcolp@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 78569 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug
	  2007) | 4 lines (closes issue #10335) Reported by: adamgundy
	  Update sip.conf to include another scenario where directrtpsetup
	  will fail. ........

2007-08-07 23:04 +0000 [r78541]  Russell Bryant <russell@digium.com>

	* main/pbx.c, pbx/pbx_spool.c, main/sha1.c, res/res_features.c,
	  res/res_crypto.c, utils/smsq.c, include/asterisk/features.h: Add
	  another big set of doxygen documentation improvements from
	  snuffy. (closes issue #9892) (closes issue #10395)

2007-08-07 22:13 +0000 [r78521]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, include/asterisk/manager.h: Use the linkedlists.h
	  macros for the manager action list.

2007-08-07 21:00 +0000 [r78489]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 78488 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78488 | russell | 2007-08-07 15:57:54 -0500 (Tue, 07
	  Aug 2007) | 2 lines Fix the build of this module on 64-bit
	  platforms ........

2007-08-07 19:44 +0000 [r78451]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 78450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78450 | mmichelson | 2007-08-07 14:43:57 -0500 (Tue, 07 Aug
	  2007) | 5 lines The logic behind inboxcount's return value was
	  reversed in has_voicemail and message_count. (closes issue
	  #10401, reported by st1710, patched by me) ........

2007-08-07 19:36 +0000 [r78442]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 78437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78437 | tilghman | 2007-08-07 14:34:25 -0500 (Tue, 07 Aug 2007)
	  | 2 lines Don't free the environment handle when the connection
	  fails, because other connections might be depending upon it
	  ........

2007-08-07 19:14 +0000 [r78417]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c, /, apps/app_directory.c,
	  apps/app_voicemail.c: Merged revisions 78415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78415 | tilghman | 2007-08-07 14:09:38 -0500 (Tue, 07 Aug 2007)
	  | 2 lines Reconnection doesn't happen automatically when a DB
	  goes down (fixes issue #9389) ........

2007-08-07 18:26 +0000 [r78378]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 78375 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78375 | qwell | 2007-08-07 13:25:15 -0500 (Tue, 07 Aug
	  2007) | 3 lines Properly check the capabilities count to avoid a
	  segfault. (ASA-2007-019) ........

2007-08-07 17:46 +0000 [r78372]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 78371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r78371 | russell | 2007-08-07 12:45:30 -0500
	  (Tue, 07 Aug 2007) | 12 lines Merged revisions 78370 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07
	  Aug 2007) | 4 lines Revert patch committed for issue #9660. It
	  broke E&M trunks. (closes issue #10360) (closes issue #10364)
	  ........ ................

2007-08-07 16:17 +0000 [r78346-78347]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Can't forget outsignaling!

	* channels/chan_zap.c: Just for jsmith... make signaling a valid
	  option that acts like signalling.

2007-08-07 16:04 +0000 [r78342]  Russell Bryant <russell@digium.com>

	* res/res_eventtest.c (removed): Remove some test code from trunk
	  as it doesn't need to be here. I'm just going to keep it with a
	  bunch of other changes i have sitting in a branch.

2007-08-07 15:40 +0000 [r78338]  Joshua Colp <jcolp@digium.com>

	* main/frame.c: (closes issue #10225) Reported by: klaus3000 Clean
	  up AST_FORMAT_LIST list. It may have mattered in the old days to
	  have undefined entries but these days it does not.

2007-08-06 23:00 +0000 [r78312]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c: Add a TalkingToChan to the response of the
	  "agents" manager action. This is similar to the existing "talking
	  to" that you see what using the "agent show" CLI command. Closes
	  issue #10102

2007-08-06 21:59 +0000 [r78276-78279]  Joshua Colp <jcolp@digium.com>

	* apps/app_senddtmf.c: Fix bug where a NULL timeout would make
	  things explode if SendDTMF was called with it.

	* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
	  res/res_features.c, apps/app_test.c, main/app.c,
	  include/asterisk/channel.h, apps/app_senddtmf.c: Extend the
	  ast_senddigit and ast_dtmf_stream API calls to allow the duration
	  of the DTMF digit(s) to be specified and make the SendDTMF
	  application have the capability to use it.

	* main/channel.c, /: Merged revisions 78275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2
	  lines Add additional DTMF log messages to help when debugging
	  issues. ........

2007-08-06 20:45 +0000 [r78243]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 78242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78242 | russell | 2007-08-06 15:44:09 -0500 (Mon, 06 Aug 2007) |
	  4 lines Fix an issue where dynamic threads can get free'd, but
	  still exist in the dynamic thread list. (closes issue #10392,
	  patch from Mihai, with credit to his colleague, Pete) ........

2007-08-06 19:52 +0000 [r78227]  Doug Bailey <dbailey@digium.com>

	* main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c,
	  main/fskmodem.c: Change the fsk filter used in CID and TDD decode
	  to an integer based implementation

2007-08-06 17:51 +0000 [r78186-78192]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fixing a compiler warning which warns that a
	  variable may be used unitialized. Thanks to mvanbaak for pointing
	  this out.

	* /, channels/chan_sip.c, include/asterisk/config.h, main/config.c:
	  Merged revisions 78103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug
	  2007) | 7 lines Changed the behavior of sip's realtime_peer
	  function to match the corresponding way of matching for
	  non-realtime peers. Now matches are made on both the IP address
	  and port number, or if the insecure setting is set to "port" then
	  just match on the IP address. In order to accomplish this, I also
	  added a new API call, ast_category_root, which returns the first
	  variable of an ast_category struct ........

2007-08-06 16:51 +0000 [r78185]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 78184 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) |
	  5 lines Fix the return value of AST_LIST_REMOVE(). This shouldn't
	  be causing any problems, though, because the only code that uses
	  the return value only checks to see if it is NULL. (closes issue
	  #10390, pointed out by mihai) ........

2007-08-06 16:34 +0000 [r78183]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 78182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78182 | file | 2007-08-06 13:32:44 -0300 (Mon, 06 Aug 2007) | 2
	  lines It is possible for a transfer to occur before the remote
	  device has our tag in which case they send none in the transfer.
	  In this case we need to not fail the transfer dialog lookup.
	  ........

2007-08-06 16:31 +0000 [r78179-78181]  Jason Parker <jparker@digium.com>

	* /, main/config.c: Merged revisions 78180 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #9938) ........ r78180 | qwell | 2007-08-06 11:30:51 -0500
	  (Mon, 06 Aug 2007) | 5 lines Fix an issue with using UpdateConfig
	  (manager action) where escaped semicolons in a config would be
	  converted to just semicolons (\; to ;) Issue 9938 ........

	* channels/chan_skinny.c, configs/skinny.conf.sample: Implement
	  setvar functionality in chan_skinny Closes issue #10379, patch by
	  mvanbaak.

2007-08-06 15:28 +0000 [r78167-78173]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 78172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4
	  lines (closes issue #10355) Reported by: wdecarne Now that we
	  pass through RTP timestamp information we need to make the
	  allowed timestamp skew considerably less. There are situations
	  where a source may change and due to the timestamp difference the
	  receiver will experience an audio gap since we did not indicate
	  by setting the marker bit that the source changed. ........

	* apps/app_externalivr.c: (closes issue #10381) Reported by: yehavi
	  Use the filename we parsed using the standard parsing when
	  launching the application specified to ExternalIVR.

	* /, configure, configure.ac: Merged revisions 78166 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78166 | file | 2007-08-06 11:18:20 -0300 (Mon, 06 Aug
	  2007) | 4 lines (closes issue #10383) Reported by: rizzo Include
	  stdlib.h so NULL gets defined for gethostbyname_r checks.
	  ........

2007-08-05 04:16 +0000 [r78142-78144]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 78143 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r78143 | russell | 2007-08-04 23:15:31 -0500 (Sat, 04
	  Aug 2007) | 2 lines Fix compilation failure when MALLOC_DEBUG is
	  enabled, but DEBUG_THREADS is not ........

	* apps/app_exec.c: Make this module build on my mac

2007-08-05 03:42 +0000 [r78140]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 78139 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78139 | tilghman | 2007-08-04 22:29:01 -0500 (Sat, 04 Aug 2007)
	  | 2 lines If peer is not found, the error message is misleading
	  (should be peer not found, not ACL failure) ........

2007-08-05 03:14 +0000 [r78138]  Russell Bryant <russell@digium.com>

	* include/asterisk/linkedlists.h: Fix building res_crypto on
	  systems that init locks with constructors. The problem was that
	  res_crypto now has a RWLIST named "keys". The macro for defining
	  this list defines a function used as a constructor for the list
	  called "init_keys". However, there was another function called
	  init_keys in this module for a CLI command. The fix is just to
	  prepend the generated functions with underscores.

2007-08-03 20:21 +0000 [r78029-78102]  Russell Bryant <russell@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 78101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78101 | russell | 2007-08-03 15:14:06 -0500 (Fri, 03 Aug 2007) |
	  10 lines (closes issue #10194) Reported by: blitzrage Patches:
	  bug0010194 uploaded by vovochka Tested by: blitzrage Fix a
	  problem when you call Voicemail() with multiple mailboxes
	  specified and ODBC_STORAGE is in use. The audio part of the
	  message was only given to the first mailbox specified. ........

	* /, main/utils.c, include/asterisk/lock.h, main/astmm.c: Merged
	  revisions 78095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) |
	  28 lines Add some improvements to lock debugging. These changes
	  take effect with DEBUG_THREADS enabled and provide the following:
	  * This will keep track of which locks are held by which thread as
	  well as which lock a thread is waiting for in a thread-local data
	  structure. A reference to this structure is available on the
	  stack in the dummy_start() function, which is the common entry
	  point for all threads. This information can be easily retrieved
	  using gdb if you switch to the dummy_start() stack frame of any
	  thread and print the contents of the lock_info variable. * All of
	  the thread-local structures for keeping track of this lock
	  information are also stored in a list so that the information can
	  be dumped to the CLI using the "core show locks" CLI command.
	  This introduces a little bit of a performance hit as it requires
	  additional underlying locking operations inside of every
	  lock/unlock on an ast_mutex. However, the benefits of having this
	  information available at the CLI is huge, especially considering
	  this is only done in DEBUG_THREADS mode. It means that in most
	  cases where we debug deadlocks, we no longer have to request
	  access to the machine to analyze the contents of ast_mutex_t
	  structures. We can now just ask them to get the output of "core
	  show locks", which gives us all of the information we needed in
	  most cases. I also had to make some additional changes to astmm.c
	  to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
	  enabled. I disabled tracking of one of the locks in astmm.c
	  because it gets used inside the replacement memory allocation
	  routines, and the lock tracking code allocates memory. This
	  caused infinite recursion. ........

	* /, channels/chan_iax2.c: Merged revisions 78063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78063 | russell | 2007-08-03 12:01:07 -0500 (Fri, 03 Aug 2007) |
	  4 lines Only pass through HOLD and UNHOLD control frames when the
	  mohinterpret option is set to "passthrough". This was pointed out
	  by Kevin in the middle of a training session. ........

	* /, channels/chan_iax2.c: Merged revisions 78028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r78028 | russell | 2007-08-02 21:04:22 -0500 (Thu, 02 Aug 2007) |
	  6 lines Don't reuse the timespec that was set to 0 in the
	  previous timedwait as it will just return immediately. Also, fix
	  some logic so the thread's lock isn't unlocked twice in the weird
	  case of dynamic threads getting acquired right after a timeout.
	  (pointed out by SteveK) ........

2007-08-02 21:54 +0000 [r77994-77997]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged
	  revisions 77996 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #9779) ........ r77996 | qwell | 2007-08-02 16:53:39 -0500
	  (Thu, 02 Aug 2007) | 5 lines Make sure we actually allow 6 chars
	  to be sent. Also make note of the "A" option of date format.
	  Issue 9779, modifications by DEA, wedhorn, and myself. ........

	* /, channels/chan_skinny.c: Merged revisions 77993 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10325) ........ r77993 | qwell | 2007-08-02 15:22:40 -0500
	  (Thu, 02 Aug 2007) | 5 lines If a device disconnects, the session
	  will go away. If this happens during call setup, we need to give
	  up. Issue 10325. ........

2007-08-02 19:26 +0000 [r77950]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 77949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77949 | russell | 2007-08-02 14:25:14 -0500 (Thu, 02 Aug 2007) |
	  5 lines Fix the case where a dynamic thread times out waiting for
	  something to do during the first time it runs. This shouldn't
	  ever happen, but we should account for it anyway. (pointed out by
	  pete, who works with mihai) ........

2007-08-02 18:43 +0000 [r77948]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 77947 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10299) ........ r77947 | qwell | 2007-08-02 13:42:36 -0500
	  (Thu, 02 Aug 2007) | 5 lines Make sure we clear the prompt status
	  message on a hangup. Also rearrange messages to better fit with
	  what a wireshark trace shows it should be. Issue 10299, initial
	  patch and solution by sbisker, modified by me to fit with
	  wireshark trace. ........

2007-08-02 18:32 +0000 [r77946]  Steve Murphy <murf@digium.com>

	* /, main/fskmodem.c: Merged revisions 77945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r77945 | murf | 2007-08-02 12:21:40 -0600 (Thu,
	  02 Aug 2007) | 9 lines Merged revisions 77942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
	  line This patch hopefully solves 10141; The user is running with
	  it, and it doesn't appear to harm asterisk's operation, and may
	  prevent a crash. I'll store it in 1.2, as we have shut down
	  support on 1.2, but since I developed the patch before support
	  finished, and it might affect 1.4 and trunk, I'm going ahead with
	  it. ........ ................

2007-08-02 18:05 +0000 [r77940-77944]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 77943 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77943 | russell | 2007-08-02 13:04:43 -0500 (Thu, 02 Aug 2007) |
	  9 lines Fix another race condition in the handling of dynamic
	  threads. If the dynamic thread timed out waiting for something to
	  do, but was acquired to perform an action immediately afterwords,
	  then wait on the condition again to give the other thread a
	  chance to finish setting up the data for what action this thread
	  should perform. Otherwise, if it immediately continues, it will
	  perform the wrong action. (reported on IRC by mihai, patch by me)
	  (related to issue #10289) ........

	* channels/chan_iax2.c: Fix an issue that Simon pointed out to me
	  on IRC. There were cases in the trunk version of
	  find_idle_thread() where the old full frame processing
	  information was not cleared out. This would have caused full
	  frames to get deferred for processing by threads that weren't
	  actually processing frames for that call. Nice catch!!

	* /, channels/chan_iax2.c: Merged revisions 77939 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77939 | russell | 2007-08-02 11:56:04 -0500 (Thu, 02 Aug 2007) |
	  4 lines Add another sanity check to vnak_retransmit(). This check
	  ensures that frames that have already been marked for deletion
	  don't get retransmitted. (closes issue #10361, patch from mihai)
	  ........

2007-08-02 15:16 +0000 [r77891-77895]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 77894 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10358) ........ r77894 | qwell | 2007-08-02 10:15:45 -0500
	  (Thu, 02 Aug 2007) | 5 lines Make sure that we show the correct
	  extension if dialed from a macro "From: 5555" rather than "From:
	  s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
	  tested by sbisker ........

	* /, channels/chan_skinny.c: Merged revisions 77890 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10291) ........ r77890 | qwell | 2007-08-01 17:28:56 -0500
	  (Wed, 01 Aug 2007) | 4 lines Put in some additional debug
	  information for softkey/stimulus messages. Issue 10291, patch by
	  DEA. ........

2007-08-01 22:24 +0000 [r77889]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 77887 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77887 | russell | 2007-08-01 17:16:17 -0500 (Wed, 01 Aug 2007) |
	  23 lines Fix some race conditions which have been causing weird
	  problems in chan_iax2. The most notable problem is that people
	  have been seeing storms of VNAK frames being sent due to really
	  old frames mysteriously being in the retransmission queue and
	  never getting removed. It was possible that a dynamic thread got
	  created, but did not acquire its lock before the thread that
	  created it signals it to perform an action. When this happens,
	  the thread will sleep until it hits a timeout, and then get
	  destroyed. So, the action never gets performed and in some cases,
	  means a frame doesn't get transmitted and never gets freed since
	  the scheduler never gets a chance to reschedule transmission.
	  Another less severe race condition is in the handling of a
	  timeout for a dynamic thread. It was possible for it to be
	  acquired to perform at action at the same time that it hit a
	  timeout. When this occurs, whatever action it was acquired for
	  would never get performed. (patch contributed by Mihai and
	  SteveK) (closes issue #10289) (closes issue #10248) (closes issue
	  #10232) (possibly related to issue #10359) ........

2007-08-01 22:19 +0000 [r77888]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 77886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77886 | tilghman | 2007-08-01 17:14:47 -0500 (Wed, 01 Aug 2007)
	  | 2 lines Voicemail with ODBC_STORAGE defined does not compile
	  cleanly (missing def) ........

2007-08-01 21:12 +0000 [r77879-77884]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 77883 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77883 | qwell | 2007-08-01 16:08:42 -0500 (Wed, 01 Aug
	  2007) | 7 lines Fix an issue that caused one-way audio on some
	  newer devices (specifically the 7921), due to sending packets in
	  the wrong order during hangup. Also make sure we clear
	  tones/messages on the correct line/instance. Issue 10291, patch
	  by DEA, tested by sbisker and myself. ........

	* apps/app_queue.c, doc/tex/queuelog.tex: Add the Ring time in the
	  CONNECT on the queue_log and on the Manager event AgentConnect
	  Closes issue #10349, patch by eliel

2007-08-01 19:37 +0000 [r77864-77878]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, configure, configure.ac, main/asterisk.c: Instead of
	  adding the SOLARIS check to each HAVE_SYSINFO check let's just
	  make the sysinfo autoconf logic a bit pickier about what it
	  considers a usable sysinfo.

	* main/pbx.c, main/asterisk.c: Solaris does not have a sysinfo like
	  we know of on Linux.

	* configure, configure.ac: Don't look for /dev/urandom when cross
	  compiling. Just assume it is not available.

	* /, utils/smsq.c, channels/chan_iax2.c,
	  include/asterisk/threadstorage.h, channels/chan_mgcp.c,
	  apps/app_voicemail.c: Merged revisions 77869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77869 | file | 2007-08-01 14:56:59 -0300 (Wed, 01 Aug 2007) | 2
	  lines Add some fixes for building on Solaris. ........

	* /, main/utils.c: Merged revisions 77867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77867 | file | 2007-08-01 14:52:11 -0300 (Wed, 01 Aug 2007) | 2
	  lines Whoops, I meant R_5 not R5. ........

	* /, configure, configure.ac: Merged revisions 77865 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77865 | file | 2007-08-01 14:42:52 -0300 (Wed, 01 Aug
	  2007) | 2 lines And for my last trick... make sure that if
	  gethostbyname_r is exported by a library that it is used.
	  ........

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: Merged revisions 77863 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77863 | file | 2007-08-01 14:22:35 -0300 (Wed, 01 Aug 2007) | 2
	  lines Extend autoconf logic to determine which version of
	  gethostbyname_r is on the system. ........

2007-08-01 15:39 +0000 [r77858]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, main/autoservice.c, main/pbx.c,
	  apps/app_osplookup.c, channels/chan_local.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, res/res_features.c,
	  apps/app_zapras.c, apps/app_macro.c, pbx/pbx_dundi.c,
	  apps/app_queue.c: Convert code that checks the _softhangup member
	  of ast_channel directory to use the ast_check_hangup() funciton.
	  This function takes scheduled hangups into account. (closes issue
	  #10230, patch by Juggie)

2007-08-01 15:28 +0000 [r77857]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: Convert CLI helpers list to rwlist.

2007-08-01 14:09 +0000 [r77853-77855]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 77854 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77854 | mmichelson | 2007-08-01 09:08:57 -0500 (Wed, 01 Aug
	  2007) | 8 lines Fixes an issue I introduced to queues wherein a
	  queue with joinempty=yes would kick people out of the queue
	  because of erroneously thinking the 'n' option was in use.
	  (closes issue #10320, reported by jfitzgibbon, patched by me,
	  tested by blitzrage and me) Thank you blitzrage for all the
	  testing you've done lately with queues! It's much appreciated!
	  ........

	* /, apps/app_queue.c: Merged revisions 77852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77852 | mmichelson | 2007-08-01 08:59:59 -0500 (Wed, 01 Aug
	  2007) | 7 lines If a queue uses dynamic realtime members, then
	  the member list should be updated after each attempt to call the
	  queue. This fixes an issue where if a caller calls into a queue
	  where no one is logged in, they would wait forever even if a
	  member logged in at some point. (closes issue #10346, reported by
	  and tested by blitzrage, patched by me) ........

2007-08-01 04:36 +0000 [r77851]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Twould help if we actually defined ->mod before
	  comparing against it (reported and fixed by Juggie via IRC).

2007-07-31 21:33 +0000 [r77847]  Steve Murphy <murf@digium.com>

	* /, contrib/scripts/ast_grab_core: Merged revisions 77844 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r77844 | murf | 2007-07-31 14:59:10 -0600 (Tue,
	  31 Jul 2007) | 9 lines Merged revisions 77842 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
	  line This probably isn't super-general, but it's a first stab at
	  using kill -11 to generate a core file instead of gcore. ........
	  ................

2007-07-31 18:50 +0000 [r77834-77838]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_lock.c, CHANGES: Add some documentation detailing an
	  aspect of dialplan functions, as requested by Russell

	* funcs/func_lock.c (added), UPGRADE.txt: Add func_lock, which
	  creates dialplan mutexes, and note that the Macro apps are now
	  deprecated. (Closes issue #10264)

2007-07-31 16:21 +0000 [r77833]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/speech.h, res/res_speech.c: Merged revisions
	  77831 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77831 | file | 2007-07-31 13:17:09 -0300 (Tue, 31 Jul 2007) | 2
	  lines Add a flag to the speech API that allows an engine to set
	  whether it received results or not. ........

2007-07-31 15:59 +0000 [r77829]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: thanks to Russel, for pointing out that the
	  dialoglist_lock/unlock routines also need to be macros if
	  DETECT_DEADLOCKS is set

2007-07-31 15:54 +0000 [r77828]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/cflags.xml, /: Merged revisions 77827 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77827 | kpfleming | 2007-07-31 10:53:42 -0500 (Tue, 31
	  Jul 2007) | 2 lines DETECT_DEADLOCKS can't be enabled without
	  DEBUG_THREADS or it does nothing ........

2007-07-31 15:22 +0000 [r77825]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 77824 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul
	  2007) | 6 lines This patch makes Asterisk send 100 Trying
	  provisional responses upon receipt of re-invites. This makes it
	  so that if there are two or more Asterisk servers between
	  endpoints, the Asterisk servers will not keep retransmitting the
	  re-invites. (closes issue #10274, reported by cstadlmann, patched
	  by me with approval from file) ........

2007-07-31 15:01 +0000 [r77819-77821]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: there is no use in having functions that
	  have no code in them, and hide the locking info when
	  DEBUG_THREADS is enabled... i could have fixed this to be
	  dependent on DEBUG_THREADS, but it would be just as easy for
	  someone to add their test/debugging code to the macros as it
	  would have been to the functions

	* channels/chan_sip.c: use a different method for overriding the
	  send_digit_begin pointer, as the old one fails to compile on my
	  64-bit system with gcc-4.1 and --enable-dev-mode turned on

	* apps/app_senddtmf.c: umm... let's build with --enable-dev-mode,
	  mmkay?

2007-07-31 03:32 +0000 [r77810]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: Discovered in experiments on core files: if
	  you wrap the lock and unlock calls with sip_pvt_lock and
	  sip_pvt_unlock, you lose the tracing info you would normally get
	  via DETECT_DEADLOCKS; so I turn these two functions into macros
	  when DETECT_DEADLOCKS is called. This way, you get meaningful
	  stuff in the file and func slots in the lock_info struct.

2007-07-31 01:10 +0000 [r77808]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, apps/app_dictate.c, apps/app_record.c,
	  apps/app_authenticate.c, apps/app_sayunixtime.c,
	  apps/app_userevent.c, apps/app_chanisavail.c, apps/app_image.c,
	  apps/app_followme.c, apps/app_controlplayback.c,
	  funcs/func_enum.c, funcs/func_odbc.c, apps/app_minivm.c,
	  res/res_agi.c, apps/app_amd.c, apps/app_url.c,
	  apps/app_directory.c, apps/app_rpt.c, apps/app_parkandannounce.c,
	  apps/app_read.c, funcs/func_timeout.c, apps/app_page.c,
	  apps/app_festival.c, apps/app_privacy.c,
	  apps/app_waitforsilence.c, apps/app_disa.c, apps/app_transfer.c,
	  apps/app_talkdetect.c, apps/app_queue.c, apps/app_playback.c,
	  res/res_monitor.c, apps/app_speech_utils.c, funcs/func_curl.c,
	  funcs/func_channel.c, funcs/func_cdr.c, apps/app_sendtext.c,
	  apps/app_macro.c, apps/app_sms.c, apps/app_senddtmf.c,
	  apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_stack.c,
	  apps/app_voicemail.c: Mostly cleanup of documentation to
	  substitute the pipe with the comma, but a few other formatting
	  cleanups, too.

2007-07-30 20:42 +0000 [r77801]  Joshua Colp <jcolp@digium.com>

	* main/dial.c, include/asterisk/dial.h: Add support for call
	  forwarding and timeouts to the dialing API.

2007-07-30 20:36 +0000 [r77797-77800]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Change another unnecessary use of the
	  increment operator to explicitly set the var to 1

	* channels/chan_iax2.c: Explicitly set a variable to 1 instead of
	  using the increment operator.

	* /, channels/chan_iax2.c: Merged revisions 77794 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) |
	  8 lines Fix an issue that could potentially cause corruption of
	  the global iax frame queue. In the network_thread() loop, it
	  traverses the list using the AST_LIST_TRAVERSE_SAFE macro.
	  However, to remove an element of the list within this loop, it
	  used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I
	  believe could leave some of the internal variables of the SAFE
	  macro invalid. Mihai says that he already made this change in his
	  local copy and it didn't help his VNAK storm issues, but I still
	  think it's wrong. :) ........

2007-07-30 20:19 +0000 [r77796]  Jason Parker <jparker@digium.com>

	* /, main/say.c: Merged revisions 77795 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10083) ........ r77795 | qwell | 2007-07-30 15:17:08 -0500
	  (Mon, 30 Jul 2007) | 6 lines Applications like SayAlpha() should
	  not hang up the channel if you request an "unknown" character
	  such as a comma. Instead, skip the character and move on. Issue
	  10083, initial patch by jsmith, modified by me. ........

2007-07-30 19:42 +0000 [r77793]  Luigi Rizzo <rizzo@icir.org>

	* main/channel.c: print formats as 0x%x instead of %d in a warning
	  message. Being bitmasks, it is a lot easier to read this way.

2007-07-30 19:39 +0000 [r77789-77792]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Fix the return value of ast_agi_fdprintf() to
	  include the result from ast_carefulwrite()

	* res/res_agi.c: Improve ast_agi_fdprintf() by using the ast_str()
	  API. * Use a thread local ast_str for building the string that
	  will be written out to the console for debug, and to the FD for
	  the AGI itself, instead of allocating a buffer on the heap every
	  time the function is called. * Use the information contained
	  within the ast_str to determine how many bytes need to be written
	  instead of calling strlen().

	* main/manager.c: Remove an XXX comment noting that it would be
	  nice for a declaration to be inside of a function. (Yes, it
	  would!) Replace it with a note that explains why it can't be done
	  using the way that the AST_THREADSTORAGE macro is currently
	  defined.

	* include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 77788
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) |
	  10 lines (closes issue #10279) Reported by: seanbright Patches:
	  res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright
	  (license 71) res_agi.carefulwrite.trunk.07252007.patch uploaded
	  by seanbright (license 71) Allow the "agi_network: yes" line to
	  be printed out in the AGI debug output. Also, allow partial
	  writes to be handled when writing out this line just like it is
	  for all of the others. ........

2007-07-30 19:11 +0000 [r77787]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/agi.h, res/res_agi.c: Cleanup of res_agi,
	  ensuring thread safety (closes issue #10288)

2007-07-30 18:56 +0000 [r77786]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 77785 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) |
	  3 lines file and I both committed changes for issue #10301.
	  Remove a duplicated assignment to restore the original value of
	  the previous channel. ........

2007-07-30 18:45 +0000 [r77784]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 77783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r77783 | tilghman | 2007-07-30 13:43:55 -0500
	  (Mon, 30 Jul 2007) | 10 lines Merged revisions 77782 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30
	  Jul 2007) | 2 lines Revert change in revision 71656, even though
	  it fixed a bug, because many people were depending upon the
	  (broken) behavior. ........ ................

2007-07-30 17:31 +0000 [r77781]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 77780 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) |
	  16 lines (closes issue #10301) Reported by: fnordian Patches:
	  asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
	  Additional changes by me Fix some problems in
	  channel_find_locked() which can cause an infinite loop. The
	  reference to the previous channel is set to NULL in some cases.
	  These changes ensure that the reference to the previous channel
	  gets restored before needing it again. I'm not convinced that the
	  code that is setting it to NULL is really the right thing to do.
	  However, I am making these changes to fix the obvious problem and
	  just leaving an XXX comment that it needs a better explanation
	  that what is there now. ........

2007-07-30 17:12 +0000 [r77772-77779]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 77778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4
	  lines (closes issue #10327) Reported by: kkiely Instead of
	  directly mucking with the extension/context/priority of the
	  channel we are transferring when it has a PBX simply call
	  ast_async_goto on it. This will ensure that the channel gets
	  handled properly and sent to the right place. ........

	* apps/app_followme.c: Minor clean up of app_followme.

	* main/channel.c, /: Merged revisions 77771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6
	  lines (closes issue #10301) Reported by: fnordian Patches:
	  asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
	  Restore previous behavior where if we failed to lock the channel
	  we wanted we would return to exactly the same point as if we had
	  just reentered the function. ........

2007-07-30 15:22 +0000 [r77770]  Russell Bryant <russell@digium.com>

	* cdr/cdr_adaptive_odbc.c: Resolve some compiler warnings so that I
	  can build under dev mode

2007-07-30 14:53 +0000 [r77769]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_macro.c: Merged revisions 77768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r77768 | file | 2007-07-30 11:51:44 -0300 (Mon,
	  30 Jul 2007) | 12 lines Merged revisions 77767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
	  lines (closes issue #10334) Reported by: ramonpeek Pass through
	  the return value from macro_exec through the MacroIf application.
	  ........ ................

2007-07-30 10:55 +0000 [r77616-77766]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: minor code rearrangements: + place the link
	  field at the beginning of struct sip_pvt, and not somewhere in
	  the middle; + in __sip_reliable_xmit, remove a duplicate
	  assignment, and put the statements in a more logical order (i.e.
	  first copy the payload and associated info, then copy arguments
	  from the caller, then finish initializing the headers...) nothing
	  to backport.

	* channels/chan_sip.c: rename handle_request() to
	  handle_incoming(), as the former was misleading - the function
	  deals with all incoming packets, be them requests or responses.

	* channels/chan_sip.c: move some dialog-only flags to proper
	  variables, namely SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE,
	  SIP_PAGE2_NOTEXT, SIP_PAGE2_OUTGOING_CALL These are seldom used
	  so the diff is relatively small. Note that 'OUTGOING_CALL' is
	  dangerously similar to another dialog flag, 'SIP_OUTGOING', so
	  the description will need to clarify the different meaning of the
	  two. Also note that the description of NOTEXT is a bit unclear -
	  does it mean we don't support it, or 'not requested or not
	  supported' ? On passing fix a comment referring to video instead
	  of text. Finally, mark with XXX a possibly misleading debugging
	  message. (maybe the latter is worth backporting).

	* channels/chan_sip.c: use a function, cli_yesno(), to produce the
	  output Yes or No for CLI lines. This helps maintaining
	  consistency on output, slightly improves readability, and maybe
	  one day will make it easier to translate the output in other
	  languages (though i have a hard time believing that a CLI user
	  who needs 'yes' and 'no' to be translated can actually figure out
	  what he/she is doing!)

	* channels/chan_sip.c: move the two remaining peer flags to proper
	  variables.

	* channels/chan_sip.c: move RT_FROMCONTACT to a proper sip_peer
	  field.

	* channels/chan_sip.c: Move some global 'flags' to individual
	  variables. Start putting these variables in a single struct
	  (called 'sip_cfg' for the time being, but it could as well be
	  'global' or some other name) so it is easy, when reading the
	  code, to figure out what they are for. The downside of using
	  struct fields instead of individual global variables is that the
	  compiler cannot tell if there are unused fields. But the
	  advantage of not cluttering the namespace and manilpulating all
	  these variables at once certainly overcome the disadvantagess.
	  Nothing to backport, again.

	* channels/chan_sip.c: minor simplification of a conditional
	  statement

	* channels/chan_sip.c: build the version of sip_tech with no
	  send_digit_begin at load time instead of duplicating the
	  initializer. This should remove the risk of forgetting fields in
	  the initializer.

	* channels/chan_sip.c: remove bit position from description of
	  SIP_* flags. use AST_FORMAT_AUDIO_MASK instead of playing with
	  AST_FORMAT_MAX_AUDIO to determine audio formats. There is a
	  dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call() which
	  surely needs fixing, namely: /* mask request with some set of
	  allowed formats. * XXX this needs to be fixed. * The original
	  code uses AST_FORMAT_AUDIO_MASK, but it is * unclear what to use
	  here. We have global_capabilities, which is * configured from
	  sip.conf, and sip_tech.capabilities, which is * hardwired to all
	  audio formats. */ The latter is possibly something to backport
	  when fixed.

	* channels/chan_sip.c: back on cleaning up the usage of flags. Move
	  together flags used in the same way (e.g. dialog only,
	  dialog-peer, ...) so it will become easier to deal with them in a
	  more systematic way. This is being done in stages so it will be
	  easier to detect breakage, if any should occur.

	* channels/chan_sip.c: more documentation on internal
	  representation of incoming SIP messages. Remove definitions for
	  now-unused flags, and add references to print routines for other
	  flags.

	* channels/chan_sip.c: make register_unref() return NULL so it is
	  easy to cleanup the original pointer while calling the function.
	  on passing add some comments on one of the places where it is
	  used, and explain why it is safe there. again, a no-op for
	  practical purposes.

	* channels/chan_sip.c: add some documentation to auto_congest(),
	  and some dialog_ref/unref (they are a no-op at the moment). Also
	  clean a pointer after freeing memory to avoid dangling
	  references, and write a for() loop in canonical form. In
	  practice, everything in this commit is a no-op.

	* channels/chan_sip.c: more dialog_ref()/dialog_unref() calls

	* channels/chan_sip.c: more dialog_ref()/dialog_unref() calls

	* channels/chan_sip.c: start introducing hooks for reference counts
	  on dialog descriptors. This commit is, for all practical
	  purposes, a no-op, as it only introduces the dialog_ref() and
	  dialog_unref() methods, and uses them in a few places (not all
	  the places where they would be needed). The goal is to start
	  annotating the code with these calls, so the transition to a
	  proper container will be easier. Nothing to backport.

	* channels/chan_sip.c: remove an unused string

	* channels/chan_sip.c: simplify a conditional expression using S_OR

	* channels/chan_sip.c: make use of received= and rport= fields in
	  sip replies. In a nutshell, these fields are used to tell a sip
	  entity the address and port its request came from, and are
	  extremely useful in the presence of NATs, especially with
	  symmetric NATs where STUN is totally ineffective. This patch
	  stores the address and port in the 'ourip' field of the dialog
	  descriptor, so they can be reused in subsequent transactions. As
	  it is, it works well for things like REGISTER requiring
	  authentication, because the second REGISTER request (with auth
	  credentials) will carry the correct address. Maybe it can also be
	  useful, in case of an address change, to do one or both of the
	  following: + propagate the new address to the parent user/peer
	  descriptor so that new dialogs will use the correct address from
	  the beginning. This is trivial to implement, I am just waiting
	  for feedback on this. + re-issue a request in case of an address
	  change. This a lot less trivial, maybe unnecessary, and probably
	  covered by the previous item. I would seriously consider this
	  patch for addition to 1.4 and 1.2. The code is very little
	  intrusive, and it would solve in a correct way the nat traversal
	  problems for which externip/externaddr/stunaddr are only a
	  partial and expensive workaround.

2007-07-27 23:21 +0000 [r77572-77603]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Some ODBC drivers don't set the
	  CHAR_OCTET_LENGTH field correctly.

	* Makefile: Target asterisk.pdf stopped building when the build was
	  moved to the doc directory.

	* /, res/res_odbc.c: Merged revisions 77571 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77571 | tilghman | 2007-07-27 13:15:58 -0500 (Fri, 27 Jul 2007)
	  | 2 lines Missing newline ........

2007-07-27 17:05 +0000 [r77537-77541]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 77540 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77540 | file | 2007-07-27 14:04:08 -0300 (Fri, 27 Jul 2007) | 6
	  lines (closes issue #10310) Reported by: prashant_jois Patches:
	  cdr_pgsql.patch uploaded by prashant (license 114) Finish the
	  Postgresql connection after the log messages are printed so we
	  don't access invalid memory. ........

	* channels/chan_sip.c: Turn 4 lines of code into 1 line that does
	  the same thing.

	* /, channels/chan_sip.c: Merged revisions 77536 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6
	  lines (closes issue #10323) Reported by: julianjm Patches:
	  chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm
	  (license 99) Clear ONHOLD flag when decrementing the onHold peer
	  count. If we did not do this the count may keep decreasing.
	  ........

2007-07-27 16:20 +0000 [r77534]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_config.c: 'dialplan save' shouldn't be converting '|'
	  back to ',' anymore.

2007-07-27 15:46 +0000 [r77520]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, pbx/pbx_ael.c: These fixes take care of two
	  problems: a complaint in asterisk-dev that goto's aren't working
	  in trunk, a side effect of the move to commas as arg seps in apps
	  and funcs; and a problem I spotted myself with dial's 'e' option,
	  where gotos were off by one, because I forgot to set the AUTOLOOP
	  flag in the peer channel.

2007-07-27 14:31 +0000 [r77491]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 77490 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul
	  2007) | 3 lines "re-invite" was misspelled ........

2007-07-26 23:20 +0000 [r77461]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 77460 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4
	  lines (closes issue #10302) Reported by: litnialex If a DTMF end
	  frame comes from a channel without a begin and it is going to a
	  technology that only accepts end frames (aka INFO) then use the
	  minimum DTMF duration if one is not in the frame already.
	  ........

2007-07-26 22:17 +0000 [r77432]  Kevin P. Fleming <kpfleming@digium.com>

	* /, doc/tex/mp3.tex, sounds/Makefile: Merged revisions 77424,77429
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007)
	  | 2 lines use new canonical name for download server ........
	  r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007)
	  | 2 lines change protocol for downloads as well ........

2007-07-26 21:24 +0000 [r77411]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 77410 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) |
	  10 lines AST_DEVMODE was defined in trunk, but not in 1.4. When
	  Asterisk is compiled under dev mode, AST_DEVMODE will get defined
	  in buildopts.h. Change 1.4 to define it in the same way that
	  trunk does. Also, revert the change that added this define in the
	  Makefile The advantage to doing it this way is that buildopts.h
	  gets installed when you install Asterisk. Then, when building any
	  out of tree modules, or building asterisk-addons, these modules
	  know which options the rest of Asterisk was built with. ........

2007-07-26 20:39 +0000 [r77381]  Mark Michelson <mmichelson@digium.com>

	* Makefile, /, main/logger.c: Merged revisions 77380 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26
	  Jul 2007) | 7 lines Fixes to get ast_backtrace working properly.
	  The AST_DEVMODE macro was never defined so the majority of
	  ast_backtrace never attempted compilation. The makefile now
	  defines AST_DEVMODE if configure was run with --enable-dev-mode.
	  Also, changes were made to acccomodate 64 bit systems in
	  ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
	  their roles in allowing me to get this committed ........

2007-07-26 19:33 +0000 [r77349-77351]  Tilghman Lesher <tlesher@digium.com>

	* /, main/logger.c: Merged revisions 77350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007)
	  | 2 lines Missed one ........

	* /, main/logger.c: Merged revisions 77348 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007)
	  | 2 lines Oops, that builtin define should be all-lowercase.
	  ........

2007-07-26 18:31 +0000 [r77319]  Mark Michelson <mmichelson@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 77318 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul
	  2007) | 8 lines Two consecutive calls to PQfinish could occur,
	  meaning free gets called on the same variable twice. This patch
	  sets the connection to NULL after calls to PQfinish so that the
	  problem does not occur. Also in this patch, prashant_jois
	  informed me that it is safe to pass a null pointer to PQfinish,
	  so I have removed the check for conn's existence from
	  my_unload_module. (closes issue 10295, reported by junky, patched
	  by me with input from prashant_jois) ........

2007-07-26 15:49 +0000 [r77268-77299]  Russell Bryant <russell@digium.com>

	* main/udptl.c, res/res_features.c, main/say.c,
	  codecs/codec_adpcm.c, apps/app_alarmreceiver.c,
	  cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c,
	  main/indications.c, main/config.c, main/loader.c, res/res_smdi.c,
	  pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_zapscan.c,
	  apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c,
	  apps/app_amd.c, cdr/cdr_odbc.c, res/res_speech.c,
	  apps/app_dial.c, codecs/codec_g722.c, funcs/func_timeout.c,
	  codecs/codec_speex.c, channels/chan_agent.c, codecs/codec_g726.c,
	  channels/iax2-provision.c, apps/app_db.c, channels/chan_misdn.c,
	  main/srv.c, apps/app_waitforring.c, apps/app_macro.c,
	  apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_vpb.cc,
	  apps/app_meetme.c, res/res_snmp.c, codecs/codec_gsm.c,
	  res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c,
	  res/res_jabber.c, main/channel.c, main/cdr.c,
	  channels/chan_phone.c, main/dial.c, res/res_config_odbc.c,
	  main/manager.c, funcs/func_odbc.c, res/res_agi.c, main/app.c,
	  main/image.c, apps/app_rpt.c, apps/app_parkandannounce.c,
	  channels/chan_mgcp.c, apps/app_adsiprog.c, apps/app_while.c,
	  codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c,
	  channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c,
	  main/translate.c, codecs/codec_alaw.c, apps/app_waitforsilence.c,
	  res/res_crypto.c, apps/app_queue.c, apps/app_getcpeid.c,
	  channels/chan_oss.c, main/rtp.c, apps/app_flash.c,
	  main/abstract_jb.c, main/file.c, channels/chan_h323.c,
	  codecs/codec_ulaw.c, pbx/pbx_dundi.c, apps/app_sms.c,
	  pbx/pbx_gtkconsole.c: Do a massive conversion for using the
	  ast_verb() macro (closes issue #10277, patches by mvanbaak)
	  Basically, this changes ... if (option_verbose > 2)
	  ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3,
	  "Something\n");

	* doc/tex/odbcstorage.tex, doc/tex/hardware.tex, doc/tex/mp3.tex,
	  doc/tex/channelvariables.tex, doc/tex/qos.tex,
	  doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex,
	  doc/tex/dundi.tex, doc/tex/enum.tex, doc/tex/asterisk-conf.tex,
	  doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/imapstorage.tex,
	  doc/tex/privacy.tex, LICENSE, doc/tex/app-sms.tex,
	  doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Merge a big batch of
	  documentation fixes for escaping, marking URLs, places where
	  verbatim text went off the end of the page on the PDF, and
	  various other improvements (closes issue #10307, IgorG)

	* channels/chan_sip.c: Revert some changes to call abs() on the
	  result of ast_random(). * random() is defined to return a
	  positive result, and now ast_random() will always do so as well

	* main/utils.c: Ensure that the read from /dev/urandom returns a
	  positive result (closes issue #10308, reported by yehavi, patched
	  by me)

2007-07-26 13:19 +0000 [r77267]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Things expecting a positive result from
	  ast_random() should not be surprised (closes #10308)

2007-07-26 13:10 +0000 [r77266]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Add a link to the list of assigned RTP payload types
	  for convenience.

2007-07-26 05:35 +0000 [r77233-77248]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: document how the RTP marker bit is passed for video
	  frames, and why this does not overwrite useful information.

	* main/rtp.c: add an entry for h263plus in an empty slot of the rtp
	  types.

2007-07-26 01:33 +0000 [r77217-77218]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: The upgrade of application argument separators
	  to comma has an effect on AEL; I commented out the code that
	  substitutes commas with vertbars, so we can get apps to parse
	  their args correctly.

	* apps/app_meetme.c: Merged revisions 77191 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1
	  line This fix solves problem with intense squelch noise when
	  someone joins conf in bug 9430; We repro'd the problem with
	  meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm
	  applying it. It looks like playing the recorded username will
	  louse up the next thing played into the channel. Josh rearranged
	  the code so as to start things over before playing data directly
	  into the conference. ........

2007-07-25 22:18 +0000 [r77182]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 77176 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul
	  2007) | 4 lines (closes issue #10303) Reported by: jtodd Add
	  SPEECH_DTMF_TERMINATOR variable so the user can specify the digit
	  to terminate a DTMF string with. If none is specified then no
	  terminator will be used. ........

2007-07-25 21:58 +0000 [r77156]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_iax2.c: silence a warning in ast-devmode on a
	  potentially uninitialized var. At first sight (but the function
	  is very large so i am not 100% sure) the code seems correct, so
	  maybe my compiler is just not smart enough to figure that out at
	  the optimization level it has. Not worthwhile merging to 1.4 i
	  believe.

2007-07-25 21:53 +0000 [r77155]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 77154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul
	  2007) | 3 lines chan->emulate_dtmf_duration is an unsigned int,
	  not a signed int, so use %u instead of %d in the format string
	  ........

2007-07-25 17:16 +0000 [r77072]  Joshua Colp <jcolp@digium.com>

	* /, configure, acinclude.m4: Merged revisions 77071 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul
	  2007) | 2 lines Fix autoconf logic for finding OpenH323 when it
	  is not in the first place searched (/usr/share/openh323).
	  ........

2007-07-25 14:13 +0000 [r77023-77054]  Luigi Rizzo <rizzo@icir.org>

	* main/translate.c: change the debug level to 3 for an exceedingly
	  annoying message (3-deep nested loop)

	* main/rtp.c: Merged revisions 77022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3
	  lines set the sequence number in a frame for all frame types
	  ........

2007-07-25 01:06 +0000 [r76985]  Russell Bryant <russell@digium.com>

	* CHANGES: remove a couple of entries that got duplicated and snuck
	  into the SIP section. Also, align the NAT/STUN entry with the
	  others.

2007-07-25 00:34 +0000 [r76984]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 76983 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue,
	  24 Jul 2007) | 9 lines Merged revisions 76978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
	  line this fixes bug 10293, where the error message because
	  defaultzone or loadzone was not defined was confusing ........
	  ................

2007-07-24 22:13 +0000 [r76874-76940]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 76937 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r76937 | tilghman | 2007-07-24 17:12:43 -0500
	  (Tue, 24 Jul 2007) | 10 lines Merged revisions 76934 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
	  Jul 2007) | 2 lines Oops, res contains the error code, not errno.
	  I was wondering why a mutex was reporting "No such file or
	  directory"... ........ ................

	* build_tools/cflags.xml: Add the flag to trigger an intentional
	  crash on mutex errors

	* doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/jitterbuffer.tex,
	  doc/tex/odbcstorage.tex, doc/tex/hardware.tex,
	  doc/tex/privacy.tex, doc/tex/billing.tex, doc/tex/ael.tex,
	  doc/tex/channelvariables.tex, doc/tex/qos.tex,
	  doc/tex/realtime.tex, doc/tex/asterisk.tex, doc/tex/queuelog.tex:
	  Fix escaping and some of the formattting (closes issue #10285)

2007-07-24 17:43 +0000 [r76841-76852]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Revert trivial whitespace change (for
	  testing)

	* channels/chan_skinny.c: Trivial whitespace change to test
	  comitting...

2007-07-24 17:05 +0000 [r76807]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 76803 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3
	  lines Don't create the Asterisk channel until we are starting the
	  PBX on it. (ASA-2007-018) ........

2007-07-24 16:42 +0000 [r76804]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 76801 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul
	  2007) | 13 lines Added a membercount variable to call_queue
	  struct which keeps track of the number of logged in members in a
	  particular queue. This makes it so that the 'n' option for
	  Queue() can act properly depending on which strategy is used. If
	  the strategy is roundrobin, rrmemory, or ringall, we want to ring
	  each phone once before moving on in the dialplan. However, if any
	  other strategy is used, we will only ring one phone since it
	  cannot be guaranteed that a different phone will ring on
	  subsequent attempts to ring a phone. As a side effect of this,
	  the QUEUE_MEMBER_COUNT dialplan function now just reads the
	  membercount variable instead of traversing through the member
	  list to figure out how many members there are. Special thanks to
	  blitzrage for helping to test this out. (closes issue #10127,
	  reported by bcnit, patched by me, tested by blitzrage) ........

2007-07-24 16:09 +0000 [r76791]  Joshua Colp <jcolp@digium.com>

	* sounds/Makefile: Don't download/install the sound packages if
	  already installed.

2007-07-24 15:35 +0000 [r76785]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: The chan_skinny Dial() syntax was funky.
	  You had to do Dial(Skinny/line@device) This allows you to just
	  Dial(Skinny/line), as long as line isn't ambiguous. Note that
	  this does not remove or deprecate the "old" syntax, as it's still
	  quite useful - even moreso if shared lines get implemented.
	  Initial patch by me, with some changes and suggestions from
	  wedhorn. (closes issue #10263)

2007-07-24 14:49 +0000 [r76755-76770]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: two small fixes when using stun (reported by
	  Marta Carbone): + externexpire was not initialized properly; +
	  stunaddr was not handled properly on a sip reload

	* CHANGES: add documentation on nat/stun support in chan_sip

2007-07-24 02:59 +0000 [r76710-76712]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Move manager users list over to an rwlist.

	* res/res_agi.c: You need to put static in front of a static RWLIST
	  declaration to make it really static... and don't call
	  AST_RWLIST_HEAD_DESTROY on a statically declared list.

	* main/manager.c: Don't bother calling AST_RWLIST_EMPTY on a list
	  before AST_RWLIST_TRAVERSE, it's just a double check.

2007-07-23 22:41 +0000 [r76707-76709]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 76708 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007)
	  | 4 lines It was our stated intention for 1.4 that files created
	  in app_voicemail should depend upon the umask. Unfortunately,
	  mkstemp() creates files with mode 0600, regardless of the umask.
	  This corrects that deficiency. ........

	* include/asterisk/agi.h, res/res_agi.c: Enhance AGI with several
	  fixes: - Makes the structures handling external AGI commands a
	  bit more thread-safe - Makes AGI transparently work with both
	  live and hungup channels - DeadAGI is hence no longer necessary
	  and is deprecated - CLI bug fixes - Commands will refuse to run
	  if the channel is dead and the command is nonsensical for dead
	  channels.

2007-07-23 21:42 +0000 [r76706]  Joshua Colp <jcolp@digium.com>

	* res/res_crypto.c: Clean up res_crypto module. It now uses an
	  rwlist to keep the keys and it should also be thread safe now.

2007-07-23 20:27 +0000 [r76703-76704]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c, UPGRADE.txt: Missed one conversion to comma
	  delimiter (thanks, Juggie) and add documentation on the change to
	  the Local channel name.

	* funcs/func_rand.c, apps/app_readfile.c, channels/chan_local.c,
	  apps/app_record.c, funcs/func_env.c, funcs/func_strings.c,
	  funcs/func_vmcount.c, include/asterisk/aes.h, funcs/func_logic.c,
	  apps/app_exec.c, apps/app_controlplayback.c, funcs/func_odbc.c,
	  apps/app_skel.c, apps/app_zapras.c, apps/app_url.c,
	  apps/app_externalivr.c, apps/app_parkandannounce.c,
	  apps/app_dial.c, main/pbx.c, apps/app_page.c,
	  apps/app_softhangup.c, UPGRADE.txt, funcs/func_cut.c,
	  apps/app_talkdetect.c, apps/app_queue.c, funcs/func_realtime.c,
	  include/asterisk/app.h, apps/app_channelredirect.c,
	  apps/app_macro.c, pbx/pbx_config.c, apps/app_verbose.c,
	  apps/app_chanspy.c, funcs/func_callerid.c, apps/app_voicemail.c:
	  Merge the dialplan_aesthetics branch. Most of this patch simply
	  converts applications using old methods of parsing arguments to
	  using the standard macros. However, the big change is that the
	  really old way of specifying application and arguments separated
	  by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the
	  way that has been recommended since long before 1.0 will become
	  the only method available (e.g. NoOp(foo,bar).

2007-07-23 19:00 +0000 [r76657]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 76656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul
	  2007) | 3 lines Fix some incorrect softkey labels in messages.
	  Don't try to play dialtone in some unimplemented features.
	  ........

2007-07-23 18:31 +0000 [r76655]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 76654 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r76654 | file | 2007-07-23 15:29:48 -0300 (Mon,
	  23 Jul 2007) | 12 lines Merged revisions 76653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4
	  lines (closes issue #5866) Reported by: tyler Do not force
	  channel format changes when a generator is present. The generator
	  may have changed the formats itself and changing them back would
	  cause issues. ........ ................

2007-07-23 17:58 +0000 [r76621]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 76620 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10276) ........ r76620 | qwell | 2007-07-23 12:57:53 -0500
	  (Mon, 23 Jul 2007) | 4 lines Don't try to queue up hold/unhold
	  frames on a non-existent channel. Issue 10276. ........

2007-07-23 17:49 +0000 [r76619]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_morsecode.c: Merged revisions 76618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2
	  lines Allow app_morsecode to build on PPC Linux by putting the
	  value of the digit char in an int. ........

2007-07-23 14:45 +0000 [r76564]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: add two missing entries in the replica of
	  the sip_tech that does not use DTMF BEGIN frames. 1.4 seems
	  correct (it does not have the two fields). However, as this bug
	  shows, the current way of creating the sip_tech replica is too
	  error-prone, one can easily forget to update one of the two
	  entries. Perhaps it would be better to create sip_tech_info
	  expliclty at module load, by doing sip_tech_info = sip_tech;
	  sip_tech_info.send_digit_begin = NULL (in this case, this is
	  something applicable to 1.4 as well).

2007-07-23 14:38 +0000 [r76563]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r76561 | file | 2007-07-23 11:34:21 -0300 (Mon,
	  23 Jul 2007) | 14 lines Merged revisions 76560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
	  lines (closes issue #10236) Reported by: homesick Patches:
	  rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
	  Remote Party ID on guest calls. ........ ................

2007-07-23 14:37 +0000 [r76555-76562]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Mark str2dtmfmode() as currently unused to
	  resolve a compiler warning and allow building under dev mode

	* include/asterisk.h, res/res_snmp.c, channels/chan_sip.c,
	  res/res_crypto.c, res/res_convert.c, main/devicestate.c,
	  include/jitterbuf.h, res/res_config_sqlite.c, main/enum.c,
	  res/res_monitor.c, include/asterisk/file.h,
	  include/asterisk/doxyref.h, res/res_config_odbc.c,
	  res/res_indications.c, main/asterisk.c, res/res_clioriginate.c:
	  (closes issue #10271) Reported by: snuffy Patches:
	  doxygen-updates.diff uploaded by snuffy (license 35) Another big
	  batch of doxygen documentation updates

	* CHANGES: note the debug and verbose changes in CHANGES

	* include/asterisk/logger.h, main/pbx.c, main/logger.c,
	  include/asterisk/options.h, main/asterisk.c, main/cli.c: (closes
	  issue #10192) Reported by: bbryant Patches:
	  20070720__core_debug_by_file.patch uploaded by bbryant (license
	  36) (with some modifications by me) Tested by: russell, bbryant
	  This set of changes introduces the ability to set the core debug
	  or verbose levels on a per-file basis. Interestingly enough, in
	  1.4, you have the ability to set core debug for a single file,
	  but that functionality was accidentally lost in the conversion of
	  the CLI commands to the new format. This patch improves upon what
	  was in 1.4 by letting you set it for more than 1 file, and by
	  also supporting verbose. *** Janitor Project *** This patch also
	  introduces a new macro, ast_verb(), which is similar to
	  ast_debug(). Setting the per file verbose value only works for
	  messages that use this macro. Converting existing uses of
	  ast_verbose() can be done like: if (option_debug > 2)
	  ast_verbose(VERBOSE_PREFIX_3 "Something useful\n"); ...
	  ast_verb(3, "Something useful\n");

2007-07-23 14:18 +0000 [r76547]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: introduce two functions, map_x_s() and
	  map_s_x(), to map between integers and strings using a single
	  translation table, and use them in a few places instead of ad-hoc
	  routines that duplicate the table. On passing, note that
	  REFER_CONFIRMED is never used, and add a few comments. Nothing to
	  backport here.

2007-07-23 14:02 +0000 [r76524]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove an unused function to resolve a
	  compiler warning

2007-07-23 13:46 +0000 [r76523]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Use autoconf
	  logic to determine byte swapping macro presence. This should now
	  also use other macros if present.

2007-07-23 13:29 +0000 [r76521]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: move "sip prunte realtime ..." and "sip set
	  debug ... " to NEW_CLI style.

2007-07-23 13:24 +0000 [r76520]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 76519 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul
	  2007) | 6 lines (closes issue #10268) Reported by: mvanbaak
	  Patches: chan_skinny_openbsd.diff uploaded by mvanbaak (license
	  7) Add another OS that has to use the Macros for byte ordering.
	  ........

2007-07-23 12:29 +0000 [r76486]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 76485 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76485 | russell | 2007-07-23 07:25:01 -0500 (Mon, 23 Jul 2007) |
	  6 lines Use a signed integer for storing the number of bytes in
	  the packet read from the network. Using an unsigned value here
	  made it impossible to handle an error returned from recvfrom().
	  Furthermore, in the case that recvfrom() did return an error,
	  this would cause a crash due to a heap overflow. (closes issue
	  #10265, reported by and fix suggested by timrobbins) ........

2007-07-23 03:10 +0000 [r76313-76467]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: Add some documentation on the sipregistry
	  states and the handling of the sip_register structures. This
	  commit only changes comments and whitespace.

	* channels/chan_sip.c: add a bit of comments on internal functions.

	* channels/chan_sip.c: rewrite "sip show {channels|subscriptions}"
	  CLI handler using the new-style cli format. No functional
	  changes, nothing to backport.

	* channels/chan_sip.c: Make sip_destroy() return NULL so the caller
	  can do things like foo = sip_destroy(foo); and reduce the chance
	  of bugs due to dangling pointers. Also remove a duplicate
	  prototype for the function. nothing to backport.

	* channels/chan_sip.c: add two comment blocks, one on reusing
	  nonces, and one on the handling of an 'authpeer' local variable.

	* channels/chan_sip.c: comment and slightly restructure
	  handle_request() in the part that handles responses, so that
	  there is a common exit point. Mark two places where probably we
	  could return -1 instead of 0 to report an error to the caller.
	  (change triggered by investigations on how the 'SIP_PKT_IGNORE'
	  field was used). nothing to backport from this commit

	* channels/chan_sip.c: remove unused argument from
	  handle_invite_replaces(), and also leftover SIP_PKT_* stuff from
	  the previous commit.

	* channels/chan_sip.c: Cleanup of flags used in struct sip_request,
	  moving them to individual variables. Apart from SIP_PKT_IGNORE
	  which was used a zillion times, the other two are used seldom. On
	  passing: - move the arrays to the end of struct sip_request, so a
	  (small) buffer overflow is less likely to overwrite the other
	  fields; - note that the 'ignore' argument to
	  handle_invite_replaces() is not used and should be removed (will
	  be done in a separate commit). Nothing to backport in this
	  change.

	* channels/chan_sip.c: move two per-packet flags to proper
	  variables.

	* channels/chan_sip.c: minor clarification on the usage of SIP_*
	  flags. Also correct some items that were misclassified.

	* channels/chan_sip.c: document the way sipdebug works, and
	  implement it through variables and not flags. NOTE: The old
	  behaviour (preserved in this commit) is that if sipdebug is set
	  in the config file, it can only be disabled by reloading the
	  config. I am not sure if this is accidental or voluntary, but it
	  is really unconvenient and I think it should be handled in the
	  same way as other options i.e. consider requests from the config
	  file or the cli (or the command line) to be fully equivalent and
	  act on the same status variable.

	* channels/chan_sip.c: move the SIP_REALTIME flag to a field in the
	  user/peer structure.

	* channels/chan_sip.c: Add a note to document how the temporary
	  'pvt' should be initialized before using it. I am unclear on the
	  details right now so i hope someone can comment more. The obvious
	  (and lazy) approach would be to bzero() all of it (except for the
	  string pool), but isn't that too much work ? Feedback wanted
	  here...

2007-07-21 14:39 +0000 [r76296]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/utils.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Add
	  support for using /dev/urandom to get random numbers on systems
	  that support it.

2007-07-21 09:35 +0000 [r76229-76279]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: whoops... was setting needdestroy on the
	  wrong dialog. (spotted by a diff with my own branch)

	* channels/chan_sip.c: more two more flags to proper variables:
	  ALREADYGONE and NEEDDESTROY.

	* channels/chan_sip.c: use explicit variables for things that don't
	  need to be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
	  replaced by a 'do_history' field in the sip_pvt structure.

	* channels/chan_sip.c: Use ast_str_append() instead of
	  ast_build_string() to construct the sdp messages. Overall the
	  code is slightly more readable (because the string is fully
	  described by a single pointer), and more efficient (because the
	  length is stored explicitly so you don't need to do strlen()). (I
	  have been using this code for almost a year now.) I wish we had
	  infix string operators to do this sort of things! Nothing to
	  backport from this change.

2007-07-21 01:25 +0000 [r76224]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: We have two 'technology' descriptors for a
	  SIP channel, so define and use a macro to determine whether we
	  are pointing to one of them, so when one goes away (or a new one
	  appears) we don't have to touch all the code.

2007-07-21 01:08 +0000 [r76222]  Steve Murphy <murf@digium.com>

	* apps/app_queue.c: One small documentation update made to
	  accompany 10154, the upgrading of the queue ringing to allow
	  periodic announcments

2007-07-21 01:01 +0000 [r76221]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c, configs/sip.conf.sample: Enhance NAT support
	  as discussed on the -dev list, i.e.: + extensive documentation
	  changes both in sip.conf.sample and in the source; + allow
	  "externip" and "externhost" to include a port number as well; +
	  allow "bindaddr" to have a port number (making bindport
	  unnecessary, even though it is still present for backward
	  compatibility); + introduce the new "stunaddr" parameter to
	  specify an STUN server to be used from the main SIP socket; +
	  extend the "sip show settings" output to show all the above.
	  Internally: + change related data structures from struct in_addr
	  to struct sockaddr_in to store the port numbers as well; +
	  reorganize ast_sip_ouraddrfor() (should also be renamed to
	  sip_ouraddrfor() because it is not a generic API, though it might
	  become so if called with a socket as an additional argument, in
	  which case it can be moved elsewhere). As mentioned in the
	  documentation, media sessions still do not use STUN so the port
	  numbers may still be incorrect when Asterisk is behind a NAT On
	  passing, some of the debugging messages printing media addresses
	  are probably using the wrong values, but this will be
	  checked/fixed in a subsequent commit if needed. Part of the
	  following chunk in the function that handles a "sip reload" is
	  probably needed on previous versions as well, to avoid leaking
	  the memory used for the "localaddr" list: @@ -17244,13 +17274,17
	  @@ /* Reset IP addresses */ memset(&bindaddr, 0,
	  sizeof(bindaddr)); + memset(&stunaddr, 0, sizeof(stunaddr)); +
	  memset(&internip, 0, sizeof(internip)); + /* Free memory for
	  local network address mask */ + ---> ast_free_ha(localaddr);
	  <----- memset(&localaddr, 0, sizeof(localaddr));
	  memset(&externip, 0, sizeof(externip)); memset(&default_prefs, 0
	  , sizeof(default_prefs));

2007-07-21 00:57 +0000 [r76220]  Steve Murphy <murf@digium.com>

	* apps/app_queue.c: This update was supplied in 10154; to allow
	  announcemnts if the 'r' option (ringing) is provided.

2007-07-20 22:25 +0000 [r76216]  Jason Parker <jparker@digium.com>

	* configs/say.conf.sample, apps/app_playback.c: Add support for
	  default "say mode" (whether to use the "old" method or "new"
	  method. "new" method being config file) Add support for
	  autocomplete of "say load" CLI command. Patch by IgorG (closes
	  issue #10243)

2007-07-20 21:41 +0000 [r76213]  Steve Murphy <murf@digium.com>

	* /, sounds/Makefile: Merged revisions 76211 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76211 | murf | 2007-07-20 15:36:05 -0600 (Fri, 20 Jul 2007) | 1
	  line This patch from 10249 is worth applying! It prevents
	  downloading sound files if they are already downloaded. Darn
	  Practical, if you ask me ........

2007-07-20 21:04 +0000 [r76175-76179]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 76174 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r76174 | qwell | 2007-07-20 15:32:55 -0500 (Fri, 20 Jul
	  2007) | 2 lines It's possible for sub->owner to be NULL here if
	  you cancel the call immediately after/during sending a digit.
	  ........

2007-07-20 18:44 +0000 [r76140]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_directory.c: Merged revisions 76139 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul
	  2007) | 6 lines When using users.conf for the entries in the
	  directory, if multiple users had the same last name, only the
	  first user listed would be available in the directory. (closes
	  issue #10200, reported by mrskippy, patched by me) ........

2007-07-20 18:28 +0000 [r76138]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 76132 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) |
	  6 lines Use the define that specifies the default length of an
	  artificially created DTMF digit in the ast_senddigit() function.
	  The define is set to 100ms by default, which is the same thing
	  that this function was using. But, using the define lets changes
	  take effect in this case, as well as the others where it was
	  already used. ........

2007-07-20 17:21 +0000 [r76055-76091]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r76087 | file | 2007-07-20 14:20:09 -0300 (Fri,
	  20 Jul 2007) | 14 lines Merged revisions 76080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
	  lines (closes issue #10247) Reported by: fkasumovic Patches:
	  chan_sip.patch uploaded by fkasumovic (license #101) Drop any
	  peer realm authentication entries when reloading so multiple
	  entries do not get added to the peer. ........ ................

	* /, res/res_convert.c: Merged revisions 76067 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r76067 | file | 2007-07-20 14:10:17 -0300 (Fri, 20 Jul 2007) | 6
	  lines (closes issue #10246) Reported by: fkasumovic Patches:
	  res_conver.patch uploaded by fkasumovic (license #101) Use the
	  last occurance of . to find the extension, not the first
	  occurance. ........

	* channels/chan_sip.c: It is impossible for the externhost variable
	  to not exist, it is however possible for it to be empty.

2007-07-20 15:06 +0000 [r76034-76037]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: Don't use a field size for the last argument
	  of printf format, because in this case the string is left-aligned
	  and it is not truncated anyways. Omitting the field size prevents
	  the generation of trailing whitespace, which makes the string fit
	  in smaller windows.

	* channels/chan_sip.c: Extend the 'network settings' section with
	  indication on the localnet settings (requires the change in SVN
	  76034), and also give an indication on whether/why/how the
	  remapping of addresses in SIP message is done or not. I think
	  this is especially useful for debugging the configuration, as the
	  address remapping depends on a combination of at least 3
	  parameters (localnet, externhost, externip) and successful DNS
	  lookup. An example of the output of this section is below:
	  Network Settings: --------------------------- SIP address
	  remapping: Enabled using externhost Externhost: foo.dyndns.net
	  Externip: 80.64.128.23:0 Externrefresh: 10 Internal IP:
	  12.34.56.78:5060 Localnet: 192.168.0.0/255.255.0.0
	  10.0.0.0/255.0.0.0 I leave to the community the judgement if the
	  above info is a useful addition for 1.4. It is not a bugfix, but
	  it is neither a new feature, only a useful diagnostic tool. Note
	  that I would like to move there also the bindaddress/port
	  information, in the usual addr:port format e.g. Bindaddress:
	  0.0.0.0:5060 so that network information is all in one place.

	* include/asterisk/acl.h, main/acl.c: expose struct ast_ha so
	  external code can do things such as printing it (e.g. chan_sip.c
	  in a subsequent commit). Obviously exposing the internals of a
	  data structure is far from ideal (especially in a case like this
	  where the implementation is very inefficient and will need to be
	  changed at some point). On the other hand, it was also unclear
	  what additional APIs should we provide instead, and because
	  exposing the stucture has no impact on source and binary
	  compatibility, this seemed to me the best option at this time.

2007-07-20 01:54 +0000 [r76015]  Tilghman Lesher <tlesher@digium.com>

	* main/logger.c: Reduce some logging contention by switching
	  several locks over to rwlocks

2007-07-19 23:24 +0000 [r75982-75983]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c,
	  channels/chan_sip.c, include/asterisk/dundi.h,
	  res/res_features.c, include/asterisk/chanspy.h,
	  include/asterisk/speech.h, channels/iax2-provision.c,
	  include/asterisk/cdr.h, include/asterisk/channel.h,
	  res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c,
	  channels/iax2-provision.h, main/loader.c,
	  include/asterisk/abstract_jb.h, include/asterisk/features.h,
	  main/channel.c, include/asterisk/app.h, funcs/func_odbc.c,
	  include/asterisk/module.h, include/asterisk/jabber.h,
	  apps/app_minivm.c, main/app.c, pbx/pbx_dundi.c,
	  apps/app_mixmonitor.c, apps/app_voicemail.c: After some study,
	  thought, comparing, etc. I've backed out the previous universal
	  mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit
	  version of ast_flags (ast_flags64), and 64-bit versions of the
	  test-flag, set-flag, etc. macros, and an app_parse_options64
	  routine, and I use these in app_dial alone, to eliminate the
	  30-option limit it had grown to meet. There is room now for 32
	  more options and flags. I was heavily tempted to implement some
	  of the other ideas that were presented, but this solution does
	  not intro any new versions of dial, doesn't have a different API,
	  has a minimal/zero impact on code outside of dial, and doesn't
	  seriously (I hope) affect the code structure of dial. It's the
	  best I can think of right now. My goal was NOT to rewrite dial. I
	  leave that to a future, coordinated effort.

	* apps/app_queue.c: This repairs a 'warning: ISO C90 forbids mixed
	  declarations and code' message that cripples my dev-mode enabled
	  build

2007-07-19 19:02 +0000 [r75977-75979]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 75978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75978 | mmichelson | 2007-07-19 13:59:30 -0500 (Thu, 19 Jul
	  2007) | 3 lines The diff on this looks pretty big but all I did
	  was remove a pointless if statement (always evaluates true).
	  ........

	* /, apps/app_queue.c: Merged revisions 75969 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul
	  2007) | 10 lines Changes in handling return values of several
	  functions in app_queue. This all started as a fix for issue
	  #10008 but now includes all of the following changes: 1.
	  Simplifying the code to handle positive return values from ast
	  API calls. 2. Removing the background_file function. 3. The fix
	  for issue #10008 (closes issue #10008, reported and patched by
	  dimas) ........

2007-07-19 15:59 +0000 [r75911-75930]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: (closes issue #10210, reported and patched by
	  juggie) This merges the trunk only part of the patches from this
	  issue. In 1.4, res_agi will issue a warning if you try to use
	  DeadAGI on a channel that is not hung up. Now, in trunk, it just
	  plain won't let you do it.

	* /, channels/chan_iax2.c: Merged revisions 75928 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75928 | russell | 2007-07-19 10:53:15 -0500
	  (Thu, 19 Jul 2007) | 14 lines Merged revisions 75927 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19
	  Jul 2007) | 6 lines When processing full frames, take sequence
	  number wraparound into account when deciding whether or not we
	  need to request retransmissions by sending a VNAK. This code
	  could cause VNAKs to be sent erroneously in some cases, and to
	  not be sent in other cases when it should have been. (closes
	  issue #10237, reported and patched by mihai) ........
	  ................

	* main/acl.c: Remove some debug code that was added in revision
	  75894, which removed some other debug code. :)

2007-07-19 12:38 +0000 [r75873-75894]  Luigi Rizzo <rizzo@icir.org>

	* main/acl.c: comment out some terribly expensive debugging code in
	  the body of ast_apply_ha()

	* channels/chan_sip.c: print more of the network settings
	  (externip, externhost etc.) in the "sip show settings" cli
	  output. I have put these in a separate section, probably even
	  bindaddr and SIP port should go there. There are more things to
	  add here e.g. localnet and so on.

	* channels/chan_sip.c: document the use of externip, externhost and
	  other nat-related options, as well as the handling of the sip
	  socket.

	* channels/chan_sip.c: ast_sip_ouraddrfor() never fails, so make it
	  void and remove the code that would never be called.

	* channels/chan_sip.c: portability fix: use %f instead of %lf when
	  printing double. The l is useless.

2007-07-19 04:45 +0000 [r75841-75857]  Tilghman Lesher <tlesher@digium.com>

	* channels/misdn/ie.c, channels/misdn/isdn_lib.c: Allow chan_misdn
	  to build in dev-mode

	* apps/app_rpt.c: Fix trunk where I broke it earlier (for
	  ast_strftime branch)

2007-07-18 23:00 +0000 [r75808]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75807 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r75807 | qwell | 2007-07-18 17:59:18 -0500 (Wed, 18 Jul
	  2007) | 1 line Need to make sure we set milliseconds and
	  timestamp - pointed out by the recent ast_ time stuff from
	  Tilghman ........

2007-07-18 22:52 +0000 [r75806]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: I thought I noticed a memory leak earlier
	  when I saw that the contents of this list were not destroyed when
	  the module is unloaded. However, after reading the code related
	  to the use of this list a lot today, I realized that it isn't
	  necessary. So, I have added a comment to explain why it isn't
	  necessary.

2007-07-18 22:40 +0000 [r75805]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Change IAX variables to use datastores
	  (closes issue #9315)

2007-07-18 21:10 +0000 [r75761]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75759 | russell | 2007-07-18 16:09:46 -0500
	  (Wed, 18 Jul 2007) | 13 lines Merged revisions 75757 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18
	  Jul 2007) | 5 lines When traversing the queue of frames for
	  possible retransmission after receiving a VNAK, handle sequence
	  number wraparound so that all frames that should be retransmitted
	  actually do get retransmitted. (issue #10227, reported and
	  patched by mihai) ........ ................

2007-07-18 20:43 +0000 [r75750]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 75749 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75749 | tilghman | 2007-07-18 15:40:18 -0500
	  (Wed, 18 Jul 2007) | 10 lines Merged revisions 75748 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18
	  Jul 2007) | 2 lines Store prior to copy (closes issue #10193)
	  ........ ................

2007-07-18 20:18 +0000 [r75714-75734]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75732 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul
	  2007) | 1 line Umm, why are we transmitting dialtone on cfwdall?
	  ........

	* /, channels/chan_skinny.c: Merged revisions 75711 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #9245) ........ r75711 | qwell | 2007-07-18 14:54:32 -0500
	  (Wed, 18 Jul 2007) | 4 lines Fixes for 7935/7936 conference
	  phones. Issue 9245, patch by slimey. ........

2007-07-18 19:51 +0000 [r75710]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75707 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #9887) ........ r75707 | qwell | 2007-07-18 14:48:12 -0500
	  (Wed, 18 Jul 2007) | 4 lines Fix issues with new 79x1 phones.
	  Issue 9887, patches by DEA ........

2007-07-18 19:50 +0000 [r75709]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: convert some lines indented with spaces to
	  tabs

2007-07-18 19:47 +0000 [r75706]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c, funcs/func_strings.c, main/utils.c,
	  apps/app_alarmreceiver.c, include/asterisk/localtime.h,
	  cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c,
	  main/loader.c, main/cli.c, cdr/cdr_csv.c, main/cdr.c,
	  channels/chan_phone.c, main/manager.c, channels/chan_skinny.c,
	  cdr/cdr_sqlite.c, apps/app_minivm.c, channels/misdn/ie.c,
	  main/logger.c, main/http.c, main/stdtime/localtime.c,
	  cdr/cdr_odbc.c, apps/app_rpt.c, include/asterisk/options.h,
	  channels/chan_mgcp.c, cdr/cdr_manager.c, main/pbx.c,
	  channels/chan_zap.c, funcs/func_timeout.c, channels/chan_sip.c,
	  channels/chan_agent.c, channels/iax2-parser.c,
	  apps/app_playback.c, cdr/cdr_tds.c, main/callerid.c,
	  res/snmp/agent.c, apps/app_sms.c, include/asterisk/strings.h,
	  main/asterisk.c, apps/app_voicemail.c: Merge in ast_strftime
	  branch, which changes timestamps to be accurate to the
	  microsecond, instead of only to the second

2007-07-18 17:59 +0000 [r75659]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, apps/app_queue.c: Merged revisions 75658 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75658 | dhubbard | 2007-07-18 12:56:30 -0500
	  (Wed, 18 Jul 2007) | 9 lines Merged revisions 75657 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18
	  Jul 2007) | 1 line removed the word 'pissed' from ast_log(...)
	  function call for BE-90 ........ ................

2007-07-18 15:45 +0000 [r75586-75624]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 75623 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2
	  lines Few more places that needs to check for onhold state.
	  ........

	* /, channels/chan_sip.c: Merged revisions 75621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5
	  lines (closes issue #10165) Reported by: elandivar It is possible
	  for hold status to exist without call limits set, so we need to
	  ensure update_call_counter is executed regardless. ........

	* /, channels/chan_h323.c: Merged revisions 75619 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2
	  lines Don't bother reloading chan_h323 if it did not load
	  successfully in the first place. This would otherwise cause a
	  crash. ........

	* funcs/func_curl.c: Clean up func_curl a bit.

2007-07-18 14:35 +0000 [r75585]  Steve Murphy <murf@digium.com>

	* main/channel.c, channels/chan_sip.c, res/res_features.c,
	  pbx/pbx_dundi.c, main/rtp.c, apps/app_voicemail.c: This corrects
	  the problem with flags and %lld formats on 64-bit machines, where
	  uint64_t is NOT acceptable for %lld, and also works on 32-bit
	  machines. At least, with gcc.

2007-07-18 14:20 +0000 [r75566-75584]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 75583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75583 | file | 2007-07-18 11:18:53 -0300 (Wed, 18 Jul 2007) | 5
	  lines (closes issue #10224) Reported by: irroot Record the
	  threadid of each running thread before shutting them down as the
	  thread themselves may change the value. ........

	* channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_realtime.c,
	  apps/app_voicemail.c: Minor code tweaks. Variables were being
	  checked wrong in some situations and didn't need to be checked in
	  others.

2007-07-18 12:38 +0000 [r75530]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_meetme.c: Merged revisions 75529 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75529 | tilghman | 2007-07-18 07:29:41 -0500 (Wed, 18 Jul 2007)
	  | 2 lines Using a freed frame causes crashes (closes issue #9317)
	  ........

2007-07-17 21:52 +0000 [r75505]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Spotted this bug today myself, trying to reproduce
	  a BE bug. Use a vert bar instead of a comma, when calling RAND.

2007-07-17 20:58 +0000 [r75446-75451]  Russell Bryant <russell@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75450 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75450 | russell | 2007-07-17 15:57:56 -0500
	  (Tue, 17 Jul 2007) | 11 lines Merged revisions 75449 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
	  Jul 2007) | 3 lines Properly check for the length in the skinny
	  packet to prevent an invalid memcpy. (ASA-2007-016) ........
	  ................

	* channels/iax2-parser.h, /, channels/chan_iax2.c,
	  channels/iax2-parser.c: Merged revisions 75445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75445 | russell | 2007-07-17 15:48:21 -0500
	  (Tue, 17 Jul 2007) | 13 lines Merged revisions 75444 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17
	  Jul 2007) | 5 lines Ensure that when encoding the contents of an
	  ast_frame into an iax_frame, that the size of the destination
	  buffer is known in the iax_frame so that code won't write past
	  the end of the allocated buffer when sending outgoing frames.
	  (ASA-2007-014) ........ ................

2007-07-17 20:42 +0000 [r75438-75442]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75441 | russell | 2007-07-17 15:42:12 -0500
	  (Tue, 17 Jul 2007) | 12 lines Merged revisions 75440 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17
	  Jul 2007) | 4 lines After parsing information elements in IAX
	  frames, set the data length to zero, so that code later on does
	  not think it has data to copy. (ASA-2007-015) ........
	  ................

2007-07-17 20:05 +0000 [r75406]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /: Merged revisions 75405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul
	  2007) | 6 lines Fixing an error I made earlier. ast_fileexists
	  can return -1 on failure, so I need to be sure that we only enter
	  the if statement if it is successful. Related to my fix to issue
	  #10186 ........

2007-07-17 20:01 +0000 [r75402-75404]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 75403 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) |
	  12 lines (closes issue #10209) Reported by: juggie Patches:
	  10209-trunk-2.patch uploaded by juggie Tested by: juggie,
	  blitzrage In ast_pbx_run(), mark a channel as hung up after an
	  application returned -1, or when it runs out of extensions to
	  execute. This is so that code can detect that this channel has
	  been hung up for things like making sure DeadAGI is used on
	  actual dead channels, and is beneficial for other things, like
	  making sure someone doesn't try to start spying on a channel that
	  is about to go away. ........

	* /, res/res_agi.c: Merged revisions 75401 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75401 | russell | 2007-07-17 14:45:07 -0500 (Tue, 17 Jul 2007) |
	  3 lines Remove a duplicated newline character in AGI debug
	  output. (closes issue #10207, patch by seanbright) ........

2007-07-17 19:40 +0000 [r75400]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c,
	  channels/chan_sip.c, include/asterisk/dundi.h,
	  res/res_features.c, include/asterisk/chanspy.h,
	  include/asterisk/speech.h, channels/iax2-provision.c,
	  include/asterisk/cdr.h, include/asterisk/channel.h,
	  res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c,
	  channels/iax2-provision.h, main/loader.c,
	  include/asterisk/features.h, include/asterisk/abstract_jb.h,
	  main/channel.c, funcs/func_odbc.c, include/asterisk/module.h,
	  include/asterisk/jabber.h, apps/app_minivm.c, utils/ael_main.c,
	  pbx/pbx_dundi.c, apps/app_mixmonitor.c, utils/check_expr.c,
	  apps/app_voicemail.c: via 10206, I have added an option (e) to
	  Dial to allow the h exten to get run on peer. Had to upgrade
	  ast_flag stuff to 64 bits to do this.

2007-07-17 14:48 +0000 [r75381]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/config.h: Make trunk build once again.

2007-07-17 14:32 +0000 [r75365-75379]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/config.h, main/config.c: Introduce
	  ast_parse_arg() , a generic function to parse strings in a
	  consistent way. This is meant to replace the custom code which is
	  repeated all over the place in the various files when parsing
	  config files, CLI entries and other string information. Right now
	  the code supports parsing int32, uint32 and sockaddr_in with
	  optional default values and bound checks. It contains minimal
	  error checking, but that can be easily extended as the need
	  arises. Being a new API i am introducing this only in trunk,
	  though I believe that once the interface has been ironed out it
	  might become a worthwhile addition to 1.4 as well - basically,
	  the first time we will need to fix a piece of argument parsing
	  code, we might as well bring in this change and use the new API
	  instead.

	* apps/app_minivm.c: Initialize a variable to avoid a warning when
	  the compiler (and/or the optimization level) may think it is used
	  uninitialized. The code was indeed correct, but unfortunately the
	  result of some compiler checks such as -Wunused and
	  -Wuninitialized depends heavily on the optimization level.

2007-07-17 12:01 +0000 [r75351]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c: Fix an incorrect parenthesization (TODO: Find a
	  better word) in app_dial Pointed out by Fanzhou Zhao Closes issue
	  #10216

2007-07-16 20:58 +0000 [r75307]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/dns.c: Merged revisions 75306 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75306 | kpfleming | 2007-07-16 15:53:24 -0500
	  (Mon, 16 Jul 2007) | 11 lines Merged revisions 75304 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16
	  Jul 2007) | 3 lines provide proper copyright/license attribution
	  for this structure that was copied from a BSD-licensed header
	  file long, long ago... ........ ................

2007-07-16 18:38 +0000 [r75255-75260]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: Change the function name
	  slightly... just for kpfleming!

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Add in
	  check for the GCC attribute deprecated. It may be used soon!

	* funcs/func_enum.c, funcs/func_rand.c, main/pbx.c,
	  funcs/func_curl.c, funcs/func_version.c, funcs/func_cut.c,
	  funcs/func_vmcount.c, include/asterisk/pbx.h,
	  funcs/func_realtime.c: For my next trick I will make it so
	  dialplan functions no longer need to call ast_module_user_add and
	  ast_module_user_remove. These are now called in the ast_func_read
	  and ast_func_write functions outside of the module.

2007-07-16 18:18 +0000 [r75254]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /: Merged revisions 75253 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul
	  2007) | 8 lines Restoring functionality from 1.2 wherein
	  Retrydial will not exit if there is no announce file specified.
	  This change makes it so that if there is no announce file
	  specified, the application will continue until finished (or
	  caller hangs up). If a bogus announce file is specified, then a
	  warning message will be printed saying that the file could not be
	  found, but execution will still continue. (closes issue #10186,
	  reported by jon, patched by me) ........

2007-07-16 15:57 +0000 [r75183-75227]  Joshua Colp <jcolp@digium.com>

	* apps/app_verbose.c: I found this sillyness when I did my
	  ast_module_user conversion. Return immediately if no data was
	  passed to the Verbose application.

	* apps/app_readfile.c, apps/app_record.c, apps/app_sayunixtime.c,
	  apps/app_test.c, apps/app_alarmreceiver.c, apps/app_image.c,
	  apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
	  apps/app_skel.c, apps/app_zapscan.c, apps/app_dumpchan.c,
	  apps/app_zapras.c, apps/app_amd.c, apps/app_url.c,
	  apps/app_externalivr.c, apps/app_milliwatt.c, apps/app_dial.c,
	  main/pbx.c, apps/app_page.c, apps/app_privacy.c, apps/app_echo.c,
	  apps/app_softhangup.c, apps/app_disa.c, apps/app_morsecode.c,
	  apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
	  apps/app_playback.c, apps/app_speech_utils.c,
	  apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
	  apps/app_macro.c, apps/app_zapateller.c, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, apps/app_cdr.c, apps/app_voicemail.c,
	  apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c,
	  apps/app_userevent.c, apps/app_followme.c,
	  apps/app_controlplayback.c, apps/app_osplookup.c,
	  apps/app_setcallerid.c, apps/app_minivm.c, apps/app_mp3.c,
	  apps/app_directory.c, apps/app_rpt.c, apps/app_ivrdemo.c,
	  apps/app_parkandannounce.c, apps/app_adsiprog.c,
	  apps/app_while.c, apps/app_nbscat.c, apps/app_read.c,
	  apps/app_festival.c, apps/app_system.c, apps/app_getcpeid.c,
	  apps/app_queue.c, apps/app_channelredirect.c, apps/app_forkcdr.c,
	  apps/app_flash.c, apps/app_directed_pickup.c, apps/app_sms.c,
	  include/asterisk/pbx.h, apps/app_senddtmf.c, apps/app_stack.c,
	  apps/app_verbose.c: Applications no longer need to call
	  ast_module_user_add and ast_module_user_remove. This is now taken
	  care of in the pbx_exec function outside of the application.

	* apps/app_readfile.c, res/res_features.c, apps/app_record.c,
	  apps/app_sayunixtime.c, apps/app_test.c,
	  apps/app_alarmreceiver.c, apps/app_image.c,
	  apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
	  apps/app_zapscan.c, apps/app_dumpchan.c, apps/app_zapras.c,
	  apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c,
	  apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
	  apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c,
	  apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c,
	  apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c,
	  apps/app_speech_utils.c, funcs/func_curl.c, apps/app_zapbarge.c,
	  apps/app_waitforring.c, apps/app_sendtext.c, apps/app_macro.c,
	  apps/app_zapateller.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
	  apps/app_cdr.c, apps/app_voicemail.c, apps/app_meetme.c,
	  apps/app_authenticate.c, apps/app_userevent.c,
	  funcs/func_vmcount.c, apps/app_followme.c, funcs/func_enum.c,
	  res/res_config_odbc.c, apps/app_setcallerid.c,
	  apps/app_osplookup.c, apps/app_minivm.c, res/res_agi.c,
	  apps/app_mp3.c, res/res_realtime.c, apps/app_rpt.c,
	  apps/app_ivrdemo.c, apps/app_parkandannounce.c,
	  apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c,
	  res/res_config_pgsql.c, apps/app_read.c, apps/app_festival.c,
	  apps/app_waitforsilence.c, apps/app_system.c, apps/app_queue.c,
	  apps/app_getcpeid.c, funcs/func_realtime.c, apps/app_forkcdr.c,
	  apps/app_channelredirect.c, apps/app_flash.c,
	  funcs/func_blacklist.c, apps/app_sms.c, apps/app_senddtmf.c,
	  apps/app_stack.c, apps/app_verbose.c: It is no longer required
	  for each module that deals with a channel to call
	  ast_module_user_hangup_all in it's unload function. The loader
	  will automatically perform this action for it.

2007-07-16 02:51 +0000 [r75163-75164]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h, include/asterisk/dundi.h,
	  include/asterisk/enum.h, include/asterisk/config.h,
	  include/asterisk/io.h, include/asterisk/cli.h,
	  include/asterisk/channel.h, include/asterisk/cdr.h,
	  include/asterisk/manager.h, include/asterisk/tdd.h,
	  include/asterisk/abstract_jb.h, include/asterisk/file.h,
	  include/asterisk/res_odbc.h, include/asterisk/adsi.h,
	  include/asterisk/crypto.h, include/asterisk/doxyref.h,
	  include/asterisk/image.h, include/asterisk/musiconhold.h,
	  include/asterisk/jabber.h, include/asterisk/linkedlists.h,
	  include/asterisk/module.h, include/asterisk/strings.h,
	  include/asterisk/pbx.h, include/asterisk/frame.h,
	  include/asterisk/say.h, include/asterisk/translate.h: Merge a
	  bunch of doxygen updates to header files. This includes changes
	  to use the \retval tag for documenting return values, fixing
	  various warnings when generating the documentation, and various
	  other things. (closes issue #10203, snuffy)

	* funcs/func_iconv.c: Cast the 2nd argument to iconv() to a void *,
	  as some systems define it as a (const char *), while others
	  define it as (char *). This is done to suppress compiler warnings
	  about it.

2007-07-13 20:37 +0000 [r75109]  Russell Bryant <russell@digium.com>

	* /: Merged revisions 75108 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75108 | russell | 2007-07-13 15:36:16 -0500
	  (Fri, 13 Jul 2007) | 11 lines Merged revisions 75107 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
	  Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
	  load_moh_classes() could return without destroying the loaded
	  configuration. ........ ................

2007-07-13 20:16 +0000 [r75082]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 75078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75078 | mmichelson | 2007-07-13 15:15:30 -0500
	  (Fri, 13 Jul 2007) | 13 lines Merged revisions 75066 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13
	  Jul 2007) | 5 lines Fixed an issue where chanspy flags were
	  uninitialized if no options were passed. What triggered this
	  investigation was an IRC chat where some people's quiet flags
	  were set while others' weren't even though none of them had
	  specified the q option. ........ ................

2007-07-13 20:15 +0000 [r75054-75077]  Russell Bryant <russell@digium.com>

	* main/rtp.c: resolve a compiler warning

	* /, res/res_musiconhold.c: Merged revisions 75067 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75067 | russell | 2007-07-13 15:10:40 -0500
	  (Fri, 13 Jul 2007) | 14 lines Merged revisions 75059 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
	  Jul 2007) | 6 lines Ensure that adding a user to the list of
	  users of a specific music on hold class is not done at the same
	  time as any of the other operations on this list to prevent list
	  corruption. Using the global moh_data lock for this is not ideal,
	  but it is what is used to protect these lists everywhere else in
	  the module, and I am only changing what is necessary to fix the
	  bug. ........ ................

	* channels/chan_zap.c, /: Merged revisions 75053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r75053 | russell | 2007-07-13 14:11:26 -0500
	  (Fri, 13 Jul 2007) | 20 lines Merged revisions 75052 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13
	  Jul 2007) | 12 lines (closes issue #9660) Reported by: mmacvicar
	  Patches submitted by: bbryant, russell Tested by: mmacvicar,
	  marco, arcivanov, jmhunter, explidous When using a TDM400P (and
	  probably other analog cards) there was a chance that you could
	  hang up and pick the phone back up where it has been long enough
	  to be not considered a flash hook, but too soon such that the
	  device reports that it is busy and the person on the phone will
	  only hear silence. This patch makes chan_zap more tolerant of
	  this and gives the device a couple of seconds to succeed so the
	  person on the phone happily gets their dialtone. ........
	  ................

2007-07-13 16:22 +0000 [r75034]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/rtp.h, main/rtp.c: Small improvement to the STUN
	  support so it can be used by sockets other than RTP ones. The
	  main change is a new API function in main/rtp.c (see there for a
	  description) int ast_stun_request(int s, struct sockaddr_in *dst,
	  const char *username, struct sockaddr_in *answer) which can be
	  used to send an STUN request on a socket, and optionally wait for
	  a reply and store the STUN_MAPPED_ADDRESS into the 'answer'
	  argument (obviously, the version that waits for a reply is
	  blocking, but this is no different from DNS resolutions).
	  Internally there are minor modifications to let
	  stun_handle_packet() be somewhat configurable on how to parse the
	  body of responses. At the moment i am not committing any change
	  to the clients, but adding STUN client support is extremely
	  simple, e.g. chan_sip.c could do something like this: + add a
	  variable to store the stun server address; static struct
	  sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add
	  code to parse a config file of the form
	  "stunaddr=my.stun.server.org:3478" (not shown for brevity); +
	  right after binding the main sip socket, talk to the stun server
	  to determine the externally visible address if
	  (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock,
	  &stunaddr, NULL, &externip); so now 'externip' is set with the
	  externally visible address. so it is really trivial. Similarly
	  ast_stun_request could be called when creating the RTP socket
	  (possibly adding a struct sockaddr_in field in the struct ast_rtp
	  to store the externalip).

2007-07-12 23:02 +0000 [r74999]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 74997 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ ........

2007-07-12 20:46 +0000 [r74956]  Steve Murphy <murf@digium.com>

	* /, channels/chan_sip.c: Merged revisions 74955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1
	  line This patch resolves 10143; thanks to irroot for the patch;
	  looked acceptable. Let the community decide if it messes things
	  up ........

2007-07-12 19:19 +0000 [r74891-74923]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 74922 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2
	  lines Whoops... didn't want this to be returned to 0 each
	  iteration. ........

	* main/channel.c, /: Merged revisions 74888 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2
	  lines When waiting for a digit ensure that a begin frame was
	  received with it, not just an end frame. (issue #10084 reported
	  by rushowr) ........

2007-07-12 16:54 +0000 [r74865-74867]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 74866 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul
	  2007) | 1 line It helps if I actually add this stuff for the 7921
	  too - otherwise it won't actually do much of anything. ........

	* /, channels/chan_skinny.c: Merged revisions 74864 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul
	  2007) | 1 line Add device ID for 7921 wireless skinny phone
	  ........

2007-07-12 16:21 +0000 [r74850]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: more cleanup, this time to stun_handle_packet().
	  Among other things: + mark a potentially dangerous
	  write-past-end-of-buffer + localize some variables in the block
	  generating stun replies. As before, not ready yet for a merge to
	  1.4

2007-07-12 15:55 +0000 [r74816]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 74815 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74815 | file | 2007-07-12 12:53:55 -0300 (Thu,
	  12 Jul 2007) | 10 lines Merged revisions 74814 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul 2007) | 2
	  lines Only print out a warning for situations where it is
	  actually helpful. (issue #10187 reported by denke) ........
	  ................

2007-07-12 15:42 +0000 [r74813]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: a little bit of code cleanup to rtp.c, mostly to
	  function ast_rtp_new_with_bindaddr(): 1. add comments to the
	  logic of the main loop; 2. use a common exit point on failure so
	  the cleanup is done only in one place; 3. handle failures in
	  rtp_socket() in the main loop of the function; No functional
	  changes except for #3 above, so it is not yet worthwhile merging
	  this and other changes to 1.4 Once the cleanup work on this file
	  will be complete (which among other things should include some
	  extensions to the stun support) it might be a good thing to push
	  all the changes to 1.4

2007-07-11 23:05 +0000 [r74769]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 74767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74767 | russell | 2007-07-11 17:57:07 -0500
	  (Wed, 11 Jul 2007) | 13 lines Merged revisions 74766 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11
	  Jul 2007) | 5 lines The function make_trunk() can fail and return
	  -1 instead of a valid new call number. Fix the uses of this
	  function to handle this instead of treating it as the new call
	  number. This would cause a deadlock and memory corruption.
	  (possible cause of issue #9614 and others, patch by me) ........
	  ................

2007-07-11 21:15 +0000 [r74726]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 74722 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74722 | mmichelson | 2007-07-11 16:14:09 -0500
	  (Wed, 11 Jul 2007) | 13 lines Merged revisions 74719 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
	  Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
	  did not work...at all. Now it does. (closes issue #10178,
	  reported and patched by makoto, with slight modification for 1.4
	  and trunk by me) ........ ................

2007-07-11 21:09 +0000 [r74703-74713]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Code cleanup of res_agi

	* res/res_smdi.c: Code cleanup of res_smdi

	* pbx/pbx_spool.c: Clean up pbx_spool. So many nested if
	  statements...

	* main/udptl.c, include/asterisk/udptl.h: Use linkedlist macros for
	  UDPTL protocol list.

2007-07-11 18:35 +0000 [r74658]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c: Merged revisions 74657 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74657 | russell | 2007-07-11 13:34:51 -0500
	  (Wed, 11 Jul 2007) | 12 lines Merged revisions 74656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
	  Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
	  the condition that uses LIKE. This fixes realtime extensions with
	  ODBC. (closes issue #10175, reported by stuarth, patch by me)
	  ........ ................

2007-07-11 18:21 +0000 [r74636-74648]  Steve Murphy <murf@digium.com>

	* Makefile, /: Merged revisions 74642 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74642 | murf | 2007-07-11 12:18:42 -0600 (Wed, 11 Jul 2007) | 1
	  line This fixes 10172, where the entire man8 dir gets removed
	  during an uninstall of asterisk ........

	* /: blocking 74628 from trunk... only applied to 1.4

2007-07-11 17:34 +0000 [r74575-74616]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/speech.h, res/res_speech.c,
	  apps/app_speech_utils.c: Use the linkedlists.h AST_LIST_NEXT
	  macro for modifying the list of results.

	* channels/chan_phone.c, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
	  74572 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2
	  lines Instead of figuring out kernel versions that have
	  compiler.h and not... let's just use autoconf to check for it's
	  presence. (issue #10174 reported by francesco_r) ........

2007-07-11 16:24 +0000 [r74571]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: add a bit of documentation on what the stun code in
	  rtp.c does (which is very little, at the moment). Eventually,
	  when the functionality is extended, the changes can be merged
	  back to 1.4. At the moment this is pointless. Note, this change
	  is whitespace only.

2007-07-11 16:19 +0000 [r74516-74570]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/speech.h, res/res_speech.c,
	  apps/app_speech_utils.c: Allow the native formats of a channel to
	  influence the audio that is going to the engine. The best format
	  will try to be chosen with an ultimate fallback to signed linear
	  if possible.

	* res/res_speech.c: Can't forget to remember what format is in use
	  for writing.

	* include/asterisk/speech.h, res/res_speech.c: Change the speech
	  API to allow passing the format through to the engine.

	* channels/misdn/isdn_lib_intern.h: Change header a bit to get rid
	  of a doxygen parse error. (issue #10177 reported by snuffy)

	* channels/chan_phone.c, /: Merged revisions 74515 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul
	  2007) | 2 lines Only check if we need to do a SIGMA based tone
	  generation if we have a card. (issue #10179 reported by mikowhy)
	  ........

2007-07-10 23:34 +0000 [r74477]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 74476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74476 | mmichelson | 2007-07-10 18:32:52 -0500 (Tue, 10 Jul
	  2007) | 5 lines Forwarding a message with IMAP storage was
	  storing the message in the sender's box instead of the forwarded
	  mailbox. (closes issue #10138, reported and patched by jaroth)
	  ........

2007-07-10 20:02 +0000 [r74375-74429]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 74428 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10158) ................ r74428 | qwell | 2007-07-10
	  14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines Merged revisions
	  74427 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
	  lines Fix an issue where it was possible to have a service level
	  of over 100% Between the time recalc_holdtime and update_queue
	  was called, it was possible that the call could have been hungup.
	  Move both additions to the same place, so this won't happen.
	  Issue 10158, initial patch by makoto, modified by me. ........
	  ................

	* /, main/dns.c: Merged revisions 74388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74388 | qwell | 2007-07-10 14:10:36 -0500 (Tue, 10 Jul 2007) | 4
	  lines Don't use #if to check if something is defined - use #ifdef
	  instead. Pointed out by kpfleming ........

	* /, channels/chan_agent.c: Merged revisions 74379 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10169) ................ r74379 | qwell | 2007-07-10
	  14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions
	  74376 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4
	  lines Fix an issue with wrapuptime not working when using
	  AgentLogin. Issue 10169, patch by makoto, with a minor mod by me
	  to not re-break issue 9618 ........ ................

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/dns.c: Merged revisions 74374 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
	  issue #10133) ................ r74374 | qwell | 2007-07-10
	  13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines Merged revisions
	  74373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
	  lines Use res_ndestroy on systems that have it. Otherwise, use
	  res_nclose. This prevents a memleak on NetBSD - and possibly
	  others. Issue 10133, patch by me, reported and tested by scw
	  ........ ................

2007-07-10 16:01 +0000 [r74324]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 74323 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74323 | russell | 2007-07-10 11:00:11 -0500 (Tue, 10
	  Jul 2007) | 1 line fix an uninitialized variable ........

2007-07-10 15:41 +0000 [r74318-74319]  Jason Parker <jparker@digium.com>

	* /: svn revert != svn resolved Fix merged property...

	* apps/app_voicemail.c: Merged revisions 74317 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
	  issue #10170) ................ r74317 | qwell | 2007-07-10
	  10:38:32 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions
	  74316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
	  lines Fix a small typo in description in of Voicemail()
	  application. Issue 10170, patch by casper. ........
	  ................

2007-07-10 15:32 +0000 [r74315]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 74314 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74314 | russell | 2007-07-10 10:31:41 -0500
	  (Tue, 10 Jul 2007) | 11 lines Merged revisions 74313 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
	  Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
	  #10075, this part reported by jmls on IRC, patch by me) ........
	  ................

2007-07-10 15:07 +0000 [r74272]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c, include/asterisk/monitor.h,
	  apps/app_queue.c, res/res_monitor.c: Fix building that was broken
	  by recent monitor.h changes. Thanks Russell for pointing this out
	  (and pointing out what I probably did to prevent gcc from fixing
	  it - don't ctrl-C builds)

2007-07-10 14:51 +0000 [r74263-74266]  Joshua Colp <jcolp@digium.com>

	* /, main/app.c: Merged revisions 74265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r74265 | file | 2007-07-10 11:50:00 -0300 (Tue,
	  10 Jul 2007) | 10 lines Merged revisions 74264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
	  lines Ensure the group information category exists before trying
	  to do a string comparison with it. (issue #10171 reported by
	  mlegas) ........ ................

2007-07-09 21:32 +0000 [r74212]  Russell Bryant <russell@digium.com>

	* /, configure, configure.ac: Merged revisions 74211 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74211 | russell | 2007-07-09 16:31:30 -0500 (Mon, 09
	  Jul 2007) | 5 lines Update the configure script to check for a
	  required function that is not present in the 1.2 version of
	  libpri. This will prevent the configure script from thinking that
	  it has compatible libpri support for Asterisk 1.4, when it
	  actually does not because the installed version is from 1.2.
	  ........

2007-07-09 20:58 +0000 [r74164]  Jason Parker <jparker@digium.com>

	* include/asterisk/monitor.h, res/res_monitor.c: (closes issue
	  #7596) Reported by: julien23 Patches submitted by: julien23 Add
	  the ability to disable recording the input or output streams in
	  res_monitor.

2007-07-09 20:54 +0000 [r74163]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 74162 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09
	  Jul 2007) | 9 lines (closes issue #10123) Reported by: blitzrage
	  Patches submitted by: juggie, qwell, me Tested by: blitzrage When
	  trying to find a music on hold class to use, try all of the
	  options, instead of only the first one that is set. Also, change
	  the MusicOnHold applications to not hang up on the channel when a
	  class can not be found. ........

2007-07-09 20:21 +0000 [r74160]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 74159 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 Closes issue
	  #9186 ................ r74159 | qwell | 2007-07-09 15:19:28 -0500
	  (Mon, 09 Jul 2007) | 16 lines Merged revisions 74158 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul
	  2007) | 8 lines Several chan_zap options were not working on
	  reload because they were arbitrarily disallowed when reloading
	  some/most PRI options (such as signalling) was disallowed.
	  Options such as polarityonanswerdelay and answeronpolarityswitch
	  can safely be changed on a reload. This corrects that behavior.
	  Issue 9186, patch by tzafrir. ........ ................

2007-07-09 18:58 +0000 [r74125]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: remove an unused variable

2007-07-09 18:43 +0000 [r74121-74123]  Mark Michelson <mmichelson@digium.com>

	* /: Merged revisions 74122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74122 | mmichelson | 2007-07-09 13:38:28 -0500 (Mon, 09 Jul
	  2007) | 3 lines Forgot to get rid of an extraneous debug message.
	  ........

	* /, apps/app_queue.c: Merged revisions 74120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul
	  2007) | 6 lines The n option for Queue should make the queue exit
	  immediately after failure to reach any members and should not be
	  dependent on the timeout value passed to Queue (closes issue
	  #10127, reported by bcnit, repaired by me) ........

2007-07-09 16:35 +0000 [r74084]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Add Queue and DestinationChannel headers to the
	  AgentCalled manager event to be more like the rest of the events
	  in this module. (closes issue #10114, patch by kwakwaversal)

2007-07-09 15:34 +0000 [r74083]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 74082 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul
	  2007) | 2 lines Only destroy the scheduler context if it was
	  allocated. (issue #10124 reported by gzero) ........

2007-07-09 14:58 +0000 [r74048]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 74047 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74047 | mmichelson | 2007-07-09 09:57:41 -0500 (Mon, 09 Jul
	  2007) | 4 lines Fixed a logic error in leave_voicemail. Pass the
	  mailbox instead of the context to inbox_count when the context is
	  "default." (closes issue #10135, reported by yannj, repaired by
	  me) ........

2007-07-09 14:50 +0000 [r74044-74046]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_skinny.c, pbx/pbx_dundi.c: Merged revisions
	  74045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2
	  lines Few minor thread synchronization tweaks. (issue #10124
	  reported by gzero) ........

	* /, configure, acinclude.m4: Merged revisions 74043 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r74043 | file | 2007-07-09 11:34:33 -0300 (Mon, 09 Jul
	  2007) | 2 lines Use AC_CHECK_HEADER to check for ptlib/openh323
	  to allow for cross compiling. (issue #9675 reported by zandbelt)
	  ........

2007-07-09 08:30 +0000 [r74024-74025]  Olle Johansson <oej@edvina.net>

	* CHANGES: Update with new features

	* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
	  include/asterisk/channel.h: Implementation of a feature that will
	  disable "missed calls" counters on SIP phones. If the call is
	  answered by another phone, other phones won't display the call as
	  "missed". You can also add an option to the dial command so that
	  you can have a "followme" scenario and not count the calls as
	  "missed" when you cancel the call. Thanks to Ramon and Frank for
	  feedback on this feature.

2007-07-09 04:09 +0000 [r73994]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, /, channels/chan_sip.c,
	  main/ast_expr2f.c, include/asterisk/channel.h,
	  funcs/func_devstate.c, apps/app_voicemail.c: Merged revisions
	  73985 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007)
	  | 2 lines Doxygen formatting fixes; fixes errors while 'make
	  progdocs'. (Closes issue #10104) ........

2007-07-09 03:14 +0000 [r73931-73983]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c, /: Merged revisions 73980 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73980 | file | 2007-07-09 00:13:19 -0300 (Mon, 09 Jul 2007) | 2
	  lines Give Agent channel names priority when doing CDR merging.
	  (issue #10011 reported by krtorio) ........

	* res/res_features.c: Use linkedlist macros for parking.

	* main/manager.c: Make sure the idText variable is empty, and put
	  it in the right place for the manager ack packet. (issue #10152
	  reported by srt)

	* /, pbx/pbx_config.c: Merged revisions 73930 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73930 | file | 2007-07-08 22:13:57 -0300 (Sun, 08 Jul 2007) | 2
	  lines Add a few sanity checks when writing out the dialplan.
	  (issue #10157 reported by dome) ........

2007-07-08 21:01 +0000 [r73911]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
	  main/ast_expr2.y, configure.ac, main/ast_expr2.c: Restore EXP2
	  and LOG2 functions, by providing mathematical identify functions,
	  when the underlying C functions are not available.

2007-07-08 13:22 +0000 [r73886]  Russell Bryant <russell@digium.com>

	* res/res_features.c: ast_exists_extension() does not return an
	  ast_device_state, so change this function to explicitly check for
	  the int return value. Also, make a few other minor changes such
	  as removing a variable.

2007-07-08 09:49 +0000 [r73850]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 73849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2
	  lines While tracking down a bug, I need some more history.
	  Dumphistory is very useful, indeed. ........

2007-07-07 16:44 +0000 [r73821]  Steve Murphy <murf@digium.com>

	* configure, include/asterisk/autoconfig.h.in, main/ast_expr2.y,
	  configure.ac, bootstrap.sh, main/ast_expr2.c: These changes fix
	  10145 and 10150, a prob with BSD and exp2/log2 not existing, as
	  well as the bootstrap needing a small upgrade for openbsd. Many
	  thanks to mvanbaak

2007-07-06 23:05 +0000 [r73771]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73769 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73769 | russell | 2007-07-06 18:02:58 -0500
	  (Fri, 06 Jul 2007) | 12 lines Merged revisions 73768 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06
	  Jul 2007) | 4 lines If a sip_pvt struct has already registered an
	  extension state callback, remove the old one before adding a new
	  one. If this isn't done, Asterisk will crash. (issue #10120)
	  ........ ................

2007-07-06 16:39 +0000 [r73728]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 73727 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73727 | mmichelson | 2007-07-06 11:36:17 -0500 (Fri, 06 Jul
	  2007) | 8 lines Fixing a rare case which causes voicemail to
	  crash when compiled with IMAP storage. inboxcount has the
	  possibility of finding an "interactive" vm_state when no
	  persistent "non-interactive" vm_state exists for that mailbox. If
	  this should happen when someone attempts to leave a message, it
	  results in a crash. This patch, along with my commit in revision
	  72670 fix issue 10053, reported by jaroth. closes issue #10053
	  ........

2007-07-06 16:30 +0000 [r73726]  Kevin P. Fleming <kpfleming@digium.com>

	* main/minimime/mimeparser.yy.c, main/minimime/mimeparser.h,
	  main/minimime/mimeparser.tab.c, main/minimime/mimeparser.y,
	  main/minimime/Makefile, main/minimime/mimeparser.l,
	  main/minimime/mimeparser.tab.h, main/minimime/mm_parse.c:
	  eliminate another batch of compiler warnings (and a bug, although
	  in code we aren't using)... note that this required manually
	  editing the lexer output code (generated by flex), so some of
	  them will come back if the lexer is rebuilt

2007-07-06 16:14 +0000 [r73680-73701]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 73696 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73696 | russell | 2007-07-06 11:12:51 -0500
	  (Fri, 06 Jul 2007) | 16 lines Merged revisions 73684 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
	  Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
	  Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
	  with MSSQL 2005 by explicitly stating that '\' is being used as
	  an escape character. ........ ................

	* /, channels/chan_sip.c: Merged revisions 73679 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73679 | russell | 2007-07-06 10:57:25 -0500
	  (Fri, 06 Jul 2007) | 15 lines Merged revisions 73678 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06
	  Jul 2007) | 7 lines (closes issue #10125) Reported by: makoto
	  Patches submitted by: makoto This fixes a crash in chan_sip that
	  happens when the bindaddr setting is not valid on Asterisk
	  startup, gets fixed, and then a reload gets issued. ........
	  ................

2007-07-06 15:47 +0000 [r73677]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/busy.h (added), channels/ringtone.h (added),
	  channels/Makefile, channels: it really seems pointless to run
	  gentone to create these header files every time we build
	  Asterisk...

2007-07-06 15:28 +0000 [r73676]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 73675 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73675 | mmichelson | 2007-07-06 10:27:28 -0500
	  (Fri, 06 Jul 2007) | 13 lines Merged revisions 73674 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
	  Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
	  (issue 9618, reported by jiddings, patched by moi) closes issue
	  #9618 ........ ................

2007-07-06 03:48 +0000 [r73557-73633]  Russell Bryant <russell@digium.com>

	* CHANGES: Redistribute a lot of the items that were in the Misc.
	  section

	* CHANGES: note TLS support for manager and HTTP in CHANGES

	* CREDITS: Philippe was listed twice

	* /, BUGS: Merged revisions 73629 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73629 | russell | 2007-07-05 22:34:46 -0500 (Thu, 05 Jul 2007) |
	  1 line fix a little spelling error ........

	* /, channels/chan_sip.c: Merged revisions 73598 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) |
	  3 lines Fix a crash in chan_sip. Don't try to stop the monitor
	  thread if it was never started. (closes issue #10124, reported by
	  gzero, fixed by me) ........

	* /, channels/chan_iax2.c: Merged revisions 73555 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73555 | russell | 2007-07-05 18:05:33 -0500 (Thu, 05 Jul 2007) |
	  3 lines copy from the correct buffer when deferring a full frame
	  (related to issue #9937) ........

2007-07-05 22:48 +0000 [r73553]  Kevin P. Fleming <kpfleming@digium.com>

	* main/minimime/mm_contenttype.c, main/minimime/mm_envelope.c,
	  main/minimime/mm_mimepart.c, main/minimime/mm_param.c,
	  main/minimime/mm_context.c, main/minimime/mm_mimeutil.c: comment
	  out some code that is not used and does not have prototypes

2007-07-05 22:32 +0000 [r73552]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 73551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73551 | russell | 2007-07-05 17:31:31 -0500 (Thu, 05 Jul 2007) |
	  6 lines * Store the call number that a thread is processing
	  without the full frame bit set to ease debugging * When deferring
	  a full frame for processing, stick it into the queue for the
	  thread that is processing frames for that call, not the one that
	  read the current frame and is about to go back into the idle list
	  (related to issue #9937) ........

2007-07-05 22:29 +0000 [r73550]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73548 | kpfleming | 2007-07-05 17:20:44 -0500
	  (Thu, 05 Jul 2007) | 10 lines Merged revisions 73547 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05
	  Jul 2007) | 2 lines we shouldn't allow G.723.1 endpoints to use
	  VAD, just like we don't support it for G.729 ........
	  ................

2007-07-05 22:23 +0000 [r73549]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Add the ability to play an announcement to
	  queue caller just before bridging Issue 7479, patch by
	  tristan_mahe.

2007-07-05 20:52 +0000 [r73513-73514]  Russell Bryant <russell@digium.com>

	* main/ast_expr2.y, main/ast_expr2.c: resolve a compiler warning so
	  i can build in dev mode

	* /, res/res_features.c: Merged revisions 73512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) |
	  5 lines Pass HOLD and UNHOLD frames to the other channel when
	  they are returned from a native bridge function. This fixes a
	  problem where when two zap channels are natively bridged and one
	  does a flash hook, the other channel did not receive music on
	  hold. (Reported to me directly by Doug Bailey at Digium) ........

2007-07-05 19:20 +0000 [r73468]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73467 | file | 2007-07-05 16:18:02 -0300 (Thu,
	  05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
	  lines Copy language information to the dialog structure when
	  calling a peer for situations where a PBX may be started on the
	  dialed channel. (issue #10121 reported by clegall_proformatique)
	  ........ ................

2007-07-05 18:15 +0000 [r73449]  Steve Murphy <murf@digium.com>

	* main/pbx.c, utils/expr2.testinput, main/ast_expr2.h,
	  main/ast_expr2.y, main/ast_expr2f.c, include/asterisk/ast_expr.h,
	  pbx/pbx_ael.c, UPGRADE.txt, doc/tex/channelvariables.tex,
	  utils/ael_main.c, main/ast_expr2.fl, main/ast_expr2.c,
	  utils/check_expr.c: In regards to changes for 9508, expr2 system
	  choking on floating point numbers, I'm adding this update to
	  round out (no pun intended) and make this FP-capable version of
	  the Expr2 stuff interoperate better with previous integer-only
	  usage, by providing Functions syntax, with 20 builtin functions
	  for floating pt to integer conversions, and some general floating
	  point math routines that might commonly be used also. Along with
	  this, I made it so if a function was not a builtin, it will try
	  and find it in the ast_custom_function list, and if found,
	  execute it and collect the results. Thus, you can call system
	  functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs,
	  without having to wrap them in $\{...\} (curly brace) notation.
	  Did a valgrind on the standalone and made sure there's no mem
	  leaks. Looks good. Updated the docs, too.

2007-07-05 17:21 +0000 [r73432]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Remove directory creation of directories
	  we've never used.

2007-07-05 16:05 +0000 [r73402]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 73400 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73400 | mmichelson | 2007-07-05 10:59:41 -0500 (Thu, 05 Jul
	  2007) | 5 lines Correcting a minor CLI bug I found. When issuing
	  the queue show command, if you type queue show and then press
	  tab, you can continue pressing tab and it will keep
	  auto-completing queue names even though only 1 queue can be used
	  as an argument. ........

2007-07-05 15:29 +0000 [r73399]  Russell Bryant <russell@digium.com>

	* channels/chan_vpb.cc, /, channels/Makefile: Merged revisions
	  73398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r73398 | russell | 2007-07-05 10:28:27 -0500 (Thu, 05 Jul 2007) |
	  2 lines Make this module build for me in dev-mode ........

2007-07-05 14:22 +0000 [r73317-73359]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /, apps/app_chanspy.c: Merged revisions 73355 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73355 | file | 2007-07-05 11:21:44 -0300 (Thu,
	  05 Jul 2007) | 10 lines Merged revisions 73349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
	  lines Tweak spy locking. (issue #9951 reported by welles)
	  ........ ................

	* channels/chan_local.c, /: Merged revisions 73319 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73319 | file | 2007-07-05 10:27:40 -0300 (Thu,
	  05 Jul 2007) | 10 lines Merged revisions 73318 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2
	  lines Actually check to make sure a PBX was started on one of the
	  Local channels instead of blindly assuming it was. (issue #10112
	  reported by makoto) ........ ................

	* /, apps/app_queue.c: Merged revisions 73316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73316 | file | 2007-07-05 10:22:13 -0300 (Thu,
	  05 Jul 2007) | 10 lines Merged revisions 73315 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
	  lines Reset ServicelevelPerf variable back to 0 if we are unable
	  to calculate it each time... otherwise we will get previous
	  values. (issue #10117 reported by noriyuki) ........
	  ................

2007-07-05 07:45 +0000 [r73209-73298]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
	  configs/misdn.conf.sample, channels/misdn_config.c: added general
	  Jitterbuffer Implementation. #9960

	* /, channels/misdn/isdn_lib.c: Merged revisions 73253 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73253 | crichter | 2007-07-04 16:53:48 +0200
	  (Mi, 04 Jul 2007) | 9 lines Merged revisions 73252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
	  Jul 2007) | 1 line bchannel configurations like echocancel and
	  volume control, need to be setuped on inbound calls too. ........
	  ................

	* channels/chan_misdn.c, /: Merged revisions 73208 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73208 | crichter | 2007-07-04 10:27:44 +0200
	  (Mi, 04 Jul 2007) | 9 lines Merged revisions 73207 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
	  Jul 2007) | 1 line bad bug in overlapdial case, we called
	  start_pbx multiple times, because the state wasn't changed..
	  ........ ................

2007-07-03 22:17 +0000 [r73191]  Steve Murphy <murf@digium.com>

	* /: blocking 73143 (revert of 9508 bug fix for 1.4) -- don't want
	  it backed out of trunk, too

2007-07-03 21:44 +0000 [r73144-73175]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: mkstemp doesn't specify a file mode, so we
	  should chmod it to VOICEMAIL_FILE_MODE Taken from a larger patch
	  by ltd - the rest of which is no longer necessary in trunk.
	  Closes issue #9231

	* apps/app_meetme.c: Fix a build warning, and potential issue if
	  option p is not set at all.

	* apps/app_meetme.c: Add support for changing the exit key from #
	  to any DTMF. This does not break existing configs - the arguments
	  to p are optional. Issue 8827, initial patch by junky, mostly
	  rewritten by fw to re-use option p, further modified by me.

2007-07-03 18:25 +0000 [r73127]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix up the device state processing thread in
	  app_queue so that it's not possible for there to be entries in
	  the queue and the thread is just sleeping (Thanks to mmichelson
	  for bringing the problem to my attention)

2007-07-03 12:40 +0000 [r73054]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, /: Merged revisions 73053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500
	  (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03
	  Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it
	  does not, because atoi does not distinguish between 0 and error
	  (closes issue #10106) ........ ................

2007-07-03 08:22 +0000 [r73006]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 73005 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r73005 | crichter | 2007-07-03 10:17:06 +0200
	  (Di, 03 Jul 2007) | 9 lines Merged revisions 73004 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
	  Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
	  be called from mISDN Source channels.. #9449 ........
	  ................

2007-07-03 05:21 +0000 [r73003]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Typo (closes issue 10105)

2007-07-03 02:51 +0000 [r72987]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Correct an issue where the wrong type was being
	  used to start sasl. Pointed out by and patch provided by mog.

2007-07-02 23:02 +0000 [r72982-72986]  Russell Bryant <russell@digium.com>

	* main/pbx.c, doc/tex/ast_funcdocs.tex (removed), main/manager.c,
	  doc/tex/ast_cli_commands.tex (removed), res/res_agi.c,
	  doc/tex/ast_appdocs.tex (removed), doc/tex/asterisk.tex,
	  doc/tex/ast_manager_actiondocs.tex (removed),
	  doc/tex/ast_agi_commands.tex (removed), main/cli.c: After some
	  discussion on the asterisk-dev list, we determined that this
	  approach for extracting application, function, manager, and agi
	  documentation is the wrong one to take. The most severe problem
	  is that the output depends on which modules are loaded as well as
	  compile time options, which both determine which parts are
	  available.

	* doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
	  doc/tex/ast_cli_commands.tex (added), doc/tex/ast_appdocs.tex
	  (added), doc/tex/realtime.tex (added), doc/qos.tex (removed),
	  doc/queues-with-callback-members.tex (removed), doc/tex/dundi.tex
	  (added), doc/ajam.tex (removed), doc/tex/cliprompt.tex (added),
	  doc/misdn.tex (removed), doc/manager.tex (removed),
	  doc/tex/chaniax.tex (added), doc/sla.tex (removed),
	  doc/billing.tex (removed), doc/tex/app-sms.tex (added),
	  build_tools/prep_tarball, doc/tex/ices.tex (added),
	  doc/localchannel.tex (removed), doc/cdrdriver.tex (removed),
	  doc/tex/asterisk.tex (added), doc/tex/queuelog.tex (added),
	  doc/freetds.tex (removed), doc/odbcstorage.tex (removed),
	  doc/tex/hardware.tex (added), doc/tex/mp3.tex (added), doc/tex
	  (added), doc/channelvariables.tex (removed), doc/ael.tex
	  (removed), doc/enum.tex (removed), doc/tex/configuration.tex
	  (added), doc/security.tex (removed), doc/tex/asterisk-conf.tex
	  (added), Makefile, doc/imapstorage.tex (removed),
	  doc/tex/ast_funcdocs.tex (added), doc/privacy.tex (removed),
	  doc/tex/ast_manager_actiondocs.tex (added),
	  doc/ast_agi_commands.tex (removed), doc/tex/jitterbuffer.tex
	  (added), doc/ast_cli_commands.tex (removed),
	  doc/tex/extensions.tex (added), doc/ast_appdocs.tex (removed),
	  doc/tex/queues-with-callback-members.tex (added), doc/tex/qos.tex
	  (added), doc/realtime.tex (removed), doc/dundi.tex (removed),
	  doc/tex/ajam.tex (added), doc/cliprompt.tex (removed),
	  doc/tex/manager.tex (added), doc/tex/misdn.tex (added),
	  doc/chaniax.tex (removed), doc/tex/README.txt (added),
	  doc/tex/sla.tex (added), doc/app-sms.tex (removed),
	  doc/tex/billing.tex (added), doc/ices.tex (removed),
	  doc/tex/localchannel.tex (added), doc/tex/cdrdriver.tex (added),
	  doc/asterisk.tex (removed), doc/queuelog.tex (removed),
	  doc/tex/odbcstorage.tex (added), doc/tex/freetds.tex (added),
	  doc/hardware.tex (removed), doc/mp3.tex (removed),
	  doc/tex/channelvariables.tex (added), doc/tex/ael.tex (added),
	  doc/tex/enum.tex (added), doc/configuration.tex (removed),
	  doc/tex/security.tex (added), doc/asterisk-conf.tex (removed),
	  doc/tex/imapstorage.tex (added), doc/ast_funcdocs.tex (removed),
	  doc/tex/privacy.tex (added), doc/tex/Makefile (added),
	  doc/ast_manager_actiondocs.tex (removed),
	  doc/tex/ast_agi_commands.tex (added): * Move LaTeX docs into a
	  tex/ subdirectory of the doc/ dir * Add a Makefile in doc/tex/
	  for generating PDF and HTML * Add a README.txt file to doc/tex/
	  to document which tools are used and what web sites to visit for
	  getting them. * Update build_tools/prep_tarball to put the proper
	  Asterisk version string in the automatically generated PDF for
	  release tarballs

2007-07-02 21:50 +0000 [r72940]  Steve Murphy <murf@digium.com>

	* utils/expr2.testinput, /, main/Makefile, main/ast_expr2.h,
	  main/ast_expr2.y, main/ast_expr2f.c, UPGRADE.txt,
	  main/ast_expr2.fl, main/ast_expr2.c: Merged revisions 72933 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1
	  line support for floating point numbers added to ast_expr2
	  $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp
	  numbers. The MATH function returns fp numbers by default, so this
	  fix is considered necessary. ........

2007-07-02 20:45 +0000 [r72937-72939]  Russell Bryant <russell@digium.com>

	* res/res_agi.c, doc/ast_agi_commands.tex: Fix up the AGI doc dump
	  CLI command and update the AGI commands tex file to not include a
	  bunch of empty entries.

	* doc/ast_cli_commands.tex (added), doc/asterisk.tex: Add CLI
	  commands to the docs

	* main/cli.c: Add a CLI command to output docs on CLI commands to a
	  file

2007-07-02 20:35 +0000 [r72935-72936]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Yet another Solaris tweak...

	* res/res_limit.c: Fix building under Solaris.

2007-07-02 19:31 +0000 [r72920-72932]  Russell Bryant <russell@digium.com>

	* doc/asterisk.tex, doc/ast_agi_commands.tex (added): Add AGI
	  commands to the documentation

	* res/res_agi.c: Add a CLI command to export the AGI command docs

	* res/res_agi.c: Add a note that the AGI commands array is not
	  handled in a thread-safe way

	* doc/asterisk.tex, doc/ast_manager_actiondocs.tex (added): Update
	  the documentation to include a manager action reference

	* main/manager.c: Add a CLI command to dump the built-in manager
	  action documentation

	* main/manager.c, /: Merged revisions 72926 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72926 | russell | 2007-07-02 13:18:46 -0500 (Mon, 02 Jul 2007) |
	  3 lines Remove a bogus comment and add proper locking to the
	  handler function for the CLI command to show information on
	  manager actions. ........

	* doc/ast_funcdocs.tex (added), doc/asterisk.tex: update
	  documentation to include dialplan functions

	* main/pbx.c: Add "core dump funcdocs" CLI command

	* main/pbx.c: change the "core dump appdocs" CLI command to use the
	  new API for creating CLI commands

	* doc/ast_appdocs.tex: update application documentation dump

2007-07-02 14:39 +0000 [r72889]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 72888 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2
	  lines Added additional DTMF debug messages for when emulation
	  occurs. ........

2007-07-02 09:34 +0000 [r72867-72869]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 72852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72852 | crichter | 2007-07-02 10:41:08 +0200
	  (Mo, 02 Jul 2007) | 9 lines Merged revisions 72585 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29
	  Jun 2007) | 1 line check if the bchannel stack id is already
	  used, if so don't use it a second time. Also added a release_chan
	  lock, so that the same chan_list object cannot be freed twice.
	  chan_misdn does not crash anymore on heavy load with these
	  changes. ........ ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  Merged revisions 72851 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72851 | crichter | 2007-07-02 10:27:19 +0200
	  (Mo, 02 Jul 2007) | 9 lines Merged revisions 72099 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27
	  Jun 2007) | 1 line simplified generation for dummy bchannels,
	  also we mark them as dummies, so they are not used later as
	  real-bchannels, optimized the RESTART mechanisms, we block a
	  channel now on cause:44, and send out a RESTART automatically,
	  then on reception of RESTART_ACKNOWLEDGE we unblock the channel
	  again. ........ ................

	* channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Merged
	  revisions 72850 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72850 | crichter | 2007-07-02 10:14:43 +0200
	  (Mo, 02 Jul 2007) | 9 lines Merged revisions 72087 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27
	  Jun 2007) | 1 line simplified channel finding and locking a lot.
	  removed unnecessary #ifdefed areas. ........ ................

2007-07-01 23:53 +0000 [r72807]  Russell Bryant <russell@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 72806 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72806 | russell | 2007-07-01 18:52:45 -0500
	  (Sun, 01 Jul 2007) | 13 lines Merged revisions 72805 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01
	  Jul 2007) | 5 lines When appending lines to call files to keep
	  track of retries, write a leading newline just in case the
	  original call file did not have a newline at the end. This fix is
	  in response to a problem I saw reported on the asterisk-users
	  mailing list. ........ ................

2007-06-30 16:53 +0000 [r72767]  Russell Bryant <russell@digium.com>

	* /, configure, configure.ac: Merged revisions 72766 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r72766 | russell | 2007-06-30 11:50:40 -0500 (Sat, 30
	  Jun 2007) | 3 lines Tweak the configure script so that error
	  output isn't spewed to the console when searching for GTK2 libs,
	  and they aren't found. ........

2007-06-29 21:37 +0000 [r72741]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Add support
	  for regcontext and regexten to chan_skinny Issue 9762, patch by
	  mvanbaak.

2007-06-29 21:24 +0000 [r72738]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/http.c: Fix my recent change for sending large files via the
	  http server. This code *must* write the file to the FILE *, and
	  not the raw fd. Otherwise, it breaks TLS support. Thanks to rizzo
	  for catching this!

2007-06-29 21:14 +0000 [r72727]  Luigi Rizzo <rizzo@icir.org>

	* main/minimime/Makefile: As the comment in the code says: Use
	  weaker error checking because we have some automatically
	  generated files. However just mask out -Werror, because other
	  warnings below: -Wundef -Wstrict-prototypes
	  -Wmissing-declarations -Wmissing-prototypes may actually be
	  important and spot out real bugs.

2007-06-29 20:56 +0000 [r72701-72706]  Russell Bryant <russell@digium.com>

	* /, formats/format_pcm.c: Merged revisions 72705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72705 | russell | 2007-06-29 15:56:18 -0500 (Fri, 29 Jun 2007) |
	  1 line give format_pcm a more concise destription ........

	* include/asterisk/http.h, main/manager.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, main/http.c:
	  Merge changes from team/russell/http_filetxfer Handle
	  transferring large files from the built-in http server.
	  Previously, the code attempted to malloc a block as large as the
	  file itself. Now it uses the sendfile() system call so that the
	  file isn't copied into userspace at all if it is available.
	  Otherwise, it just uses a read/write of small chunks at a time.

2007-06-29 20:33 +0000 [r72700]  Luigi Rizzo <rizzo@icir.org>

	* main/Makefile: Make sure that we properly recurse in
	  subdirectories to check dependencies for libraries. Because these
	  targets (e.g. minimime/libmmime.a) are real ones, declaring them
	  .PHONY would cause them to be rebuilt every time (see e.g. SVN
	  64355). As a workaround I am using the following CHECK_SUBDIR
	  target: CHECK_SUBDIR: # do nothing, just make sure that we
	  recurse in the subdir/ minimime/libmmime.a: CHECK_SUBDIR @cd
	  minimime && $(MAKE) libmmime.a which seems to do a better job
	  than .PHONY (probably because .PHONY forces the rebuild even if
	  the recursive make does not think it is necessary). If this turns
	  out to be the correct approach, we can then merge it back into
	  1.4

2007-06-29 20:02 +0000 [r72670]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Found a grievous logical error in
	  get_vm_state_by_imapuser. The imapuser being passed in was never
	  getting compared to imapusers of any of the vm_states in the
	  vmstates list. I also found some places in the code where I used
	  my typical brace style and changed it to match the typical
	  Asterisk brace style.

2007-06-29 19:09 +0000 [r72666]  Luigi Rizzo <rizzo@icir.org>

	* /: 72665 not applicable to trunk

2007-06-29 04:56 +0000 [r72555-72557]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 72556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72556 | tilghman | 2007-06-28 23:47:11 -0500 (Thu, 28 Jun 2007)
	  | 2 lines Issue 10055 - Change memory allocation to use the heap
	  for a command, since the output has the potential to overflow the
	  stack (as it did here) ........

2007-06-28 21:31 +0000 [r72539]  Jason Parker <jparker@digium.com>

	* Makefile, configure, configure.ac, makeopts.in: Apparently some
	  builds of gcc don't have declaration-after-statement. This checks
	  for it in configure, and only uses it if it's available. If it's
	  wrong, somebody please yell at me and tell me why.

2007-06-28 20:52 +0000 [r72524]  Dwayne M. Hubbard <dhubbard@digium.com>

	* funcs/func_math.c: Added AND, OR, and XOR bitwise operations to
	  MATH for issue 9891, thanks jcmoore

2007-06-28 19:41 +0000 [r72492]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, res/res_config_odbc.c,
	  include/asterisk/strings.h: Remove the ill-advised ast_restrdupa
	  API call and related structures

2007-06-28 19:35 +0000 [r72490-72491]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Fix building with
	  -Wdeclaration-after-statement, here too

	* res/res_jabber.c: Fix building with -Wdeclaration-after-statement

2007-06-28 19:07 +0000 [r72452-72466]  Luigi Rizzo <rizzo@icir.org>

	* /: 72462 is not applicable to trunk

	* res/res_features.c, apps/app_sms.c: move variable declarations to
	  the beginning of a block. Not applicable to previous branches.

	* channels/chan_skinny.c: move variable declarations to the
	  beginning of the block

	* apps/app_minivm.c: move variable declarations to the beginning of
	  a block. Not applicable to previous branches

	* /: 72453 was already applied to trunk some time ago

	* Makefile: Add -Wdeclaration-after-statement to AST_DEVMODE to
	  detect declarations in the middle of a block. Approved by:
	  Russel, Kevin The fallout will be fixed in separate commits. I am
	  doing this only on trunk only for the time being, because 1.4
	  still requires a bit more polishing to give a clean compile (at
	  least on FreeBSD).

2007-06-28 16:35 +0000 [r72437]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Fix bug where point code gets corrupted on
	  CPG

2007-06-27 23:30 +0000 [r72384]  Brett Bryant <bbryant@digium.com>

	* /, main/asterisk.c: Merged revisions 72383 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72383 | bbryant | 2007-06-27 18:29:14 -0500
	  (Wed, 27 Jun 2007) | 11 lines Merged revisions 72373 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27
	  Jun 2007) | 3 lines Reinstating patch. This actually fixes the
	  problem, however I was running a development branch without it
	  and mistakenly thought it wasn't fixed. Fixes issue #10010, and
	  #9654: 100% CPU usage caused by an asterisk console losing it's
	  controlling terminal. ........ ................

2007-06-27 23:26 +0000 [r72354-72382]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 72381 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed,
	  27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2
	  lines Update documentation to clarify variable usage with
	  MixMonitor. (issue #9494 reported by netoguy) ........
	  ................

	* channels/chan_jingle.c: Silly jingle...

	* channels/chan_sip.c, CHANGES: Add SIPREFERRINGCONTEXT and
	  SIPREFERREDBYHDR variables when a transfer takes place. (issue
	  #8378 reported by jcovert)

2007-06-27 23:04 +0000 [r72337]  Brett Bryant <bbryant@digium.com>

	* /, main/asterisk.c: Merged revisions 72335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72335 | bbryant | 2007-06-27 18:03:01 -0500
	  (Wed, 27 Jun 2007) | 10 lines Merged revisions 72333 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27
	  Jun 2007) | 2 lines Reverted changes for earlier revisions 72259
	  to 72261. Issue #9654, #10010 ........ ................

2007-06-27 22:58 +0000 [r72330-72332]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 72331 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun
	  2007) | 2 lines Make payload IDs for iLBC/Speex match to our
	  list. Since these are dynamic payloads the other side shouldn't
	  care. (issue #9426 reported by irroot) ........

	* /, apps/app_queue.c: Merged revisions 72328 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72328 | file | 2007-06-27 18:45:49 -0400 (Wed,
	  27 Jun 2007) | 10 lines Merged revisions 72327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
	  lines Fix issue where queue log events might be missing. (issue
	  #7765 reported by mtryfoss) ........ ................

2007-06-27 22:47 +0000 [r72329]  Mark Michelson <mmichelson@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Added ability to customize which buttons control forward,
	  reverse, pause, and stop during message playback. (closes issue
	  9474, reported and patched by jaroth with modifications by me)

2007-06-27 22:27 +0000 [r72325-72326]  Jason Parker <jparker@digium.com>

	* main/cli.c: Fix a segfault when trying to tab complete the "core
	  show uptime" command. Reported in #asterisk-dev on IRC by
	  jcmoore, fixed by me.

	* main/say.c: Add support for Thai language in say.c Issue 9417,
	  patch by dome, with some cleanup done by me.

2007-06-27 21:44 +0000 [r72304]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Let's NOT create a deadlock scenario here

2007-06-27 21:09 +0000 [r72274]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 72272 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72272 | russell | 2007-06-27 16:08:34 -0500
	  (Wed, 27 Jun 2007) | 13 lines Merged revisions 72267 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27
	  Jun 2007) | 5 lines Fix a minor issue with parsing the priority
	  number. You could have as much whitespace as you want around a
	  numeric priority, but you couldn't have any whitespace around a
	  special priority like "n" or "hint". (issue #10039, reported by
	  mitheloc, fixed by me) ........ ................

2007-06-27 20:47 +0000 [r72261]  Brett Bryant <bbryant@digium.com>

	* /, main/asterisk.c: Merged revisions 72260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72260 | bbryant | 2007-06-27 15:46:12 -0500
	  (Wed, 27 Jun 2007) | 12 lines Merged revisions 72259 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27
	  Jun 2007) | 4 lines Fixes 100% load when controlling terminal
	  disappears. Issue #9654, #10010 ........ ................

2007-06-27 20:26 +0000 [r72233-72258]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 72257 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72257 | file | 2007-06-27 16:25:24 -0400 (Wed,
	  27 Jun 2007) | 10 lines Merged revisions 72256 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
	  lines I may possibly get shot for doing this... but... defer CDR
	  processing until after the channel has been dealt with. This
	  should eliminate all of the issues with channels going funky
	  (SIP/PRI) when you are posting CDRs to a database that is either
	  slow or unavailable and do not want to enable batching. ........
	  ................

	* /: Fix up properties.

	* main/logger.c: Fix -T option. (issue #10073 reported by xylome)

2007-06-27 19:50 +0000 [r72232]  Mark Michelson <mmichelson@digium.com>

	* /, configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Adding feature to support the storage and retrieval of voicemail
	  greetings using IMAP storage. This feature may be turned on by
	  adding imapgreetings=yes to the general section of voicemail.conf
	  voicemail.conf.sample has details on the options added. As a
	  result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
	  In addition to the IMAP greeting changes, I also have added an
	  enum for the voicemail folders and so now the code should be
	  easier to understand and maintain when it comes to this area.

2007-06-27 19:13 +0000 [r72207]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 72205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72205 | kpfleming | 2007-06-27 14:13:21 -0500 (Wed, 27 Jun 2007)
	  | 2 lines use the proper type for storing group number bits so
	  that if someone specifies 'group=42' it will actually work
	  instead of being silently ignored ........

2007-06-27 18:37 +0000 [r72183]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 72182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72182 | qwell | 2007-06-27 13:36:56 -0500 (Wed, 27 Jun 2007) | 4
	  lines Fix another problem in voicemail with missing symbols.
	  Issue 10074, patch by kryptolus, extended to include #if 0'd
	  blocks (just in case) ........

2007-06-27 17:34 +0000 [r72149]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 72148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2
	  lines Make the ast_read_noaudio API call behave better under
	  circumstances where DTMF emulation was happening and a generator
	  was setup. (issue #10065 reported by stevefeinstein) ........

2007-06-27 17:14 +0000 [r72134]  Jason Parker <jparker@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 72125 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun
	  2007) | 4 lines Don't modify a variable that we don't want
	  modified. Make a copy of it instead. Issue 10029, patch by
	  phsultan with slight modifications by me (to remove needless
	  casts). Note: chan_jingle in trunk does not appear to have the
	  same bug. ........

2007-06-27 16:38 +0000 [r72113]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 72112 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) |
	  3 lines Only output debug information related to RTCP timestamps
	  when RTCP debug is turned on (issue #10066, patch by me) ........

2007-06-27 08:08 +0000 [r72052]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/isdn_lib.c: Merged revisions 72042 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r72042 | crichter | 2007-06-27 09:58:06 +0200
	  (Mi, 27 Jun 2007) | 13 lines Merged revisions 72040-72041 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
	  1 line for inbound TE calls, we setup the bchannel when we get
	  the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
	  removed some #if 0 areas which weren't used anymore. ........
	  r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
	  1 line isdn_lib.c didn't compile ........ ................

2007-06-27 01:00 +0000 [r71988-72007]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 72006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r72006 | file | 2007-06-26 20:58:35 -0400 (Tue, 26 Jun 2007) | 2
	  lines Make unloading of pbx_dundi actually work. ........

	* channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add rtpdest
	  option to SIP CHANNEL() dialplan function to return the IP
	  address and port that RTP (be it audio/video/text) is going to.

2007-06-26 23:03 +0000 [r71952-71954]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 71953 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71953 | mmichelson | 2007-06-26 18:02:09 -0500 (Tue, 26 Jun
	  2007) | 4 lines Removing a pointless line. This variable was
	  already set earlier and between then and this line, there is no
	  way that the values on the right side of the assignment could
	  have changed. ........

	* apps/app_voicemail.c: The variable msgnum was being overwritten
	  if IMAP storage was enabled. Put necessary #ifndef's around the
	  line which would overwrite.

2007-06-26 20:36 +0000 [r71916]  Jason Parker <jparker@digium.com>

	* /, main/rtp.c: Merged revisions 71915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4
	  lines Don't dereference a pointer that may be NULL here. Issue
	  10017. ........

2007-06-26 20:34 +0000 [r71883-71914]  Mark Michelson <mmichelson@digium.com>

	* apps/app_record.c: Create directory if it does not exist. (Closes
	  issue 10061, Reported and patched by eliel)

	* /, apps/app_voicemail.c: Merged revisions 71877 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71877 | mmichelson | 2007-06-26 14:00:05 -0500 (Tue, 26 Jun
	  2007) | 11 lines A few changes, the ultimate goal of which is to
	  keep better track of the number of messages that a mailbox
	  currently has. A description of the changes: 1. Changed the
	  "updated" field of the vm_state struct to act more as a binary
	  semaphore than a counting semaphore, since its current
	  implementation made the inboxcount function not work properly.
	  This change falls in line with a change made by UPenn with their
	  IMAP setup and helps to sync our changes with theirs. 2.
	  Eliminated some redundant calls to get_vm_state_by_mailbox inside
	  leave_voicemail 3. Use the play_folder variable to keep track of
	  the number of old and new messages in a mailbox as the messages
	  are deleted 4. Added an increment to the number of new messages
	  that was not there previously in the leave_voicemail function
	  ........

2007-06-26 16:39 +0000 [r71830]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Simplify some code in res_jabber relating to
	  SASL support. Issue 9988, patch by phsultan.

2007-06-26 15:50 +0000 [r71797]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 71796 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71796 | mmichelson | 2007-06-26 10:47:31 -0500 (Tue, 26 Jun
	  2007) | 5 lines Fixing bug where the authuser was mistakenly
	  pulled from the mailbox string instead of the IMAP user. (closes
	  issue 10054, reported and patched by jaroth) ........

2007-06-26 12:30 +0000 [r71752]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 71751 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71751 | tilghman | 2007-06-26 07:27:47 -0500
	  (Tue, 26 Jun 2007) | 10 lines Merged revisions 71750 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26
	  Jun 2007) | 2 lines Issue 10062 - Trying to move a message
	  without selecting one first results in memory corruption ........
	  ................

2007-06-26 00:10 +0000 [r71721-71732]  Mark Michelson <mmichelson@digium.com>

	* configure, configure.ac: Fixes a problem where Asterisk would not
	  compile if IMAP_STORAGE was enabled. Needed to add a space
	  between file name and options.

	* apps/app_voicemail.c: In my commit earlier today, I accidentally
	  left a prototype that isn't defined. This gets rid of that
	  prototype.

2007-06-25 19:20 +0000 [r71688]  Russell Bryant <russell@digium.com>

	* doc/imapstorage.tex, configure, configure.ac,
	  apps/app_voicemail.c: Allow compilation off app_voicemail with
	  IMAP_STORAE against a system installed version of the c-client
	  library. (issue #10047, jcollie)

2007-06-25 18:20 +0000 [r71658]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 71657 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71657 | tilghman | 2007-06-25 13:14:59 -0500
	  (Mon, 25 Jun 2007) | 10 lines Merged revisions 71656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25
	  Jun 2007) | 2 lines Issue 10035 - handle_exec returns a result
	  inconsistent with all of the other AGI commands ........
	  ................

2007-06-25 16:43 +0000 [r71637]  Steve Murphy <murf@digium.com>

	* main/cdr.c: Luigi's suggestion to move the llfrom decl was a good
	  one. Done.

2007-06-25 16:13 +0000 [r71630]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Using inboxcount instead of countmessages.

2007-06-25 15:35 +0000 [r71577-71613]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Tweak CLI command completion and some help
	  text. (issue #10049 reported by IgorG)

	* /, channels/chan_h323.c: Merged revisions 71576 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71576 | file | 2007-06-25 10:13:45 -0400 (Mon, 25 Jun 2007) | 2
	  lines Build a peer as well when hash323 is enabled in users.conf
	  (issue #9599 reported by asagage) ........

2007-06-25 13:42 +0000 [r71557]  Russell Bryant <russell@digium.com>

	* main/say.c, main/rtp.c, main/sched.c: Convert so more logging to
	  ast_debug (issue #10045, dimas)

2007-06-25 13:04 +0000 [r71521-71525]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 71522 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r71522 | file | 2007-06-25 09:03:03 -0400 (Mon, 25 Jun
	  2007) | 2 lines Minor tweak for queueing up the unhold frame...
	  this will teach me to do bugs while half asleep. (issue #10046
	  reported by dimas) ........

	* res/res_agi.c: Minor header inclusion tweak for new usage of
	  stat()

2007-06-25 12:40 +0000 [r71520]  Russell Bryant <russell@digium.com>

	* doc/asterisk-mib.txt, /: Merged revisions 71519 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71519 | russell | 2007-06-25 07:40:06 -0500 (Mon, 25 Jun 2007) |
	  2 lines Fix a typo in the Asterisk mib. (issue #10048, Matti)
	  ........

2007-06-25 09:46 +0000 [r71475-71500]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions 71214 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71214 | crichter | 2007-06-23 00:44:42 +0200
	  (Sa, 23 Jun 2007) | 9 lines Merged revisions 70341 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
	  Jun 2007) | 1 line fixed a bug that was introduced by copy and
	  paste in the last commit ..bchannels weren't cleaned properly.
	  ........ ................

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 71123 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71123 | crichter | 2007-06-22 17:38:08 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 70672 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21
	  Jun 2007) | 1 line we activate the bchannels in TE mode on
	  incoming calls only when we want to connect the call. ........
	  ................

	* /, channels/misdn/isdn_lib.c: Merged revisions 71122 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71122 | crichter | 2007-06-22 17:34:31 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 70342 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
	  Jun 2007) | 1 line forgot one place .. ........ ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions 71121 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71121 | crichter | 2007-06-22 17:32:54 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 70311 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
	  Jun 2007) | 1 line on receiption of cause:44 we mark the channel
	  as in use and inform the user about the situation, we need to
	  test the RESTART stuff then. Also shuffled the
	  empty_chan_in_stack function after the bchannel cleaning
	  functions, to avoid race conditions. ........ ................

	* channels/chan_misdn.c, /: Merged revisions 71120 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71120 | crichter | 2007-06-22 17:30:08 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 69887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
	  Jun 2007) | 1 line when we send out a SETUP, but get no response,
	  we should cleanup everything after reception of a hangup.
	  ........ ................

	* /, channels/misdn/isdn_msg_parser.c: Merged revisions 71118 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71118 | crichter | 2007-06-22 17:27:53 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 69053 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13
	  Jun 2007) | 1 line restart indicator 0x80 is correct, at least
	  that's what libpri does. ........ ................

	* channels/chan_misdn.c, /: Merged revisions 71106 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71106 | crichter | 2007-06-22 17:22:06 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 68887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
	  Jun 2007) | 1 line if the bridged partner is mISDN too we should
	  not send dtmf tones, they are transmitted inband always ........
	  ................

	* channels/chan_misdn.c, /: Merged revisions 71096 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71096 | crichter | 2007-06-22 17:17:04 +0200
	  (Fr, 22 Jun 2007) | 9 lines Merged revisions 68874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
	  Jun 2007) | 1 line if we have already some digits, we just stop
	  the tones. ........ ................

2007-06-25 01:11 +0000 [r71413-71434]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 71430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71430 | file | 2007-06-24 21:10:06 -0400 (Sun,
	  24 Jun 2007) | 10 lines Merged revisions 71414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
	  lines Ignore other URIs after the first in a 300 Multiple Choice
	  response. (issue #10041 reported by homesick) ........
	  ................

	* main/cdr.c, /: Merged revisions 71422 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71422 | file | 2007-06-24 21:07:31 -0400 (Sun, 24 Jun 2007) | 2
	  lines Fix it so 1.4 actually compiles on my box. ........

	* /, channels/chan_agent.c: Merged revisions 71412 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r71412 | file | 2007-06-24 20:49:21 -0400 (Sun, 24 Jun
	  2007) | 2 lines Check to make sure the channel pointer is present
	  before queueing up an unhold frame on it. (issue #10046 reported
	  by dimas) ........

2007-06-24 20:17 +0000 [r71338-71372]  Russell Bryant <russell@digium.com>

	* /, build_tools/prep_tarball: Merged revisions 71371 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r71371 | russell | 2007-06-24 15:16:32 -0500 (Sun, 24
	  Jun 2007) | 3 lines Include the menuselect-tree file in tarballs
	  to make builds from tarballs a little bit faster ........

	* /, main/asterisk.c: Merged revisions 71362 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71362 | russell | 2007-06-24 15:06:31 -0500
	  (Sun, 24 Jun 2007) | 10 lines Merged revisions 71358 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24
	  Jun 2007) | 2 lines Revert the patch from issue 9654 due to an
	  unexpected side effect ........ ................

	* main/udptl.c, apps/app_meetme.c, main/say.c, main/translate.c,
	  main/jitterbuf.c, apps/app_test.c, main/rtp.c, main/loader.c,
	  main/io.c, main/manager.c, apps/app_skel.c, apps/app_minivm.c,
	  main/logger.c, main/http.c, apps/app_rpt.c, main/sched.c:
	  Conversions to ast_debug() (issue #9984, patches from eliel and
	  dimas)

2007-06-24 17:51 +0000 [r71268-71292]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_features.c: Merged revisions 71291 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71291 | tilghman | 2007-06-24 12:50:24 -0500 (Sun, 24 Jun 2007)
	  | 2 lines Issue 10044 - chan->cdr is NULL here, so peer->cdr is
	  what we really wanted to use ........

	* main/manager.c, /, main/db.c: Merged revisions 71289 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71289 | tilghman | 2007-06-24 12:39:34 -0500
	  (Sun, 24 Jun 2007) | 10 lines Merged revisions 71288 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
	  Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
	  be able to set variables to the empty string. ........
	  ................

	* apps/app_mixmonitor.c: Issue 9970 - Ensure directory exists
	  before trying to write an output file

2007-06-23 03:32 +0000 [r71231]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /, res/res_features.c: Merged revisions 71230 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71230 | murf | 2007-06-22 21:29:48 -0600 (Fri, 22 Jun 2007) | 1
	  line This patch is meant to fix 8433; where clid and src are lost
	  via bridging. ........

2007-06-22 19:53 +0000 [r71190]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_sms.c: Code cleanups

2007-06-22 16:19 +0000 [r71146-71158]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Use stat to determine whether the file exists or
	  not. (issue #10038 reported by Mike Anikienko)

	* main/rtp.c: Behold the magic of casting!

2007-06-22 15:15 +0000 [r71093]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /, main/rtp.c: Merged revisions 71063 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun
	  2007) | 1 line My conditions for merging amaflags info was naive;
	  DOCUMENTATION is the default, although null is possible; theft of
	  user-settable fields is not good. Just copy them, leave them
	  alone. This is for bug 10016. (plus a small fix to rtp, to elim a
	  compiler warning (dev mode)) ........

2007-06-22 15:03 +0000 [r71069]  Jason Parker <jparker@digium.com>

	* /, res/res_agi.c, main/file.c, apps/app_speech_utils.c: Merged
	  revisions 71068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri,
	  22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
	  lines Fix a few silly usages of ast_playstream() - it only ever
	  returns 0... Issue 10035 ........ ................

2007-06-22 14:56 +0000 [r71067]  Brett Bryant <bbryant@digium.com>

	* /, main/asterisk.c: Merged revisions 71066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r71066 | bbryant | 2007-06-22 09:53:08 -0500
	  (Fri, 22 Jun 2007) | 18 lines Merged revisions 71064 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22
	  Jun 2007) | 10 lines Fixed infinite loop when controlling
	  terminal was lost and return value of input function wasn't
	  checked for errors. This would cause 100% cpu to be taken up.
	  (closes issue #9654, issue #10010) Reported by: mnicholson, and
	  eserra Idea for the patch from mnicholson, patched by me ........
	  ................

2007-06-22 04:35 +0000 [r71040]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, include/asterisk/utils.h, pbx/pbx_spool.c,
	  apps/app_dictate.c, apps/app_minivm.c, apps/app_test.c,
	  main/logger.c, main/utils.c, apps/app_sms.c, res/res_monitor.c,
	  apps/app_voicemail.c: Issue 9990 - New API ast_mkdir, which
	  creates parent directories as necessary (and is faster than an
	  outcall to mkdir -p)

2007-06-22 04:13 +0000 [r71024]  Jason Parker <jparker@digium.com>

	* build_tools/cflags.xml, main/asterisk.c: Nothing to see here.

2007-06-22 03:15 +0000 [r71004]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 71003 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r71003 | russell | 2007-06-21 22:14:41 -0500 (Thu, 21 Jun 2007) |
	  3 lines Fix a small typo which ... well ... completely broke
	  chan_iax2. oops! (issue #9937, patch by me) ........

2007-06-21 23:07 +0000 [r70961]  Jason Parker <jparker@digium.com>

	* main/manager.c, configs/manager.conf.sample,
	  include/asterisk/manager.h, main/rtp.c: Add manager events for
	  RTCP statistics. Also adds a new "reporting" permission for
	  manager, since it can be incredibly spammy. This permission was
	  discussed on the -dev mailing list some months back. Issue 8613,
	  patch by johann8384, with some minor changes by me.

2007-06-21 22:41 +0000 [r70951]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 70949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70949 | murf | 2007-06-21 16:34:41 -0600 (Thu,
	  21 Jun 2007) | 9 lines Merged revisions 70948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
	  line This little fix is in response to bug 10016, but may not
	  cure it. The code is wrong, clearly. In a situation where you set
	  the CDR's amaflags, and then ForkCDR, and then set the new CDR's
	  amaflags to some other value, you will see that all CDRs have had
	  their amaflags changed. This is not good. So I fixed it. ........
	  ................

2007-06-21 21:41 +0000 [r70900]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 70899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70899 | file | 2007-06-21 17:40:19 -0400 (Thu,
	  21 Jun 2007) | 10 lines Merged revisions 70898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
	  lines Don't explode if the gain option is specified without a
	  value. (issue #9274 reported by mfarver) ........
	  ................

2007-06-21 21:16 +0000 [r70877-70887]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 70883 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70883 | russell | 2007-06-21 16:14:53 -0500 (Thu, 21 Jun 2007) |
	  3 lines Put the thread reading from the socket back in the idle
	  list if it deferred the processing of a full frame to another
	  thread ........

	* /, channels/chan_iax2.c: Merged revisions 70866 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70866 | russell | 2007-06-21 16:07:04 -0500 (Thu, 21 Jun 2007) |
	  5 lines If a full frame is received while one of the iax2 threads
	  is in the middle of handling a full frame for the same call,
	  queue it up for processing by that same thread later instead of
	  dropping it. (issue #9937, patch by me) ........

2007-06-21 20:28 +0000 [r70857]  Steve Murphy <murf@digium.com>

	* /, cdr/cdr_custom.c: Merged revisions 70841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70841 | murf | 2007-06-21 14:19:36 -0600 (Thu,
	  21 Jun 2007) | 9 lines Merged revisions 70804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
	  line it was pointed out that the cdr_custom config load could get
	  a lock, and under certain circumstances, would never release it.
	  I also noted that the situation where more than one mapping spec
	  was warned about, but did not ignore further mappings as it had
	  promised. I think I have fixed both situations. ........
	  ................

2007-06-21 19:54 +0000 [r70809]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 70808 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70808 | mmichelson | 2007-06-21 14:49:44 -0500 (Thu, 21 Jun
	  2007) | 4 lines When volgain is used don't leave a temporary file
	  behind. (Closes Issue 8514, Reported and patched by ulogic, code
	  reviewed by Jason Parker) ........

2007-06-21 19:08 +0000 [r70794]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_buildopts_h: when we are building modules that
	  other modules depend on, create preprocessor defines (in
	  buildopts.h) marking that those modules were built

2007-06-21 18:40 +0000 [r70783]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Merge changes from team/russell/sla_reload *
	  Add support for the reload of sla.conf (closes issue #9481, patch
	  by me)

2007-06-21 18:03 +0000 [r70769]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Remove deprecated function call

2007-06-21 15:58 +0000 [r70729-70731]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Expand AGISTATUS variable to include NOTFOUND
	  which is set when the AGI file could not be found. (issue #9285
	  reported by srdjan)

	* /, main/rtp.c: Merged revisions 70727 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2
	  lines Do not Packet2Packet bridge if packetization settings do
	  not allow it. (issue #9117 reported by phsultan) ........

2007-06-21 15:23 +0000 [r70728]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 70726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) |
	  2 lines Remove a couple of duplicate unlocks ........

2007-06-21 14:00 +0000 [r70678]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 70677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70677 | file | 2007-06-21 09:58:36 -0400 (Thu, 21 Jun 2007) | 2
	  lines Fix building with ODBC storage enabled. (issue #10025
	  reported by denisgalvao) ........

2007-06-21 13:18 +0000 [r70676]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 70656 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70656 | murf | 2007-06-21 07:00:39 -0600 (Thu, 21 Jun 2007) | 1
	  line Via complaints aired in asterisk-users, I submit these
	  changes, which allow cdr updates to see macro context/exten,
	  whether hung up or not ........

2007-06-20 23:33 +0000 [r70613]  Jason Parker <jparker@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 70612 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70612 | qwell | 2007-06-20 18:32:39 -0500 (Wed, 20 Jun 2007) | 4
	  lines Fix some potential memory leaks in cdr_pgsql. Issue 10020,
	  patch by me, with credit to prashant_jois for pointing out the
	  problem. ........

2007-06-20 23:31 +0000 [r70611]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removed an extraneous debug message I'd
	  left in my previous commit

2007-06-20 23:31 +0000 [r70610]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, apps/app_queue.c: Fix trunk brokenness; also,
	  optimize application registration

2007-06-20 23:26 +0000 [r70607]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/pbx.c, apps/app_queue.c: Cleaning up a
	  small disaster I created earlier

2007-06-20 22:55 +0000 [r70555-70561]  Jason Parker <jparker@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 70560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70560 | qwell | 2007-06-20 17:55:21 -0500 (Wed, 20 Jun 2007) | 1
	  line Fix a stupid mistake in my last cdr_pgsql race condition fix
	  ........

	* /, cdr/cdr_pgsql.c: Merged revisions 70554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70554 | qwell | 2007-06-20 17:31:35 -0500 (Wed, 20 Jun 2007) | 4
	  lines Fix a race condition in cdr_pgsql that can occur when
	  reloading the module. Issue 10022, patch by me, with credit to
	  prashant_jois for finding the bug. ........

2007-06-20 22:24 +0000 [r70553]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 70552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70552 | file | 2007-06-20 18:22:20 -0400 (Wed,
	  20 Jun 2007) | 10 lines Merged revisions 70551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
	  lines Don't overwrite the configured username setting upon a
	  REGISTER. (issue #8565 reported by jsmith) ........
	  ................

2007-06-20 21:38 +0000 [r70531]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, apps/app_queue.c: As per 9228, now app_queue
	  should have the proper machinery to do gosubs.

2007-06-20 21:31 +0000 [r70530]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Main fix: Fixing a bug which caused
	  VoiceMailMain to always report that you had 0 messages when using
	  IMAP storage. Secondary fixes: adding locks to list access in
	  several places Big thanks to Russell Bryant for helping out with
	  this.

2007-06-20 20:54 +0000 [r70493-70495]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 70494 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r70494 | qwell | 2007-06-20 15:53:16 -0500 (Wed, 20 Jun
	  2007) | 4 lines Make sure we clear the previously dialed number
	  if it did not exist. Issue 9958. ........

	* main/http.c: Revert the change made in revision 45474, since this
	  causes other issues. Issue 10021.

2007-06-20 20:10 +0000 [r70461]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael_lex.c,
	  pbx/pbx_ael.c, doc/ael.tex, include/asterisk/ael_structs.h,
	  pbx/ael/ael.tab.h, CHANGES, pbx/ael/ael.flex: This finishes the
	  changes for making Macro args LOCAL to the call, and allowing
	  users to declare local variables.

2007-06-20 19:30 +0000 [r70446]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, /: Merged revisions 70445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70445 | tilghman | 2007-06-20 14:29:23 -0500
	  (Wed, 20 Jun 2007) | 10 lines Merged revisions 70444 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20
	  Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong
	  channel ........ ................

2007-06-20 18:48 +0000 [r70398]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 70397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70397 | russell | 2007-06-20 13:46:49 -0500
	  (Wed, 20 Jun 2007) | 13 lines Merged revisions 70396 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20
	  Jun 2007) | 5 lines Fix a problem where an established call would
	  not be properly disconnected when a PRI disconnect is received
	  depending on which cause code was received. (issue #9588,
	  original patch by softins, updated patch from jtexter3, and some
	  additional feedback from mhardeman) ........ ................

2007-06-20 17:55 +0000 [r70361]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /, main/rtp.c: Merged revisions 70360 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun
	  2007) | 2 lines Put the speex packetization values back in but
	  disable it when setting up the smoother. ........

2007-06-20 17:35 +0000 [r70358]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, pbx/pbx_ael.c: Merge work to make U(...) option
	  work for Dial

2007-06-20 14:33 +0000 [r70310]  Olle Johansson <oej@edvina.net>

	* channels/chan_zap.c: Show TDD status in "zap show channels"
	  Inspired by work at Omnitor in Sweden

2007-06-20 13:00 +0000 [r70253-70291]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Oops, shouldn't have taken that last shortcut
	  (also add some checks)

	* apps/app_stack.c: Another method of doing local variables,
	  hopefully a little closer to what codefreeze had in mind

	* apps/app_stack.c: Local variables for codefreeze

2007-06-20 02:13 +0000 [r70234]  Russell Bryant <russell@digium.com>

	* /, contrib/scripts/ast_grab_core: Merged revisions 70164 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70164 | russell | 2007-06-19 19:03:22 -0500 (Tue, 19 Jun 2007) |
	  2 lines don't delete the backtrace in ast_grab_core ........

2007-06-20 00:26 +0000 [r70199]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 70198 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70198 | file | 2007-06-19 20:24:36 -0400 (Tue, 19 Jun 2007) | 2
	  lines Don't do packetization/smoother stuff with speex, it
	  doesn't work. ........

2007-06-19 23:38 +0000 [r70122-70162]  Steve Murphy <murf@digium.com>

	* CHANGES: Added a little verbage to CHANGES

	* apps/app_dial.c, apps/app_queue.c, apps/app_rpt.c: Via bug9228,
	  no way to create macros via AEL, and some of the apps allow you
	  to call macros..., I modded the apps that allow macro calls to
	  allow gosubs calls also, to make them AEL compliant.

	* UPGRADE.txt, CHANGES: Moved those comments from UPGRADE.txt to
	  CHANGES. Ooops.

	* UPGRADE.txt: Some UPGRADE.txt comments to cover some enhancements
	  added today.

	* configs/cdr_manager.conf.sample, cdr/cdr_manager.c: This
	  enhancement provided via bug 9993, a patch to upgrade cdr_manager
	  to have cdr_custom capabilities. Many thanks to eserra for this
	  contribution

2007-06-19 19:15 +0000 [r70088]  Russell Bryant <russell@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 70084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) |
	  3 lines Only attempt to queue a hangup on the owner channel if it
	  actually exists. (issue #9795, patch from zandbelt) ........

2007-06-19 18:31 +0000 [r70063]  Steve Murphy <murf@digium.com>

	* main/channel.c, /: Merged revisions 70062 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue,
	  19 Jun 2007) | 9 lines Merged revisions 70053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
	  line This fixes 9246, where channel variables are not available
	  in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
	  consolidate the channel variables during a masquerade, and then
	  copy the merged variables back onto the clone, so the zombie has
	  the same vars that the 'original' has. ........ ................

2007-06-19 17:09 +0000 [r70006]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 70003 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue,
	  19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
	  lines Handle the CC field in the RTP header. (issue #9384
	  reported by DoodleHu) ........ ................

2007-06-19 17:07 +0000 [r70001]  Steve Murphy <murf@digium.com>

	* include/asterisk/callerid.h, channels/chan_zap.c,
	  doc/India-CID.txt (added), configs/zapata.conf.sample: These
	  changes were submitted via bug 6683, to allow CID detection in
	  India, with carriers that do Polarity/DTMF CID signalling.

2007-06-19 16:25 +0000 [r69988]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 69987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69987 | file | 2007-06-19 12:24:31 -0400 (Tue,
	  19 Jun 2007) | 10 lines Merged revisions 69986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
	  lines Update BRIDGEPEER variable if set to the new channel name
	  when a masquerade happens. (issue #9699 reported by dimas)
	  ........ ................

2007-06-19 15:27 +0000 [r69945]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) |
	  10 lines Fix a crash that could occur when handing device state
	  changes. When the state of a device changes, the device state
	  thread tells the extension state handling code that it changed.
	  Then, the extension state code calls the callback in chan_sip so
	  that it can update subscriptions to that extension. A pointer to
	  a sip_pvt structure is passed to this function as the call which
	  needs a NOTIFY sent. However, there was no locking done to ensure
	  that the pvt struct didn't disappear during this process. (issue
	  #9946, reported by tdonahue, patch by me, patch updated to trunk
	  to use the sip_pvt lock wrappers by eliel) ........

2007-06-19 15:14 +0000 [r69943]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: Add support for
	  setting nature of address, presentation, and other related SS7
	  number options (#10000)

2007-06-19 13:56 +0000 [r69850-69896]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 69895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69895 | file | 2007-06-19 09:55:25 -0400 (Tue,
	  19 Jun 2007) | 10 lines Merged revisions 69894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
	  lines Perform an extra hangup check just in case. (issue #9589
	  reported by bcnit) ........ ................

	* /, res/res_features.c: Merged revisions 69847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69847 | file | 2007-06-19 09:00:57 -0400 (Tue,
	  19 Jun 2007) | 10 lines Merged revisions 69846 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
	  lines Add parked call extension AFTER the parking slot has been
	  announced, otherwise two threads will try to handle the same
	  channel and it will go kaboom. (issue #9191 reported by japple)
	  ........ ................

2007-06-18 23:28 +0000 [r69808-69809]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Undoing my last commit. I misread the code
	  before.

	* apps/app_voicemail.c: Cleaned up a section where there were two
	  consecutive identical if statements. Combined the bodies of the
	  two into one if. I blame svn merging for this.

2007-06-18 22:23 +0000 [r69807]  Brett Bryant <bbryant@digium.com>

	* apps/app_queue.c: Fixed issue where 'stop gracfeully' was hanging
	  ...

2007-06-18 21:58 +0000 [r69806]  Joshua Colp <jcolp@digium.com>

	* /, main/callerid.c: Merged revisions 69805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69805 | file | 2007-06-18 17:57:10 -0400 (Mon, 18 Jun 2007) | 2
	  lines Fix for building on PowerPC under Linux. ........

2007-06-18 19:52 +0000 [r69797]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69796 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007)
	  | 2 lines Issue 10005 - Segfault with missing arguments, plus fix
	  a missing define for SIP INFO channels ........

2007-06-18 19:02 +0000 [r69779-69795]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69794 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2
	  lines Don't count RTP timeout when involved in a T38 fax session.
	  (issue #9222 reported by ivoc) ........

	* /, channels/chan_sip.c: Merged revisions 69775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69775 | file | 2007-06-18 14:18:12 -0400 (Mon,
	  18 Jun 2007) | 10 lines Merged revisions 69765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
	  lines Set the peer name on the dialog to the one configured in
	  sip.conf and NOT the username to be used for authentication
	  attempts. (issue #9967 reported by achauvin) ........
	  ................

2007-06-18 17:50 +0000 [r69745-69746]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/safe_asterisk: Merged revisions 69744 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69744 | tilghman | 2007-06-18 12:46:40 -0500
	  (Mon, 18 Jun 2007) | 10 lines Merged revisions 69743 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18
	  Jun 2007) | 2 lines Issue 9998 - Remove SIG prefix, since it's
	  not supported by ksh ........ ................

	* apps/app_rpt.c: Janitor for ast_localtime

2007-06-18 16:56 +0000 [r69705-69709]  Joshua Colp <jcolp@digium.com>

	* main/dnsmgr.c, /: Merged revisions 69708 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69708 | file | 2007-06-18 12:51:36 -0400 (Mon, 18 Jun 2007) | 2
	  lines Remember the DNS lookup done when dnsmgr is called for the
	  first time so that it does not needlessly spit out changed
	  messages when the host really didn't change. ........

	* main/cdr.c, main/dnsmgr.c, main/asterisk.c: Few more rwlist
	  conversions... why not.

2007-06-18 16:35 +0000 [r69691-69703]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /, build_tools/menuselect-deps.in,
	  configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in,
	  configure.ac, cdr/cdr_odbc.c, res/res_odbc.c,
	  apps/app_voicemail.c: Merged revisions 69702 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) |
	  6 lines To prevent 92138749238754 more reports of "I have
	  unixodbc installed, but still can't build *_odbc.so!", check for
	  ltdl directly, instead of just listing it as another library to
	  include in the unixodbc check in the configure script. This also
	  makes ltdl show up as a dependency in menuselect so people know
	  what to go install. (related to issue #9989, patch by me)
	  ........

	* /, build_tools/prep_moduledeps: Merged revisions 69689 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69689 | russell | 2007-06-18 11:15:12 -0500 (Mon, 18 Jun 2007) |
	  5 lines Change the use of "echo -e" to "printf". On systems where
	  /bin/sh is not bash, most of the lines in menuselect-tree were
	  getting a "-e" at the beginning of every line. I'm surprised
	  nobody noticed this, but I think the XML parser was being very
	  nice and ignoring them. ........

2007-06-18 16:06 +0000 [r69663-69672]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69668 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2
	  lines Don't defer the BYE till later on a transfer when the
	  transfer itself goes kaboom and has no hope of working. ........

	* /, channels/chan_sip.c: Merged revisions 69661 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2
	  lines Few minor transfer tweaks. We can't unlock something we
	  never locked, and better handle a specific scenario with doing an
	  attended transfer between two non-bridged calls. ........

2007-06-18 15:46 +0000 [r69662]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 69660 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69660 | russell | 2007-06-18 10:46:14 -0500 (Mon, 18 Jun 2007) |
	  2 lines Tweak paths for BSD systems (issue #10001, stuarth)
	  ........

2007-06-18 13:57 +0000 [r69626]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69625 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2
	  lines Fix issue where it would be possible for the negotiated
	  codecs to get set back to nothing. (issue #9992 reported by
	  yehavi) ........

2007-06-15 20:21 +0000 [r69583]  Russell Bryant <russell@digium.com>

	* /: This was only an issue in 1.4. This issue was fixed in trunk
	  as a part of bbryant's patch to support named dynamic feature
	  groups. Merged revisions 69579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69579 | russell | 2007-06-15 15:18:58 -0500 (Fri, 15 Jun 2007) |
	  5 lines Fix a silly deadlock in res_features that I found while
	  debugging on one of blitzrage's test machines. It was one of the
	  situations where he was seeing hung channels, and may be the
	  cause of some of the reports from other people. (related to issue
	  #9235) ........

2007-06-15 19:25 +0000 [r69559]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 69558 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri,
	  15 Jun 2007) | 2 lines Add support for setting the maximum length
	  of acceptable DTMF in SpeechBackground.

2007-06-15 15:36 +0000 [r69519]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 69518 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) |
	  5 lines The SLATRUNK_STATUS variable indicated "SUCCESS" for both
	  an answer of the incoming call on the trunk, or if the trunk
	  reached its ring timeout. This patch changes the variable to say
	  "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle,
	  patch by me) ........

2007-06-14 23:23 +0000 [r69471]  Jason Parker <jparker@digium.com>

	* /, main/config.c: Merged revisions 69470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69470 | qwell | 2007-06-14 18:22:51 -0500 (Thu,
	  14 Jun 2007) | 12 lines Merged revisions 69469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
	  lines Fix an issue where the line number in an unterminated
	  comment block error message would show the wrong line number.
	  "Reported" to me on #asterisk (somebody posted an error message,
	  and I happened to catch it) ........ ................

2007-06-14 23:01 +0000 [r69436]  Russell Bryant <russell@digium.com>

	* main/pbx.c, channels/chan_vpb.cc, apps/app_meetme.c,
	  res/res_features.c, channels/iax2-provision.c, main/enum.c,
	  res/res_monitor.c, apps/app_speech_utils.c, main/loader.c,
	  main/cli.c, main/channel.c, channels/chan_misdn.c,
	  apps/app_minivm.c, main/http.c, main/file.c,
	  channels/chan_h323.c, res/res_indications.c,
	  apps/app_directory.c, main/asterisk.c: Convert uses of strdup()
	  to ast_strdup() (issue #9983, eliel)

2007-06-14 22:56 +0000 [r69435]  Jason Parker <jparker@digium.com>

	* /, sounds/Makefile: Merged revisions 69434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69434 | qwell | 2007-06-14 17:56:09 -0500 (Thu, 14 Jun 2007) | 1
	  line Update to latest versions of sound files. ........

2007-06-14 22:09 +0000 [r69394-69405]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/utils.h, main/pbx.c, /, main/say.c,
	  cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c,
	  cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
	  main/manager.c, cdr/cdr_sqlite.c, apps/app_minivm.c,
	  main/callerid.c, main/logger.c, main/stdtime/localtime.c,
	  cdr/cdr_odbc.c, main/asterisk.c, cdr/cdr_manager.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions
	  69392 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007)
	  | 2 lines use ast_localtime() in every place localtime_r() was
	  being used ........

	* formats/format_ogg_vorbis.c: oops... somebody patched this module
	  without compile-testing it... bad :-)

2007-06-14 21:09 +0000 [r69327-69360]  Russell Bryant <russell@digium.com>

	* /, main/say.c: Merged revisions 69358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69358 | russell | 2007-06-14 16:08:23 -0500 (Thu, 14 Jun 2007) |
	  3 lines Fix some problems with saying dates and times for the
	  "tw" langauge (issue #9964, ljmid) ........

	* CHANGES: update CHANGES for tw support in voicemail

	* apps/app_voicemail.c: Add support for the tw language in
	  voicemail (issue #9964, ljmid)

	* funcs/func_rand.c, main/frame.c, channels/chan_local.c,
	  res/res_features.c, apps/app_record.c, funcs/func_strings.c,
	  apps/app_test.c, main/devicestate.c, apps/app_alarmreceiver.c,
	  apps/app_ices.c, channels/chan_iax2.c, main/config.c,
	  res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c,
	  apps/app_zapras.c, apps/app_amd.c, channels/chan_alsa.c,
	  cdr/cdr_odbc.c, main/db.c, apps/app_dial.c, formats/format_wav.c,
	  channels/chan_agent.c, apps/app_disa.c,
	  formats/format_ogg_vorbis.c, channels/iax2-provision.c,
	  apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c,
	  apps/app_zapbarge.c, channels/chan_misdn.c,
	  channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
	  formats/format_g726.c, apps/app_chanspy.c, main/asterisk.c,
	  apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c,
	  res/res_musiconhold.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
	  apps/app_followme.c, codecs/codec_zap.c, cdr/cdr_radius.c,
	  res/res_jabber.c, res/res_config_sqlite.c, main/enum.c,
	  cdr/cdr_csv.c, main/cdr.c, main/channel.c, main/dial.c,
	  channels/chan_phone.c, apps/app_osplookup.c, apps/app_minivm.c,
	  res/res_agi.c, apps/app_mp3.c, main/app.c, apps/app_rpt.c,
	  main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c,
	  res/res_config_pgsql.c, funcs/func_version.c,
	  channels/chan_zap.c, funcs/func_db.c, channels/chan_sip.c,
	  apps/app_festival.c, apps/app_waitforsilence.c, res/res_crypto.c,
	  res/res_adsi.c, main/acl.c, apps/app_queue.c, cdr/cdr_tds.c,
	  channels/chan_jingle.c, apps/app_channelredirect.c,
	  apps/app_directed_pickup.c, main/adsistub.c, main/callerid.c,
	  main/file.c, channels/chan_h323.c, channels/chan_nbs.c,
	  apps/app_stack.c, main/dsp.c: Add a massive set of changes for
	  converting to use the ast_debug() macro. (issue #9957, patches
	  from mvanbaak, caio1982, critch, and dimas)

2007-06-14 16:41 +0000 [r69308]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Clean up debug messages a little bit for ss7
	  linkset debugging

2007-06-14 15:43 +0000 [r69261]  Brett Bryant <bbryant@digium.com>

	* main/manager.c: Couple of manager ssl options weren't loading
	  because of a typo.

2007-06-14 15:25 +0000 [r69260]  Jason Parker <jparker@digium.com>

	* funcs/func_groupcount.c, /: Merged revisions 69259 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69259 | qwell | 2007-06-14 10:21:29 -0500 (Thu,
	  14 Jun 2007) | 12 lines Merged revisions 69258 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4
	  lines Change a quite broken while loop to a for loop, so
	  "continue;" works as expected instead of eating 99% CPU... Issue
	  9966, patch by me. ........ ................

2007-06-13 21:20 +0000 [r69223]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 69221 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69221 | file | 2007-06-13 17:17:28 -0400 (Wed, 13 Jun 2007) | 2
	  lines Let's make chan_iax2 media only native transfers actually
	  work. (issue #9376 reported by simone cittadini) ........

2007-06-13 20:03 +0000 [r69187]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69183 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) |
	  9 lines Move the logic for destroying a call when no response is
	  received to a BYE outside of the block that checks for FLAG_FATAL
	  to be set. This flag is only set when the packet is transmitted
	  with the reliability set to XMIT_CRITICAL when the original
	  packet is transmitted. A BYE is always sent with it set to
	  XMIT_RELIABLE, meaning this code could never be encountered. This
	  resulted in seeing some SIP channels that would never go away
	  with the last packet sent being a BYE. (part of issue #9235,
	  patch from jcmoore) ........

2007-06-13 20:00 +0000 [r69185]  Joshua Colp <jcolp@digium.com>

	* /, channels/iax2-parser.c: Merged revisions 69184 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r69184 | file | 2007-06-13 15:58:59 -0400 (Wed, 13 Jun
	  2007) | 2 lines Add TXMEDIA to list so that it is properly
	  displayed during iax2 packet output. ........

2007-06-13 19:47 +0000 [r69182]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 69181 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69181 | mmichelson | 2007-06-13 14:41:13 -0500 (Wed, 13 Jun
	  2007) | 5 lines Contains a patch for fixing an encoding problem
	  when using Outlook to view voicemail emails and attachments. This
	  fix has also been tested on Thunderbird, Evolution, Pine, and
	  Mutt. (Issue 9336, reported by marwick, patched by mutterc)
	  ........

2007-06-13 19:10 +0000 [r69147]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 69144 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2
	  lines Really ignore NULL frames and check whether the channel
	  hungup or not. (issue #9912 reported by junky) ........

2007-06-13 19:05 +0000 [r69137]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c: Completely remove callback mode and all
	  references to it from chan_agent. Issue 9969, patch by eliel.

2007-06-13 18:23 +0000 [r69129-69130]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/app.h, funcs/func_groupcount.c, main/app.c,
	  main/cli.c: Use read/write lock based lists for group counting.

	* /, main/app.c: Merged revisions 69128 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r69128 | file | 2007-06-13 14:16:00 -0400 (Wed,
	  13 Jun 2007) | 10 lines Merged revisions 69127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
	  lines Return group counting to previous behavior where you could
	  only have one group per category. (issue #9711 reported by
	  irroot) ........ ................

2007-06-13 17:37 +0000 [r69081-69108]  Jason Parker <jparker@digium.com>

	* res/res_config_pgsql.c: Continuation of issue 9968 (revision
	  69081). This should be the last one.

	* main/pbx.c, channels/chan_sip.c: Fixes for ast_strlen_zero()
	  janitor project. Issue 9968, patch by eliel.

2007-06-13 16:59 +0000 [r69017-69072]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 69071 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) |
	  3 lines Clarify a bit of logic. This doesn't change behavior in
	  any way, but it is helpful when following the logic to debug
	  problems like 9235. ........

	* /, channels/chan_iax2.c: Merged revisions 69069 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69069 | russell | 2007-06-13 11:29:12 -0500 (Wed, 13 Jun 2007) |
	  3 lines Fix a place where a chan_iax2 pvt struct was accessed
	  without the lock held. This issue was reported to me via email by
	  Dmitry Mishchenko. Thanks! ........

	* res/snmp/agent.c: Simplify some logic and convert spaces to tabs

	* res/snmp/agent.c: The variable used for the return value must be
	  declared as static. I broke this when applying the patch, sorry!
	  (issue #9637, jeffg)

	* include/asterisk/logger.h: Put parenthesis around the level
	  argument to ast_debug()

	* /, cdr/cdr_pgsql.c: Merged revisions 69016 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69016 | russell | 2007-06-12 14:40:17 -0500 (Tue, 12 Jun 2007) |
	  4 lines Fix a memory leak pointed out by prashant_jois in
	  #asterisk-bugs. PQclear() was not called on the result structure
	  after doing a PQexec(). Also, fix up some formatting in passing.
	  ........

2007-06-12 19:38 +0000 [r69013-69015]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 69014 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69014 | file | 2007-06-12 15:36:29 -0400 (Tue, 12 Jun 2007) | 2
	  lines Change the full frame dropping log message to debug to
	  avoid future bug reports. ........

	* /, channels/chan_iax2.c: Merged revisions 69012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69012 | file | 2007-06-12 15:26:38 -0400 (Tue, 12 Jun 2007) | 2
	  lines Schedule the sending of a PING packet a second later than
	  previously so that it does not collide with the LAGRQ. ........

2007-06-12 19:19 +0000 [r68970-69011]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 69010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) |
	  12 lines In ast_channel_make_compatible(), just return if the
	  channels' read and write formats already match up. There are code
	  paths that call this function on a pair of channels multiple
	  times. This made calls fail that were using g729 in some cases.
	  The reason is that codec_g729a will unregister itself from the
	  list of available translators will all licenses are in use. So,
	  the first time the function got called, the right translation
	  path was allocated. However, the second time it got called, the
	  code would not find a translation path to/from g729 and make the
	  call fail, even if the channel actually already had a g729
	  translation path allocated. (SPD-32) ........

	* main/pbx.c: Convert pbx.c to use ast_debug() for debug logging.
	  (issue #9925, dimas)

	* include/asterisk/logger.h: Add a new macro, ast_debug(), which
	  combines the check of the value of option_debug and the actual
	  call to ast_log(). (issue #9925, dimas)

	* doc/ast_appdocs.tex: update the dump of application docs

	* apps/app_dial.c, apps/app_privacy.c, apps/app_authenticate.c,
	  channels/chan_agent.c, apps/app_image.c, apps/app_chanisavail.c,
	  apps/app_transfer.c, apps/app_system.c, apps/app_queue.c,
	  apps/app_playback.c, apps/app_controlplayback.c,
	  apps/app_osplookup.c, apps/app_sendtext.c, apps/app_minivm.c,
	  apps/app_url.c, pbx/pbx_config.c, include/asterisk/options.h,
	  apps/app_voicemail.c: Completely remove all of the code related
	  to jumping to priority n + 101. yay! (issue #9926, caio1982)

2007-06-12 14:26 +0000 [r68900-68923]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 68922 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue,
	  12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
	  lines Bring RTP back to Asterisk at the end of a native bridge no
	  matter what. ........ ................

	* main/autoservice.c, main/app.c: Even more minor code cleanup!

	* main/channel.c: Minor code cleanup.

	* channels/chan_agent.c: Remove old stuff from the
	  AgentCallbackLogin days and only autocomplete agents in the agent
	  logoff CLI command that are logged in. (issue #9952 reported by
	  eliel)

2007-06-11 22:31 +0000 [r68855]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/frame.c: corrected CLI 'core show codecs' syntax for issue
	  9945, thanks eserra.

2007-06-11 22:21 +0000 [r68854]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_disa.c, UPGRADE.txt: Issue 8971 - Allow DISA input to be
	  ended with a '#'.

2007-06-11 22:07 +0000 [r68816-68831]  Jason Parker <jparker@digium.com>

	* main/manager.c, configs/manager.conf.sample: Change
	  displayconnects option in manager.conf to be per-user. Issue
	  9932, patch by eliel

	* /, include/asterisk/time.h: Merged revisions 68814 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r68814 | qwell | 2007-06-11 16:20:15 -0500 (Mon, 11 Jun
	  2007) | 2 lines Solaris 10 sometimes (?) needs this include in
	  order to have NULL defined. ........

2007-06-11 20:51 +0000 [r68782]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_directory.c: Merged revisions 68781 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68781 | tilghman | 2007-06-11 15:45:53 -0500 (Mon, 11 Jun 2007)
	  | 2 lines Issue 9947 - fn2 was unused / incorrectly used ........

2007-06-11 17:05 +0000 [r68740]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 68733 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68733 | crichter | 2007-06-11 18:57:59 +0200
	  (Mo, 11 Jun 2007) | 9 lines Merged revisions 68732 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11
	  Jun 2007) | 1 line added check for NULL Pointer when calling
	  misdn_new. Asterisk does not allow us to create channels anymore
	  when stop gracefully is used :). also modified the
	  restart_indicator to 0 ........ ................

2007-06-11 14:41 +0000 [r68662-68685]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Change channel list to read/write list... I'm
	  crazy.

	* main/channel.c, /: Merged revisions 68683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68683 | file | 2007-06-11 10:33:12 -0400 (Mon,
	  11 Jun 2007) | 10 lines Merged revisions 68682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
	  lines Improve deadlock handling of the channel list. (issue #8376
	  reported by one47) ........ ................

	* main/manager.c: Add username completion for manager show user CLI
	  command. (issue #9929 reported by eliel)

	* configs/sip.conf.sample: Update documentation for proper CLI
	  commands. (issue #9936 reported by eserra)

2007-06-11 11:40 +0000 [r68661]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  channels/misdn/isdn_lib.c: Merged revisions 68644 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68644 | crichter | 2007-06-11 12:29:18 +0200
	  (Mo, 11 Jun 2007) | 9 lines Merged revisions 68631 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11
	  Jun 2007) | 1 line fixed problem that the dummybc chanels had no
	  lock, checking for the lock now. Also fixed the channel restart
	  stuff, we can now specify and restart particular channels too.
	  ........ ................

2007-06-11 04:28 +0000 [r68596]  Tilghman Lesher <tlesher@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 68595 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68595 | tilghman | 2007-06-10 23:21:30 -0500 (Sun, 10 Jun 2007)
	  | 2 lines "dialplan save" produced garbage in the config file
	  ........

2007-06-09 01:06 +0000 [r68575]  Jason Parker <jparker@digium.com>

	* channels/chan_misdn.c: Fix compile errors in chan_misdn.c
	  Reported by d1mas in #asterisk-bugs on IRC.

2007-06-08 22:23 +0000 [r68473-68528]  Russell Bryant <russell@digium.com>

	* /, apps/app_dictate.c: Merged revisions 68527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68527 | russell | 2007-06-08 17:23:22 -0500
	  (Fri, 08 Jun 2007) | 12 lines Merged revisions 68526 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08
	  Jun 2007) | 4 lines Don't automatically hang up after running
	  Dictate so that callers can exit cleanly using '#' (closes issue
	  #9577, patch from Thomas Andrews) ........ ................

	* doc/asterisk-mib.txt, res/snmp/agent.c: Add support for
	  retrieving the number of channels that are currently bridged via
	  SNMP. (closes issue #9637, initial patch from jeffg, modified by
	  me)

	* include/asterisk/app.h, res/res_agi.c, main/app.c,
	  apps/app_controlplayback.c, apps/app_voicemail.c: Add an option
	  for ControlPlayback to be able to start at an offset from the
	  beginning of the file. Also, add a channel variable that
	  indicates the location in the file where the Playback was
	  stopped. (closes issue #7655, patch from sharkey)

	* main/pbx.c: Add channel variable manager event (issue #7291,
	  patch from tonyh and jontow)

2007-06-08 16:03 +0000 [r68453]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 68450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68450 | kpfleming | 2007-06-08 10:52:47 -0500 (Fri, 08 Jun 2007)
	  | 2 lines actually remember the type/subclass of full frames that
	  are in process ........

2007-06-08 15:51 +0000 [r68449]  Jason Parker <jparker@digium.com>

	* res/res_config_sqlite.c: Fix incorrect logic for param count.
	  Issue 9918.

2007-06-08 15:32 +0000 [r68448]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Minor formatting change to test changes to
	  mantis auto-closing issues (closes issue #6000)

2007-06-08 00:18 +0000 [r68374-68405]  Joshua Colp <jcolp@digium.com>

	* /, main/say.c: Merged revisions 68401 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68401 | file | 2007-06-07 20:17:04 -0400 (Thu,
	  07 Jun 2007) | 10 lines Merged revisions 68397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
	  lines Don't call ast_waitstream_full when the control file
	  descriptor and audio file descriptor are not set, simply call
	  ast_waitstream! (issue #8530 reported by rickead2000) ........
	  ................

	* main/dnsmgr.c, /: Merged revisions 68370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68370 | file | 2007-06-07 20:02:34 -0400 (Thu,
	  07 Jun 2007) | 10 lines Merged revisions 68368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
	  lines Do a DNS lookup immediately upon calling the dnsmgr
	  function, don't wait until a refresh happens. (issue #9097
	  reported by plack) ........ ................

2007-06-07 23:17 +0000 [r68339-68359]  Russell Bryant <russell@digium.com>

	* /, main/say.c: Merged revisions 68354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68354 | russell | 2007-06-07 18:14:45 -0500
	  (Thu, 07 Jun 2007) | 11 lines Merged revisions 68351 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07
	  Jun 2007) | 3 lines Fix a problem where saying a character
	  wouldn't properly break out when the caller pressed '#' (issue
	  #8113, reported by patbaker82, patch from jamesgolovich (hey,
	  long time no see!) and patbaker82) ........ ................

	* include/asterisk/devicestate.h, channels/chan_sip.c,
	  contrib/asterisk-ng-doxygen, main/devicestate.c,
	  include/asterisk/manager.h, res/res_config_sqlite.c, main/rtp.c,
	  include/asterisk/http.h, include/asterisk/doxyref.h,
	  main/manager.c, include/asterisk/event.h, funcs/func_shell.c,
	  apps/app_skel.c, channels/chan_h323.c,
	  include/asterisk/strings.h, include/asterisk/stringfields.h: Fix
	  a bunch of doxygen errors and document more things (issue #9842,
	  snuffy)

2007-06-07 23:00 +0000 [r68327]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 68326 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68326 | qwell | 2007-06-07 18:00:01 -0500 (Thu, 07 Jun 2007) | 5
	  lines Fix incorrect French syntax of "old messages". Request for
	  feedback was sent to asterisk-dev mailing list, with little
	  response. Issue 9118, patch by junky. ........

2007-06-07 22:38 +0000 [r68325]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a couple of places that got missed in
	  the conversion to using the new API call for creating detached
	  threads. (issue #9915, reported by elguro, fixed by me)

2007-06-07 22:18 +0000 [r68321]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 68313 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68313 | kpfleming | 2007-06-07 17:14:35 -0500 (Thu, 07 Jun 2007)
	  | 6 lines some improvements to the IAX2 full frame dropping logic
	  recently added: - use inaddrcmp(), since we have it - output the
	  type of frame and subclass being dropped, and the type/subclass
	  that is already being processed (which caused the drop) ........

2007-06-07 21:22 +0000 [r68284-68289]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: Doxygenify a lot of the functions in res_jabber
	  (issue #9886, snuffy)

	* /, channels/chan_agent.c, apps/app_queue.c: Merged revisions
	  68280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68280 | russell | 2007-06-07 16:16:07 -0500 (Thu, 07 Jun 2007) |
	  4 lines Fix loading persistent queue members when using realtime
	  configuration for queues. Also, remove an unneeded leading slash
	  for the astdb family. (issue #9911, patch by atis) ........

2007-06-07 20:25 +0000 [r68220-68251]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 68249 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r68249 | qwell | 2007-06-07 15:25:18 -0500 (Thu, 07 Jun
	  2007) | 4 lines Fix an issue with newer phones which require
	  packets be padded out to the correct length. Issue 9887, patch by
	  DEA. ........

	* /, apps/app_voicemail.c: Merged revisions 68211 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68211 | qwell | 2007-06-07 15:06:00 -0500 (Thu,
	  07 Jun 2007) | 12 lines Merged revisions 68204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
	  lines Don't try to save voicemail greetings unless the user
	  presses '1' to accept/save. Issue 9904, patch by me. ........
	  ................

2007-06-07 19:51 +0000 [r68201]  Olle Johansson <oej@edvina.net>

	* CREDITS: Adding Philippe to CREDITS for hard work on detecting
	  bugs in our jabber/jingle integration

2007-06-07 19:50 +0000 [r68200]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 68198 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun
	  2007) | 5 lines Submitting a fix for Issue 8016. Added a check to
	  make sure that greetings get stored properly. (Issue 8016,
	  reported by edhorton, patched by alamantia with modification by
	  me. Thanks to Jason Parker for the advice on this). ........

2007-06-07 19:49 +0000 [r68195-68199]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_features.c: Merged revisions 68196 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r68196 | oej | 2007-06-07 21:46:10 +0200 (Thu, 07 Jun
	  2007) | 2 lines Disable chan_features by default in menuselect
	  ........

	* channels/chan_sip.c: - Doxygen updates - Adding docs on flags to
	  be able to clean up a bit

2007-06-07 19:31 +0000 [r68193]  Russell Bryant <russell@digium.com>

	* /, main/strcompat.c: Merged revisions 68192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68192 | russell | 2007-06-07 14:30:30 -0500 (Thu, 07 Jun 2007) |
	  3 lines Include stdarg.h for build issues on Solaris (issue
	  #9381) ........

2007-06-07 18:41 +0000 [r68138-68158]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 68157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2
	  lines Fix logic when doing a name based channel search for a
	  structure when you want to start from a specific point in the
	  channel list. (issue #9324 reported by slavon) ........

	* doc/queues-with-callback-members.tex: AEL in trunk now uses GOSUB
	  so we have to update the queues with callback members example.
	  (issue #9813 reported by Mike Anikienko)

2007-06-07 15:48 +0000 [r68118]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: Minor formatting change ... testing mantis
	  stuff to see if we're done (issue #9790) (closes issue #9816)

2007-06-07 14:23 +0000 [r68072]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 68071 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r68071 | file | 2007-06-07 10:21:59 -0400 (Thu,
	  07 Jun 2007) | 10 lines Merged revisions 68070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
	  lines Allow the 'g' option to work if used with the 'S' option.
	  (issue #9888 reported by gasparz) ........ ................

2007-06-07 10:06 +0000 [r67991-68040]  Olle Johansson <oej@edvina.net>

	* /, res/res_jabber.c: Merged revisions 68030 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68030 | oej | 2007-06-07 12:00:17 +0200 (Thu, 07 Jun 2007) | 2
	  lines Adding a few Todo's to res_jabber so we don't forget.
	  ........

	* /, res/res_jabber.c: Merged revisions 68028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68028 | oej | 2007-06-07 11:55:13 +0200 (Thu, 07 Jun 2007) | 4
	  lines Ok, we found out that this is not about if you have any
	  *active* clients using TLS, but if you have initialized TLS at
	  all during the lifetime of the module. So if you reload to
	  disable TLS, it won't help. ........

	* /, res/res_jabber.c: Merged revisions 68027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r68027 | oej | 2007-06-07 11:42:26 +0200 (Thu, 07 Jun 2007) | 8
	  lines If you have a jabber client that uses TLS, refuse unload.
	  Bad fix, but will prevent crashes while we are trying to find a
	  workaround. Iksemel development seems to have stalled and we
	  might have to stop using the TCP/TLS connections in that library
	  and use our own, which would scale better from a poll/select
	  perspective I guess. It would also make it easier to migrate to
	  OpenSSL and stop Asterisk from depending on both OpenSSL and
	  GnuTLS. ........

	* /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions
	  67993 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6
	  lines Issue #9738 - Make sure we can unload res_jabber. Patch by
	  phsultan - thanks! Due to a bug in the iksemel library, this will
	  not work if you are using GTLS in the connection. That's being
	  investigated. If you figure out a way to handle that without us
	  having to patch iksemel, let us know in the bug report. Thanks.
	  ........

	* res/res_jabber.c: Simplification of res_jabber code (done at
	  Inria, Paris with Philippe)

	* main/strcompat.c: Reverting part of #67864 to be able to compile
	  agi/eagi-test that relies on this without having ast_log and
	  other asterisk api functions available - I could not compile on
	  OS/X without reverting this.

2007-06-07 00:12 +0000 [r67925-67944]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 67941 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67941 | file | 2007-06-06 20:10:48 -0400 (Wed,
	  06 Jun 2007) | 10 lines Merged revisions 67938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
	  lines Only notify the devicestate system of a peer state change
	  when the peer is built from the config file. (issue #9900
	  reported by arkadia) ........ ................

	* /, main/file.c: Merged revisions 67924 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67924 | file | 2007-06-06 19:38:15 -0400 (Wed, 06 Jun 2007) | 2
	  lines Properly handle cases where a stream can't be written to.
	  (issue #9757 reported by junky) ........

2007-06-06 23:12 +0000 [r67920]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Allow overlapdialing directions to be
	  configurable. Bug #8554

2007-06-06 22:35 +0000 [r67901]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_iax2.c: added CLI 'iax2 unregister <peername>' for
	  issue 9812, thanks eliel

2007-06-06 22:27 +0000 [r67875-67895]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Remove our little
	  joke that was making fun of email disclaimers which nobody else
	  seemed to think was very funny. Oh well ... :)

	* /, res/res_snmp.c: Merged revisions 67872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67872 | russell | 2007-06-06 17:08:02 -0500 (Wed, 06 Jun 2007) |
	  6 lines Disable reload functionality in res_snmp. It is not
	  possible to initialize the snmp library more than once without
	  completely unloading the module and loading it again. (issue
	  #9571, reported by hristo, additional helpful debug information
	  from festr, patch from me) ........

2007-06-06 21:20 +0000 [r67864]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, main/autoservice.c, main/frame.c,
	  channels/chan_local.c, apps/app_readfile.c, res/res_features.c,
	  main/threadstorage.c, main/say.c, funcs/func_strings.c,
	  apps/app_alarmreceiver.c, main/devicestate.c,
	  cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c,
	  main/indications.c, main/config.c, main/loader.c, main/cli.c,
	  res/res_smdi.c, channels/chan_skinny.c, main/strcompat.c,
	  main/http.c, apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c,
	  res/res_speech.c, apps/app_milliwatt.c, main/sched.c,
	  apps/app_dial.c, main/pbx.c, channels/chan_agent.c,
	  channels/iax2-provision.c, channels/iax2-parser.c,
	  main/chanvars.c, res/res_monitor.c, main/netsock.c,
	  apps/app_speech_utils.c, channels/chan_misdn.c,
	  funcs/func_curl.c, main/fixedjitterbuf.c, apps/app_macro.c,
	  res/res_indications.c, apps/app_mixmonitor.c, main/asterisk.c,
	  res/res_odbc.c, main/dlfcn.c, apps/app_voicemail.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, main/utils.c,
	  res/res_musiconhold.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c,
	  apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c,
	  res/res_config_sqlite.c, main/enum.c, channels/misdn_config.c,
	  main/io.c, main/channel.c, main/cdr.c, funcs/func_enum.c,
	  main/dial.c, main/manager.c, apps/app_osplookup.c, main/tdd.c,
	  funcs/func_odbc.c, cdr/cdr_sqlite.c, res/res_agi.c,
	  apps/app_minivm.c, main/app.c, apps/app_directory.c,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
	  codecs/codec_lpc10.c, res/res_config_pgsql.c,
	  channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c,
	  apps/app_festival.c, main/translate.c, main/jitterbuf.c,
	  main/acl.c, apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
	  cdr/cdr_tds.c, main/file.c, main/callerid.c, main/event.c,
	  funcs/func_devstate.c, funcs/func_callerid.c, main/dsp.c: Issue
	  9869 - replace malloc and memset with ast_calloc, and other
	  coding guidelines changes

2007-06-06 21:16 +0000 [r67813-67863]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 67862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) |
	  4 lines Fix a crash when doing call pickups with SIP phones. The
	  code unlocked the channel when it should not have. (issue #9652,
	  reported by corruptor, fixed by me) ........

	* res/res_features.c, include/asterisk/features.h: Constify the
	  return values of ast_parking_ext() and ast_pickup_ext()

	* main/manager.c: Minor formatting change to test closing mantis
	  issues through commit tags (closes issue #9828)

	* main/manager.c: Minor formatting change to test closing mantis
	  issues through commit tags (closes issue #9828)

	* apps/app_voicemail.c: Please forgive this flood of tiny changes
	  ... this will be cool when it works how we want it to :) (testing
	  mantis+svn) (issue #9828)

2007-06-06 19:46 +0000 [r67808]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 67804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun
	  2007) | 10 lines Fix for Issue 9810. There was a segfault under a
	  specific set of circumstances: 1. VoiceMailMain was configured in
	  the dialplan with an extension as its argument 2. A message was
	  left for this mailbox 3. Tried to call VoiceMailMain but hung up
	  before entering password. This was fixed by checking that a
	  pointer was non-null prior to trying to dereference it. (Issue
	  9810, reported by xmarksthespot, patched by Corydon76 with
	  modifications by me). ........

2007-06-06 19:44 +0000 [r67787-67807]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: minor formatting change ... testing
	  mantis/svn (issue #9828)

	* apps/app_voicemail.c: Don't try to check the result of alloca ...
	  ... testing mantis/svn stuff ... (issue #9828)

	* main/dsp.c: Yet another minor change to test mantis/svn (issue
	  #9828)

	* main/dsp.c: minor formatting change ... testing mantis/svn (issue
	  #9828)

	* main/dsp.c: minor formatting change ... testing mantis/svn (issue
	  #9828)

	* main/app.c: Formatting change ... testing (issue #9828)

2007-06-06 19:02 +0000 [r67784]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing a crash wherein Asterisk would
	  segfault when attempting to leave a voicemail when IMAP storage
	  was enabled. Though no bug was reported to the bugtracker, there
	  was mention of this made as a note on bug 9810 by edhorton.

2007-06-06 19:00 +0000 [r67697-67782]  Russell Bryant <russell@digium.com>

	* main/app.c: Make another formatting change ... testing mantis/svn
	  stuff (issue #9828)

	* main/app.c: Another minor formatting change ... testing
	  mantis/svn (issue #9828)

	* main/app.c: Minor formatting change ... testing mantis/svn (issue
	  #9828)

	* channels/chan_iax2.c: Make another small tweak ... mantis/svn
	  testing (issue #9828)

	* res/res_features.c: Another tiny formatting change for testing
	  ... (issue #9828)

	* main/app.c: More random formatting changes to test Mantis/SVN
	  integration (issue #9828)

	* main/app.c: Make a completely arbitrary formatting change to test
	  out some Mantis/SVN integration stuff. (issue #9828)

	* main/channel.c, /, include/asterisk/linkedlists.h: Merged
	  revisions 67716 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67716 | russell | 2007-06-06 11:55:59 -0500
	  (Wed, 06 Jun 2007) | 13 lines Merged revisions 67715 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06
	  Jun 2007) | 5 lines We have some bug reports showing crashes due
	  to a double free of a channel. Add a sanity check to
	  ast_channel_free() to make sure we don't go on trying to free a
	  channel that wasn't found in the channel list. (issue #8850, and
	  others...) ........ ................

	* res/res_features.c: Change "show parkedcalls" to "parkedcalls
	  show" and mark the previous command as deprecated. Also, convert
	  the CLI command to the new style. (issue #9861, patch from eliel)

2007-06-06 13:32 +0000 [r67595-67651]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 67650 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed,
	  06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
	  lines Reinvite the RTP back to the Asterisk machine when the
	  timeout happens. (issue #9888 reported by gasparz) ........
	  ................

	* /, main/translate.c: Merged revisions 67631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67631 | file | 2007-06-06 09:18:39 -0400 (Wed, 06 Jun 2007) | 2
	  lines Fix plc_samples warning when registering a translator.
	  (issue #9897 reported by xylome) ........

	* /, apps/app_directed_pickup.c: Merged revisions 67626 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67626 | file | 2007-06-06 09:16:34 -0400 (Wed, 06 Jun 2007) | 2
	  lines Include macroexten while searching for a channel to pick up
	  in case they are in a macro. (issue #9491 reported by jamesb63)
	  ........

	* /, res/res_agi.c: Merged revisions 67597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67597 | file | 2007-06-06 08:34:06 -0400 (Wed, 06 Jun 2007) | 2
	  lines Make the new "agi debug off" CLI command work. (issue #9890
	  reported by eliel) ........

	* channels/chan_zap.c: When SS7 is enabled add w/SS7 to the end.
	  (issue #9893 reported by Mike Anikienko)

	* /, main/devicestate.c: Merged revisions 67594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67594 | file | 2007-06-06 08:20:27 -0400 (Wed,
	  06 Jun 2007) | 10 lines Merged revisions 67593 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
	  lines Revert channel name splitting fix for Zap. The moral of the
	  story is don't use - in your user/peer names. (issue #9668
	  reported by stevedavies) ........ ................

2007-06-05 23:02 +0000 [r67560]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 67558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) |
	  5 lines Fix some crashes related to the use of the "meetme" CLI
	  command. The code for this command was not locking the conference
	  list at all. (issue #9351, reported by and patch submitted by
	  Junk-Y, committed patch is different and by me) ........

2007-06-05 22:59 +0000 [r67557]  Mark Michelson <mmichelson@digium.com>

	* main/cli.c: Found a bug where when "core set debug #" is used,
	  the verbosity is read as the old value instead of the old debug
	  value, leading to an erroneous status message after setting. This
	  was purely a cosmetic issue and had no other underlying problems.

2007-06-05 22:04 +0000 [r67529]  Steve Murphy <murf@digium.com>

	* utils/Makefile, /, pbx/ael/ael.tab.c, pbx/ael/ael.y,
	  pbx/pbx_ael.c, pbx/Makefile: Merged revisions 67526 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r67526 | murf | 2007-06-05 15:30:18 -0600 (Tue, 05 Jun
	  2007) | 1 line this fixes bug 9883, wherein macros were not
	  allowing the includes construct. fixed and tested, looks OK. Now
	  includes can serve as an adjunct to catch. ........

2007-06-05 20:55 +0000 [r67493]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 67492 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) |
	  16 lines This bug has been hanging over my head ever since I
	  wrote this SLA code. Every time I tried to go debug it by adding
	  some debug output, the behavior would change. It turns out I
	  wasn't crazy. I had the following piece of code: if (remove)
	  AST_LIST_REMOVE_CURRENT(...); Well, AST_LIST_REMOVE_CURRENT was
	  not wrapped in braces, so my conditional statement didn't do much
	  good at all. It always ran at least all of the macro minus the
	  first statement, so I was seeing list entries magically disappear
	  when they weren't supposed to. After many hours of debugging, I
	  have come to this extremely irritating fix. :) (issues #9581,
	  #9497) ........

2007-06-05 20:16 +0000 [r67486]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Merged revisions 67424 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun
	  2007) | 5 lines Fix for bug number 9786, wherein voicemails saved
	  to IMAP storage using extensions other than gsm were unable to be
	  played over the phone. (Issue 9786, reporter: xmarksthespot,
	  Patched by xmarksthe spot with revisions by me, reviewed by
	  Russell Bryant). ........

2007-06-05 19:50 +0000 [r67458]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 67457 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67457 | russell | 2007-06-05 14:48:02 -0500 (Tue, 05 Jun 2007) |
	  2 lines Suppress a bunch of debug output unless option_debug is
	  on ........

2007-06-05 18:23 +0000 [r67423]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 67420 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67420 | murf | 2007-06-05 12:17:28 -0600 (Tue, 05 Jun 2007) | 1
	  line Added code to automatically add a default case to switches
	  that don't have one. In some cases, rather than fall thru, it
	  results in a goto with -1 result, which terminates the extension;
	  a sort of dialplan seqfault, sort of. This was required to fix
	  bug reported in 9881 ........

2007-06-05 18:19 +0000 [r67398-67422]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 67421 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r67421 | qwell | 2007-06-05 13:18:24 -0500 (Tue, 05 Jun
	  2007) | 4 lines Correctly update date/time on devices throughout
	  the life of the device, instead of just at registration. Issue
	  9152, yet another patch by DEA. ........

	* main/manager.c: Make sure we default allowmultiplelogin to
	  on/yes, per the default stated in the config. Issue 9885, patch
	  by eliel.

2007-06-05 17:24 +0000 [r67397]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/misdn/isdn_msg_parser.c: changed #if DEBUG to #ifdef
	  DEBUG to fix make failure when configured with --enable-dev-mode

2007-06-05 17:11 +0000 [r67361-67380]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Improve the way that the zaptel channel name
	  is created by using the Asterisk strings API and by only
	  allocating space on the stack

	* /: Merged revisions 67360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67360 | russell | 2007-06-05 11:56:36 -0500 (Tue, 05 Jun 2007) |
	  5 lines Fix a problem that showed itself by causing Zap channel
	  names to be completely bogus on my machine.
	  ast_safe_string_alloc() was broken. It called vsnprintf() on a
	  va_args list twice without re-initializing it. After the first
	  usage, va_end() and va_start() must be called again. ........

2007-06-05 16:21 +0000 [r67345-67350]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/chan_misdn_config.h: Merged revisions 67334 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67334 | crichter | 2007-06-05 18:14:07 +0200
	  (Di, 05 Jun 2007) | 9 lines Merged revisions 67307 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05
	  Jun 2007) | 1 line briding is a bool, fixed copy and paste issue.
	  ........ ................

	* channels/chan_misdn.c, /: Merged revisions 67329 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67329 | crichter | 2007-06-05 18:11:57 +0200
	  (Di, 05 Jun 2007) | 9 lines Merged revisions 67306 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
	  Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
	  cb_events function a lot. Commented the different possibilities a
	  bit and made functions of shared code. When the dialed extension
	  does not exist in the extensions.conf we'll jump into the 'i'
	  extension if this does exist, else we disconnect the call with
	  the cause:1 = No Route to Destination. ........ ................

2007-06-05 15:54 +0000 [r67310]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/module.h, main/asterisk.c, main/loader.c:
	  Merged revisions 67308 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) |
	  5 lines When shutting down "gracefully", go through and run the
	  unload() callbacks for all of the modules. "stop now" is
	  considered a non-graceful shutdown and will not go through this
	  process. (issue #9804, reported by chrisost, patch by me)
	  ........

2007-06-05 15:24 +0000 [r67305]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 67304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67304 | file | 2007-06-05 12:22:30 -0300 (Tue, 05 Jun 2007) | 2
	  lines Only muck with the thread structure if an idle one was
	  found/created. ........

2007-06-05 14:59 +0000 [r67272-67273]  Russell Bryant <russell@digium.com>

	* doc/CODING-GUIDELINES: add a note about inline comments

	* channels/chan_iax2.c: Doxygenify the comments for new members of
	  the iax2_thread struct

2007-06-05 14:45 +0000 [r67271]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 67270 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007)
	  | 3 lines ensure that a burst of full frames (AST_FRAME_DTMF
	  being the prime example) will not be processed out of order...
	  this is a brute force fix, but seems to be the safest fix for now
	  (thanks to the Digium PQ department for finding this bug)
	  ........

2007-06-05 11:48 +0000 [r67240]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
	  channels/misdn_config.c: Merged revisions 67210 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67210 | crichter | 2007-06-05 12:25:32 +0200
	  (Di, 05 Jun 2007) | 9 lines Merged revisions 67209 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05
	  Jun 2007) | 1 line added possibility to deactivate bridging per
	  port ........ ................

2007-06-04 23:45 +0000 [r67164]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_math.c: Merged revisions 67162 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67162 | tilghman | 2007-06-04 18:43:01 -0500
	  (Mon, 04 Jun 2007) | 10 lines Merged revisions 67161 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04
	  Jun 2007) | 2 lines According to MATH, 0+1181000386 = 1181000448.
	  Oops. ........ ................

2007-06-04 23:32 +0000 [r67160]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 67158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) |
	  5 lines Fix up a bunch of places where the iax2 pvt structure can
	  disappear and the code did not account for it and crashes.
	  (issues #9642, #9569, #9666, probably others ... based on the
	  work by stevedavies and mihai, with additional changes from me)
	  ........

2007-06-04 23:29 +0000 [r67122-67157]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 67156 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun
	  2007) | 6 lines Fix for skinny keepalives. If there is no traffic
	  from the phone for (keep_alive * 1100) ms (arbitrarily adding 10%
	  for network issues, etc), unregister the device. Issue 8394,
	  patch by DEA. ........

	* /, channels/chan_mgcp.c: Merged revisions 67121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4
	  lines Fixes for dtmf/dialing with mgcp (similar to the recent fix
	  for chan_skinny) Issue 9855, patch by DEA. ........

2007-06-04 22:29 +0000 [r67120]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 67119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) |
	  6 lines Add comments for two functions that get called with the
	  appropriate call locked, but perform operations that could result
	  in the pvt structure getting destroyed before returning again,
	  causing numerous seg faults all over the module. (inspired by
	  issues #9642, #9569, and #9666, and the work done by stevedavies
	  and mihai) ........

2007-06-04 22:15 +0000 [r67095]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 67073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67073 | murf | 2007-06-04 15:59:34 -0600 (Mon, 04 Jun 2007) | 1
	  line This typo has been here since 1.4 forked. It has been the
	  source of heartburn to many a dialplan/CDR programmer. ........

2007-06-04 21:48 +0000 [r67070-67072]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 67071 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) |
	  2 lines Add a missing \n. (pointed out by jcmoore on IRC)
	  ........

	* channels/chan_iax2.c: Remove a leftover unlock and lock of the
	  iax2 pvt struct lock that was left over from my attempt at
	  putting pvt structs in a hash table. It can cause seg faults, and
	  has no reason to stay. (issue #9642, pointed out by stevedavies)

2007-06-04 19:32 +0000 [r67063-67069]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 67068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2
	  lines Better handle SIP devices that say they have SDP content...
	  but really don't. (issue #9398 reported by mthomasslo) ........

	* apps/app_dial.c, /: Merged revisions 67066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2
	  lines Initialize cidname variable to nothing since it may be used
	  without having been touched. (issue #9661 reported by dimas)
	  ........

	* /, res/res_features.c: Merged revisions 67064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67064 | file | 2007-06-04 13:41:59 -0400 (Mon, 04 Jun 2007) | 2
	  lines Returning a value that indicates the parking of a call was
	  a success when it really wasn't (because the parking slot
	  selected was in use) is the wrong thing to do. (issue #9723
	  reported by mdu113) ........

	* apps/app_directed_pickup.c: Minor clean up. Constify a few
	  variables and use ast_strlen_zero in a few places.

2007-06-04 17:12 +0000 [r67062]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk, /,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.mandrake.zaptel,
	  contrib/init.d/rc.slackware.asterisk: Merged revisions 67061 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r67061 | tilghman | 2007-06-04 12:11:43 -0500
	  (Mon, 04 Jun 2007) | 10 lines Merged revisions 67060 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04
	  Jun 2007) | 2 lines Add revision Id tags (by request of tzafrir)
	  ........ ................

2007-06-04 16:03 +0000 [r67024-67029]  Russell Bryant <russell@digium.com>

	* /, configure, configure.ac: Merged revisions 67026 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r67026 | russell | 2007-06-04 11:02:31 -0500 (Mon, 04
	  Jun 2007) | 6 lines Change the configure script to build a test
	  program against libcurl to make sure the results from curl-config
	  can be used to compile successfully. This is intended to help
	  prevent a situation where you are cross compiling, and the
	  configure script finds the curl library installed on the host.
	  (issue #9865, reported and patched by zandbelt) ........

	* main/ast_expr2f.c, pbx/ael/ael_lex.c, main/app.c: Change javadoc
	  style code documentation to the same format we use elsewhere.
	  (issue #9864, patch from snuffy)

2007-06-04 15:53 +0000 [r67023]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_jabber.c: Merged revisions 67021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67021 | tilghman | 2007-06-04 10:50:16 -0500 (Mon, 04 Jun 2007)
	  | 2 lines Issue 9739 - Malformed jid causes a crash ........

2007-06-04 15:50 +0000 [r67016-67022]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 67020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) |
	  7 lines Resolve a deadlock in chan_iax2. When handling an
	  implicit ACK to a frame that was marked as the final transmission
	  for a call, don't call iax2_destroy() for that call while the
	  global frame queue is still locked. There is a very nice
	  explanation of the deadlock in the report. (issue #9663, thorough
	  report and patch from stevedavies, additional positive test
	  reports from mihai and joff_oconnell) ........

	* include/asterisk/stringfields.h: Fix some compiler warnings in
	  C++ modules. (issue #9866, reported by osk, patch by Corydon76)

	* channels/chan_sip.c, main/netsock.c: Fix a couple of places where
	  "tos" was used instead of "cos". (issue #9540, patch by IgorG)

2007-06-04 11:48 +0000 [r66998]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Add support for autocompleting start/stop
	  options of the mixmonitor CLI command. (issue #9862 reported by
	  eliel)

2007-06-03 06:10 +0000 [r66981]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_gtalk.c, channels/chan_nbs.c, channels/chan_mgcp.c:
	  ast_calloc janitor (Inspired by issue 9860)

2007-06-01 23:39 +0000 [r66957-66959]  Russell Bryant <russell@digium.com>

	* main/pbx.c: remove a bogus comment that came from copy/paste

	* include/asterisk/devicestate.h, include/asterisk.h, main/pbx.c,
	  include/asterisk/event_defs.h, main/devicestate.c,
	  include/asterisk/pbx.h, apps/app_queue.c, main/asterisk.c: Merge
	  major changes to the way device state is passed around Asterisk.
	  The two places that cared about device states were app_queue and
	  the hint code in pbx.c. The changes include converting it to use
	  the Asterisk event system, as well as other efficiency
	  improvements. * app_queue: This module used to register a
	  callback into devicestate.c to monitor device state changes. Now,
	  it is just a subscriber to Asterisk events with the type, device
	  state. * pbx.c hints: Previously, the device state processing
	  thread in devicestate.c would call ast_hint_state_changed() each
	  time the state of a device changed. Then, that code would go
	  looking for all the hints that monitor that device, and call
	  their callbacks. All of this blocked the device state processing
	  thread. Now, the hint code is a subscriber of Asterisk events
	  with the type, device state. Furthermore, when this code receives
	  a device state change event, it queues it up to be processed by
	  another thread so that it doesn't block one of the event
	  processing threads.

	* channels/chan_iax2.c: Remove 80 bytes in the iax2_registry struct
	  that weren't being used

2007-06-01 21:49 +0000 [r66920]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_odbc.c: Merged revisions 66919 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66919 | tilghman | 2007-06-01 16:45:44 -0500 (Fri, 01 Jun 2007)
	  | 2 lines On some drivers, deallocating the statement handle
	  isn't enough. We also have to clear the cursor (nice, Oracle)
	  ........

2007-06-01 21:33 +0000 [r66910-66918]  Mark Michelson <mmichelson@digium.com>

	* /: Merged revisions 66916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  ........

	* /, apps/app_voicemail.c: Merged revisions 66897 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun
	  2007) | 3 lines Submitting a fix for voicemail with IMAP storage.
	  Attachments with format specified as gsm were duplicated (i.e.
	  two attachments) were left. Thank you very much to xmarksthespot
	  for submitting the patch that fixed this. (Issues 9787 and 8873,
	  Reported by xmarksthespot and jerjer, patched by xmarksthespot)
	  ........

2007-06-01 19:42 +0000 [r66880-66882]  Russell Bryant <russell@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 66881 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01
	  Jun 2007) | 6 lines Changes to the way DTMF is handled in the
	  core broke dialing in chan_skinny. This patch makes chan_skinny
	  usable again. I did not end up testing this, but there are
	  multiple positive test reports listed in the bug report. (issue
	  #9596, reported by pj, testing by pj and mvanbaak, and the fix
	  was written by DEA) ........

	* /, apps/app_page.c: Merged revisions 66879 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) |
	  2 lines List app_meetme as a module that app_page depends on.
	  ........

2007-06-01 18:36 +0000 [r66878]  Jason Parker <jparker@digium.com>

	* res/res_config_sqlite.c: Documentation fixes for
	  res_config_sqlite. Issue 9854, patch by tzafrir.

2007-06-01 13:48 +0000 [r66856]  Russell Bryant <russell@digium.com>

	* configs/sip.conf.sample: Add some more information about the SIP
	  Disclaimer header.

2007-05-31 23:04 +0000 [r66822]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/asterisk.8: Merged revisions 66821 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66821 | tilghman | 2007-05-31 18:03:28 -0500 (Thu, 31 May 2007)
	  | 2 lines Issue 9850 - update preferred command line syntax
	  ........

2007-05-31 21:23 +0000 [r66772-66818]  Russell Bryant <russell@digium.com>

	* configs/sip.conf.sample: fix a typo.

	* channels/chan_sip.c, configs/sip.conf.sample: To satisfy some
	  legal concerns, add an option for chan_sip to include a
	  disclaimer along with SIP messages in the header, X-Disclaimer.
	  This is off by default. Also, the text of the disclaimer can be
	  customized in sip.conf.

	* include/asterisk/app.h, /, include/asterisk/speech.h,
	  res/res_speech.c: Merged revisions 66775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66775 | russell | 2007-05-31 13:41:58 -0500 (Thu, 31 May 2007) |
	  3 lines Change a couple of header files to not use "new", which
	  is a reserved keyword in C++. (issue #9830, reported by osk)
	  ........

	* res/res_features.c, CHANGES, configs/features.conf.sample: Add
	  support for configuring named groups of custom call features in
	  features.conf. This allows you to create a feature one time, and
	  then map it into groups for various different key mappings for
	  the same feature, as well as easy access control to groups of
	  features. (patch from bbryant)

	* res/res_features.c, configs/features.conf.sample: Revert changes
	  that snuck in with revision 66724.

	* apps/app_minivm.c: - Don't check if the list is empty needlessly
	  - Don't free structures before calling load_config(), because
	  load_config() already does it - Use the existing functions for
	  freeing the minivm structures instead of replicating the code
	  (issue #9846, patch from eliel)

2007-05-31 17:16 +0000 [r66771]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 66770 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r66770 | tilghman | 2007-05-31 12:15:09 -0500
	  (Thu, 31 May 2007) | 10 lines Merged revisions 66744 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31
	  May 2007) | 2 lines Issue 9818 - Fix for issue 8329 breaks
	  pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime,
	  but only because we lack core API to do it. ........
	  ................

2007-05-31 16:18 +0000 [r66769]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 66768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r66768 | file | 2007-05-31 12:14:48 -0400 (Thu,
	  31 May 2007) | 10 lines Merged revisions 66764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
	  lines It is now possible for this path of execution to have the
	  frame pointer be NULL, therefore we need to check for it before
	  trying to access it. (issue #9836 reported by barthpbx) ........
	  ................

2007-05-31 15:05 +0000 [r66734]  Tilghman Lesher <tlesher@digium.com>

	* configs/func_odbc.conf.sample, funcs/func_odbc.c: Issue 9799 -
	  Multirow results for func_odbc

2007-05-31 14:52 +0000 [r66724]  Russell Bryant <russell@digium.com>

	* res/res_features.c, apps/app_minivm.c,
	  configs/features.conf.sample: Fix a crash on reload by using
	  calloc() instead of malloc() to ensure that data is properly
	  initialized. (issue #9765, reported by MatsK, patch from eliel)

2007-05-31 10:26 +0000 [r66705]  Olle Johansson <oej@edvina.net>

	* include/asterisk/app.h, apps/app_osplookup.c,
	  include/asterisk/event.h, apps/app_meetme.c, channels/chan_sip.c,
	  include/asterisk/event_defs.h, apps/app_skel.c,
	  apps/app_minivm.c, res/res_jabber.c: Issue #9842 - Doxygen
	  updates by snuffy. Thanks! (Committed from Media Plaza in
	  Utrecht, Netherlands - Open Source VoIP conference)

2007-05-30 23:44 +0000 [r66672]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 66671 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66671 | mmichelson | 2007-05-30 18:26:39 -0500 (Wed, 30 May
	  2007) | 2 lines Fixed seg-faults when recording greetings in
	  voicemail with IMAP enabled. (Issue No. 9734, reported by
	  xmarksthespot, patched by me) ........

2007-05-30 17:23 +0000 [r66603-66638]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, channels/chan_features.c: This concludes my
	  tweaking of things.

2007-05-30 05:17 +0000 [r66539-66585]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_channelredirect.c, channels/chan_vpb.cc,
	  res/res_config_odbc.c, funcs/func_shell.c, funcs/func_cdr.c,
	  apps/app_zapras.c, res/res_indications.c, apps/app_transfer.c,
	  apps/app_stack.c, funcs/func_devstate.c, res/res_config_sqlite.c,
	  res/res_odbc.c: Issue 9477 - Improve menuselect labels

	* /, funcs/func_strings.c: Merged revisions 66538 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r66538 | tilghman | 2007-05-29 16:56:07 -0500
	  (Tue, 29 May 2007) | 10 lines Merged revisions 66537 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29
	  May 2007) | 2 lines If the value of a variable passed to FIELDQTY
	  is blank, then FIELDQTY should return 0, not 1. ........
	  ................

	* funcs/func_enum.c: Shorten description to a much more reasonable
	  length

2007-05-29 19:53 +0000 [r66502-66505]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: oops. Thanks Terry.

	* /, channels/chan_sip.c: Merged revisions 66503 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2
	  lines Properly handle 408 request timeout - according to the RFC,
	  the dialog dies if a request in a dialog gets this response.
	  ........

	* /, channels/chan_sip.c: Merged revisions 66474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2
	  lines Don't issue hangup on hangup on hangup on hangup (for
	  jcmoore) ........

2007-05-29 19:00 +0000 [r66471]  Doug Bailey <dbailey@digium.com>

	* main/dsp.c: Changed the dtmf detection to integer based goertzel
	  algorithm.

2007-05-29 16:46 +0000 [r66438]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 66437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2
	  lines Handle cases where a frame may have no data. (issue #9519
	  reported by dmb) ........

2007-05-29 16:19 +0000 [r66432-66433]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 66414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2
	  lines Don't reset hangupcause if we already have one ........

	* /, channels/chan_sip.c: Merged revisions 66404 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2
	  lines Tracking down hanging channels, killing them one by one.
	  Issue #9235 and related ........

2007-05-29 15:44 +0000 [r66399]  Joshua Colp <jcolp@digium.com>

	* /, doc/datastores.txt: Merged revisions 66398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66398 | file | 2007-05-29 11:43:16 -0400 (Tue, 29 May 2007) | 2
	  lines Update datastores documentation. (issue #9801 reported by
	  mnicholson) ........

2007-05-29 10:02 +0000 [r66367]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 66363 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue,
	  29 May 2007) | 10 lines Merged revisions 66349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
	  lines Issue #9802 - Change inuse counter on CANCEL ........
	  ................

2007-05-28 23:28 +0000 [r66313-66315]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't try to unregister a peer using the sip
	  unregister CLI command if they are not registered. (issue #9811
	  reported by eliel)

	* channels/chan_sip.c: Due to the way stringfields work the value
	  of the url pointer will always be non-NULL so we have to use
	  ast_strlen_zero to make sure it is not empty. (issue #9821
	  reported by pj)

2007-05-28 18:50 +0000 [r66295]  Olle Johansson <oej@edvina.net>

	* apps/app_voicemail.c: - Don't re-invent existing headers (some
	  already existed in chan_sip) - Rename command so taht module name
	  comes first

2007-05-28 15:59 +0000 [r66278]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_iconv.c (added): Issue 7021 - Add ICONV function for
	  converting between character sets

2007-05-26 19:35 +0000 [r66225]  Joshua Colp <jcolp@digium.com>

	* apps/app_minivm.c: Unlock the minivmlock when no configuration is
	  found. (issue #9814 reported by eliel)

2007-05-26 06:07 +0000 [r66208]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Since this code now uses the API call for
	  creating a detached thread, there is no reason to keep a thread
	  attribute structure on the conference structure. (Pointed out by
	  Tony Mountifield on the asterisk-dev list)

2007-05-25 15:08 +0000 [r66175-66178]  Kevin P. Fleming <kpfleming@digium.com>

	* /: block change that is already here

	* channels/chan_jingle.c, configure, configure.ac: more minor fixes

2007-05-25 14:49 +0000 [r66161]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c: Merged revisions 66159 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r66159 | tilghman | 2007-05-25 09:41:27 -0500
	  (Fri, 25 May 2007) | 10 lines Merged revisions 66127 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25
	  May 2007) | 2 lines Issue 9791 - Fix pronunciation of seconds in
	  Dutch ........ ................

2007-05-25 14:37 +0000 [r66158]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_jingle.c, /, configure, configure.ac,
	  channels/chan_gtalk.c, makeopts.in, res/res_jabber.c: Merged
	  revisions 66157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007)
	  | 3 lines handle the GNUTLS library properly in the configure
	  script and build system don't build in OSP support unless we have
	  found and are allowed to use SSL support ........

2007-05-25 13:26 +0000 [r66109-66126]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Minor tweak... drop translation path if one
	  exists when we get an already signed linear frame in. Chances are
	  the stream has then switched to signed linear and we no longer
	  need the path.

	* /, main/slinfactory.c: Merged revisions 66074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66074 | file | 2007-05-24 18:16:58 -0400 (Thu, 24 May 2007) | 2
	  lines Fix slinfactory logic when dealing with frames coming in
	  that may already be in the signed linear format. ........

2007-05-24 22:25 +0000 [r66072-66077]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 66076 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) |
	  1 line if the string field init fails, clean up the stuff that
	  was allocated already ........

	* main/channel.c, /: Merged revisions 66070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) |
	  2 lines Check the result of ast_string_field_init() in
	  ast_channel_alloc() ........

2007-05-24 22:07 +0000 [r66071]  Kevin P. Fleming <kpfleming@digium.com>

	* main/aescrypt.c, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, include/asterisk/aes_internal.h
	  (added), configure.ac, main/aestab.c, include/asterisk/aes.h,
	  main/aeskey.c, pbx/pbx_dundi.c, channels/chan_iax2.c,
	  makeopts.in: use the OpenSSL AES implementation if it's available
	  (unless configured not to)

2007-05-24 20:55 +0000 [r66031]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac: Merged revisions 66029-66030 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r66029 | qwell | 2007-05-24 15:53:18 -0500 (Thu, 24 May 2007) | 2
	  lines Following moving strip to AC_PATH_TOOL, we need to do
	  something similar for ar. ........ r66030 | qwell | 2007-05-24
	  15:54:16 -0500 (Thu, 24 May 2007) | 2 lines Rebuild configure
	  script for previous ar fix. ........

2007-05-24 20:51 +0000 [r66028]  Joshua Colp <jcolp@digium.com>

	* CHANGES, apps/app_voicemail.c: Add ListAllVoicemailUsers manager
	  command. (issue #8112 reported by Tony Zhao)

2007-05-24 20:44 +0000 [r65982-66027]  Russell Bryant <russell@digium.com>

	* /, configure, configure.ac: Merged revisions 66026 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r66026 | russell | 2007-05-24 15:42:53 -0500 (Thu, 24
	  May 2007) | 3 lines Checking for the strip application needs to
	  be done with AC_PATH_TOOL instead of AC_PATH_PROG to properly
	  handle cross compilation environments. ........

	* doc/CODING-GUIDELINES: add a note about using the intenal API for
	  creating detached threads

	* Makefile, /: Merged revisions 65978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65978 | russell | 2007-05-24 14:05:08 -0500 (Thu, 24 May 2007) |
	  3 lines Clear CFLAGS before running make for menuselect. (issue
	  #9784, reported by ovi, patch by me) ........

2007-05-24 19:05 +0000 [r65979]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 65965-65967 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007)
	  | 2 lines don't use uninitialized variables ........ r65966 |
	  kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2
	  lines don't reference GnuTLS headers and functions unless the
	  configure script found it ........ r65967 | kpfleming |
	  2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use
	  #ifdef instead of #if ........

2007-05-24 18:30 +0000 [r65964-65968]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/utils.h, channels/chan_zap.c,
	  channels/chan_sip.c, apps/app_meetme.c, main/utils.c,
	  channels/chan_iax2.c, main/cdr.c, main/manager.c,
	  pbx/pbx_spool.c, channels/chan_skinny.c, main/http.c,
	  channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_rpt.c,
	  apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: Add
	  a new API call for creating detached threads. Then, go replace
	  all of the places in the code where the same block of code for
	  creating detached threads was replicated. (patch from bbryant)

	* main/rtp.c: Make this build on *my* machine again, and hopefully
	  not break others.

2007-05-24 15:35 +0000 [r65906]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, funcs/func_math.c: Merged revisions 65866 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65866 | dhubbard | 2007-05-24 10:08:56 -0500 (Thu, 24 May 2007)
	  | 1 line merged qwell's func_math patch for issue 9507 ........

2007-05-24 15:30 +0000 [r65905]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, /: Merged revisions 65902 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65902 | file | 2007-05-24 11:27:23 -0400 (Thu, 24 May 2007) | 2
	  lines Add the ability to blacklist certain commands from being
	  executed using the Command AMI action. (issue #9240 reported by
	  junky) ........

2007-05-24 15:29 +0000 [r65904]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_gtalk.c: Merged revisions 65901 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May
	  2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump
	  since the GnuTLS interface did not support multithreading
	  correctly. ........

2007-05-24 15:28 +0000 [r65903]  Jason Parker <jparker@digium.com>

	* /, codecs/codec_speex.c, main/translate.c, codecs/codec_ilbc.c,
	  .cleancount, include/asterisk/translate.h: Merged revisions 65877
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4
	  lines Fix handling of zero-length frames when a codec is capable
	  of native PLC. Issue 9183, patch by Mihai. ........

2007-05-24 15:23 +0000 [r65894-65898]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_gtalk.c: Merged revisions 65892 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May
	  2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by
	  phsultan. Thanks! ........

	* /, channels/chan_gtalk.c: Merged revisions 65857 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May
	  2007) | 2 lines Issue 7686, fix by phsultan, NAT issues when
	  calling from gtalk to SIP over nat. ........

2007-05-24 15:10 +0000 [r65869]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 65863 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2
	  lines I like it when the RTP stack compiles myself... ........

2007-05-24 15:04 +0000 [r65855]  Russell Bryant <russell@digium.com>

	* /, apps/app_festival.c: Merged revisions 65853 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) |
	  4 lines Ensure that frames are fully initialized. This will
	  probably fix getting weird timestamp log messages in logs when
	  using the Festival app. (issue #9781, patch by me) ........

2007-05-24 14:52 +0000 [r65844]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_gtalk.c: Merged revisions 65841 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May
	  2007) | 2 lines Issue #8536 - Caller ID not set in CDR for jingle
	  ........

2007-05-24 14:50 +0000 [r65843]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 65842 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) |
	  5 lines Fix the calculation of the RTT for RTCP. The previous
	  code would result in oscillating and incorrect data.
	  Additionally, the RTT would sometimes report negative values due
	  to incorrect calculations. (issue #9601, patch from davetroy)
	  ........

2007-05-24 14:43 +0000 [r65840]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 65839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65839 | file | 2007-05-24 10:42:12 -0400 (Thu,
	  24 May 2007) | 10 lines Merged revisions 65837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
	  lines Allow RFC2833 to be negotiated when an INVITE comes in
	  without SDP and is not matched to a user or peer. (issue #9546
	  reported by mcrawford) ........ ................

2007-05-24 14:41 +0000 [r65838]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, res/res_jabber.c: Issue #8409 and
	  accidentally a fix to chan_sip that wasn't supposed to be there
	  but is still ok... Sorry. Lack of Tea, really.

2007-05-24 11:38 +0000 [r65814]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Yes Virginia, there is a reason why we have
	  stringfields in the sip_pvt structure...

2007-05-24 09:51 +0000 [r65769]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65768 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65768 | crichter | 2007-05-24 11:37:32 +0200
	  (Do, 24 Mai 2007) | 9 lines Merged revisions 65767 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
	  Mai 2007) | 1 line we should only activate the generator in
	  chan_misdn, when asterisk hask not yet taken the call
	  (WAITING4DIGS state). Alerting audio will be generated fomr
	  asterisk for example. ........ ................

2007-05-24 03:28 +0000 [r65749]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: - Remove debug variable that was only used
	  in one place - convert string handling to the ast_str API -
	  Convert strdup() to ast_strdup() and check the result - Minor
	  formatting changes

2007-05-24 03:27 +0000 [r65748]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Oops, should have released this when we
	  were done with it.

2007-05-24 02:23 +0000 [r65731]  Mark Spencer <markster@digium.com>

	* channels/chan_sip.c: Add SendURL support

2007-05-23 21:01 +0000 [r65678-65688]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 65685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007)
	  | 2 lines start the delayed PBX when receive voice or video full
	  frames as well, and comment this delayed-PBX activity ........

	* /, channels/chan_sip.c: Merged revisions 65683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65683 | kpfleming | 2007-05-23 16:51:56 -0400
	  (Wed, 23 May 2007) | 10 lines Merged revisions 65682 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23
	  May 2007) | 2 lines ensure that variables are set on a newly
	  created channel before we start a PBX on it ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 65679-65680 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007)
	  | 2 lines don't start a PBX on a new incoming IAX2 channel until
	  we have some sort of response to our ACCEPT (ACK or anything
	  else) ........ r65680 | kpfleming | 2007-05-23 16:35:50 -0400
	  (Wed, 23 May 2007) | 2 lines clear the 'delay PBX' flag when we
	  are ready to start the PBX ........

	* /, channels/chan_iax2.c: Merged revisions 65677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65677 | kpfleming | 2007-05-23 16:07:59 -0400
	  (Wed, 23 May 2007) | 10 lines Merged revisions 65676 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23
	  May 2007) | 2 lines if we are going to set variables on a newly
	  created channel, it should be done *before* we start the PBX on
	  it ........ ................

2007-05-23 17:17 +0000 [r65659]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Don't check for MWI event subscribers
	  before creating the MWI event in voicemail. MWI events get
	  cached, so go ahead and always generate them so the cache gets
	  populated.

2007-05-23 15:37 +0000 [r65640]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we get the cause code in the REL

2007-05-23 13:10 +0000 [r65591]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 65589 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65589 | russell | 2007-05-23 08:07:13 -0500
	  (Wed, 23 May 2007) | 11 lines Merged revisions 65588 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23
	  May 2007) | 3 lines Revert revision 62417 as someone reported
	  problems with it to Mark. This was related to issue #9588.
	  ........ ................

2007-05-23 13:07 +0000 [r65590]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Fix compiling of res_musiconhold under dev
	  mode.

2007-05-23 02:55 +0000 [r65573]  Russell Bryant <russell@digium.com>

	* main/devicestate.c: Fix a couple minor spelling mistakes.

2007-05-22 20:26 +0000 [r65542]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/make_version: Merged revisions 65541 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r65541 | kpfleming | 2007-05-22 16:25:41 -0400 (Tue, 22
	  May 2007) | 2 lines when building a version string for a
	  developer branch, include the base branch in the version string
	  ........

2007-05-22 18:52 +0000 [r65502-65505]  Russell Bryant <russell@digium.com>

	* main/channel.c, configs/musiconhold.conf.sample,
	  include/asterisk/channel.h, res/res_musiconhold.c, CHANGES: Add a
	  new feature for Music on Hold. If you set the "digit" option for
	  a class in musiconhold.conf, a caller on hold may press this
	  digit to switch to listening to that music class. This involved
	  adding a new callback for generators, which allow generators to
	  get notified of DTMF from the channel they are running on. Then,
	  a callback was implemented for the music on hold generators.
	  (patch from bbryant)

	* channels/chan_zap.c, /, apps/app_voicemail.c: Merged revisions
	  65501 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) |
	  3 lines List res_smdi as a dependency for app_voicemail and
	  chan_zap (Thanks to mnicholson for pointing it out) ........

2007-05-22 15:25 +0000 [r65455]  BJ Weschke <bweschke@btwtech.com>

	* /, apps/app_followme.c: Merged revisions 65408 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65408 | bweschke | 2007-05-22 10:02:56 -0400 (Tue, 22 May 2007)
	  | 3 lines Fix a problem with flag recognition. ........

2007-05-22 15:08 +0000 [r65451-65454]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: Use ast_strlen_zero where possible. (issue
	  #9774 reported by eliel)

	* main/cdr.c: Make my compiler happy! Yay!

2007-05-22 12:58 +0000 [r65376]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Don't overwrite a pointer to the first
	  channel... that is bad. (issue #9770 reported by tfbu)

2007-05-22 12:52 +0000 [r65375]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix a couple of spots in the handling of device
	  states that could lead to a double free. (issue #9772, reported
	  by Mike Anikienko, fix by me)

2007-05-22 08:21 +0000 [r65343]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65342 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65342 | crichter | 2007-05-22 10:12:20 +0200
	  (Di, 22 Mai 2007) | 9 lines Merged revisions 65328 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
	  Mai 2007) | 1 line we stop the tones only when we're in the
	  pre-call phase, otherwise e.g. when in CONNECTED state we should
	  not stop tones when we receive an Information Message ........
	  ................

2007-05-22 02:41 +0000 [r65313]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Fix for 64-bit platform

2007-05-21 06:56 +0000 [r65298]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: I know we have talked about rewriting app_queue
	  for Asterisk 1.6, but once I saw this, I couldn't help myself
	  from changing it. Previously, for *every* device state change,
	  app_queue would spawn a thread to handle it. Now, the device
	  state callback just puts the state change in a queue and it gets
	  handled by a single state change processing thread.

2007-05-21 02:05 +0000 [r65283]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c: Comment a few more things, and remove an
	  unnecessary db connection check

2007-05-20 18:01 +0000 [r65233-65253]  Joshua Colp <jcolp@digium.com>

	* /, res/res_agi.c: Merged revisions 65250 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65250 | file | 2007-05-20 13:59:58 -0400 (Sun, 20 May 2007) | 2
	  lines res_agi needs to export two symbols (ast_agi_register and
	  ast_agi_unregister) for usage by others. (issue #9755 reported by
	  mnicholson) ........

	* res/res_crypto.c, res/res_musiconhold.c: Music on hold and crypto
	  no longer need their symbols globally exported. They register the
	  function pointers upon loading with their respective stubs.

	* main/adsistub.c, main/cryptostub.c: Clean up adsistub file a bit
	  (just spacing) and change over the crypto sub to use this
	  build_stub macro strategy.

	* main/Makefile, main/adsistub.c, res/res_adsi.c: Add the adsistub
	  file to the Asterisk makefile, fix a stub definition, and no
	  longer make the symbols from res_adsi global since they don't
	  need to be.

2007-05-18 22:35 +0000 [r65202-65203]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 65201 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1
	  line Ugh. The svnmerge didn't catch the shift from cdr.c to
	  main/cdr.c, and neither did I. This is the remainder of the 9717
	  patch, the fix for the run-away FAIL status for a call ........

	* apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
	  65200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri,
	  18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
	  line This update will fix the situation that occurs as described
	  by 9717, where when several targets are specified for a dial, if
	  any one them reports FAIL, the whole call gets FAIL, even though
	  others were ringing OK. I rearranged the priorities, so that a
	  new disposition, NULL, is at the lowest level, and the
	  disposition get init'd to NULL. Then, next up is FAIL, and next
	  up is BUSY, then NOANSWER, then ANSWERED. All the related set
	  routines will only do so if the disposition value to be set to is
	  greater than what's already there. This gives the intended
	  effect. So, if all the targets are busy, you'd get BUSY for the
	  call disposition. If all get BUSY, but one, and that one rings is
	  not answered, you get NOANSWER. If by some freak of nature, the
	  NULL value doesn't get overridden, then the disp2str routine will
	  report NOANSWER as before. ........ ................

2007-05-18 20:21 +0000 [r65169]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c (added),
	  configs/cdr_adaptive_odbc.conf.sample (added): Merge
	  cdr_adaptive_odbc from developer branch

2007-05-18 18:18 +0000 [r65077-65124]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Related to issue #9235 btw. Merged
	  revisions 65123 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri,
	  18 May 2007) | 10 lines Merged revisions 65122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
	  lines Not getting an ACK to a 200 OK in the initial invite is
	  critical to the call. ........ ................

	* /, channels/chan_sip.c: Merged revisions 65076 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri,
	  18 May 2007) | 13 lines Merged revisions 65075 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
	  lines Issue 9235 - part of the problem, maybe not all. Please
	  retry with this patch (and no other patch) if you have problems
	  with hanging SIP channels. Thank you. A special Thank You to
	  WeBRainstorm that gave me access to his system. ........
	  ................

2007-05-18 12:43 +0000 [r65006-65040]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
	  revisions 65039 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r65039 | crichter | 2007-05-18 14:40:46 +0200
	  (Fr, 18 Mai 2007) | 9 lines Merged revisions 65007 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18
	  Mai 2007) | 1 line fixed a warning regarding Keypad encoding.
	  encode the IE sending_complete at the right position. ........
	  ................

	* channels/chan_misdn.c, /: Merged revisions 64904 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64904 | crichter | 2007-05-18 10:58:51 +0200
	  (Fr, 18 Mai 2007) | 9 lines Merged revisions 64902 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
	  Mai 2007) | 1 line we *need* to send a PROCEEDING when
	  sending_complete is set, even if need_more_infos is requested.
	  ........ ................

2007-05-18 10:41 +0000 [r64973-64975]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 64974 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2
	  lines Issue 9487 - stop media flows at hangup of call ........

	* channels/chan_sip.c: Makeup, darling.

2007-05-18 10:03 +0000 [r64951-64963]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 64515 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64515 | crichter | 2007-05-16 10:44:51 +0200
	  (Mi, 16 Mai 2007) | 9 lines Merged revisions 64513 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
	  Mai 2007) | 1 line in the case immediate=yes, we directly jump
	  into the dialplan, where people can use PlayTones to indicate a
	  Dialtone, so we don't need to to that by ourself. also we should
	  not do a dialtone_indicate for incoming calls on a TE port in
	  overlapdialmode. ........ ................

	* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
	  channels/misdn/isdn_lib.c: Merged revisions 63534 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63534 | crichter | 2007-05-09 15:17:10 +0200
	  (Mi, 09 Mai 2007) | 17 lines Merged revisions 62945,63402,63519
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
	  1 line when we're in state WAITING4DIGS, we use the asterisk
	  tone-generator which prods us, so we can't just return -1 in
	  misdn_write in this case. Added a MISDN_KEYPAD channel variable,
	  and fixed the sending of keypad. this enables us to modify the
	  call forward parameters in the switch. ........ r63402 | crichter
	  | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
	  application misdn_check_l2l1 which tries to pull up the L1/L2 on
	  all ports that have the layers down in a group. It waits then for
	  a timeout. This helps for scenarios where multiple PMP BRIs are
	  grouped together, or where a provider has a faulty PTP
	  Implementation, that looses the L2 after a while. ........ r63519
	  | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
	  release_chan frees ch, so we should never touch ch after
	  release_chan, this may cause segfaults. ........ ................

	* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  Merged revisions 62912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62912 | crichter | 2007-05-03 16:36:32 +0200
	  (Do, 03 Mai 2007) | 17 lines Merged revisions 61357,61770,62885
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
	  1 line some fixes for PMP Hold/Retrieve, it should work now, when
	  briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
	  (Di, 24 Apr 2007) | 1 line added lock for sending messages to
	  avoid double sending. shuffled some empty_chans after the
	  cb_event calls, this avoids that a release_complete from a quite
	  different call releases a fresh created setup by accident.
	  ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
	  Mai 2007) | 1 line fixed the problem that misdn_write did not
	  return -1 when called with 0 samples in a frame this resultet in
	  a deadlock in some circumstances, when the call ended because of
	  a busy extension. added encoding of keypad. ........
	  ................

	* channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
	  revisions 59774 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59774 | crichter | 2007-04-03 09:20:27 +0200
	  (Di, 03 Apr 2007) | 17 lines Merged revisions 59623-59624,59639
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
	  1 line we can now make 30 channels on a PRI (before we forgot
	  chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
	  (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
	  r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
	  1 line added option which allows us to accept incoming SETUP
	  Messages without automatically sending Proceeding or Setup
	  Acknowledge, this is useful with some broken switches and if you
	  want to Release incoming calls without previously having
	  acknowledged them. The new option is
	  noautorespond_on_setup=yes|no default is no, so we don't break
	  the existing behaviour ........ ................

	* channels/chan_misdn.c, /: Merged revisions 59254 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59254 | crichter | 2007-03-27 17:00:10 +0200
	  (Di, 27 Mär 2007) | 9 lines Merged revisions 59252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
	  Mär 2007) | 1 line fixed #9355 ........ ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h, channels/misdn/isdn_lib.c,
	  channels/misdn_config.c: Merged revisions 59064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59064 | crichter | 2007-03-20 14:16:06 +0100
	  (Di, 20 Mär 2007) | 21 lines Merged revisions
	  58849-58850,59062-59063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
	  1 line added method standard_dec for dialing out on groups, to
	  avoid conflicts, which caused issues with some ISDN providers
	  ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
	  Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
	  crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
	  avoid sending a disconnect when we already received one. ........
	  r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
	  1 line modified a loglevel ........ ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  channels/misdn/isdn_lib.c: Merged revisions 58825-58826 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58825 | crichter | 2007-03-12 13:43:24 +0100
	  (Mo, 12 Mär 2007) | 1 line added UU transceiving and corect
	  handling for rdnis ................ r58826 | crichter |
	  2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines Merged
	  revisions 57034,57523,57753,58558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
	  1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
	  bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
	  19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
	  r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
	  1 line fixed another place where the out_cause was hardcoded to
	  16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
	  Mar 2007) | 1 line we can free channel 31 as well, since we can
	  occupy it ........ ................

2007-05-18 09:10 +0000 [r64903-64921]  Olle Johansson <oej@edvina.net>

	* include/asterisk/adsi.h, main/adsistub.c (added), res/res_adsi.c,
	  apps/app_voicemail.c: Issue #5930 - Remove dependencies on
	  res_adsi.so - clwade A big THANK YOU to clwade for this patch.
	  Minor modifications by me.

	* channels/chan_sip.c: Another fix for the support for recordings
	  controlled by INFO-packets We still lack a setting to
	  enable/disable this per peer

2007-05-18 02:55 +0000 [r64869-64870]  Russell Bryant <russell@digium.com>

	* CHANGES: Add ENUMQUERY and ENUMRESULT to the CHANGES file.

	* /, apps/app_queue.c: Merged revisions 64868 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) |
	  5 lines Fix a small bug I noticed while working on something
	  else. app_queue did not unregister its device state monitoring
	  callback in unload_module(). So, this would make Asterisk crash
	  on the first device state change after you unload the module.
	  ........

2007-05-17 21:20 +0000 [r64821]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 64820 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64820 | tilghman | 2007-05-17 16:19:34 -0500
	  (Thu, 17 May 2007) | 10 lines Merged revisions 64819 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17
	  May 2007) | 2 lines How is it that we never caught that this is
	  returning the opposite of our documentation, until now? ........
	  ................

2007-05-17 17:12 +0000 [r64786]  Russell Bryant <russell@digium.com>

	* main/manager.c, configs/manager.conf.sample: Add an option that
	  lets you only allow one connection at a time for each manager
	  user. (issue #8664, reported and original patch by ssokol, patch
	  updated by bkruse, and further updated by me)

2007-05-17 16:54 +0000 [r64762]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 64761 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64761 | qwell | 2007-05-17 11:53:27 -0500 (Thu,
	  17 May 2007) | 12 lines Merged revisions 64758 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
	  lines If we have a negative current message, we shouldn't go back
	  even further... Issue 9727. ........ ................

2007-05-17 16:53 +0000 [r64757-64760]  Russell Bryant <russell@digium.com>

	* /, contrib/scripts/astxs (removed): Merged revisions 64759 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64759 | russell | 2007-05-17 11:52:53 -0500 (Thu, 17 May 2007) |
	  3 lines Remove script that is no longer functional since the
	  build system was redone. (issue #9340, reported by junky)
	  ........

	* apps/app_dial.c, /: Merged revisions 64756 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) |
	  3 lines Increase the size of a buffer to support longer dial
	  strings for channels. (issue #9291, reported and fix suggested by
	  meni) ........

2007-05-17 16:11 +0000 [r64721-64755]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 64754 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2
	  lines Even more direct RTP setup fixes! Don't allow a codec that
	  isn't supported to creep into the SDP of either side. (issue
	  #9446 reported by marcelbarbulescu) ........

	* /, apps/app_voicemail.c: Merged revisions 64720 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64720 | file | 2007-05-17 09:48:44 -0400 (Thu, 17 May 2007) | 2
	  lines Fix authuser support. (issue #9740 reported by
	  xmarksthespot) ........

2007-05-17 06:14 +0000 [r64657-64687]  Russell Bryant <russell@digium.com>

	* README, /: Merged revisions 64686 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64686 | russell | 2007-05-17 01:13:53 -0500 (Thu, 17 May 2007) |
	  3 lines Update the main README to reflect the new build process
	  for 1.4 and above. (issue #9725, patch by eliel) ........

	* main/app.c: Ignore this ... playing with jira (AST-1)

2007-05-16 11:01 +0000 [r64494-64611]  Olle Johansson <oej@edvina.net>

	* /: Blocking patch

	* /, channels/chan_sip.c: Below patches with some re-structuring
	  for trunk --- Merged revisions 64602 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2
	  lines Issue #9681 - Handle www-auth on BYE ........

	* /, channels/chan_sip.c: Merged revisions 64578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2
	  lines Final part of issue #9483 - fixing transfer() of sip calls
	  in the dial plan (twilson) ........

	* /: Blocking patch that was already committed to trunk

	* /, channels/chan_sip.c: Merged revisions 64543 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed,
	  16 May 2007) | 10 lines Merged revisions 64535 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
	  lines Support SIP uri's starting with SIP: and sip: (reported by
	  Tony Mountfield on the mailing list. Thanks!) ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 64516 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed,
	  16 May 2007) | 17 lines Merged following patch with a lot of
	  changes for 1.4 ------ Merged revisions 64514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
	  lines Issue #9726 - rlister - Better logging for ACL denials
	  While at it, also added better logging and handling of peers that
	  are not supposed to register. My patch, stole the issue report
	  from Russell. My apologies, Russell :-) ........ ................

	* channels/chan_sip.c: Issue #9304 - Update help text to match
	  functionality. Patch by kshumard with changes by oej

	* channels/chan_sip.c, configs/sip.conf.sample: Issue #6789 -
	  Marquis - Add option to support regexten removal when host
	  becomes unreachable

	* main/event.c: This file really needs more documentation... When
	  we implement new API's - please include a small general overview
	  in Doxygen

	* main/dial.c: Small doxygen updates

2007-05-15 23:05 +0000 [r64469-64480]  Russell Bryant <russell@digium.com>

	* funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: Add two
	  new dialplan functions: ENUMQUERY and ENUMRESULT. These functions
	  allow you to initiate an ENUM query using ENUMQUERY, and then
	  access the details of all of the results using ENUMRESULT.
	  Previously, if you wanted to access multiple results, Asterisk
	  would have to do a new DNS lookup every time. (patch by bbryant)

	* pbx/pbx_dundi.c: Make sure that DUNDIRESULT is given an ID.

2007-05-15 20:45 +0000 [r64455]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: XXX-XXX-XXX
	  appears to be the standard ANSI pointcode format

2007-05-15 19:57 +0000 [r64427]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 64426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64426 | russell | 2007-05-15 14:52:18 -0500 (Tue, 15 May 2007) |
	  3 lines Properly fix a problem that occurs when you set
	  PARKINGEXTEN to an exten where a call is already parked. (issue
	  #9723, patch by me) ........

2007-05-14 23:43 +0000 [r64399]  Kevin P. Fleming <kpfleming@digium.com>

	* /: this does not belong here

2007-05-14 22:25 +0000 [r64384]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Only print the SS7 UP once. Not every time
	  we get the test messages on the line.

2007-05-14 21:51 +0000 [r64355]  Jason Parker <jparker@digium.com>

	* main/Makefile: With libmmime.a as a .PHONY target, asterisk gets
	  rebuilt every time, but without proper ASTCFLAGS. This caused a
	  problem with the buildinfo.o file not being able to find
	  asterisk/build.h This was affecting DESTDIR, but I *think* that
	  if asterisk had never been installed before, it would've failed
	  also.

2007-05-14 21:17 +0000 [r64354]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 64353 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64353 | russell | 2007-05-14 16:16:39 -0500 (Mon, 14 May 2007) |
	  4 lines When someone requests a specific parking space using the
	  PARKINGEXTEN variable, ensure that no other caller is already
	  there. (issue #9723, reported by mdu113, patch by me) ........

2007-05-14 19:35 +0000 [r64323-64325]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 64324 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2
	  lines Change -2 to XMIT_ERROR to clarify a bit more ........

	* /: Blocking patch already committed to trunk

2007-05-14 19:21 +0000 [r64322]  Russell Bryant <russell@digium.com>

	* /, channels/chan_alsa.c: Merged revisions 64306 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) |
	  3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it.
	  An unknown indication will trigger an error and cause sounds to
	  stop, which in this case, is ringing. ........

2007-05-14 18:49 +0000 [r64274-64279]  Joshua Colp <jcolp@digium.com>

	* /, codecs/codec_speex.c: Merged revisions 64278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64278 | file | 2007-05-14 14:48:33 -0400 (Mon, 14 May 2007) | 2
	  lines Properly set datalen field when doing PLC in codec_speex.
	  (issue #9722 reported by mihai) ........

	* /, main/devicestate.c: Merged revisions 64276 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r64276 | file | 2007-05-14 14:36:34 -0400 (Mon,
	  14 May 2007) | 10 lines Merged revisions 64275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
	  lines Only perform stripping of - strings from the channel name
	  for Zap channels. Anywhere else we might remove a legitimate part
	  of a device name. (issue #9668 reported by stevedavies) ........
	  ................

	* channels/chan_sip.c: If no port is specified in the outboundproxy
	  setting then use the standard SIP port. (issue #9665 reported by
	  tootai)

2007-05-14 18:14 +0000 [r64243-64273]  Jason Parker <jparker@digium.com>

	* configs/queues.conf.sample: oops - silly typo there

	* configs/queues.conf.sample, apps/app_queue.c: Don't allow
	  rounding seconds to weird values that may cause "unexpected"
	  results. Issue 9514.

	* apps/app_queue.c: Add 'c' option to app_queue which allows for
	  continuing in the dialplan if the callee hangs up. Issue 9284,
	  patch by lyl, modified a little bit by me (I felt 'continue' was
	  better than 'keepalive')

2007-05-14 17:25 +0000 [r64242]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 64240 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2
	  lines Fix scenario where if a phone that simply called Echo() put
	  itself on hold it could never get off hold. ........

2007-05-14 16:08 +0000 [r64225-64226]  Russell Bryant <russell@digium.com>

	* configure: Regenerate configure script after last change to
	  acinclude.m4

	* acinclude.m4: Remove an extra space from the macro that checks
	  for C defines. (issue #9715, tzafrir)

2007-05-14 14:13 +0000 [r64208]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, channels/chan_local.c, /: Merged
	  revisions 64193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1
	  line As per 9570, worrisome CDR warnings have been removed, that
	  are either not helpful, or not relevant. ........

2007-05-14 10:40 +0000 [r64142-64158]  Olle Johansson <oej@edvina.net>

	* main/channel.c, /: Merged revisions 64157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2
	  lines Add hangupcause when we lack codecs for transcoding
	  ........

	* channels/chan_sip.c: Improve handling network errors on
	  transmission to hosts that don't reply or are unreachable With
	  this code, the call will fail as soon as we get a network error.
	  This may happen on first xmit or a later one, so the retransmit
	  code handles this too.

2007-05-12 22:28 +0000 [r64087-64115]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 64114 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2
	  lines This concludes my final adventure with bitmasks and the
	  onhold flag. Would anyone care for some peanuts? ........

	* /, channels/chan_sip.c: Merged revisions 64086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2
	  lines Tweak hold flags some more. They can be of three states
	  when active: active, inactive, one direction. ........

2007-05-12 19:38 +0000 [r64072]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_enum.c: Issue 9716 - doc/enum.txt no longer exists in
	  trunk

2007-05-12 16:33 +0000 [r64045]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 64044 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2
	  lines Ensure the onhold flag is set no matter what when being put
	  on hold. ........

2007-05-11 22:52 +0000 [r63967-64030]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Add/fix
	  support for Redial, Speeddial, and Messages buttons. Combined
	  effort by DEA and mvanbaak.

	* main/asterisk.c: oops.. Fix the logic of the last commit.

	* Makefile, main/asterisk.c: Better fallback method for
	  autosystemname. Issue 9713, patch by Juggie with minor mods by
	  me.

	* main/manager.c, /: Merged revisions 63982 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7
	  lines Hide manager password from "manager show user foo". I
	  realize that there are other ways to get this, but we really
	  don't need to just show it in plain text so easily. Issue 9273,
	  patch by junky ........

	* Makefile, main/asterisk.c: Add autosystemname setting to
	  asterisk.conf When enabled, it will set the systemname to be the
	  hostname of the system Issue 9713, patch by Juggie - slightly
	  modified by me, to "failover" to localhost

2007-05-11 18:31 +0000 [r63946]  Russell Bryant <russell@digium.com>

	* doc/qos.tex: Fix some syntax errors.

2007-05-11 16:37 +0000 [r63906]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, /, contrib/scripts/safe_asterisk: Merged revisions
	  63905 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63905 | tilghman | 2007-05-11 11:35:51 -0500
	  (Fri, 11 May 2007) | 10 lines Merged revisions 63903 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11
	  May 2007) | 2 lines Issue 9121 - fixups for safe_asterisk script
	  ........ ................

2007-05-11 16:21 +0000 [r63901-63902]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 63886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) |
	  6 lines When MD5 authentication is not possible because there is
	  no challenge present, either because the Challenge action was
	  never issued, or some other reason, give a proper error message
	  and return an error instead of claiming that the user wasn't
	  found. (reported by jsmith on IRC) ........

	* res/res_agi.c: Add gender support for AGI SAY NUMBER. (issue
	  #9537, patch by chappell)

2007-05-11 15:48 +0000 [r63873]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 63872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63872 | file | 2007-05-11 11:43:14 -0400 (Fri, 11 May 2007) | 2
	  lines Make the PARKINGEXTEN feature of parking actually work.
	  (issue #9708 reported by mdu113) ........

2007-05-10 23:16 +0000 [r63832]  Jason Parker <jparker@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 63830 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu,
	  10 May 2007) | 12 lines Merged revisions 63828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
	  lines Fix an issue with trying to kill a thread before it gets
	  created. Issue 9709, patch by nic_bellamy. ........
	  ................

2007-05-10 22:25 +0000 [r63805]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 63804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63804 | russell | 2007-05-10 17:23:42 -0500 (Thu, 10 May 2007) |
	  4 lines Strip terminal escape sequences from CLI command output
	  that is going to be sent out over the manager interface. (issue
	  #9659, reported by pari, fixed by me) ........

2007-05-10 21:25 +0000 [r63786]  Doug Bailey <dbailey@digium.com>

	* main/callerid.c: Added check for negative offset in cid spill to
	  prevent infinite loops

2007-05-10 20:51 +0000 [r63730-63751]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 63749 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu,
	  10 May 2007) | 12 lines Merged revisions 63748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
	  lines Do not allocate SIP pvt's for PEERs we can not reach. This
	  was seen as a lot of dialogs being created then immediately
	  destroyed at reload/restart of the SIP channel. ........
	  ................

	* apps/app_minivm.c: Fixing reload. Thanks to Mats Karlsson!

2007-05-09 19:24 +0000 [r63699]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 63698 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2
	  lines Use the DTMF frame on the channel when returning a DTMF
	  frame from AST_FRAME_NULL or AST_FRAME_VOICE. ........

2007-05-09 19:21 +0000 [r63697]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 63612 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) |
	  5 lines Modify ast_senddigit_begin() to use the same assumptions
	  used elsewhere in the code in that if a channel does not have a
	  send_digit_begin() callback, it only cares about DTMF END events.
	  (pointed out by Michael Neuhauser on the asterisk-dev list)
	  ........

2007-05-09 17:35 +0000 [r63655]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Merged revisions 63654 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed,
	  09 May 2007) | 10 lines Merged revisions 63653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
	  lines Make sure we only create a DSP if it's requested on
	  SUB_REAL ........ ................

2007-05-09 16:56 +0000 [r63613]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 63611 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63611 | file | 2007-05-09 12:54:56 -0400 (Wed,
	  09 May 2007) | 10 lines Merged revisions 63610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
	  lines Properly handle hints that point to multiple devices in
	  chan_sip. Why chan_sip is even doing this I have no idea but I
	  would rather not go into a rant. (issue #9536 reported by
	  rlister) ........ ................

2007-05-09 16:44 +0000 [r63609]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 63608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) |
	  5 lines Only call ast_senddigit_begin() in ast_senddigit() if the
	  channel has a send_digit_begin() callback. Checking the
	  END_DTMF_ONLY flag was the wrong thing to do, because that flag
	  indicates that a *bridged* channel only wants DTMF END events
	  coming from this channel. ........

2007-05-09 14:52 +0000 [r63567]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_directory.c: Merged revisions 63566 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63566 | tilghman | 2007-05-09 09:50:33 -0500
	  (Wed, 09 May 2007) | 10 lines Merged revisions 63565 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09
	  May 2007) | 2 lines Replicate fix from 51158 (app_voicemail) to
	  app_directory (Issue 9224) ........ ................

2007-05-09 13:24 +0000 [r63536]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 63535 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63535 | russell | 2007-05-09 08:24:03 -0500 (Wed, 09 May 2007) |
	  6 lines I have seen multiple people post questions trying to
	  figure out what the message "The configure script must be
	  executed before running 'make'" means. So, add another like that
	  says to specifically run ./configure. If this isn't obvious
	  enough, then they should be using something like AsteriskNOW and
	  not installing from source. ........

2007-05-09 13:07 +0000 [r63533]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 63532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2
	  lines Don't retransmit 200 OK's on ignore status. (Reported on
	  asterisk-users) ........

2007-05-08 22:40 +0000 [r63479]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 63478 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63478 | tilghman | 2007-05-08 17:38:02 -0500
	  (Tue, 08 May 2007) | 10 lines Merged revisions 63477 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08
	  May 2007) | 2 lines Issue 9602 - segfault in app_macro ........
	  ................

2007-05-08 16:54 +0000 [r63404-63449]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 63448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63448 | russell | 2007-05-08 11:53:09 -0500 (Tue, 08 May 2007) |
	  4 lines I mixed up the use of the find_feature() function, so I
	  renamed it find_dynamic_feature, and changed the code to use the
	  correct lock when using it. ........

	* channels/chan_sip.c, res/res_features.c,
	  include/asterisk/features.h: I noted this on the dev list but got
	  no response, so I just did it myself. Lock the call features when
	  being used in chan_sip.

	* /, res/res_features.c: Merged revisions 63445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63445 | russell | 2007-05-08 11:30:43 -0500 (Tue, 08 May 2007) |
	  2 lines Use a read/write lock when accessing the built-in
	  features. ........

	* contrib/scripts/realtime_pgsql.sql (added), /,
	  contrib/realtime_pgsql.sql (removed): Merged revisions 63403 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63403 | russell | 2007-05-08 10:10:37 -0500 (Tue, 08 May 2007) |
	  3 lines Move realtime_pgsql.sql to contrib/scripts to be with the
	  rest of the sql examples. (issue #9676, suretec) ........

2007-05-08 06:26 +0000 [r63361]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 63360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63360 | tilghman | 2007-05-08 01:22:37 -0500
	  (Tue, 08 May 2007) | 10 lines Merged revisions 63359 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08
	  May 2007) | 2 lines Issue 9527 - upon entering a folder, no
	  message is selected (curmsg == -1), so deleting causes memory
	  corruption (beyond bounds) ........ ................

2007-05-07 22:32 +0000 [r63319-63330]  Russell Bryant <russell@digium.com>

	* /, contrib/realtime_pgsql.sql (added),
	  configs/res_pgsql.conf.sample (added): Merged revisions 63329 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) |
	  3 lines Add a sample configuration file and example tables for
	  use with res_config_pgsql. (issue #9676, suretec) ........

	* apps/app_meetme.c: Make a minor tweak to admin_exec() - don't
	  lock the conference list until it is actually necessary.

	* apps/app_meetme.c, CHANGES: Add a new application,
	  MeetMeChannelAdmin, which is similar to MeetMeAdmin, except it
	  lets you operate on a channel by name instead of conference
	  member number. It is very useful in combination with the 'X'
	  option to ChanSpy. (issue #9671, patch by mnicholson, with some
	  small modifications by me)

2007-05-07 21:47 +0000 [r63284-63287]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
	  revisions 63286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon,
	  07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
	  lines Properly handle what happens during a masquerade in
	  relation to group counting. (issue #9657 reported by ramonpeek)
	  ........ ................

2007-05-07 20:07 +0000 [r63228-63255]  Olle Johansson <oej@edvina.net>

	* /, main/config.c: Merged revisions 63254 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63254 | oej | 2007-05-07 22:05:15 +0200 (Mon, 07 May 2007) | 2
	  lines Don't remove configuration from memory just because one
	  section failed. ........

	* include/asterisk/module.h, main/loader.c: Constifications

	* channels/chan_jingle.c, res/res_jabber.c: Adding external
	  referenses for doxygen See
	  http://www.asterisk.org/doxygen/trunk/extref.html

	* channels/chan_misdn.c: Adding external reference

	* channels/chan_misdn.c: Doxyfication... There's a shortage of
	  comments in this file...

2007-05-06 20:09 +0000 [r63182]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Lock iax2 pvt structure when passing off to
	  the AMI function, and make sure it exists. (issue #9674 reported
	  by arabe)

2007-05-06 13:11 +0000 [r63168]  Olle Johansson <oej@edvina.net>

	* /, main/file.c: Merged revisions 63152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63152 | oej | 2007-05-06 14:28:38 +0200 (Sun, 06 May 2007) | 2
	  lines Stop the video stream when you stop playback of all streams
	  for a call ........

2007-05-05 08:05 +0000 [r63136]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Adding some missing spaces - Correcting
	  error messages - Disabling code that doesn't do anything - Making
	  sure we always respond to this request, happily

2007-05-04 20:11 +0000 [r63105]  Pari Nannapaneni <paripurnachand@digium.com>

	* /, configs/manager.conf.sample: Merged revisions 63047 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1
	  line explanation for httptimeout in manager.conf ........

2007-05-04 20:06 +0000 [r63104]  Jason Parker <jparker@digium.com>

	* /, res/res_jabber.c: Merged revisions 63099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r63099 | qwell | 2007-05-04 15:03:49 -0500 (Fri, 04 May 2007) | 4
	  lines Fix a crash when checking version attribute in an incoming
	  XML caps element. Issue 9667, patch by phsultan. ........

2007-05-04 19:48 +0000 [r63089]  Russell Bryant <russell@digium.com>

	* main/manager.c: Convert spaces to tabs for indentation.

2007-05-04 18:47 +0000 [r63046-63076]  Steve Murphy <murf@digium.com>

	* res/res_features.c: According to my testing, it's better if the
	  ast_find_call_feature func ran this way instead, as far as the
	  snom record button is concerned

	* doc/CODING-GUIDELINES, channels/chan_sip.c, res/res_features.c,
	  include/asterisk/features.h: a small upgrade to the coding
	  standard, and an update to the code that triggered the upgrade.

	* channels/chan_sip.c, res/res_features.c, UPGRADE.txt,
	  include/asterisk/features.h: Added a small bit of code to support
	  the SNOM 360's Record button. Made the find_feature func in
	  res_features.c public, so I could use it to find the automon dial
	  sequence as configured by the user. When the INFO packet has a
	  Record: header with on/off, the sequence is sent as consecutive
	  DTMF frames on the phone's channel, triggering the automon
	  functionality. The user has to configure the automon in
	  features.conf, and set up his dialplan accordingly.

2007-05-04 13:56 +0000 [r63030-63032]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, channels/chan_iax2.c: Add the new
	  ChannelUpdate event to inform manager clients about the PVT ID
	  and some other channel driver data that is needed to follow the
	  call through the PBX.

	* main/manager.c: Add "CoreStatus" - from the moremanager branch.
	  This can be extended with more information, ideas and patches are
	  welcome, as usual :-)

	* include/asterisk.h, main/manager.c, include/asterisk/manager.h,
	  include/asterisk/options.h: - Add manager command CoreSettings -
	  Add missing option to options.h - Add missing variables to
	  asterisk.h - Move manager version to manager.h include file

2007-05-03 16:45 +0000 [r62990]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 62989 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu,
	  03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
	  lines When a peer is seeded or built tell the devicestate core to
	  update it's status. This is easier then having chan_sip load
	  before pbx_config. (issue #9658 reported by dlynes) ........
	  ................

2007-05-03 16:43 +0000 [r62988]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/loader.c: Merged revisions 62986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007)
	  | 2 lines improve loader a bit, by avoiding trying to initialize
	  embedded modules twice and avoiding trying to load modules from
	  disk when they have been loaded already during the 'preload' pass
	  (reported by blitzrage on IRC, patch by me) ........

2007-05-03 15:23 +0000 [r62943]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 62942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) |
	  17 lines Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant,
	  2007, TM, Patent Pending). This set of changes came from a
	  debugging session I had with Dwayne Hubbard. When he called into
	  his home FXO, ran the Echo application, and pressed a digit, the
	  digit would be echoed back and would never end. This is fixed,
	  along with a couple other little improvements. * When chan_zap is
	  in the middle of playing a digit to a channel, it feeds back null
	  frames, not voice frames. So, I have modified ast_read to check
	  the timing on emulated DTMF when it receives null frames, in
	  addition to where it was doing this on voice frames. * Make a
	  tweak to setting the duration on emulated DTMF digits. If there
	  was no duration specified, it set it to be the minimum, instead
	  of the default. * Instead of timing the emulated digits off of
	  the number of samples in audio frames that pass through, just use
	  time values. Now there is no code in this section that assumes
	  8kHz audio. ........

2007-05-03 14:44 +0000 [r62911-62914]  Steve Murphy <murf@digium.com>

	* /: blocking 62913 (1.4) from trunk, as it's already done here

	* /, pbx/ael/ael.tab.c, pbx/ael/ael.y,
	  pbx/ael/ael-test/ref.ael-test20 (added), pbx/ael/ael.tab.h,
	  pbx/ael/ael-test/ael-test20/extensions.ael (added),
	  pbx/ael/ael-test/ael-test20 (added): Merged revisions 62883 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62883 | murf | 2007-05-03 07:54:56 -0600 (Thu, 03 May 2007) | 1
	  line These mods fix bug 9623, where an '@' in the eswitch
	  contents causes a syntax error. I also updated the regressions.
	  ........

2007-05-03 00:25 +0000 [r62824-62843]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 62842 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62842 | kpfleming | 2007-05-02 20:23:37 -0400
	  (Wed, 02 May 2007) | 10 lines Merged revisions 62841 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
	  May 2007) | 2 lines doh... initializing the pointer variable will
	  work just a bit better ........ ................

	* main/minimime: ignore the archive we build in this directory

	* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
	  revisions 62797,62807 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62797 | kpfleming | 2007-05-02 19:57:23 -0400
	  (Wed, 02 May 2007) | 7 lines improve static Realtime config
	  loading from PostgreSQL: don't request sorting on fields that are
	  pointless to sort on use ast_build_string() instead of snprintf()
	  don't request the list of fieldnames that resulted from the query
	  when we both knew what they were before we ran the query _AND_ we
	  aren't going to do anything with them anyway (patch by me,
	  inspired by blitzrage's bug report about res_config_odbc)
	  ................ r62807 | kpfleming | 2007-05-02 20:02:57 -0400
	  (Wed, 02 May 2007) | 15 lines Merged revisions 62796 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
	  May 2007) | 7 lines increase reliability and efficiency of static
	  Realtime config loading via ODBC: don't request fields we aren't
	  going to use don't request sorting on fields that are pointless
	  to sort on explicitly request the fields we want, because we
	  can't expect the database to always return them in the order they
	  were created (reported by blitzrage in person (!), patch by me)
	  ........ ................

2007-05-02 23:50 +0000 [r62791-62795]  Russell Bryant <russell@digium.com>

	* CHANGES: Fix some bad grammar.

	* apps/app_meetme.c, CHANGES: When a conference is created, the
	  UNIQUEID of the channel that caused it to be created will now be
	  stored. Then, every channel that joins the conference will have
	  the MEETMEUNIQUEID channel variable set with this ID. This can be
	  used to relate callers that come and go from long standing
	  conferences. (issue #7295, patch by softins)

	* CHANGES: Note Hungarian language support in CHANGES

	* main/say.c, configs/say.conf.sample: Add Hungarian language
	  support to say.c and say.conf. (issue #7077, patch by adomjan)

	* main/channel.c, /: Merged revisions 62789 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) |
	  20 lines Merge changes from team/russell/inband_dtmf ... Fix some
	  issues related to generating inband DTMF. There are two changes
	  here: 1) The list of DTMF tones in the senddigit_begin() function
	  explicitly specified 100ms of the tone followed by 100ms of
	  silence. This really broke things with the way that Asterisk now
	  wants complete control over when the digit begins and ends. So,
	  regardless of what Asterisk really wanted to do, this was going
	  to play out the tone at the length it wanted to. This caused
	  various problems like DTMF translation to inband to be extremely
	  unreliable. The list of tones has been changed so that the
	  correct DTMF tone is played indefinitely until Asterisk tells it
	  to stop. 2) ast_write() had to be modified to let a DTMF_END
	  frame get processed even when a generator is present. This is how
	  the tone will finally get stopped. (issues #8944, #9250, #9348,
	  maybe others. Thanks to mdu113 from #8944 for the testing and
	  feedback!) ........

2007-05-02 20:57 +0000 [r62741]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, /: Merged revisions 62738 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62738 | murf | 2007-05-02 14:46:07 -0600 (Wed,
	  02 May 2007) | 9 lines Merged revisions 62737 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1
	  line Some tweaks to satisfy CDR bug 8796, where being in 'h'
	  extension louses up the dst field ........ ................

2007-05-02 17:49 +0000 [r62693]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 62692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62692 | tilghman | 2007-05-02 12:43:48 -0500
	  (Wed, 02 May 2007) | 12 lines Merged revisions 62691 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02
	  May 2007) | 4 lines Issue 9638 - if a text frame is sent with no
	  terminating NULL through a bridged IAX connection, the remote end
	  will receive garbage characters tacked onto the end. ........
	  ................

2007-05-02 17:24 +0000 [r62690]  Steve Murphy <murf@digium.com>

	* main/channel.c, main/pbx.c, channels/chan_zap.c, /,
	  cdr/cdr_radius.c: Merged revisions 62689 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1
	  line a)In chan_zap, set the clid, src fields in channel_alloc
	  call. b)in the channel_alloc func, set the cid_num and name
	  fields from the arglist[blush]. c) don't update the channel app &
	  app data fields if you are in the 'h' extension. d)the
	  load_module func in cdr_radius needs to return DECLINE, SUCCESS.
	  ........

2007-05-02 15:46 +0000 [r62671-62673]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, CHANGES: Update the device state
	  functionality of chan_local such that it will return NOT_INUSE or
	  INUSE when Local channels are in use as opposed to just UNKNOWN.
	  It will still return INVALID if the extension doesn't exist at
	  all. (issue #8048, patch from tim_ringenbach)

	* CHANGES: Add the new options for attended transfer to the CHANGES
	  file.

	* doc/ip-tos.tex (removed), doc/qos.tex (added): For some reason
	  when I merged 802.1p support, the new documentation file was not
	  properly added. Thanks to IgorG for pointing it out! :)

2007-05-02 12:12 +0000 [r62609-62656]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add a small message that we're doing
	  something. On my systems, there's a long dead period with a
	  non-responsive CLI after I issue "load chan_sip.so"

	* channels/chan_sip.c: More username body parts to fix... If
	  working, this needs to be backported to 1.2, 1.4. But first, some
	  serious SIP testing :-)

	* channels/chan_sip.c: Handle
	  sip:username;parameter=12345@example.com;parameter=1234 URI's
	  properly

	* /, channels/chan_sip.c: Merged revisions 62624 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2
	  lines Don't unlock a channel that we already know does not exist
	  (propably isue 8228) ........

	* CREDITS: Updating CREDITS

2007-05-01 22:24 +0000 [r62549-62593]  Russell Bryant <russell@digium.com>

	* res/res_features.c, configs/features.conf.sample: In addition to
	  making it so attended transfers don't fail unnecessarily, add
	  some new options to control what happens when you hangup on an
	  attended transfer before the target extension answers the
	  transferred channel. You can now have it send the transferee back
	  to the transferer. (issue #8413, patch from sergee with very
	  minor modifications by me)

	* /, res/res_features.c: Merged revisions 62548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62548 | russell | 2007-05-01 16:57:10 -0500
	  (Tue, 01 May 2007) | 12 lines Merged revisions 62547 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01
	  May 2007) | 4 lines Remove an unnecessary check that makes it so
	  if you hang up after doing an attended transfer before the target
	  extension answers the channel, the transfer is not successful.
	  (issue #9338, patch by svanlund) ........ ................

2007-05-01 21:41 +0000 [r62546]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 62545 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007)
	  | 2 lines Bug 9590 - Memory leaks around find_user() (found by
	  rayjay, different fixes by me) ........

2007-05-01 16:27 +0000 [r62415-62498]  Russell Bryant <russell@digium.com>

	* /, configs/indications.conf.sample: Merged revisions 62497 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62497 | russell | 2007-05-01 11:26:48 -0500
	  (Tue, 01 May 2007) | 11 lines Merged revisions 62496 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01
	  May 2007) | 3 lines Add indications.conf information for the
	  Philippines. (issue #9525, reported and patched by loloski)
	  ........ ................

	* CHANGES: Add a note to CHANGES about the new support for 802.1p.
	  Thanks IgorG!

	* CHANGES, apps/app_queue.c, doc/queuelog.tex: This patch adds
	  additional information to the EXITWITHKEY and EXITWITHTIMEOUT
	  entries in the queue log. (issue #7561, reported and originally
	  patched by fkasumovic, patch slightly modified and updated to
	  trunk by me)

	* include/asterisk/acl.h, main/udptl.c, channels/chan_sip.c,
	  include/asterisk/rtp.h, main/acl.c, include/asterisk/netsock.h,
	  channels/iax2-provision.c, channels/chan_iax2.c, main/rtp.c,
	  main/netsock.c, configs/h323.conf.sample,
	  configs/iax.conf.sample, configs/mgcp.conf.sample,
	  configs/iaxprov.conf.sample, channels/chan_h323.c,
	  pbx/pbx_dundi.c, include/asterisk/udptl.h,
	  configs/sip.conf.sample, doc/asterisk.tex, channels/chan_mgcp.c:
	  Add support for setting the CoS for VLAN traffic (802.1p) in
	  Linux. The file doc/qos.tex has been updated to document the new
	  functionality. (issue #9540, patch submitted by IgorG)

	* channels/chan_zap.c, /: Merged revisions 62419 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62419 | russell | 2007-04-30 10:58:28 -0500
	  (Mon, 30 Apr 2007) | 12 lines Merged revisions 62417 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30
	  Apr 2007) | 4 lines This patch fixes an issue where depending on
	  the cause code, when the network sends a PRI disconnect, the call
	  may not be properly hung up. (issue #9588, reported and patched
	  by softins) ........ ................

	* channels/chan_sip.c: Don't crash when invalid arguments are
	  provided to the CHANNEL() function for a SIP channel. (issue
	  #9619, reported by jtodd, original patch by Corydon76, committed
	  patch slightly modified by me)

	* include/asterisk/http.h, /, main/http.c: Merged revisions 62414
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) |
	  4 lines When serving dynamic content, include a Cache-Control
	  header to instruct the browsers to not store the resulting
	  content. (issue #9621, reported by Pari, patch by me) ........

2007-04-30 14:56 +0000 [r62372]  Jason Parker <jparker@digium.com>

	* configs/iax.conf.sample, /: Merged revisions 62371 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr
	  2007) | 2 lines Remove unused (and potentially confusing)
	  jitterbuffer options from sample config. ........

2007-04-30 14:37 +0000 [r62370]  Joshua Colp <jcolp@digium.com>

	* /, main/asterisk.c: Merged revisions 62369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62369 | file | 2007-04-30 11:36:11 -0300 (Mon,
	  30 Apr 2007) | 10 lines Merged revisions 62368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
	  lines Update copyright notice. It's now the year 2007! ........
	  ................

2007-04-29 05:51 +0000 [r62219-62332]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 62331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) |
	  3 lines Fix a bug that made the "language" setting in zapata.conf
	  not functional. (issue #9626, reported and fixed by sergee)
	  ........

	* CHANGES: note MeetMe change in CHANGES

	* apps/app_meetme.c: Enable the functionality of the 'o' option to
	  "optimize talker" by default.

	* channels/iax2.h: Reformat some of iax2.h and convert comments to
	  doxygen format

	* include/asterisk.h, channels/chan_zap.c, channels/chan_sip.c,
	  main/Makefile, res/res_eventtest.c (added),
	  configs/voicemail.conf.sample, UPGRADE.txt, CHANGES,
	  channels/chan_iax2.c, main/dial.c, include/asterisk/event.h
	  (added), include/asterisk/event_defs.h (added), main/event.c
	  (added), configs/sip.conf.sample, main/asterisk.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from
	  team/russell/events This set of changes introduces a new generic
	  event API for use within Asterisk. I am still working on a way
	  for events to be shared between servers, but this part is ready
	  and can already be used inside of Asterisk. This set of changes
	  introduces the first use of the API, as well. I have restructured
	  the way that MWI (message waiting indication) is handled. It is
	  now event based instead of polling based. For example, if there
	  are a bunch of SIP phones subscribed to mailboxes, then chan_sip
	  will not have to constantly poll the mailboxes for changes.
	  app_voicemail will generate events when changes occur. See
	  UPGRADE.txt and CHANGES for some more information on the effects
	  of these changes from the user perspective. For developer
	  information, see the text in include/asterisk/event.h. As always,
	  additional feedback is welcome on the asterisk-dev mailing list.

	* doc/ast_appdocs.tex, doc/dundi.tex: Update the DUNDi section of
	  the documentation with example usage of DUNDIQUERY and
	  DUNDIRESULT. Also, update the automatically generated application
	  docs.

	* pbx/pbx_dundi.c, CHANGES: Merge changes from
	  team/russell/dundi_results This introduces two new dialplan
	  functions: DUNDIQUERY and DUNDIRESULT. DUNDIQUERY lets you
	  intitiate a DUNDi query from the dialplan. Then, DUNDIRESULT will
	  let you find out how many results there are, and access each one
	  without having to the query again.

	* include/asterisk/lock.h: Remove a message that goes to LOG_ERROR
	  that's not really an error.

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add a
	  min-announce-frequency option to queues.conf which allows you to
	  control the minimum amount of time between queue announcements
	  for use when the caller's queue position changes frequently.
	  (issue #9604, patch by Matthew Roth)

	* /, channels/chan_agent.c: Merged revisions 62218 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27
	  Apr 2007) | 11 lines Fix a weird problem where when a caller
	  talking to someone sitting behind an agent channel sent a digit,
	  the digit would be played to the agent for forever. This is
	  because chan_agent always returned -1 from its send_digit_begin
	  and _end callbacks. This non-zero return value indicates to the
	  Asterisk core that it would like an inband DTMF generator put on
	  the channel. However, this is the wrong thing to do. It should
	  *always* return 0, instead. When the digit begin and end
	  functions are called on the proxied channel, the underlying
	  channel will indicate whether inband DTMF is needed or not, and
	  the generator will be put on that one, and not the Agent channel.
	  (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
	  by me) ........

2007-04-27 16:18 +0000 [r62175]  Jason Parker <jparker@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 62174 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri,
	  27 Apr 2007) | 11 lines Merged revisions 62173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
	  lines This transcoder message needn't be a NOTICE. I've seen it
	  cause confusion more than a few times. ........ ................

2007-04-27 16:15 +0000 [r62172]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 62171 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) |
	  6 lines If no variables were passed into
	  pbx_substitute_variables_helper_full(), then don't even bother
	  creating a temporary bogus channel, since that is only for
	  allowing certain functions to operate on the variables as if they
	  were on a channel. Most importantly, this fixes a crash. (issue
	  #9613, reported by callguy, fixed by me) ........

2007-04-27 14:40 +0000 [r62096-62141]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #9545 Autocomplete for "sip
	  unregister" cli command. (eliel) Thanks!

	* /, channels/chan_sip.c: Merged revisions 62137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri,
	  27 Apr 2007) | 12 lines Merged revisions 62126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
	  lines Issue #7351 - SIP Cancel fails due to the wrong contact
	  uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
	  - THANKS!!!! THis was a hard one to catch. ........
	  ................

	* /: Blocking patch to 1.4 that was alredy in trunk

2007-04-26 16:35 +0000 [r62039]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 62038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r62038 | file | 2007-04-26 12:33:52 -0400 (Thu,
	  26 Apr 2007) | 10 lines Merged revisions 62037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
	  lines Revert previous fix for when the IAX2 channel goes funky
	  (that's the technical term). This is causing legit calls to be
	  prematurely hung up. (issue #9600 reported by justdave) ........
	  ................

2007-04-26 03:24 +0000 [r62006]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 62005 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2
	  lines Missed an ast_app_group_discard during merge. Thanks
	  blitzrage! ........

2007-04-26 01:50 +0000 [r61960-61962]  Joshua Colp <jcolp@digium.com>

	* /, res/res_monitor.c: Merged revisions 61961 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2
	  lines Don't always say that the channel is being paused if it is
	  actually being unpaused in the Manager ack message. (reported by
	  jsmith in #asterisk-bugs) ........

	* /, main/config.c: Merged revisions 61959 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61959 | file | 2007-04-25 21:27:18 -0400 (Wed,
	  25 Apr 2007) | 10 lines Merged revisions 61958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
	  lines Don't count failed include attempts against the
	  configuration include level. (issue #9593 reported by mostyn)
	  ........ ................

2007-04-25 22:34 +0000 [r61915]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61914 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61914 | kpfleming | 2007-04-25 17:29:53 -0500
	  (Wed, 25 Apr 2007) | 10 lines Merged revisions 61913 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25
	  Apr 2007) | 2 lines handle a very bizarre race condition with
	  channels being redirected before a simple switch can be started
	  on them (issue #9286) ........ ................

2007-04-25 22:01 +0000 [r61864-61876]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 61870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61870 | russell | 2007-04-25 16:59:07 -0500
	  (Wed, 25 Apr 2007) | 10 lines Merged revisions 61866 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25
	  Apr 2007) | 2 lines If the callerid= option is specified, but
	  empty, clear any previous data. ........ ................

	* /, channels/chan_iax2.c: Merged revisions 61863 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61863 | russell | 2007-04-25 16:13:15 -0500
	  (Wed, 25 Apr 2007) | 10 lines Merged revisions 61862 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25
	  Apr 2007) | 2 lines Ensure that callerid settings are reset on a
	  reload. ........ ................

2007-04-25 19:27 +0000 [r61806]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, include/asterisk/app.h, funcs/func_groupcount.c,
	  /, main/app.c, main/cli.c: Merged revisions 61805 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed,
	  25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
	  lines Merge rewritten group counting support. No more storing
	  data on the variable list of the channels. That was bad, mmmk?
	  (issue #7497 reported by sabbathbh) ........ ................

2007-04-25 16:23 +0000 [r61788-61800]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61799 | russell | 2007-04-25 11:22:07 -0500
	  (Wed, 25 Apr 2007) | 11 lines Merged revisions 61798 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25
	  Apr 2007) | 3 lines Fix a typo where cid_num got copied instead
	  of cid_ani. (issue #9587, reported and patched by xrg) ........
	  ................

	* main/manager.c, /: Merged revisions 61787 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61787 | russell | 2007-04-24 16:34:53 -0500
	  (Tue, 24 Apr 2007) | 12 lines Merged revisions 61786 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24
	  Apr 2007) | 4 lines Don't crash if a manager connection provides
	  a username that exists in manager.conf but does not have a
	  password, and also requests MD5 authentication. (ASA-2007-012)
	  ........ ................

2007-04-24 19:08 +0000 [r61784]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c, /: removed #if 0 block from chan_zap
	  restart_monitor()

2007-04-24 19:03 +0000 [r61775-61782]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/channel.h: Merged revisions
	  61781 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) |
	  6 lines Improve DTMF handling in ast_read() even more in response
	  to a discussion on the asterisk-dev mailing list. I changed the
	  enforced minimum length of a digit from 100ms to 80ms.
	  Furthermore, I made it now enforce a gap of 45ms in between
	  digits. These values are not configurable in a configuration file
	  right now, but they can be easily changed near the top of
	  main/channel.c. ........

	* main/dial.c, /: Merged revisions 61774 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) |
	  5 lines Add a few more state changes in handle_frame_ownerless()
	  so that the SLA code will get notified of these changes even when
	  an owner channel is not provided. This isn't from a specific bug
	  report, it's just something I noticed while poking around.
	  ........

2007-04-24 16:10 +0000 [r61773]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61772 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue,
	  24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
	  lines Allow RFC2833 to be sent in the response SDP when an INVITE
	  comes in without SDP. (issue #9546 reported by mcrawford)
	  ........ ................

2007-04-23 18:49 +0000 [r61760-61767]  Russell Bryant <russell@digium.com>

	* main/manager.c: When building a JSON encoded string in the
	  GetConfigJSON manager action, escape the '\' and '"' characters.
	  (issue #9475, reported by pari, patch by me)

	* main/pbx.c, /: Merged revisions 61765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) |
	  5 lines Some dialplan functions, such as CUT(), expect to operate
	  on variables on a channel. So, this little hack lets them work in
	  places where a channel doesn't exist, such as within DUNDi
	  configuration. (issue #9465, reported and patched by Corydon76,
	  testing by blitzrage) ........

	* main/channel.c, /: Merged revisions 61763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) |
	  4 lines Ensure that digits passing through Asterisk have a
	  reasonable minimum length. It is currently 100 ms. If someone
	  thinks this should be different, feel free to speak up. (related
	  to issues #8944, #9250, and #9348) ........

	* CHANGES: Add OSP support for IAX2 to the changes file. Also,
	  slightly reorganize some of the content.

2007-04-20 21:37 +0000 [r61706-61708]  Jason Parker <jparker@digium.com>

	* /, main/rtp.c: Merged revisions 61707 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8
	  lines Avoid invalid seqno cycling detection. Per comment from
	  Dave Troy: This adds back in some simple typecasting I had in an
	  earlier version which I realize now may be breaking things. Issue
	  #9554. ........

	* /, main/loader.c: Merged revisions 61705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61705 | qwell | 2007-04-20 16:15:29 -0500 (Fri,
	  20 Apr 2007) | 12 lines Merged revisions 61704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
	  lines Fix an issue that I noticed while looking over issue 9571.
	  The reload timestamp was getting set after reloading the built-in
	  stuff, and before the modules. ........ ................

2007-04-20 21:12 +0000 [r61698-61702]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.h, funcs/func_channel.c, channels/iax2.h,
	  channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from
	  team/russell/iax2_osp This set of changes adds OSP support to
	  chan_iax2. However, I have modified the patch a bit from what was
	  submitted. You now use the CHANNEL() function to get and set the
	  OSP token for IAX2. (issue #8531, reported by and original patch
	  by homesick, patch updated by me)

	* /, main/rtp.c: Merged revisions 61697 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) |
	  2 lines Remove a stray debug message introduced by a recent
	  commit. ........

2007-04-20 19:54 +0000 [r61695]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 61694 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri,
	  20 Apr 2007) | 13 lines Merged revisions 61692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
	  lines If the '* to hangup' option is not enabled, we don't need
	  to disable * as a valid exit key. If it was enabled, this
	  statement would've never been checked in the first place. Issue
	  #9552 ........ ................

2007-04-20 18:23 +0000 [r61691]  Russell Bryant <russell@digium.com>

	* main/manager.c, /, include/asterisk/config.h, main/config.c,
	  apps/app_voicemail.c: Merged revisions 61690 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) |
	  4 lines Fix the UpdateConfig manager action to properly treat
	  "variables" and "objects" differently (a=b versus a=>b). (issue
	  #9568, reported by pari, patch by me) ........

2007-04-20 08:41 +0000 [r61689]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Use the last line in the SDP, even if it
	  has no CRLF. Remember Jon Postel :-) This code exists in 1.2 and
	  1.4 but was removed from trunk for some unknown reason.

2007-04-19 04:37 +0000 [r61682-61684]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 61683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61683 | tilghman | 2007-04-18 23:36:20 -0500 (Wed, 18 Apr 2007)
	  | 2 lines Bug 9557 - simple reason why reading a function always
	  returned NULL ........

	* funcs/func_groupcount.c, /, funcs/func_timeout.c,
	  funcs/func_cdr.c, funcs/func_callerid.c: Merged revisions 61681
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61681 | tilghman | 2007-04-18 21:45:05 -0500
	  (Wed, 18 Apr 2007) | 13 lines Merged revisions 61680 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18
	  Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action
	  without a Channel parameter can cause Asterisk to crash. The
	  reason this needs to be fixed in the functions instead of in AMI
	  is because Channel can legitimately be NULL, such as when
	  retrieving global variables. ........ ................

2007-04-18 22:11 +0000 [r61679]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 61678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61678 | kpfleming | 2007-04-18 17:10:23 -0500 (Wed, 18 Apr 2007)
	  | 2 lines allow external build systems to extract the required
	  sound file versions ........

2007-04-18 20:48 +0000 [r61671-61677]  Olle Johansson <oej@edvina.net>

	* /, main/rtp.c: Merged revisions 61676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2
	  lines Clean upp formatting, add some doxygen stuff while we're in
	  cleaning mode... Thanks Kevin! ........

	* /, main/rtp.c: Merged revisions 61674 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2
	  lines Issue #9554 - Improve RTCP (Dave Troy) ........

	* apps/app_minivm.c (added), configs/extensions_minivm.conf.sample
	  (added), configs/minivm.conf.sample (added): Mini-voicemail - an
	  embryo for a new voicemail system based on building blocks
	  instead of one large monolithic app. Supports multiple templates
	  and is designed mostly for voicemail delivery over e-mail.
	  There's a todo with a list of ideas in the source code if you
	  want to contribute. Feedback is appreciated!

2007-04-16 15:40 +0000 [r61667]  Olle Johansson <oej@edvina.net>

	* include/asterisk/rtp.h: Doxygen changes

2007-04-14 18:22 +0000 [r61661]  Claude Patry <cpatry@gmail.com>

	* main/say.c: test my new trunk access ;)

2007-04-13 21:23 +0000 [r61660]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_sip.c: added CLI 'sip unregister <peer>' for issue
	  9326. thanks eliel

2007-04-13 21:22 +0000 [r61659]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 61658 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61658 | murf | 2007-04-13 15:17:20 -0600 (Fri, 13 Apr 2007) | 1
	  line This is a fix to the way CDR merge handles the data that
	  results from ForkCDR. ........

2007-04-13 19:18 +0000 [r61649-61657]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 61656 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61656 | file | 2007-04-13 15:17:08 -0400 (Fri,
	  13 Apr 2007) | 10 lines Merged revisions 61655 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
	  lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
	  the same as OUTBOUND_GROUP except it will get unset after use so
	  it won't get accidentally inherited. (issue #BE-140) ........
	  ................

	* /, apps/app_speech_utils.c: Merged revisions 61651 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr
	  2007) | 2 lines Do not bother looking for a result if none are
	  present. ........

	* /, channels/chan_sip.c: Merged revisions 61648 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2
	  lines For those very verbose SIP implementations that attach tons
	  of info to the Contact header... let's increase our variable
	  sizes. (issue #9535 reported by jeffg) ........

2007-04-13 17:15 +0000 [r61647]  Russell Bryant <russell@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 61645 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61645 | russell | 2007-04-13 12:10:19 -0500 (Fri, 13 Apr 2007) |
	  3 lines Eliminate a compiler warning with ODBC_STORAGE enabled so
	  that it will build under dev-mode. ........

2007-04-13 17:11 +0000 [r61646]  Steve Murphy <murf@digium.com>

	* /, channels/chan_oss.c: Merged revisions 61644 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1
	  line A fix for chan_oss that resulted from the CDR changes; it
	  helps to use the right info. ........

2007-04-13 16:35 +0000 [r61618-61642]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61641 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2
	  lines Don't assume the callid of a dialog will be set, as in some
	  circumstances it may not. (issue #9534 reported by tecnoxarxa)
	  ........

	* channels/chan_sip.c: Don't treat a host lookup as failed if
	  sipregs is not in use when doing a realtime lookup. (issue #9255
	  reported by sergee)

2007-04-11 22:19 +0000 [r61575-61599]  Dwayne M. Hubbard <dhubbard@digium.com>

	* doc/asterisk-conf.tex: clarified 'minmemfree' description in
	  doc/asterisk-conf.tex

	* main/asterisk.c, doc/asterisk-conf.tex: fixed the '-e' command
	  line option for minmemfree. updated doc/asterisk-conf.tex

	* main/pbx.c, include/asterisk/options.h, main/asterisk.c: changed
	  #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO)

	* main/pbx.c, include/asterisk/options.h, main/asterisk.c: added
	  HAVE_SYSINFO preprocessor directives for portability and general
	  happiness

2007-04-11 20:21 +0000 [r61557]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Add a
	  configure script check for sysinfo support.

2007-04-11 19:11 +0000 [r61539]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/pbx.c, include/asterisk/options.h, main/asterisk.c: added
	  option_minmemfree for use in asterisk.conf to specify the amount
	  of minimum free memory prior to accepting calls. added CLI 'core
	  show sysinfo' to display system information

2007-04-11 17:07 +0000 [r61522]  Joshua Colp <jcolp@digium.com>

	* main/logger.c: Output verbose messages to the normal logger as
	  well. (issue #9476 reported by gdalgliesh)

2007-04-11 16:06 +0000 [r61478]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61477 | russell | 2007-04-11 11:05:29 -0500
	  (Wed, 11 Apr 2007) | 13 lines Merged revisions 61476 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11
	  Apr 2007) | 5 lines If someone sets the "useragent" option in
	  sip.conf to be empty, then don't add the User-Agent header at
	  all. It is an optional header, anyway. Also, the bug report says
	  that some of Japan's SIP providers don't allow it for some weird
	  reason. (issue #9488, reported by makoto, fixed by me) ........
	  ................

2007-04-11 15:48 +0000 [r61460]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions
	  61342,61372-61373,61443 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2
	  lines AOCD's are now exported to asterisk channel variables.
	  ........ r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr
	  2007) | 2 lines Ignore facility messages in case we don't have a
	  corresponding channel object. ........ r61373 | nadi | 2007-04-11
	  15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines Export AOCD variables
	  on misdn_hangup. ........ r61443 | nadi | 2007-04-11 17:39:14
	  +0200 (Mi, 11 Apr 2007) | 2 lines Don't export AOCD variables on
	  misdn_hangup anymore, this was mainly a fix for trunk.. ........

2007-04-11 15:25 +0000 [r61379-61429]  Russell Bryant <russell@digium.com>

	* funcs/func_devstate.c: Add a minor loop optimization to the
	  custom device state callback. Once the correct device is found,
	  it should just break out of the loop ...

	* /, channels/chan_sip.c: Merged revisions 61427 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61427 | russell | 2007-04-11 10:09:39 -0500
	  (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11
	  Apr 2007) | 6 lines Fix a bug with switching between host=dynamic
	  and using specific hosts for peers. The code would only reset the
	  peer's address when it is dynamic if it was a new peer structure.
	  Now, it will also reset the address if it was already in the peer
	  list, but before the reload, it was not dynamic. (issue #9515,
	  reported by caio1982, fixed by me) ........ ................

	* /, main/http.c: Merged revisions 61407 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61407 | russell | 2007-04-11 09:48:01 -0500 (Wed, 11 Apr 2007) |
	  4 lines Add "svgz" to the mimetypes table. (issue #9510, bkruse)
	  In passing, constify the elements of the mimetypes table.
	  ........

	* /, channels/chan_sip.c: Merged revisions 61377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61377 | russell | 2007-04-11 09:04:44 -0500
	  (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11
	  Apr 2007) | 5 lines Remove the attempt at reporting configuration
	  errors in sip.conf. This can cause a bunch of improper messages
	  when using realtime. I give up. As oej tried to convince me when
	  I put this in, there is just no easy way to do it. (inspired by a
	  message on the -dev list) ........ ................

2007-04-11 14:09 +0000 [r61378]  Steve Murphy <murf@digium.com>

	* apps/app_voicemail.c: via 8119, a patch to allow voicemail data
	  to be stored in RealTime.

2007-04-11 14:01 +0000 [r61375]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove duplicate prototype declaration.
	  (issue #9517 reported by junky)

2007-04-11 13:41 +0000 [r61374]  Steve Murphy <murf@digium.com>

	* include/asterisk/config.h, main/config.c: via 8118, a RealTime
	  upgrade to make RT a complete storage abstraction. The
	  store/destroy mechanisms needed these missing peices.

2007-04-10 23:55 +0000 [r61324]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, main/manager.c, configs/manager.conf.sample,
	  include/asterisk/manager.h: Issue 6082 - New DTMF event for
	  manager

2007-04-10 22:02 +0000 [r61303]  Doug Bailey <dbailey@digium.com>

	* channels/chan_zap.c: Added zapata.conf parameter "cid_rxgain" to
	  allow the user to adjust the gain bump used during CID
	  acquisition.

2007-04-10 20:50 +0000 [r61222-61283]  Russell Bryant <russell@digium.com>

	* CHANGES: Note the bridge manager action and application in the
	  CHANGES file.

	* res/res_features.c: Merge changes from team/russell/issue_5841:
	  This patch adds a "Bridge" Manager action, as well as a "Bridge"
	  dialplan application. The manager action will allow you to steal
	  two active channels in the system and bridge them together. Then,
	  the one that did not hang up will continue in the dialplan. Using
	  the application will bridge the calling channel to an arbitrary
	  channel in the system. Whichever channel does not hang up here
	  will continue in the dialplan, as well. This patch has been
	  touched by a bunch of people over the course of a couple years.
	  Please forgive me if I have missed your name in the history of
	  things. The most recent patch came from issue #5841, but there is
	  also a reference to an earlier version of this patch from issue
	  #4297. The people involved in writing and/or reviewing the code
	  include at least: twisted, mflorrel, heath1444, davetroy,
	  tim_ringenbach, moy, tmancill, serge-v, and me. There are also
	  positive test reports from many people.

	* main/dial.c, include/asterisk/dial.h: Add an option to the dial
	  API for playing music instead of ringing to the caller. I started
	  this for use with SLA but ended up deciding not to use it.
	  However, there is no reason not to put this part in, anyway.

2007-04-10 16:07 +0000 [r61221]  Steve Murphy <murf@digium.com>

	* channels/chan_jingle.c: updated ast_channel_alloc() call to
	  include the 4 extra args everyone got. Not much info there, as
	  the config file evidently does not allow amaflags, or accountcode
	  settings; and the pvt's exten doesn't sound like what we need in
	  the cdr, either.

2007-04-10 12:47 +0000 [r61184]  Nadi Sarrar <ns@beronet.com>

	* /, channels/misdn_config.c: Merged revisions 61183 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di,
	  10 Apr 2007) | 10 lines Merged revisions 61170 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2
	  lines msns config parameter defaults to '*' ........
	  ................

2007-04-10 05:41 +0000 [r61152]  Steve Murphy <murf@digium.com>

	* main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc,
	  channels/chan_zap.c, /, channels/chan_sip.c, res/res_features.c,
	  channels/chan_agent.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
	  main/channel.c, main/cdr.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
	  apps/app_cdr.c, apps/app_voicemail.c: Merged revisions 60989 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1
	  line This is a big improvement over the current CDR fixes. It may
	  still need refinement, but this won't have as many folks
	  bothered. This also adds the mods from 1.4/r.61136; ........

2007-04-09 22:49 +0000 [r61116]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: Remove unused instances of unnamed enums.

2007-04-09 20:01 +0000 [r61073]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 61072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon,
	  09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
	  lines - Don't send ActionID before Response: header. - Don't use
	  a blank in an AMI header ........ ................

2007-04-09 19:57 +0000 [r61065-61071]  Kevin P. Fleming <kpfleming@digium.com>

	* main/minimime/mm_envelope.c, /: Merged revisions 61070 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61070 | kpfleming | 2007-04-09 14:55:14 -0500 (Mon, 09 Apr 2007)
	  | 2 lines fix up some warnings found using --enable-dev-mode
	  ........

	* /, main/minimime/tests/CVS (removed), main/minimime/Doxyfile
	  (removed), main/minimime/tests/messages/CVS (removed): Merged
	  revisions 61062 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61062 | kpfleming | 2007-04-09 14:49:09 -0500 (Mon, 09 Apr 2007)
	  | 2 lines remove some more stuff we don't need ........

2007-04-09 19:06 +0000 [r61023]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 61022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r61022 | qwell | 2007-04-09 14:05:48 -0500 (Mon, 09 Apr 2007) | 4
	  lines Use the appropriate interface name with COMPLETECALLER.
	  Issue 9395. ........

2007-04-09 19:05 +0000 [r60985-61021]  Olle Johansson <oej@edvina.net>

	* main/manager.c: Add hint to ExtensionStatus AMI event in manager

	* channels/chan_sip.c, CHANGES, channels/chan_iax2.c: use
	  "ChannelType" in events to indicate which channel driver that
	  generates the event. This replaces "ChannelDriver" and "Channel",
	  previously used to indicate channel driver. ChannelType is more
	  in line with "core show channeltypes"

	* res/res_jabber.c: Fix JabberEvents

	* /, res/res_jabber.c: Fix missing newline in JabberEvent

2007-04-09 17:23 +0000 [r60937]  Jason Parker <jparker@digium.com>

	* /, apps/app_directory.c: Merged revisions 60936 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60936 | qwell | 2007-04-09 12:22:59 -0500 (Mon,
	  09 Apr 2007) | 13 lines Merged revisions 60935 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
	  lines Allow matching on names shorter than 3 chars. This also
	  fixes the case where somebody wants to match on less then 3
	  chars. Issue 9071 ........ ................

2007-04-09 16:30 +0000 [r60917]  Dwayne M. Hubbard <dhubbard@digium.com>

	* UPGRADE.txt: updated UPGRADE.txt to include format_wav changes

2007-04-09 12:33 +0000 [r60898]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make RTP session ID and session version
	  generation random. (issue #9456 reported by tjardick)

2007-04-09 03:04 +0000 [r60848-60851]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk.h, /, main/asterisk.c: Merged revisions 60850
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500
	  (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08
	  Apr 2007) | 2 lines Don't check for error when lowering priority
	  (according to the manpage, it should never happen anyway). It
	  might could happen, though, if another thread messed with the
	  priority, so safeguard against that (reported via -dev list).
	  ........ ................

	* channels/chan_local.c, /: Merged revisions 60847 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60847 | tilghman | 2007-04-08 21:42:48 -0500
	  (Sun, 08 Apr 2007) | 10 lines Merged revisions 60846 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
	  Apr 2007) | 2 lines Bug 9505 - If the return value for
	  local_queue_frame is set, then p->lock is no longer valid.
	  ........ ................

2007-04-09 01:06 +0000 [r60763-60799]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 60798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60798 | file | 2007-04-08 21:03:14 -0400 (Sun,
	  08 Apr 2007) | 10 lines Merged revisions 60797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
	  lines When calling a device that then forwards us elsewhere... we
	  have to make our channels compatible if it is the only channel
	  being dialed. (issue #9445 reported by marcelbarbulescu) ........
	  ................

	* channels/chan_sip.c: Add counter for sip show registry CLI
	  command. (issue #9352 reported by junky)

	* /, apps/app_queue.c: Merged revisions 60762 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60762 | file | 2007-04-08 13:04:44 -0400 (Sun, 08 Apr 2007) | 2
	  lines Allow app_queue to use MONITOR_EXEC even if MONITOR_OPTIONS
	  is not set. (issue #9495 reported by cduffy) ........

2007-04-08 14:23 +0000 [r60662-60715]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 60713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60713 | tilghman | 2007-04-08 09:14:29 -0500
	  (Sun, 08 Apr 2007) | 10 lines Merged revisions 60711 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08
	  Apr 2007) | 2 lines Gosub called within a Macro resets the
	  arguments improperly and causes general weirdness. (Issue 8329)
	  ........ ................

	* /, formats/format_wav.c, main/http.c: Merged revisions 60712 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60712 | tilghman | 2007-04-08 09:12:00 -0500 (Sun, 08 Apr 2007)
	  | 2 lines Fix --enable-dev-mode ........

	* /, main/file.c: Merged revisions 60661 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60661 | tilghman | 2007-04-07 20:40:47 -0500
	  (Sat, 07 Apr 2007) | 10 lines Merged revisions 60660 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07
	  Apr 2007) | 2 lines Bug 9486 - memory leak when opening a
	  filestream ........ ................

2007-04-06 22:29 +0000 [r60641]  Dwayne M. Hubbard <dhubbard@digium.com>

	* formats/format_wav.c: removed GAIN preprocessor definition,
	  removed needsgain from struct wav_desc, removed unnecessary gain
	  code from wav_read() and wav_write()

2007-04-06 21:43 +0000 [r60566-60623]  Russell Bryant <russell@digium.com>

	* main/minimime/Makefile: Filter out -Wundef so that the
	  automatically generated C files will compile cleanly

	* main/minimime/mytest_files (removed), main/minimime/sys/CVS
	  (removed), main/minimime/.cvsignore (removed),
	  main/minimime/mm-docs (removed), main/minimime/test (removed):
	  Remove a bunch of files that weren't supposed to get added.

	* main/minimime/mm-docs/html/mm__envelope_8c.html,
	  main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
	  main/minimime/mm-docs/html/mm__context_8c.html,
	  main/minimime/sys, main/minimime/tests/Makefile,
	  main/minimime/tests/CVS/Root, main/minimime/sys/CVS/Entries,
	  main/minimime/mm-docs/latex/mm__mimeutil_8c.tex, configure,
	  main/strcompat.c, main/http.c, main/minimime/mm_error.c,
	  main/minimime/mm-docs/html/globals_func.html,
	  main/minimime/mm-docs/html/group__mimeutil.html,
	  main/minimime/mm-docs/latex/doxygen.sty,
	  main/minimime/mm_param.c, main/minimime/test/CVS, configure.ac,
	  main/minimime/.cvsignore, main/minimime/mm_init.c,
	  main/minimime/mm-docs/html/mm__queue_8h-source.html,
	  main/minimime/mm-docs/html/mm__error_8c.html,
	  main/minimime/mm-docs/html/tabs.css, main/minimime/mm_envelope.c,
	  main/minimime/mimeparser.h, main/minimime/mimeparser.l,
	  main/minimime/mm_context.c,
	  main/minimime/mm-docs/html/group__mimepart.html,
	  main/minimime/mm-docs/latex/group__envelope.tex,
	  main/minimime/tests/messages/CVS,
	  main/minimime/mm-docs/html/mm__contenttype_8c.html,
	  main/minimime/mm-docs/html/pages.html,
	  main/minimime/mm-docs/html/group__error.html,
	  main/minimime/mm-docs/latex/group__context.tex,
	  main/minimime/mimeparser.y, Makefile.moddir_rules,
	  main/minimime/sys/mm_queue.h,
	  main/minimime/mm-docs/html/bug.html,
	  main/minimime/mm-docs/html/mimeparser_8tab_8h-source.html,
	  main/minimime/tests/messages/CVS/Root,
	  main/minimime/mm_mimepart.c,
	  main/minimime/mm-docs/latex/Makefile,
	  main/minimime/mm_internal.h, main/minimime/tests/CVS,
	  main/minimime/mm-docs/latex/mm__param_8c.tex,
	  main/minimime/tests/parse.c, main/minimime/mm_base64.c,
	  main/minimime/mm.h, main/minimime/mm_header.c,
	  main/minimime/mm-docs/latex/mm__parse_8c.tex,
	  main/minimime/mm-docs/html/mimeparser_8h-source.html,
	  main/minimime/mm-docs/html/files.html,
	  main/minimime/mm-docs/latex/mm__contenttype_8c.tex,
	  main/minimime/mm-docs/html/mm__mem_8h-source.html,
	  main/minimime/mm_codecs.c,
	  main/minimime/mm-docs/latex/mm__mimepart_8c.tex,
	  main/minimime/mytest_files/mytest.c,
	  main/minimime/mm-docs/html/mm__mimeutil_8c.html,
	  main/minimime/mm-docs/latex/files.tex,
	  main/minimime/test/CVS/Entries,
	  main/minimime/mm-docs/latex/modules.tex,
	  main/minimime/tests/messages/CVS/Repository,
	  configs/http.conf.sample, main/minimime/mm_contenttype.c,
	  main/minimime/tests/messages/test1.txt,
	  main/minimime/mm-docs/html/mm__param_8c.html,
	  main/minimime/tests/messages/test3.txt,
	  main/minimime/tests/messages/test5.txt,
	  main/minimime/tests/messages/test7.txt,
	  main/minimime/mm-docs/html/group__contenttype.html,
	  main/minimime/mm-docs,
	  main/minimime/mytest_files/ast_postdata3.gz, main/minimime
	  (added), main/minimime/Make.conf,
	  main/minimime/mm-docs/latex/group__contenttype.tex,
	  main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
	  main/minimime/mm-docs/html/mm__util_8c.html,
	  main/minimime/mm-docs/html/doxygen.css, /,
	  main/minimime/mm-docs/html/mm__internal_8h.html,
	  main/minimime/tests/messages/CVS/Entries, main/minimime/Doxyfile,
	  main/minimime/minimime.c, main/minimime/mimeparser.yy.c,
	  main/minimime/tests/CVS/Entries.Log, main/minimime/test.sh,
	  include/asterisk/compat.h, main/minimime/test/CVS/Repository,
	  main/minimime/mm_mimeutil.c, main/minimime/tests,
	  main/minimime/mm-docs/latex/group__mimepart.tex,
	  main/minimime/tests/CVS/Entries, main/Makefile,
	  main/minimime/mm-docs/latex/mm__envelope_8c.tex,
	  main/minimime/mm-docs/latex/mm__util_8c.tex,
	  main/minimime/mm-docs/latex/pages.tex,
	  main/minimime/mm-docs/latex/group__mimeutil.tex,
	  main/minimime/mm-docs/latex,
	  main/minimime/mm-docs/html/mm_8h-source.html,
	  main/minimime/Makefile,
	  main/minimime/mm-docs/latex/mm__internal_8h.tex,
	  main/minimime/mm-docs/refman.pdf, include/asterisk/manager.h,
	  main/minimime/mm-docs/latex/mm__context_8c.tex,
	  main/minimime/mm-docs/latex/group__param.tex,
	  main/minimime/mm-docs/latex/group__codecs.tex,
	  main/minimime/tests/create.c, main/minimime/mm_util.c,
	  main/minimime/mm-docs/latex/bug.tex,
	  main/minimime/mimeparser.tab.c, main/minimime/mm_util.h,
	  main/minimime/mytest_files/ast_postdata,
	  main/minimime/mm-docs/html/group__envelope.html,
	  main/minimime/mm-docs/html/group__util.html,
	  main/minimime/mimeparser.tab.h,
	  main/minimime/mm-docs/html/mm__parse_8c.html,
	  main/minimime/mm-docs/html,
	  main/minimime/mm-docs/latex/group__util.tex,
	  main/minimime/mm-docs/html/group__context.html,
	  main/minimime/mm-docs/html/mm__internal_8h-source.html,
	  main/minimime/mytest_files,
	  main/minimime/mm-docs/html/mm__util_8h-source.html,
	  main/minimime/sys/CVS,
	  main/minimime/mm-docs/html/group__codecs.html, main/manager.c,
	  main/minimime/sys/CVS/Repository,
	  main/minimime/mm-docs/html/globals.html,
	  main/minimime/mm-docs/html/mm__mimepart_8c.html,
	  main/minimime/tests/CVS/Repository,
	  main/minimime/mm-docs/html/index.html,
	  main/minimime/mm-docs/html/modules.html, main/minimime/test,
	  main/minimime/mytest_files/ast_postdata2,
	  main/minimime/mm-docs/latex/group__error.tex,
	  main/minimime/mm-docs/html/mm__header_8c.html,
	  main/minimime/strlcpy.c,
	  main/minimime/mm-docs/html/group__param.html,
	  main/minimime/mm-docs/latex/refman.tex, main/minimime/mm_parse.c,
	  main/minimime/mm-docs/latex/mm__header_8c.tex,
	  main/minimime/mm-docs/latex/mm__error_8c.tex,
	  main/minimime/mm_mem.c,
	  main/minimime/mm-docs/html/mm__codecs_8c.html,
	  main/minimime/tests/messages/test2.txt,
	  main/minimime/tests/messages/test4.txt,
	  main/minimime/sys/CVS/Root,
	  main/minimime/tests/messages/test6.txt,
	  main/minimime/test/CVS/Root, main/minimime/strlcat.c,
	  main/minimime/mm_mem.h,
	  main/minimime/mm-docs/latex/mm__codecs_8c.tex: Merged revisions
	  60603 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) |
	  13 lines To be able to achieve the things that we would like to
	  achieve with the Asterisk GUI project, we need a fully functional
	  HTTP interface with access to the Asterisk manager interface. One
	  of the things that was intended to be a part of this system, but
	  was never actually implemented, was the ability for the GUI to be
	  able to upload files to Asterisk. So, this commit adds this in
	  the most minimally invasive way that we could come up with. A lot
	  of work on minimime was done by Steve Murphy. He fixed a lot of
	  bugs in the parser, and updated it to be thread-safe. The ability
	  to check permissions of active manager sessions was added by
	  Dwayne Hubbard. Then, hacking this all together and do doing the
	  modifications necessary to the HTTP interface was done by me.
	  ........

	* /, apps/app_meetme.c: Merged revisions 60565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60565 | russell | 2007-04-06 14:50:52 -0500 (Fri, 06 Apr 2007) |
	  3 lines When a station picks up a trunk that was on hold, make
	  the hints reflect that nobody has the trunk on hold anymore.
	  ........

2007-04-06 19:26 +0000 [r60531]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Use the same parameter to the two "Registry"
	  AMI events - ChannelDriver

2007-04-06 18:59 +0000 [r60522]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 60521 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) |
	  16 lines Fix a few problems with SLA. (issue #9459, reported by
	  francesco_r, fixed by me) * The original behavior was that if one
	  station put a call on hold, another one picked it up, and then
	  hung up, the code would still consider the call on hold by the
	  first station, so the trunk would not be hung up. However, to
	  better comply with what most people seem to expect it to behave,
	  it will now hang up the trunk. * Fix a problem with "barge=no".
	  This was only intended to prevent people from joining calls that
	  are in progress. However, it also prevented other people from
	  picking up a call that was on hold. This has been fixed. * When
	  there are no active stations on a trunk and it is on hold, the
	  code now indicates the HOLD and UNHOLD conditions to the trunk
	  channel. This allows music on hold to be played to the trunk when
	  it is on hold. ........

2007-04-06 18:26 +0000 [r60486-60487]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 60485 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2
	  lines Make sure we check the faxdetect option before doing fax
	  processing ........

	* channels/chan_zap.c, /: Merged revisions 60459 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri,
	  06 Apr 2007) | 10 lines Merged revisions 60456 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
	  lines There should only be one code path for doing DTMF
	  conditionals on channels. This fixes it. ........
	  ................

2007-04-06 14:53 +0000 [r60400]  Kevin P. Fleming <kpfleming@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 60399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60399 | kpfleming | 2007-04-06 09:49:51 -0500
	  (Fri, 06 Apr 2007) | 10 lines Merged revisions 60398 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06
	  Apr 2007) | 2 lines remove undocumented 'cardsmode' parameter and
	  stop searching for transcoders during reload() ........
	  ................

2007-04-06 01:29 +0000 [r60362-60363]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/speech.h, res/res_speech.c: Major res_speech
	  cleanup. It looks much better now!

	* /, include/asterisk/speech.h, res/res_speech.c,
	  apps/app_speech_utils.c: Merged revisions 60361 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2
	  lines Add support for returning different types of results (ie:
	  NBest). ........

2007-04-05 23:08 +0000 [r60326]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, formats/format_wav.c: Merged revisions 60325 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60325 | dhubbard | 2007-04-05 17:58:01 -0500 (Thu, 05 Apr 2007)
	  | 1 line modified default GAIN for issue 5823, thanks jrwalliker
	  ........

2007-04-05 22:40 +0000 [r60324]  Steve Murphy <murf@digium.com>

	* configs/cdr_custom.conf.sample, /, configs/cdr.conf.sample:
	  Merged revisions 60323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1
	  line Added some clarification to the example configs for CDRs, on
	  how to select a backend. Also, made cdr-csv the default if you
	  'make samples', and no other changes. ........

2007-04-05 16:11 +0000 [r60269]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 60268 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60268 | qwell | 2007-04-05 11:10:48 -0500 (Thu,
	  05 Apr 2007) | 13 lines Merged revisions 60267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
	  lines Just because we can't find the voicemail configuration
	  file, doesn't mean that the module failed to load. The user could
	  be using realtime. Issue #9473 ........ ................

2007-04-05 15:48 +0000 [r60266]  Russell Bryant <russell@digium.com>

	* /, main/http.c: Merged revisions 60265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60265 | russell | 2007-04-05 10:47:17 -0500 (Thu, 05 Apr 2007) |
	  2 lines Add the MIME type for gif by request from Pari ........

2007-04-05 12:57 +0000 [r60215]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 60214 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu,
	  05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
	  lines Only unlock our pvt and net locks if we are actually going
	  to try to lock the owner again. (issue #9472 reported by zoa)
	  ........ ................

2007-04-04 23:45 +0000 [r60193]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/callerid.c: ast_shrink_phone_number() must ignore
	  whitespace, otherwise my CIDCO callerid box gets LINE ERROR

2007-04-04 17:41 +0000 [r60011-60141]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 60137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60137 | russell | 2007-04-04 12:40:10 -0500
	  (Wed, 04 Apr 2007) | 14 lines Merged revisions 60134 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04
	  Apr 2007) | 6 lines It is valid to redirect channels via the
	  manager interface that are not in the UP state. Instead of
	  checking for that to prevent to ensure a dead channel doesn't get
	  redirected, just use the ast_check_hangup() API call. (issue
	  #9457, reported by Callmewind, patch by me) (related to issue
	  #8977) ........ ................

	* /, channels/chan_sip.c: Merged revisions 60112 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) |
	  3 lines Add a Content-Length of 0 to the response built by
	  transmit_response_with_unsupported(). (issue #9454, reported by
	  makoto, fixed by me) ........

	* /, channels/chan_sip.c: Merged revisions 60088 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r60088 | russell | 2007-04-04 11:39:04 -0500
	  (Wed, 04 Apr 2007) | 12 lines Merged revisions 60083 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04
	  Apr 2007) | 4 lines Fix the return value of
	  handle_common_options() so that it always properly indicates
	  whether it handled the option or not. (issue #9455, reported by
	  Netview, fixed by me) ........ ................

	* /, apps/app_meetme.c: Merged revisions 60069 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) |
	  4 lines Fix a problem where if a trunk was hung up while it was
	  on hold, all of the hints would reflect the line still on hold,
	  even though it should reflect that it is back to not in use.
	  (issue #9459, reported by francesco_r, fixed by me) ........

	* channels/chan_jingle.c, channels/chan_gtalk.c,
	  doc/rtp-packetization.txt: Add support for RTP packetization in
	  chan_jingle and chan_gtalk. (issue #9416, phsultan)

2007-04-03 19:43 +0000 [r59969]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 59963 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr
	  2007) | 2 lines Don't clash when a person both speaks and uses
	  DTMF. ........

2007-04-03 19:17 +0000 [r59854-59940]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59939 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59939 | russell | 2007-04-03 14:16:53 -0500
	  (Tue, 03 Apr 2007) | 12 lines Merged revisions 59938 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03
	  Apr 2007) | 4 lines Don't attempt to report configuration errors
	  in build_user(). oej pointed out that for a "friend" entry, this
	  won't work, because all user options are valid for peers, but not
	  the other way around. ........ ................

	* /, channels/chan_sip.c: Merged revisions 59936 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59936 | russell | 2007-04-03 13:55:57 -0500
	  (Tue, 03 Apr 2007) | 11 lines Merged revisions 59916 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03
	  Apr 2007) | 3 lines Make chan_sip report when it encounters an
	  unknown option. (issue #9440, reported by nightcrawler) ........
	  ................

	* channels/chan_sip.c: Remove a duplicate function prototype.
	  (issue #9444, junky)

	* /, main/app.c: Merged revisions 59887 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59887 | russell | 2007-04-03 13:01:49 -0500
	  (Tue, 03 Apr 2007) | 13 lines Merged revisions 59886 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03
	  Apr 2007) | 5 lines When doing a built-in blind or attended
	  transfer, restore the ability to use '#' to terminate the number
	  and immediately do the transfer instead of having to dial the
	  number and just wait for the feature digit timeout. (issue #8366,
	  xueliangliang) ........ ................

	* Makefile, /: Merged revisions 59853 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59853 | russell | 2007-04-03 11:03:35 -0500 (Tue, 03 Apr 2007) |
	  1 line Ensure that menuselect gets executed in dependency check
	  mode every time you run make. ........

2007-04-03 11:15 +0000 [r59805]  Nadi Sarrar <ns@beronet.com>

	* /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c:
	  Merged revisions 59804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di,
	  03 Apr 2007) | 15 lines Merged revisions 59788,59803 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr
	  2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a
	  port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58
	  +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the
	  4th. ........ ................

2007-04-02 19:01 +0000 [r59725]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 59724 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59724 | file | 2007-04-02 14:58:24 -0400 (Mon,
	  02 Apr 2007) | 10 lines Merged revisions 59723 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
	  lines Increase the maximum size for a string of mailboxes to
	  1024. (issue #9270 reported by rtucker) ........ ................

2007-04-02 17:40 +0000 [r59693]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: This hashing code is still causing some
	  random crashes on my system, and probably others, too. I don't
	  really have time to work on it at the moment, so I am just going
	  to revert it for now.

2007-04-02 17:38 +0000 [r59692]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 59688 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59688 | murf | 2007-04-02 11:31:32 -0600 (Mon, 02 Apr 2007) | 1
	  line continue in for-loop should go to the incrementer, not the
	  test. As per 9435, thanks to marcelbarbulescu ........

2007-04-02 16:08 +0000 [r59655]  Russell Bryant <russell@digium.com>

	* /, main/netsock.c: Merged revisions 59654 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59654 | russell | 2007-04-02 10:39:07 -0500
	  (Mon, 02 Apr 2007) | 14 lines Merged revisions 59608 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01
	  Apr 2007) | 6 lines Add the SO_REUSEADDR flag to sockets handled
	  by netsock. This is needed by the patch that went in for issue
	  7874. chan_iax2 needs to be able to create socket that is
	  lisetning on INADDR_ANY, but also be able to bind sockets to
	  specific addresses. (Thanks to Stevenson on the asterisk-dev
	  mailing list for explaining why this flag was needed.) ........
	  ................

2007-03-30 22:54 +0000 [r59574]  Jason Parker <jparker@digium.com>

	* /, configure, main/Makefile, acinclude.m4: Merged revisions 59573
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2
	  lines Add linux-uclibc host arch..."thingy". Sorry, I don't know
	  what it's called... ........

2007-03-30 20:54 +0000 [r59555]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Update to support multiple CIC groups and
	  DPCs per linkset.

2007-03-30 17:57 +0000 [r59453-59523]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c,
	  include/asterisk/cdr.h: Merged revisions 59522 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1
	  line several changes via kpflemings review ........

	* main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c,
	  include/asterisk/cdr.h: Merged revisions 59486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1
	  line These mods fix CDR issues from 8221, 8593, 8680, 8743, and
	  perhaps others. Mainly with CDRs generated from transfer
	  situations. ........

	* /, configs/extensions.conf.sample: Merged revisions 59452 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1
	  line A small clarification to keep bugs from being filed, and
	  confusion from rising, if clearglobalvars is set, and globals are
	  set in the AEL file. (9419) ........

2007-03-29 23:27 +0000 [r59364-59433]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Reduce the ridiculous number of variables
	  used in the load_config() function by just having one that can be
	  re-used. There is no functional change here (that is intentional,
	  anyway!).

	* CHANGES, apps/app_voicemail.c: Add the ability for the "voicemail
	  show users" CLI command to show users configured in realtime.

	* channels/chan_iax2.c: Fix an issue with hashing iax2 pvt
	  structures that caused random crashes on systems under heavy load
	  such as IAXtel. (possibly related to issue #9403)

	* /, res/res_jabber.c: Merged revisions 59363 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) |
	  6 lines When building a response to a subscription, the "from"
	  must be the full Jabber ID. This fixes some problems where jabber
	  users are not able to add their Asterisk account to their user
	  list, since they are unable to get Asterisk to approve their
	  subscription. (issue #8210, reported by caspy, and verified by
	  bradtem) ........

2007-03-29 17:42 +0000 [r59362]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 59361 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59361 | file | 2007-03-29 13:38:55 -0400 (Thu,
	  29 Mar 2007) | 10 lines Merged revisions 59360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
	  lines Keep a global array of variables indicating whether certain
	  conference rooms are in use. This ensures that two people going
	  into a new dynamic conference when the 'e' option is set don't go
	  into the same conference room. (issue #8835 reported by eliel)
	  ........ ................

2007-03-29 17:20 +0000 [r59305-59359]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 59358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59358 | russell | 2007-03-29 12:17:41 -0500
	  (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29
	  Mar 2007) | 5 lines If an error occurs when reading from an RTP
	  socket, and the error code does not indicate that we should try
	  again, then return NULL instead of a "null frame". This will
	  prevent Asterisk from trying over and over again, and eventually
	  causing the system to crash. (issue #8285, john) ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 59341 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) |
	  8 lines When the IAX2 read callback gets called, return NULL
	  instead of a "null frame". This will cause Asterisk to hangup the
	  call instead of keep trying whatever it was doing. Under normal
	  conditions, this function would *never* be called. However, the
	  author of this patch says an error will occur that will cause it
	  to get called every 100 thousand calls or so. When this does
	  happen, it puts the channel in a loop that eventually brings down
	  the system. So, hangup up the call is certainly a better
	  alternative. (issue #8286, john) ........

	* Makefile, /: Merged revisions 59304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59304 | russell | 2007-03-29 11:25:41 -0500 (Thu, 29 Mar 2007) |
	  2 lines Export the GTK2 library and include information to sub
	  Makefiles. ........

2007-03-29 16:08 +0000 [r59303]  Tilghman Lesher <tlesher@digium.com>

	* /, cdr/cdr_odbc.c: Merged revisions 59302 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59302 | tilghman | 2007-03-29 11:07:05 -0500
	  (Thu, 29 Mar 2007) | 11 lines Merged revisions 59301 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29
	  Mar 2007) | 3 lines Issue 9415 - No point to getting a diagnostic
	  field if we aren't doing anything with the information. (Plus, it
	  tends to crash the Postgres ODBC driver.) ........
	  ................

2007-03-28 03:40 +0000 [r59290]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 59289 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59289 | tilghman | 2007-03-27 22:38:09 -0500 (Tue, 27 Mar 2007)
	  | 2 lines Another crash that I thought we had fixed already -
	  Issue 9396 ........

2007-03-28 00:09 +0000 [r59286]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c: added filtering options to 'zap show
	  channels' to implement functionality described in issue 6520

2007-03-27 23:38 +0000 [r59282-59285]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 59284 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59284 | tilghman | 2007-03-27 18:37:31 -0500
	  (Tue, 27 Mar 2007) | 10 lines Merged revisions 59283 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27
	  Mar 2007) | 2 lines Oops ........ ................

	* /, apps/app_voicemail.c: Merged revisions 59281 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59281 | tilghman | 2007-03-27 18:32:46 -0500
	  (Tue, 27 Mar 2007) | 10 lines Merged revisions 59280 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27
	  Mar 2007) | 2 lines Fix a few remaining bad mmap(2) return values
	  ........ ................

2007-03-27 23:22 +0000 [r59274-59279]  Russell Bryant <russell@digium.com>

	* /, apps/app_directory.c: Merged revisions 59278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59278 | russell | 2007-03-27 18:20:22 -0500
	  (Tue, 27 Mar 2007) | 11 lines Merged revisions 59277 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27
	  Mar 2007) | 3 lines Fix the check of the return value from
	  mmap(). Thanks to Corydon for catching this one. ........
	  ................

	* /, apps/app_directory.c: Merged revisions 59275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59275 | russell | 2007-03-27 18:16:27 -0500 (Tue, 27 Mar 2007) |
	  3 lines Fix app_directory to actually compile with ODBC_STORAGE,
	  and update the code to the latest res_odbc API. ........

	* /, apps/Makefile: Merged revisions 59273 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59273 | russell | 2007-03-27 18:02:12 -0500 (Tue, 27 Mar 2007) |
	  4 lines Fix app_directory when ODBC_STORAGE is being used. The
	  Makefile did not properly ensure that this information got copied
	  from what was selected for app_voicemail. (issue #9224) ........

2007-03-27 20:11 +0000 [r59272]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Use better english. Renegotiate! Repeat
	  after me: renegotiate.

2007-03-27 18:21 +0000 [r59264]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 59261 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59261 | murf | 2007-03-27 12:16:32 -0600 (Tue, 27 Mar 2007) | 1
	  line via 9373 (duplicate context in AEL crashes asterisk),
	  kpfleming pointed on asterisk-dev, that DECLINE in this case the
	  proper thing to do. This change now has it doing the proper
	  thing. ........

2007-03-27 18:18 +0000 [r59257-59263]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) |
	  3 lines Fix the check that ensures that the CHANNEL function's
	  first argument is "rtpqos". Thanks, Corydon. :) ........

	* /, channels/chan_iax2.c: Merged revisions 59259 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59259 | russell | 2007-03-27 13:05:46 -0500
	  (Tue, 27 Mar 2007) | 12 lines Merged revisions 59258 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27
	  Mar 2007) | 4 lines Fix the use of the "sourceaddress" option
	  when "bindaddr" is set to 0.0.0.0 instead of having each
	  interface explicitly listed. (issue #7874, patch by stevens)
	  ........ ................

	* /, channels/chan_sip.c, funcs/func_channel.c: Merged revisions
	  59256 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) |
	  4 lines Convert the RTPQOS function to just be additional
	  parameter of the CHANNEL function. This way, it will be possible
	  for other RTP based channel drivers to expose this information in
	  the future. ........

2007-03-27 14:09 +0000 [r59233-59253]  Steve Murphy <murf@digium.com>

	* include/asterisk/config.h: Enhancement via 8118: Realtime API
	  extension: add methods store_func and destroy_func, to make
	  Realtime a complete database abstraction

	* pbx/ael/ael-test/ael-test18/extensions.ael (added),
	  pbx/ael/ael-test/ael-test18 (added),
	  pbx/ael/ael-test/ref.ael-test18 (added): added the no. 18
	  regression test

	* pbx/ael/ael-test/ael-test19/extensions.ael (added),
	  pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ael-test19
	  (added), pbx/ael/ael-test/ref.ael-test7,
	  pbx/ael/ael-test/ref.ael-test19 (added),
	  pbx/ael/ael-test/ref.ael-vtest13: updated the regressions with
	  regards to 9373, the crash on double contexts, and brought other
	  regressions up to date

	* /, pbx/pbx_ael.c: Merged revisions 59228 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59228 | murf | 2007-03-26 15:41:32 -0600 (Mon, 26 Mar 2007) | 1
	  line fix for 9373 (duplicate context in AEL crashes asterisk). I
	  turned a duplicate context from a WARNING to an ERROR. Now you
	  get a module load failure, and asterisk just exits. That's better
	  than a crash, right\? ........

2007-03-26 21:46 +0000 [r59229-59231]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007)
	  | 2 lines Change this to a single dp function to make oej happy.
	  ........

2007-03-26 20:27 +0000 [r59226]  Steve Murphy <murf@digium.com>

	* /, main/config.c: Merged revisions 59225 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59225 | murf | 2007-03-26 14:06:12 -0600 (Mon, 26 Mar 2007) | 1
	  line Fix for 9257; by eliminating the globals in main/config.c,
	  we make it thread-safe, which is a minimum requirement. ........

2007-03-26 19:35 +0000 [r59224]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 59223 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar
	  2007) | 2 lines Add ability to specify no timeout. This means as
	  soon as the prompt is done playing it moves on to the next
	  priority. ........

2007-03-26 18:34 +0000 [r59216-59218]  Russell Bryant <russell@digium.com>

	* /: Merged revisions 59217 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59217 | russell | 2007-03-26 13:33:50 -0500 (Mon, 26 Mar 2007) |
	  4 lines Somehow the code for building the email for voicemail got
	  out of sync. This change makes a few tweaks to get 1.4 in sync
	  with trunk. (issue #9301) ........

	* /, apps/app_meetme.c: Merged revisions 59215 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59215 | russell | 2007-03-26 13:28:29 -0500 (Mon, 26 Mar 2007) |
	  3 lines Fix some codec negotiation problems when CallerID support
	  is not enabled in SLA. (issue #9308, reported by twilson)
	  ........

2007-03-26 18:14 +0000 [r59214]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 59213 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar
	  2007) | 2 lines Make SpeechBackground obey the digit timeout
	  value. ........

2007-03-26 17:57 +0000 [r59211]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Merged revisions 59209 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) |
	  1 line Rename the new dialplan functions to match the variable
	  name ........

2007-03-26 17:56 +0000 [r59210]  Steve Murphy <murf@digium.com>

	* /, main/ast_expr2f.c, pbx/ael/ael.flex, main/ast_expr2.fl: Merged
	  revisions 59206 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59206 | murf | 2007-03-26 11:38:29 -0600 (Mon, 26 Mar 2007) | 1
	  line A fix for the flex input files, DONT_COMPILE, and
	  STANDALONE_AEL ........

2007-03-26 17:51 +0000 [r59208]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
	  Merged revisions 59207 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) |
	  7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully
	  functional in some because they get set in sip_hangup. So, there
	  are common situations where the variables will not be available
	  in the dialplan at all. So, this patch provides an alternate
	  method for getting to this information by introducing AUDIORTPQOS
	  and VIDEORTPQOS dialplan functions. (issue #9370, patch by
	  Corydon76, with some testing by blitzrage) ........

2007-03-26 16:48 +0000 [r59204-59205]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Fix bug in which parameter type we are
	  passing. This shouldn't be a problem since both types are the
	  same underneath.

	* channels/chan_zap.c: Small API related SS7 updates.

2007-03-26 15:59 +0000 [r59203]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, configure,
	  include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, configure.ac,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
	  revisions 59202 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4
	  lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo
	  cancellation modules, make chan_misdn use it. * add a check for
	  linux/mISDNdsp.h to configure.ac and update the autogenerated
	  files: 'configure', 'autoconfig.h.in' (the 'configure' script was
	  not in sync with the latest configure.ac, so the diff is a bit
	  bigger than expected). ........

2007-03-26 15:20 +0000 [r59201]  Joshua Colp <jcolp@digium.com>

	* /, pbx/ael/ael_lex.c: Merged revisions 59200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59200 | file | 2007-03-26 11:16:29 -0400 (Mon, 26 Mar 2007) | 2
	  lines Have ast_copy_string magically appear in the aelparse
	  binary! DONT_OPTIMIZE should now work once again. ........

2007-03-24 01:42 +0000 [r59191-59196]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59195 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59195 | file | 2007-03-23 21:39:44 -0400 (Fri,
	  23 Mar 2007) | 10 lines Merged revisions 59194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
	  lines Only try to handle a response if it has a response code.
	  (ASA-2007-011) ........ ................

	* doc/modules.txt: Update modules.txt to new loader. (issue #9358
	  reported by eliel)

2007-03-23 16:17 +0000 [r59190]  Steve Murphy <murf@digium.com>

	* /, apps/app_macro.c: Merged revisions 59188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59188 | murf | 2007-03-23 10:09:01 -0600 (Fri,
	  23 Mar 2007) | 9 lines Merged revisions 59186 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
	  line Added a few words in the Macro doc strings about the
	  behavior of macros with hangups (et al.), as per 9337 ........
	  ................

2007-03-22 23:41 +0000 [r59181-59183]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007)
	  | 2 lines don't allow string input to overrun the buffer to hold
	  it (ASA-2007-010) ........

	* channels/chan_misdn.c, /: Merged revisions 59180 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22
	  Mar 2007) | 2 lines remove variables that are no longer used
	  (--enable-dev-mode is good, developers should be using it)
	  ........

2007-03-22 14:48 +0000 [r59146]  Steve Murphy <murf@digium.com>

	* utils/Makefile, /: Merged revisions 59145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59145 | murf | 2007-03-22 08:40:53 -0600 (Thu, 22 Mar 2007) | 1
	  line The stuff in utils was compiling with -O6 even if
	  DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
	  ........

2007-03-21 18:10 +0000 [r59080-59090]  Joshua Colp <jcolp@digium.com>

	* /, main/http.c: Merged revisions 59089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59089 | file | 2007-03-21 14:08:57 -0400 (Wed, 21 Mar 2007) | 2
	  lines Add svg mimetype for pari. ........

	* /, res/res_monitor.c: Merged revisions 59087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r59087 | file | 2007-03-21 14:04:58 -0400 (Wed,
	  21 Mar 2007) | 10 lines Merged revisions 59086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
	  lines Indicate the filename changed when it is changed. (issue
	  #9311 reported by jsmith) ........ ................

	* channels/chan_sip.c: Minor tweak. Only queue up an unhold control
	  frame if we are actually on hold. This would have shown itself
	  when a call was initially being setup and the SDP data was being
	  parsed in.

	* /, channels/chan_sip.c: Merged revisions 59081 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2
	  lines Until we can do media level parsing for sendrecv/etc just
	  use the first value found. This crept up when a phone was offered
	  audio+video and returned an inactive video stream. chan_sip
	  thought the phone said to put the person on hold but that was
	  totally wrong. (issue #9319 reported by benbrown) ........

	* main/db.c: Make the database show command spit out how many
	  results it got. (issue #9332 reported by junky)

2007-03-20 21:06 +0000 [r59079]  Tilghman Lesher <tlesher@digium.com>

	* /, main/logger.c: Merged revisions 59078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59078 | tilghman | 2007-03-20 16:04:52 -0500 (Tue, 20 Mar 2007)
	  | 2 lines Fix defines for inline stack backtraces (only used by
	  developers anyway) ........

2007-03-20 20:44 +0000 [r59077]  Joshua Colp <jcolp@digium.com>

	* /, channels/iax2-parser.c: Merged revisions 59076 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar
	  2007) | 2 lines Copy len variable as well, should fix remaining
	  IAX2 DTMF issues. ........

2007-03-20 18:18 +0000 [r59071-59073]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c, include/asterisk/ael_structs.h: The fix for the
	  AEL <<security hole>> (bug 9316) is here...

	* /: blocking 59070... it was just a repair, doesn't need to be
	  here

	* /: blocking 59069... will commit these changes with separate
	  patch

2007-03-19 22:32 +0000 [r59051]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: It is possible for mod to become invalid after we
	  unload it (if it's a dynamic module) so move it around a bit.

2007-03-19 22:31 +0000 [r59050]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 59049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59049 | tilghman | 2007-03-19 17:29:56 -0500 (Mon, 19 Mar 2007)
	  | 2 lines Oops, this should have been a %d all along ........

2007-03-19 15:43 +0000 [r59041]  Tilghman Lesher <tlesher@digium.com>

	* configs/sip_notify.conf.sample, /: Merged revisions 59040 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007)
	  | 2 lines Fix unescaped semicolon (reported via -dev list)
	  ........

2007-03-18 20:39 +0000 [r59038]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 59037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3
	  lines Issue #9313, Asterisk crash on SIP return code 0 (reported
	  by qwerty1979) (ASA-2007-011) ........

2007-03-18 16:59 +0000 [r59036]  BJ Weschke <bweschke@btwtech.com>

	* /, apps/app_followme.c: Merged revisions 59035 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r59035 | bweschke | 2007-03-18 12:36:44 -0400 (Sun, 18 Mar 2007)
	  | 3 lines Don't return a non-zero return code if the profile
	  doesn't exist, to match what the documentation says it already
	  does. (#9307 Reported by kkiely) ........

2007-03-16 16:14 +0000 [r58995]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_page.c: Merged revisions 58992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2
	  lines Wait for the async thread to exit when hanging up all of
	  the paged phones under all circumstances. (issue #9181 reported
	  by PhilSmith) ........

2007-03-16 01:43 +0000 [r58954-58958]  Russell Bryant <russell@digium.com>

	* /, configs/sla.conf.sample: Merged revisions 58957 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15
	  Mar 2007) | 1 line fix a couple SLA documentation references
	  ........

	* /, build_tools/prep_tarball: Merged revisions 58953 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58953 | russell | 2007-03-15 20:12:40 -0500 (Thu, 15
	  Mar 2007) | 2 lines Add the --pdf option to the usage of rubber
	  in prep_tarball ........

2007-03-16 00:04 +0000 [r58949-58950]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, doc/ast_appdocs.tex: Merged revisions 58946 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58946 | tilghman | 2007-03-15 18:52:48 -0500 (Thu, 15 Mar 2007)
	  | 2 lines Refashion dump command to match common syntax and
	  update the resulting appdocs TeX file ........

	* main/pbx.c: Fix trunk so that it compiles again

2007-03-15 23:56 +0000 [r58942-58948]  Russell Bryant <russell@digium.com>

	* Makefile, /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  Merged revisions 58947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) |
	  3 lines Add configure script checking for GTK2 and some
	  additional Makefile targets to support gmenuselect ........

	* /, doc/asterisk.tex: Merged revisions 58941 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58941 | russell | 2007-03-15 18:24:09 -0500 (Thu, 15 Mar 2007) |
	  1 line add a link to the rubber homepage ........

2007-03-15 22:52 +0000 [r58936-58938]  Russell Bryant <russell@digium.com>

	* Makefile, /, doc/asterisk.tex: Merged revisions 58937 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58937 | russell | 2007-03-15 17:51:29 -0500 (Thu, 15 Mar 2007) |
	  2 lines Add Asterisk version information to the generated PDF
	  ........

	* /, build_tools/prep_tarball: Merged revisions 58935 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58935 | russell | 2007-03-15 17:35:52 -0500 (Thu, 15
	  Mar 2007) | 2 lines have prep_tarball attempt to build
	  asterisk.pdf ........

2007-03-15 22:33 +0000 [r58934]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_realtime.c: Merged revisions 58933 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58933 | tilghman | 2007-03-15 17:32:33 -0500 (Thu, 15
	  Mar 2007) | 2 lines Function works fine, but the documentation is
	  backwards. ........

2007-03-15 22:29 +0000 [r58932]  Russell Bryant <russell@digium.com>

	* doc/manager.txt (removed), doc/misdn.txt (removed),
	  doc/jitterbuffer.tex (added), /, doc/billing.txt (removed),
	  doc/extensions.tex (added), doc/queues-with-callback-members.tex
	  (added), doc/localchannel.txt (removed), doc/cdrdriver.txt
	  (removed), doc/00README.1st (removed), doc/ajam.tex (added),
	  doc/manager.tex (added), doc/misdn.tex (added), doc/freetds.txt
	  (removed), doc/odbcstorage.txt (removed), configure,
	  doc/model.txt (removed), doc/cygwin.txt (removed), doc/sla.tex,
	  doc/billing.tex (added), doc/ael.txt (removed),
	  doc/channelvariables.txt (removed), doc/callingpres.txt
	  (removed), doc/musiconhold-fpm.txt (removed),
	  doc/localchannel.tex (added), doc/enum.txt (removed),
	  doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
	  doc/security.txt (removed), doc/imapstorage.txt (removed),
	  doc/PEERING, main/pbx.c, doc/freetds.tex (added),
	  doc/odbcstorage.tex (added), doc/privacy.txt (removed),
	  configure.ac, doc/iax.txt (removed), doc/channelvariables.tex
	  (added), doc/ael.tex (added), doc/enum.tex (added),
	  doc/security.tex (added), doc/math.txt (removed), Makefile,
	  doc/imapstorage.tex (added), doc/privacy.tex (added),
	  doc/realtime.txt (removed), doc/dundi.txt (removed),
	  doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
	  (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
	  doc/ast_appdocs.tex (added), doc/realtime.tex (added),
	  doc/ices.txt (removed), doc/dundi.tex (added), doc/queuelog.txt
	  (removed), doc/extconfig.txt (removed), doc/radius.txt (removed),
	  doc/cliprompt.tex (added), doc/chaniax.tex (added),
	  doc/hardware.txt (removed), doc/mp3.txt (removed),
	  doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
	  (added), doc/configuration.txt (removed), doc/queuelog.tex
	  (added), doc/asterisk-conf.txt (removed), doc/sla.pdf (removed),
	  doc/ip-tos.txt (removed), doc/hardware.tex (added), doc/h323.txt
	  (removed), doc/mp3.tex (added), doc/configuration.tex (added),
	  doc/asterisk-conf.tex (added), doc/jitterbuffer.txt (removed),
	  doc/channels.txt (removed), doc/ip-tos.tex (added),
	  doc/extensions.txt (removed),
	  doc/queues-with-callback-members.txt (removed), doc/apps.txt
	  (removed), makeopts.in, doc/ajam.txt (removed): Merged revisions
	  58931 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) |
	  13 lines Merge changes from svn/asterisk/team/russell/LaTeX_docs.
	  * Convert most of the doc directory into a single LaTeX formatted
	  document so that we can generate a PDF, HTML, or other formats
	  from this information. * Add a CLI command to dump the
	  application documentation into LaTeX format which will only be
	  include if the configure script is run with --enable-dev-mode. *
	  The PDF turned out to be close to 1 MB, so it is not included.
	  However, you can simply run "make asterisk.pdf" to generate it
	  yourself. We may include it in release tarballs or have
	  automatically generated ones on the web site, but that has yet to
	  be decided. ........

2007-03-15 18:21 +0000 [r58924]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 58923 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2
	  lines Don't assume that the pvt structure will still exist after
	  calling schedule_delivery as it may not. (issue #9278 reported by
	  fmachado) ........

2007-03-14 19:19 +0000 [r58904-58907]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 58906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) |
	  4 lines Some people like to put "limitonpeer" instead of
	  "limitonpeers" in their configuration. While we're at it, support
	  "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172)
	  ........

	* /, doc/sla.tex, doc/sla.pdf: Merged revisions 58902 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14
	  Mar 2007) | 2 lines Add a more basic example setup to the
	  examples section ........

2007-03-14 17:01 +0000 [r58900-58901]  Olle Johansson <oej@edvina.net>

	* cdr/cdr_radius.c: Correct reference to Radius library THanks
	  Philippe - Greetings from Lisboa, Portugal

	* /, channels/chan_sip.c: Merged revisions 58848 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue,
	  13 Mar 2007) | 10 lines Merged revisions 58847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
	  lines Issue #9229 - No port in request URI on register to non
	  default SIP ports (neelakantan) ........ ................

2007-03-14 16:40 +0000 [r58895-58898]  Russell Bryant <russell@digium.com>

	* /, doc/security.txt: Merged revisions 58897 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58897 | russell | 2007-03-14 11:40:22 -0500
	  (Wed, 14 Mar 2007) | 11 lines Merged revisions 58896 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14
	  Mar 2007) | 3 lines Add a note to the security file that the
	  Asterisk CLI and log files may contain sensitive information, and
	  that people should keep this in mind. ........ ................

	* /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
	  58894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) |
	  8 lines By default, don't attempt to do any CallerID handling at
	  all with SLA because it is known to not work properly in some
	  situations. However, add an option to enable it for those that
	  would like to use it anyway. The short story behind this is that
	  to properly handle CallerID with SLA, we need the ability to
	  change the CallerID on an existing call, and we are not ready to
	  handle that. ........

2007-03-14 01:56 +0000 [r58881]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 58880 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58880 | tilghman | 2007-03-13 20:47:08 -0500 (Tue, 13 Mar 2007)
	  | 3 lines Issue 9162 - pbx_substitute_variables_helper assumes
	  the buffer is initialized to all zeroes. This fixes a case where
	  it wasn't. ........

2007-03-13 23:20 +0000 [r58866-58873]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 58872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) |
	  4 lines Ensure that the blinky lights show that the trunk stopped
	  ringing when the trunk hangs up before a station has answered it.
	  (issue #9234, reported by francesco_r) ........

	* /, configs/sla.conf.sample: Merged revisions 58870 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13
	  Mar 2007) | 1 line fix the reference to the SLA documentation
	  ........

	* cdr/cdr_sqlite3_custom.c (added), build_tools/menuselect-deps.in,
	  configure, include/asterisk/autoconfig.h.in,
	  configs/cdr_sqlite3_custom.conf (added),
	  doc/res_config_sqlite.txt (added), cdr/cdr_sqlite.c,
	  configs/extconfig.conf.sample, configure.ac, UPGRADE.txt,
	  CHANGES, makeopts.in, res/res_config_sqlite.c (added),
	  configs/res_config_sqlite.conf (added): Merge changes from
	  team/russell/sqlite: * Add new module, cdr_sqlite3_custom which
	  allows logging custom CDRs into a SQLite3 database. (issue #7149,
	  alerios) * Add new module, res_config_sqlite, which adds realtime
	  database configuration support for SQLite version 2. I decided
	  that this was ok since we didn't have any realtime support for
	  version 3. If someone ports this to version 3, then version 2
	  support can be removed or marked deprecated. (issue #7790,
	  rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of
	  cdr_sqlite3_custom. Also, note that there were other modules on
	  the bug tracker that did not make the cut because they provided
	  some duplicated functionality. Those are: * cdr_sqlite3 (issue
	  #6754, moy) * cdr_sqlite3 (issue #8694, bsd)

2007-03-13 10:14 +0000 [r58822-58846]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 58845 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3
	  lines Don't hangup the call on OK or errors on MESSAGE and INFO
	  inside of a dialog (like video update requests). ........

	* /, channels/chan_sip.c: Merged revisions 58843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2
	  lines Issue #9251 - Clear From URI from user attributes (tgrman)
	  ........

	* channels/chan_h323.c: Change URL to OpenH323 (thanks, Tzafrir!)

2007-03-12 01:22 +0000 [r58780-58784]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 58783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2
	  lines Allow RFC2833 compensation to compensate for even stupider
	  implementations by queueing up the end frame at the start, not
	  the actual end. (issue #8963 reported by AndrewZ) ........

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  58779 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2
	  lines Add matchexterniplocally setting which only substitutes
	  your externip/externhost setting if it matches the localnet
	  setting. I know of at least two people who need opposite
	  settings, so I made it an option! (issue #8821 reported by
	  kokoskarokoska) ........

2007-03-11 21:57 +0000 [r58761]  Kevin P. Fleming <kpfleming@digium.com>

	* main/asterisk.c: grammatical errors are bad, mmmkay?

2007-03-11 16:43 +0000 [r58742]  Jason Parker <jparker@digium.com>

	* build_tools/cflags.xml, main/asterisk.c: Add CLI command "marko
	  show birthday" to show "birthday information" for Mark Spencers
	  upcoming 30th birthday. To enable, run `make menuselect` and
	  select the option MARKO_BDAY under Compiler Flags.

2007-03-10 18:15 +0000 [r58639-58706]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 58705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) |
	  6 lines Fix a few more places in chan_iax2 where the ast_frame
	  used for receiving a frame was not properly initialized. -
	  Interpolating a frame when the jitterbuffer is in use -
	  decrypting a frame when IAX2 encryption is on - frames in an IAX2
	  trunk ........

	* /, apps/app_meetme.c: Merged revisions 58669 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) |
	  2 lines Make the compiler happy and initialize a variable.
	  ........

	* /, doc/sla.txt (removed), doc/sla.tex (added), doc/sla.pdf
	  (added): Merged revisions 58638 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) |
	  8 lines Merge some updates to the SLA documentation. I plan to
	  keep working on this to explain all of the expected behavior with
	  call handling, configuration details for specific phones, and
	  other things. However, I got tired of doing it in plain text, so
	  I switched to using LaTeX. I have included the PDF version. I
	  haven't been able to get a nice looking plain text version out of
	  it yet, but I'm not terribly concerned since this is supposed to
	  be more of the manual, while the plain text sample configuration
	  file is the reference. ........

2007-03-09 21:10 +0000 [r58592-58605]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 58604 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2
	  lines Fix spelling of unavailable in voicemail documentation.
	  (issue #9248 reported by tensai) ........

	* /, channels/chan_sip.c: Merged revisions 58584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58584 | file | 2007-03-09 15:49:47 -0500 (Fri,
	  09 Mar 2007) | 10 lines Merged revisions 58579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
	  lines If we are unable to lookup the host in a c line we have to
	  abort, otherwise the previous data is gone and we will
	  (potentially) have no data when all is said and done. ........
	  ................

2007-03-08 23:21 +0000 [r58511-58541]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 58512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) |
	  5 lines Hang up the channel that put the call on hold in the
	  event processing thread to avoid a race condition. Also, if the
	  station originated the call that it is putting on hold, don't
	  hang up the trunk if it was the only station on the call and it
	  is hanging up due to hold and not a normal hangup. ........

	* channels/chan_zap.c, /: Merged revisions 58510 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) |
	  3 lines Add a missing break statement so that handling the above
	  event does not incorrectly destroy the channel. (issue #9242,
	  andrew) ........

2007-03-08 21:34 +0000 [r58480]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 58479 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007)
	  | 2 lines Fix segfault (Issue 9236) ........

2007-03-08 20:56 +0000 [r58475]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 58474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) |
	  3 lines Refactor hold handling a bit so that it does not require
	  keeping the call up when a call is put on hold. ........

2007-03-08 18:05 +0000 [r58390-58437]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 58436 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2
	  lines Make early SDP seeding even smarter! We have to check
	  codecs in the make_compatible function too. (issue #9221 reported
	  by marcelbarbulescu) ........

	* /, main/dsp.c: Merged revisions 58389 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58389 | file | 2007-03-08 11:07:10 -0500 (Thu,
	  08 Mar 2007) | 10 lines Merged revisions 58388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
	  lines Only print out debug message if the definition that makes
	  the variables shows up was actually defined. (issue #9233
	  reported by serginuez) ........ ................

2007-03-08 13:27 +0000 [r58353-58355]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/http.c: Merged revisions 58354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007)
	  | 2 lines this change was not needed; fclose() handles closing
	  the file descriptor already ........

	* /, apps/app_meetme.c, main/http.c: Merged revisions 58351-58352
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007)
	  | 2 lines fix two cases where HTTP session file descriptors would
	  not be closed ........ r58352 | kpfleming | 2007-03-08 08:17:42
	  -0500 (Thu, 08 Mar 2007) | 2 lines fix a compiler warning, and
	  overwriting 'res' value ........

2007-03-08 01:06 +0000 [r58304-58321]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /, configure, configure.ac: Merged revisions
	  58320 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) |
	  6 lines If we receive ZT_EVENT_REMOVED, destroy the specified
	  channel. (issue #7256, tzafrir) Also, update the configure script
	  to make sure that we don't try to build chan_zap if the installed
	  version of zaptel does not include ZT_EVENT_REMOVED. ........

	* configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Add the
	  ability to dynamically specify weights for responses to DUNDi
	  queries. This can be done using a global variable or a dialplan
	  function. Using the SHELL() function will allow you to use an
	  external script to determine what the weight in the response
	  should be. This can be very useful in load balancing
	  applications. (inspired by discussions with blitzrage and jsmith
	  in #asterisk-bugs)

2007-03-07 20:05 +0000 [r58286]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: Make the loader less noisy under valgrind.

2007-03-07 18:20 +0000 [r58244]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 58243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58243 | russell | 2007-03-07 12:19:19 -0600
	  (Wed, 07 Mar 2007) | 17 lines (This bug was reported to me by
	  Kinsey Moore) Merged revisions 58242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
	  7 lines Fix a problem where the Asterisk channel name could be
	  that of the wrong IAX2 user for a call. This is because the first
	  step of choosing this name is to look for an IAX2 peer that
	  happens to have the same IP/port number that this call is coming
	  from and assuming that is it. However, this is not always
	  correct. So, I have made it change this name after authentication
	  happens since at that point, we have an exact match. ........
	  ................

2007-03-07 17:55 +0000 [r58241]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c, main/rtp.c: Merged revisions 58240 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2
	  lines Ensure we have (or should have) at least one matching codec
	  before attempting early bridge SDP seeding. (issue #9221 reported
	  by marcelbarbulescu) ........

2007-03-07 08:08 +0000 [r58224]  Olle Johansson <oej@edvina.net>

	* apps/app_ices.c: Adding reference to ices home page. Anyone that
	  has tested with ices2 ?

2007-03-07 01:07 +0000 [r58123-58208]  Russell Bryant <russell@digium.com>

	* main/file.c: Add the format of the file that is currently being
	  played to the verbose message. (issue #9105, junky)

	* main/manager.c, /: Merged revisions 58165 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58165 | russell | 2007-03-06 18:25:19 -0600
	  (Tue, 06 Mar 2007) | 12 lines Merged revisions 58164 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06
	  Mar 2007) | 4 lines If the channels acquired using the manager
	  Redirect action are not up, then don't attempt to do anything
	  with them. It could lead to weird behavior, including crashes.
	  (issue #8977) ........ ................

	* include/asterisk/utils.h: Add some documentation on the arguments
	  to the base64 encode/decode functions. (inspired by issue #9215)

	* apps/app_queue.c: Send a manager AgentComplete event when the
	  agent transfers the call, in addition to where it is already sent
	  if either side hangs up. (issue #9219, rgollent) In passing, I
	  put this code in a function so it would not be duplicated a third
	  time.

2007-03-06 23:19 +0000 [r58122]  Steve Murphy <murf@digium.com>

	* /, channels/chan_sip.c: Merged revisions 58121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue,
	  06 Mar 2007) | 9 lines Merged revisions 58115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
	  line Fix for 9220: Eyebeam cannot renew subscriptions for
	  presence info. Reason: re-SUBSCRIBE requests don't include Accept
	  headers, which the rfc says are optional (to put it tersely), (it
	  uses MAY), and luckily, the sip_pvt struct has the format info
	  stored, so we simply leave it if the format is set, and the
	  accept header null. ........ ................

2007-03-06 23:01 +0000 [r58101-58120]  Russell Bryant <russell@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 58119 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) |
	  3 lines Clarify the documentation of the dialout and
	  sendvoicemail options. (issue #9000, caio1982 and serge-v)
	  ........

	* codecs/codec_zap.c: Sync codec_zap with the one that is in the
	  1.4 branch so that it can actually build here, too.

2007-03-06 20:45 +0000 [r58054-58055]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 58053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue,
	  06 Mar 2007) | 10 lines Merged revisions 58052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
	  lines Change error message to proper message ........
	  ................

	* apps/app_stack.c: Debug control, debug control.

2007-03-06 18:02 +0000 [r58024-58025]  Russell Bryant <russell@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 58023 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06
	  Mar 2007) | 3 lines Return an error of transmit_response is
	  called without a session. (issue #9002) ........

2007-03-06 08:51 +0000 [r57979-57993]  Luigi Rizzo <rizzo@icir.org>

	* main/say.c: move declaration to the beginning of a block

	* apps/app_meetme.c: remove duplicate const

2007-03-05 20:13 +0000 [r57871-57943]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, CHANGES: Add zap show version CLI command.
	  This pulls the version/echo canceller in use directly using the
	  ZT_GETVERSION ioctl. (issue #9094 reported by tootai)

	* /, channels/chan_iax2.c: Merged revisions 57914 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2
	  lines Since chan_iax2 does not support reception of DTMF with
	  duration ensure that it is set to 0 on the frame. (issue #8521
	  reported by gdhgdh) ........

	* /, apps/app_meetme.c: Merged revisions 57872 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2
	  lines Don't create a listen channel and record the conference
	  unless the option is turned on. (issue #9204 reported by
	  francesco_r) ........

	* apps/app_meetme.c: I like it when app_meetme builds under dev
	  mode, don't you?

	* /, apps/app_voicemail.c: Merged revisions 57870 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57870 | file | 2007-03-05 12:52:03 -0500 (Mon,
	  05 Mar 2007) | 10 lines Merged revisions 57869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
	  lines Make create_dirpath use our standard for return values. -1
	  is failure, 0 is success. (issue #9205 reported by ballares)
	  ........ ................

2007-03-05 15:30 +0000 [r57827]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57826 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon,
	  05 Mar 2007) | 9 lines Merged revisions 57825 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
	  line Fixed a typo introduced via 9156 (either the gotos or their
	  doc strings are wrong) ........ ................

2007-03-05 04:21 +0000 [r57769-57799]  Joshua Colp <jcolp@digium.com>

	* /, main/slinfactory.c: Merged revisions 57798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2
	  lines Don't allow a NULL pointer to reach ast_frdup. (issue #9155
	  reported by cmaj) ........

	* configs/extensions.conf.sample: Remove no longer present CLI
	  commands from sample extensions.conf. (issue #9193 reported by
	  junky)

	* /, res/res_jabber.c: Merged revisions 57770 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2
	  lines Don't reference a potentially NULL pointer. (issue #9199
	  reported by klolik) ........

	* /, main/rtp.c: Merged revisions 57768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2
	  lines Preserve marker bit when P2P bridging. (issue #9198
	  reported by edgreenberg) ........

2007-03-03 16:43 +0000 [r57736]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Convert stack apps to use ast_storage channel
	  structure

2007-03-03 15:35 +0000 [r57708]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-vtest13:
	  updated the regression tests

2007-03-03 14:40 +0000 [r57651-57691]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, include/asterisk/channel.h: Expand datastores to
	  add the notion of inheritance. This will be needed for the
	  conversion of IAX2 variables from the current custom method to
	  ast_storage.

	* /, apps/app_voicemail.c: Merged revisions 57649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57649 | tilghman | 2007-03-03 00:45:00 -0600
	  (Sat, 03 Mar 2007) | 10 lines Merged revisions 57648 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03
	  Mar 2007) | 2 lines Memory leak of a list, if call recording was
	  abandoned ........ ................

2007-03-03 01:11 +0000 [r57621]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/say.c: Merged revisions 57620 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007)
	  | 1 line submitted patch for Georgian language, issue 9010,
	  submitted by Alexander Shaduri ........

2007-03-03 00:01 +0000 [r57557-57590]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample: Add the missing configuration template
	  to the sample config file. Thanks to Lacy Moore on the
	  asterisk-users list for pointing out that this was missing!

	* /, configure, configure.ac: Merged revisions 57556 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02
	  Mar 2007) | 3 lines Update the check that is used to determine
	  whether zaptel transcoder support is present. The interface has
	  changed. ........

2007-03-02 18:05 +0000 [r57478-57519]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Don't try to do recursive locking/unlocking when it
	  isn't supported.

	* /, channels/chan_sip.c: Merged revisions 57477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57477 | file | 2007-03-02 12:06:52 -0500 (Fri,
	  02 Mar 2007) | 10 lines Merged revisions 57475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
	  lines If a SIP message comes in and goes to a method handler that
	  requires additional values that may not be present then send back
	  an error. ........ ................

2007-03-02 17:03 +0000 [r57476]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57473 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri,
	  02 Mar 2007) | 9 lines Merged revisions 57458 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
	  line further refinement in wording of goto documentation, as per
	  9156, goto not proceeding to next instruction ........
	  ................

2007-03-02 16:59 +0000 [r57474]  Russell Bryant <russell@digium.com>

	* apps/app_dumpchan.c, main/cli.c: Add the channel's Language to
	  the "show channel" CLI command and the DumpChan application.
	  (issue #9187, Junky)

2007-03-02 05:57 +0000 [r57438]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c, utils/ael_main.c: Merged revisions 57426 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1
	  line I almost had comma escapes right, but 9184 points out the
	  problem-- the escape is removed by pbx_config, and pbx_ael should
	  also, before sending it down into the pbx engine. Also, you have
	  to insert it back in, if you are generating extensions.conf code
	  from the AEL. ........

2007-03-02 00:22 +0000 [r57365-57397]  Russell Bryant <russell@digium.com>

	* /, main/file.c: Merged revisions 57396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) |
	  4 lines Return the correct digit that interrupted the stream.
	  This fixes exiting the Background application when using the m
	  option. (issue #9176, mjagdis) ........

	* /, apps/app_meetme.c, doc/sla.txt, include/asterisk/channel.h,
	  configs/sla.conf.sample: Merged revisions 57364 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) |
	  16 lines Merge changes from svn/asterisk/team/russell/sla_updates
	  * Originally, I put in the documentation that only Zap interfaces
	  would be supported on the trunk side. However, after a discussion
	  with Qwell, we came up with a way to make IP trunks work as well,
	  using some things already in Asterisk. So, here it is, this now
	  officially supports IP trunks. * Update the SLA documentation to
	  reflect how to setup IP trunks. * Add a section in sla.txt that
	  describes how to set up an SLA system with voicemail. * Simplify
	  the way DTMF passthrough is handled in MeetMe. * Fix a bug that
	  exposed itself when using a Local channel on the trunk side in
	  SLA. The station's channel needs to be passed to the dial API
	  when dialing the trunk. * Change a WARNING message to DEBUG in
	  channel.h. This message is of no use to users. ........

2007-03-01 22:23 +0000 [r57319]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 57318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57318 | file | 2007-03-01 17:21:44 -0500 (Thu,
	  01 Mar 2007) | 10 lines Merged revisions 57317 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2
	  lines Don't even attempt to optimize things when a proxy channel
	  is involved. It will just explode in weird and unexplaineable
	  ways. (issue #9175 reported by clegall_proformatique) ........
	  ................

2007-03-01 20:24 +0000 [r57293]  Russell Bryant <russell@digium.com>

	* main/channel.c: Constify the list of codec preferences.

2007-03-01 03:01 +0000 [r57259]  TransNexus OSP Development <support@transnexus.com>

	* doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.

2007-03-01 00:08 +0000 [r57241]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Minor code cleanup... nothing to write home about.

2007-02-28 23:02 +0000 [r57204-57209]  Russell Bryant <russell@digium.com>

	* /, doc/sla.txt, configs/sla.conf.sample: Merged revisions 57207
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) |
	  2 lines minor tweaks to the sla docs ........

	* /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
	  57203 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) |
	  7 lines Merge more changes from
	  svn/asterisk/team/russell/sla_updates * Add support for private
	  hold. By setting "hold=private" for a trunk, only the station
	  that put the call on hold will be able to retrieve it from hold.
	  Also, by setting "hold=private" for a station, any call that
	  station puts on hold can only be retrieved by that station.
	  ........

2007-02-28 20:46 +0000 [r57184]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, pbx/pbx_dundi.c, include/asterisk/pbx.h,
	  pbx/pbx_config.c, apps/app_while.c: Convert the PBX core to use
	  read/write locks. This yields a nifty performance improvement
	  when it comes to simultaneous calls going through the dialplan.
	  Using murf's test the old mutex based core took an average of
	  57.3 seconds while the rwlock based core took 31.1 seconds.
	  That's a nifty 26.2 seconds performance improvement. The other
	  good part is that if we do need to switch back then we just have
	  to change the lock/unlock API calls. I converted everywhere that
	  used to touch the mutex locks directly to use them.

2007-02-28 19:59 +0000 [r57145-57147]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 57146 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) |
	  2 lines Minor formatting change ........

	* /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
	  57144 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) |
	  6 lines Merge changes from svn/asterisk/team/russell/sla_updates
	  * Add support for the "barge=no" option for trunks. If this
	  option is set, then stations will not be able to join in on a
	  call that is on progress on this trunk. ........

2007-02-28 19:30 +0000 [r57140]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57139 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57139 | murf | 2007-02-28 12:23:05 -0700 (Wed,
	  28 Feb 2007) | 9 lines Merged revisions 57118 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
	  line a small documentation update, to reflect reality in the goto
	  doc strings, as per 9156, Goto does not proceed to next prio if
	  jump fails ........ ................

2007-02-28 19:00 +0000 [r57094]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 57093 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r57093 | file | 2007-02-28 13:57:52 -0500 (Wed,
	  28 Feb 2007) | 10 lines Merged revisions 57092 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2
	  lines Fix a few more issues with the agent logoff CLI command.
	  (issue #9123 reported by arbrandes) ........ ................

2007-02-28 18:21 +0000 [r57090]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
	  57089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) |
	  8 lines Merge current set of changes from
	  svn/asterisk/team/russell/sla_updates * Add support for station
	  ring delays. Ring delays can be set globally for a station or for
	  specific trunks on the station. * Fix a few bugs in existing
	  code. * Restructure and Reorganize code to improve readability
	  and maintainability. * Improve formatting of the "sla show
	  (trunks|stations)" CLI commands. ........

2007-02-28 17:56 +0000 [r57054-57056]  Joshua Colp <jcolp@digium.com>

	* /: Merged revisions 57055 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57055 | file | 2007-02-28 12:55:03 -0500 (Wed, 28 Feb 2007) | 2
	  lines Picky compiler... ........

	* /, apps/app_speech_utils.c: Merged revisions 57053 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb
	  2007) | 2 lines Better handle timeouts when the individual speaks
	  after everything has been played but before the timeout ends.
	  ........

2007-02-28 17:22 +0000 [r57050]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 57049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r57049 | murf | 2007-02-28 10:15:27 -0700 (Wed, 28 Feb 2007) | 1
	  line I was surprised that I had not yet downgraded missing goto
	  targets and macro call defs to a warning, in case they are in
	  extensions.conf; I rectified this problem. Also, A goto in a
	  macro to a target in a catch block was not being found; I fixed
	  this too; the cause was that I needed to treat catch statements
	  like an extension in the find_match code. ........

2007-02-27 22:17 +0000 [r57011]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Properly hangup the original dialed channel, not
	  the new channel that appeared from the forwarding. (issue #9161
	  reported by PhilSmith)

2007-02-27 17:38 +0000 [r56976]  Russell Bryant <russell@digium.com>

	* /: (also issue #9159) Merged revisions 56975 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56975 | russell | 2007-02-27 11:36:09 -0600 (Tue, 27 Feb 2007) |
	  4 lines Fix voicemail email attachments. I missed the conversion
	  of one of the line endings and there was an extra one where it
	  should not have been. (issue #9128) ........

2007-02-27 00:11 +0000 [r56926-56952]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: Issue 7789 -
	  some telcos want the TON set based on the number, but without the
	  NANP prefix removed

2007-02-26 20:43 +0000 [r56889]  Russell Bryant <russell@digium.com>

	* /, channels/chan_alsa.c: Merged revisions 56888 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) |
	  4 lines Restore the behavior of Asterisk 1.2 where if a device
	  was not specified in alsa.conf, then we just use the system
	  default, instead of creating our own default of hw:0,0. (issue
	  #9139) ........

2007-02-26 20:09 +0000 [r56860]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 56856 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56856 | file | 2007-02-26 15:07:18 -0500 (Mon,
	  26 Feb 2007) | 10 lines Merged revisions 56850 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
	  lines Obey the clearglobalvars option in extensions reload (or
	  dialplan reload depending on your version). (issue #9146 reported
	  by ramonpeek) ........ ................

2007-02-26 20:04 +0000 [r56849]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 56847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) |
	  2 lines Fix a crash in my last change to iax2_indicate(). (issue
	  #9150) ........

2007-02-26 19:34 +0000 [r56811-56840]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_record.c: Merged revisions 56839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56839 | file | 2007-02-26 14:33:48 -0500 (Mon, 26 Feb 2007) | 2
	  lines Update app_record documentation to use new CLI command,
	  core show file formats. (issue #9151 reported by junky) ........

	* main/pbx.c, /: Merged revisions 56805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56805 | file | 2007-02-26 12:09:53 -0500 (Mon, 26 Feb 2007) | 2
	  lines Use ast_strlen_zero to see if the language and/or context
	  argument is not present for Background instead of just checking
	  if it is NULL. (issue #9141 reported by mjagdis) ........

2007-02-26 16:54 +0000 [r56786]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 56785 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) |
	  3 lines Do more complete locking of the chan_iax2_pvt struct in
	  the indicate callback. (Problem brought up by Ben Smithurst on
	  the asterisk-dev list) ........

2007-02-26 16:38 +0000 [r56784]  Joshua Colp <jcolp@digium.com>

	* /, main/asterisk.c: Merged revisions 56783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56783 | file | 2007-02-26 11:36:08 -0500 (Mon, 26 Feb 2007) | 2
	  lines Allow both of the show version files and core show file
	  versions CLI commands to work. (issue #9135 reported by mvanbaak)
	  ........

2007-02-26 01:05 +0000 [r56731-56742]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 56740 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56740 | russell | 2007-02-25 19:04:40 -0600 (Sun, 25 Feb 2007) |
	  2 lines Move a comment to be in the correct struct. ........

	* main/asterisk.c: Remove redundant check to ensure that LOW_MEMORY
	  is not defined. (issue #9136, mvanbaak)

	* channels/chan_iax2.c: There is no need to look in the iaxs array
	  for the pvt struct when we already have a pointer to it.

2007-02-25 14:53 +0000 [r56686]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 56685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600
	  (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25
	  Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the
	  channel list, then evaluating additional conditions (e.g. name
	  prefix) will cause a NULL dereference. ........ ................

2007-02-24 20:29 +0000 [r56623-56665]  Olle Johansson <oej@edvina.net>

	* include/asterisk/http.h, main/channel.c,
	  include/asterisk/doxyref.h, include/asterisk/utils.h,
	  include/asterisk/zapata.h, apps/app_meetme.c, res/res_limit.c,
	  include/asterisk/config.h, channels/chan_h323.c, pbx/pbx_ael.c,
	  apps/app_amd.c, include/asterisk/ael_structs.h,
	  include/asterisk/jingle.h, main/config.c, main/rtp.c: Doxygen
	  additions, corrections

	* include/asterisk/doxyref.h, channels/chan_zap.c, main/manager.c,
	  include/asterisk/frame.h: Doxygen updates and corrections

	* apps/app_osplookup.c, funcs/func_curl.c, res/res_snmp.c,
	  apps/app_festival.c, cdr/cdr_sqlite.c, codecs/codec_speex.c,
	  contrib/asterisk-ng-doxygen, include/asterisk/jabber.h,
	  res/res_crypto.c, channels/chan_h323.c, cdr/cdr_pgsql.c,
	  cdr/cdr_radius.c, apps/app_voicemail.c: Creating new doxygen
	  macro "\extref" to create page that lists external libraries and
	  URLs to these. Please help me add these references. We might want
	  to create a similar macro "\linuxpackage" to list the needed
	  Linux packages in popular distributions.

	* include/asterisk/jabber.h: Add some external references

	* include/asterisk/doxyref.h, include/asterisk/jabber.h: Doxygen
	  updates for AJI - The Asterisk Jabber API

2007-02-24 02:23 +0000 [r56574-56594]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Allow a
	  Skinny device to monitor a dialplan hint (w00t!). See
	  skinny.conf.sample for configuration example. Note: Some devices
	  (seen on 12SP+/30VIP) will lock up if they monitor too many
	  hints. This seems to be a hardware limitation - there isn't
	  anything we can do about it.

	* channels/chan_skinny.c: Support devicestate requests. Now you
	  should be able to subscribe to a Skinny device/line.

	* /, channels/chan_skinny.c: Merged revisions 56569 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb
	  2007) | 4 lines Make sure to set a speeddials parent on creation.
	  Don't crash if hold is pressed when no call is active. Don't
	  return in places that we shouldn't.. Update softkey map when call
	  is connected ........

2007-02-24 01:56 +0000 [r56564]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Make Meetme build again under dev mode.

2007-02-23 23:25 +0000 [r56487-56506]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.c: Merged revisions 56505 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56505 | russell | 2007-02-23 17:24:18 -0600
	  (Fri, 23 Feb 2007) | 16 lines Merged revisions 56504 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23
	  Feb 2007) | 8 lines Fix up a couple more signal handlers to not
	  do bad things that could cause various undesirable results. The
	  other day, I made Asterisk deadlock by hitting Control-C because
	  of a bad signal handler. Now, signal handlers just set a flag and
	  write to an alert pipe for the flag to be handled. Then, there is
	  another thread that is monitoring for these flags. If being run
	  in console mode, it is just the main thread. If Asterisk is in
	  the background, a thread is created to do it. ........
	  ................

	* channels/chan_iax2.c: Make the hashing function calculate
	  something that makes more sense. (Thanks to bmd on #asterisk-dev
	  for pointing out my pointless math).

2007-02-23 21:57 +0000 [r56458]  Joshua Colp <jcolp@digium.com>

	* /, main/sched.c: Merged revisions 56457 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2
	  lines Change log notice to debug. It is possible for a scheduled
	  item to execute and be deleted at close to the same time and
	  unavoidable. If this happens this message creeps up. ........

2007-02-23 21:20 +0000 [r56408-56447]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Merge team/russell/iax2_performance. There
	  is not a large amount of code here and the changes are not very
	  invasive. However, they should significantly improve performance
	  of chan_iax2 under load. IAX2 media frames only carry the
	  *source* call number. So, when one arrives, the correct session
	  that it is a part of has to be matched on IP address, port
	  number, and call number, instead of just a call number. Had these
	  frames carried the *destination* call number, this would not be
	  an issue, because that would be a unique identifier that would
	  make it easy to immediately identify the correct session. The way
	  that chan_iax2 did this matching was extremely inefficient. It
	  starts at the first available call number and traverses each call
	  number sequentially, locking and unlocking a mutex for each one,
	  to try to match against it. It would do this regardless of
	  whether the call number was in use or not. So, for a call with a
	  local call number of 25000, every single incoming media frame
	  would require a traversal that required 25000 mutex lock and
	  unlock operations. (Note that the max call number is about 32k).
	  I have introduced a hash table of active IAX2 calls to improve
	  this lookup process. The hash is done on the IP address, port
	  number, and call number. So, for the previous example, a few
	  lock/unlock operations may be done versus 25000 for each frame.

	* CHANGES: Note that the entries in the CHANGES file only list
	  functionality changes

	* CHANGES: Add GetConfigJSON to the CHANGES file.

	* /, channels/chan_iax2.c: Merged revisions 56407 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56407 | russell | 2007-02-23 14:20:00 -0600
	  (Fri, 23 Feb 2007) | 12 lines Merged revisions 56406 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23
	  Feb 2007) | 4 lines Don't destroy mutexes before unregistering
	  all of the entry points from the core. Also, fix a potential
	  memory leak from not destroying the locks for all of the possible
	  call numbers (about 32k of them). ........ ................

2007-02-23 19:00 +0000 [r56373]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/make_version_h: Merged revisions 56372 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56372 | kpfleming | 2007-02-23 12:59:09 -0600 (Fri, 23 Feb 2007)
	  | 2 lines build special version strings for AADK/S800i builds
	  ........

2007-02-23 18:01 +0000 [r56278-56342]  Russell Bryant <russell@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 56341 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) |
	  8 lines The IMAP storage code uses the same code to build the
	  email that is used when voicemail is sent via email using
	  something like sendmail. In the patch from bug 8033 to fix
	  various IMAP storage problems, the line endings in the email file
	  were changed in the code from "\n" to "\r\n". However, this
	  breaks sending regular voicemail to email. So, this change
	  conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
	  enabled. (issue #9128, patch by jarjarbinks, modified by me to
	  not break IMAP storage) ........

	* main/manager.c: Introduce a new manager action, GetConfigJSON,
	  which is intended to improve performance of the GUI. This encodes
	  the configuration into the JSON format in a manager header,
	  "JSON: ". The encoded information can be directly used as a
	  javascript object, so no parsing is needed. For large
	  configuration files, this can greatly improve loading times in
	  the GUI. Furthermore, the encoding takes up a lot less space when
	  being transmitted than the other alternatives. (Inspired by
	  discussion with Pari) Here is an example of what you get:
	  http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf
	  Response: Success JSON:
	  {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]}

	* main/dial.c, /, apps/app_meetme.c, doc/sla.txt,
	  configs/sla.conf.sample: Merged revisions 56277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) |
	  18 lines Merge changes from team/russell/sla_updates. This batch
	  of changes to the SLA code does a few different things. * I made
	  the SLA code event driven instead of having to act in a lot of
	  busy loops while dialing things to wait for state changes. This
	  makes the code more efficient and readable at the same time. * I
	  have implemented a couple of new features. The first is inbound
	  trunk ringing timeouts. This is an option that defines how long
	  to let an incoming call on a trunk to ring. * I have also
	  implemented ring timeouts for stations. They may be specified for
	  the entire station, meaning it is how long to let the station
	  ring before giving up. You can also specify a ring timeout for a
	  specific trunk on a station. So, you can say that you only want a
	  specific station to ring 5 seconds if it is line1 ringing, but
	  otherwise, there is no timeout. ........

2007-02-22 18:53 +0000 [r56232]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /, channels/chan_sip.c: Merged revisions 56231
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu,
	  22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
	  lines Only change the original or clone channel if it's the
	  channel behind the proxy channel, not if it's just a regular
	  bridged channel. ........ ................

2007-02-22 17:36 +0000 [r56209]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/module.h: move the ast_module_info structure
	  into the special section as well, otherwise when
	  restore_globals() is called it will lose its pointer to the
	  ast_module structure that the loader put there

2007-02-22 16:48 +0000 [r56188]  Joshua Colp <jcolp@digium.com>

	* .cleancount: Since I'm a nice guy... let's increment the clean
	  count since last night's module changes require a rebuild of
	  everything essentially.

2007-02-22 16:25 +0000 [r56187]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Fix some compilation problems in
	  app_voicemail. There was a parenthesis missing in a function
	  prototype, and "#elifdef" is not a valid preprocessor directive.
	  (issue #9122, akohlsmith)

2007-02-22 13:58 +0000 [r56156]  TransNexus OSP Development <support@transnexus.com>

	* doc/osp.txt: Update OSP documention for v1.6.

2007-02-22 10:46 +0000 [r56126]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 56125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2
	  lines Move message from verbose to debug ........

2007-02-22 02:48 +0000 [r56095]  Steve Murphy <murf@digium.com>

	* /, sounds/Makefile: Merged revisions 56094 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56094 | murf | 2007-02-21 19:39:58 -0700 (Wed, 21 Feb 2007) | 1
	  line updated the sound tarball versions in Makefile ........

2007-02-22 02:36 +0000 [r56092]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs, codecs, apps, include/asterisk/module.h,
	  Makefile.moddir_rules, Makefile.rules,
	  build_tools/make_linker_eo_script (added), cdr, pbx, res,
	  channels, formats, main/loader.c: give embedded modules a helping
	  hand by backing up and restoring their global variables when they
	  are loaded and unloaded

2007-02-22 01:26 +0000 [r56012-56056]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 56055 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) |
	  3 lines Restructure a little bit of code to reduce nesting. There
	  is no functionality change here. ........

	* /, channels/chan_sip.c: Merged revisions 56011 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r56011 | russell | 2007-02-21 18:57:36 -0600
	  (Wed, 21 Feb 2007) | 11 lines Merged revisions 56010 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21
	  Feb 2007) | 3 lines If we receive a frame that is not in any of
	  the negotiated formats, then drop it. (potentially issue #8781
	  and SPD-12) ........ ................

2007-02-22 00:38 +0000 [r56009]  Joshua Colp <jcolp@digium.com>

	* /, main/cli.c: Merged revisions 56008 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r56008 | file | 2007-02-21 19:35:55 -0500 (Wed, 21 Feb 2007) | 2
	  lines Print out deprecation notice on usage output of CLI
	  commands. (issue #8925 reported by blitzrage) ........

2007-02-22 00:05 +0000 [r55958-56005]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Make filename on email follow subject
	  message number, purely for cosmetic purposes for individuals like
	  *cough* jsmith *cough*. (issue #9111 reported by sshah)

	* /, apps/app_meetme.c: Merged revisions 55957 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55957 | file | 2007-02-21 15:35:40 -0500 (Wed,
	  21 Feb 2007) | 10 lines Merged revisions 55956 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
	  lines Change naughty warning message to provide useful
	  information. If a write now fails on a channel in meetme it will
	  tell you the channel name instead of spitting out the wrong error
	  message. ........ ................

2007-02-21 20:30 +0000 [r55955]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 55954 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4
	  lines Fix locking issue, and accept "transport-accept" as a valid
	  accept message. This should solve issues 8970 and 8503. ........

2007-02-21 20:26 +0000 [r55953]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Clarify in the doxygen docs abou RFC2833
	  compensation flag.

2007-02-21 20:23 +0000 [r55952]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 55951 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55951 | russell | 2007-02-21 14:22:33 -0600 (Wed, 21 Feb 2007) |
	  3 lines Simplify the last change to app_meetme, and move the call
	  to dispose_conf() up into the block where we know a conf exists.
	  ........

2007-02-21 20:18 +0000 [r55915-55950]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 55949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55949 | file | 2007-02-21 15:16:34 -0500 (Wed, 21 Feb 2007) | 2
	  lines Only dispose of the conference if one was created. ........

	* /, apps/app_speech_utils.c: Merged revisions 55947 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb
	  2007) | 2 lines Only start playing the next file if we have not
	  been quieted. ........

	* /, channels/chan_sip.c: Merged revisions 55914 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2
	  lines Add a flag that indicates whether a SIP dialog is an
	  outgoing call or not. SIP_OUTGOING originally did it but it was
	  repurposed to the direction of the last transaction, which can
	  cause update_call_counter to falsely decrease the wrong counters.
	  (please don't hurt me oej) (issue #8943 reported by mdu113)
	  ........

2007-02-21 14:07 +0000 [r55870]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/make_version: Merged revisions 55869 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55869 | kpfleming | 2007-02-21 08:06:47 -0600
	  (Wed, 21 Feb 2007) | 10 lines Merged revisions 55868 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
	  Feb 2007) | 2 lines use new tag version script ........
	  ................

2007-02-21 08:39 +0000 [r55835]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 55834 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2
	  lines Issue #8848 - Turn off lamp more quickly after transfer
	  (decrement inuse early on transferer's call leg) ........

2007-02-21 02:04 +0000 [r55805]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
	  revisions 55799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4
	  lines Fix segfault when buddy couldn't be found. Issue 7764,
	  patch by sailer ........

2007-02-21 01:05 +0000 [r55763]  Joshua Colp <jcolp@digium.com>

	* main/dns.c: Return trunk to a state where it compiles under
	  Darwin. The byte order stuff is ugly, if anyone wants to clean it
	  up... by all means do so.

2007-02-21 01:05 +0000 [r55762]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 55758 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55758 | russell | 2007-02-20 19:03:25 -0600 (Tue, 20 Feb 2007) |
	  4 lines Improve the reference counting to fix bugs where people
	  report seeing conferences listed that have no members. (issue
	  #9073) ........

2007-02-21 00:14 +0000 [r55671-55748]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 55741 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55741 | file | 2007-02-20 19:11:20 -0500 (Tue, 20 Feb 2007) | 2
	  lines Better handle dropped IMAP connections. (issue #9054
	  reported by bsmithurst) ........

	* /, channels/chan_sip.c: Merged revisions 55717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2
	  lines Return behavior I removed. I did not remember that you
	  could just add a localnet entry to make it work. ........

	* main/logger.c: Flush out the file pointer. (issue #9115 reported
	  by guthrie)

	* /, channels/chan_sip.c: Merged revisions 55688 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2
	  lines Don't test our own address against the localnet settings.
	  At least one person has had issues as a result of this from #7051
	  so I'm reversing it. (issue #8821 reported by kokoskarokoska)
	  ........

	* /, channels/chan_agent.c: Merged revisions 55670 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55670 | file | 2007-02-20 17:47:00 -0500 (Tue,
	  20 Feb 2007) | 10 lines Merged revisions 55669 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2
	  lines Defer clearing callback information if channels are up
	  until they are hung up. This ensures the hangup process goes
	  smoothly and no channels get hung in limbo. (issue #8088 reported
	  by kebl0155) ........ ................

2007-02-20 20:32 +0000 [r55591-55635]  Russell Bryant <russell@digium.com>

	* /, main/http.c: Merged revisions 55634 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55634 | russell | 2007-02-20 14:26:06 -0600 (Tue, 20 Feb 2007) |
	  3 lines Add the Asterisk version information to the Server header
	  in HTTP responses. (requested by Pari) ........

	* /, include/asterisk/manager.h: Merged revisions 55590 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) |
	  2 lines Increase the maximum number of manager headers to 128, at
	  the request of Pari. ........

2007-02-20 16:56 +0000 [r55556]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c,
	  res/res_jabber.c: Merged revisions 55555 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4
	  lines No need to cast nor free with strdupa (thanks file) 55555!
	  ........

2007-02-20 16:42 +0000 [r55554]  Russell Bryant <russell@digium.com>

	* /, configs/sla.conf.sample: Merged revisions 55553 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r55553 | russell | 2007-02-20 10:41:57 -0600 (Tue, 20
	  Feb 2007) | 3 lines Change the formatting of sla.conf.sample to
	  make it more readable. (issue #9112, blitzrage) ........

2007-02-20 15:19 +0000 [r55534]  Joshua Colp <jcolp@digium.com>

	* res/res_jabber.c: I like it when trunk builds, so let's make
	  res_jabber compile again!

2007-02-20 07:48 +0000 [r55514]  Olle Johansson <oej@edvina.net>

	* /, res/res_jabber.c: Merged revisions 55483 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55483 | oej | 2007-02-19 22:12:55 +0100 (Mon, 19 Feb 2007) | 3
	  lines - Not sending arguments to an application is not "out of
	  memory" - Making error messages a bit more clear ........

2007-02-19 23:27 +0000 [r55495]  Jason Parker <jparker@digium.com>

	* .cleancount: We need to bump the cleancount when we make API
	  changes...

2007-02-19 18:15 +0000 [r55436]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 55435 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55435 | tilghman | 2007-02-19 12:11:48 -0600
	  (Mon, 19 Feb 2007) | 10 lines Merged revisions 55434 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19
	  Feb 2007) | 2 lines forcename and forcegreetings options should
	  check to see if the recording already exists ........
	  ................

2007-02-19 16:01 +0000 [r55410-55414]  Joshua Colp <jcolp@digium.com>

	* CHANGES: Clarify last change for SMDI in CHANGES file.

	* configs/voicemail.conf.sample, apps/app_voicemail.c: Allow both
	  an external application and SMDI to do voicemail notification at
	  the same time. (issue #8625 reported by lters)

2007-02-19 15:24 +0000 [r55409]  Doug Bailey <dbailey@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 55397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) |
	  3 lines Changed iax2 process thread to detached to correct memory
	  leak due to left over thread context on thread exit. Modified
	  module unload process to avoid deadlocks on pthread cancels
	  ........

2007-02-18 22:07 +0000 [r55375]  Olle Johansson <oej@edvina.net>

	* apps/app_voicemail.c: Formatting changes.

2007-02-18 19:13 +0000 [r55351-55352]  Joshua Colp <jcolp@digium.com>

	* codecs/gsm/inc/proto.h: Return GSM to a state where it actually
	  builds under dev mode.

	* channels/chan_h323.c: Update chan_h323 to new set_rtp_peer
	  definition.

2007-02-18 15:11 +0000 [r55330]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Being picky...

2007-02-18 15:03 +0000 [r55329]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, channels/chan_misdn.c, main/srv.c,
	  main/editline/refresh.c, pbx/ael/ael.tab.c,
	  channels/misdn/isdn_msg_parser.c, channels/chan_oss.c,
	  main/enum.c, apps/app_voicemail.c, main/ast_expr2.c: add -Wundef
	  to the --enable-dev-mode flags, so that mistyped macro names in
	  #if expressions will be caught convert various #if expressions to
	  #ifdef for macros that may not be defined (and where the value is
	  not important) Note: two of these changes are in bison generated
	  files which is going to be inconvenient when they are regenerated

2007-02-18 15:01 +0000 [r55279-55323]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Simplify post_manager_event()

	* /, apps/app_record.c: Merged revisions 55278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55278 | oej | 2007-02-18 13:35:54 +0100 (Sun,
	  18 Feb 2007) | 10 lines Merged revisions 55277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
	  lines Documentation update (#9053, jsmith) ........
	  ................

2007-02-17 17:41 +0000 [r55220]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 55219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55219 | file | 2007-02-17 12:39:32 -0500 (Sat, 17 Feb 2007) | 2
	  lines Add missing membername option to AddQueueMember
	  documentation. (issue #9088 reported by seanbright) ........

2007-02-17 17:11 +0000 [r55218]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 55217 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb
	  2007) | 4 lines Fix an issue where callerid would not be
	  displayed on some phones. Issue 8995, initial patch and research
	  done by wedhorn ........

2007-02-17 16:48 +0000 [r55087-55198]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: We want to skip the queue if the name doesn't
	  match the specified one, not if they *do*.

	* apps/app_queue.c: Increase "queue show" buffer size from 80 to
	  240. This should be more then enough for most cases. (issue #9089
	  reported by mvanbaak)

	* apps/app_dial.c, /: Merged revisions 55154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55154 | file | 2007-02-16 22:55:30 -0500 (Fri,
	  16 Feb 2007) | 10 lines Merged revisions 55153 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
	  lines Answer the channel before recording privacy information.
	  (issue #8926 reported by lmamane) ........ ................

	* /, apps/app_queue.c: Merged revisions 55129 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55129 | file | 2007-02-16 21:59:50 -0500 (Fri, 16 Feb 2007) | 2
	  lines Make the 'i' option of Queue actually work. (issue #8986
	  reported by utis) ........

	* channels/chan_jingle.c: Update chan_jingle to new definition of
	  set_rtp_peer.

	* /, channels/chan_sip.c: Merged revisions 55086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55086 | file | 2007-02-16 20:16:59 -0500 (Fri,
	  16 Feb 2007) | 10 lines Merged revisions 55073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
	  lines Allow chan_sip to handle attended transfers from a SIP
	  phone that is sitting behind chan_agent. Yes folks, all it took
	  was one line of code. (issue #8784 reported by pzieba) ........
	  ................

2007-02-17 01:11 +0000 [r55004-55077]  Russell Bryant <russell@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 55052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) |
	  3 lines If the pg_config application is found, but there is
	  probably executing it, then consider postgres unavailable. (issue
	  #8637) ........

	* /, codecs/gsm/Makefile: Merged revisions 55050 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r55050 | russell | 2007-02-16 18:31:42 -0600 (Fri, 16 Feb 2007) |
	  3 lines Filter out yet another architecture that does not work
	  with the optimizations in the built-in libgsm. (issue 8637, ovi)
	  ........

	* /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
	  revisions 55006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55006 | russell | 2007-02-16 16:49:42 -0600
	  (Fri, 16 Feb 2007) | 17 lines Merged revisions 55005 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16
	  Feb 2007) | 9 lines Revert the change I did in revisions 54955,
	  54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a
	  conference is created from meetme.conf, it is acceptable behavior
	  that the pin can not be changed until the conference goes away. I
	  also added a note in meetme.conf to describe this behavior. We
	  still have another issue in 1.4 and trunk where some conferences
	  with no users don't go away. That is the real bug that needs to
	  be addressed here. ........ ................

	* apps/app_dumpchan.c: Print the raw read/write formats in the
	  DumpChan application. (issue #9083, junky)

2007-02-16 22:20 +0000 [r55003]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 55002 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r55002 | file | 2007-02-16 17:18:46 -0500 (Fri,
	  16 Feb 2007) | 10 lines Merged revisions 54999 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2
	  lines Do not send indications through ast_indicate in chan_agent
	  but instead go directly to the technology. This way when
	  indications are emulated they happen on the Agent channel and do
	  not screw up formats on the channels. (issue #8439 reported by
	  punkgode) ........ ................

2007-02-16 21:13 +0000 [r54970]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 54969 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r54969 | russell | 2007-02-16 15:12:18 -0600
	  (Fri, 16 Feb 2007) | 13 lines Merged revisions 54955 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16
	  Feb 2007) | 5 lines For conferences that are configured in
	  meetme.conf, check the configuration file every time someone
	  joins the conference instead of only when the conference is first
	  created. This is to ensure that changes to the pin numbers in the
	  config file are always honored. (issue #9073) ........
	  ................

2007-02-16 18:53 +0000 [r54910-54925]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 54924 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2
	  lines Need to check macro extension as well as macro context for
	  directed pickup. ........

	* res/res_features.c, configs/features.conf.sample: Allow the user
	  to specify where to enable the respective features for when a
	  parked call is picked up. (ie: transfers and parking)

2007-02-16 18:04 +0000 [r54890-54901]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 54898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54898 | russell | 2007-02-16 12:03:41 -0600 (Fri, 16 Feb 2007) |
	  4 lines Fix setting "autofallthrough" to yes by default. It was
	  set to enabled in pbx.c. However, if the option was not present
	  in extensions.conf, then pbx_config.c would set it back to
	  disabled. ........

	* /, res/res_features.c: Merged revisions 54888 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54888 | russell | 2007-02-16 11:40:38 -0600 (Fri, 16 Feb 2007) |
	  3 lines Clean up a few coding guidelines issues - spaces to tabs,
	  use sizeof() to pass the size of a static buffer, add spaces ...
	  ........

2007-02-16 17:41 +0000 [r54889]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c, CHANGES, configs/features.conf.sample: Add
	  option to features.conf that enables parking via DTMF on picked
	  up parked calls. (issue #9082 reported by francesco_r)

2007-02-16 17:26 +0000 [r54887]  Jason Parker <jparker@digium.com>

	* /, main/asterisk.c: Merged revisions 54886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54886 | qwell | 2007-02-16 11:25:21 -0600 (Fri, 16 Feb 2007) | 4
	  lines Clarify a restart message. It's silly, but the reporter had
	  a very valid point. Issue 9079 ........

2007-02-16 17:07 +0000 [r54885]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 54884 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2
	  lines Allow directed pickup to pick up the real context instead
	  of the macro context if a Macro is used. (issue #8984 reported by
	  jamesb63) ........

2007-02-16 14:31 +0000 [r54773-54862]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting, whitespace fixes

	* apps/app_voicemail.c: More cleanups of app_voicemail

	* CREDITS, main/channel.c, channels/chan_sip.c,
	  channels/chan_skinny.c, include/asterisk/rtp.h,
	  include/asterisk/channel.h, channels/chan_gtalk.c, CHANGES,
	  include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c:
	  Adding Realtime Text support (T.140) to Asterisk T.140/RFC 2793
	  is a live communication channel, originally created for IP based
	  text phones for hearing impaired. Feels very much like the old
	  Unix talk application. This code is developed and disclaimed by
	  John Martin of Aupix, UK. Tested for interoperability by myself
	  and Omnitor in Sweden, the company that wrote most of the
	  specifications. A big thank you to everyone involved in this.

	* /, channels/chan_sip.c: Merged revisions 54787 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2
	  lines Issue #7541 - Handle multipart attachments to SIP messages
	  - even if boundary is quoted. ........

	* res/res_agi.c: Issue #9068 - make sure we quote HTML characters
	  correctly too (seanbright)

	* /, res/res_agi.c: Merged revisions 54772 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r54772 | oej | 2007-02-16 12:39:55 +0100 (Fri,
	  16 Feb 2007) | 10 lines Merged revisions 54771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
	  lines Issue #9069 - If we open with TH we should not close with
	  /TD. (seanbright) ........ ................

2007-02-16 01:36 +0000 [r54711-54749]  Joshua Colp <jcolp@digium.com>

	* main/acl.c: Rely on ast_gethostbyname to handle IP addresses, not
	  inet_aton. (issue #9056 reported by pj)

	* CHANGES, apps/app_chanspy.c: Add 'o' option to Chanspy which
	  causes it to only listen to audio coming from the channel, and
	  the 'X' option which allows the user to exit to a valid single
	  digit extension. (issue #8137 reported by mnicholson)

	* /, apps/app_speech_utils.c: Merged revisions 54714 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb
	  2007) | 2 lines Don't let dtmf leak over into the engine and let
	  it skew the results... also give DTMF results priority. (issue
	  #9014 reported by surftek) ........

	* main/manager.c: Properly handle an error result from a manager
	  action. This could have left the action list permanently locked
	  for reading.

2007-02-15 20:29 +0000 [r54654-54686]  Olle Johansson <oej@edvina.net>

	* apps/app_voicemail.c: - add some notes, asking for help - insert
	  a few ast_strlen_zero - Doxygen additions - A few more spaces

	* main/io.c: Make file's new comment doxygenified

2007-02-15 16:24 +0000 [r54624]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 54623 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r54623 | file | 2007-02-15 11:19:39 -0500 (Thu,
	  15 Feb 2007) | 10 lines Merged revisions 54622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
	  lines Use a separate variable to indicate execution should
	  continue instead of the return value. (issue #8842 reported by
	  pluto70) ........ ................

2007-02-15 15:53 +0000 [r54574-54599]  Olle Johansson <oej@edvina.net>

	* CHANGES: ...and don't forget to update CHANGES

	* channels/chan_sip.c: Add callgroup and pickupgroup to SIPPEER
	  function. (thanks ramon)

	* CHANGES: Update CHANGES

	* channels/chan_sip.c, configs/extconfig.conf.sample,
	  doc/realtime.txt: Issue #7443 - amdtech - Optionally SIP
	  registrations in another realtime family.

2007-02-15 02:11 +0000 [r54489-54552]  Joshua Colp <jcolp@digium.com>

	* main/io.c: Clean up the I/O context handler.

	* apps/app_flash.c, apps/app_image.c, apps/app_exec.c: Few more
	  code clean ups.

	* apps/app_milliwatt.c: Clean up app_milliwatt code.

	* apps/app_dial.c, /: Merged revisions 54481 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2
	  lines Forward begin DTMF frames as well as end. (issue #9068
	  reported by mhardeman) ........

2007-02-14 20:45 +0000 [r54464-54466]  Olle Johansson <oej@edvina.net>

	* main/asterisk.c: Show version in "core show settings"

	* CHANGES: Updates and re-organization to make it easier to digest
	  this information

	* main/cdr.c, main/manager.c, include/asterisk/config.h,
	  include/asterisk/cdr.h, include/asterisk/manager.h,
	  main/asterisk.c, main/config.c: New CLI command "Core show
	  settings" to list some core settings

2007-02-14 17:14 +0000 [r54404]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 54375 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed,
	  14 Feb 2007) | 10 lines Merged revisions 54373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
	  lines When handling glare on a PRI, move the requested channel
	  rather than hang up the old one. Fix for 8957 and 9011. ........
	  ................

2007-02-14 17:02 +0000 [r54348-54379]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Make documentation match the source
	  code.

	* channels/chan_sip.c: Issue #9060 - host= parameter in sip.conf
	  stopped working caused by outbound proxy patch.

	* channels/chan_sip.c: Add port number to SIPPEER dialplan function

2007-02-14 08:34 +0000 [r54325]  Paul Cadach <paul@odt.east.telecom.kz>

	* codecs/codec_g722.c: I don't know how it worked earlier, but
	  valgrind produces core every time you try to load codec_g722.
	  Fixed. ;-)

2007-02-14 01:12 +0000 [r54291]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 54290 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2
	  lines Add G722 to ast_best_codec. If anyone disagrees with it's
	  placement, feel free to change it. (issue #9045 reported by gork)
	  ........

2007-02-13 22:02 +0000 [r54067-54261]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h, apps/app_meetme.c,
	  res/res_features.c, include/asterisk/cli.h, main/devicestate.c,
	  CHANGES, apps/app_queue.c, funcs/func_devstate.c (added),
	  main/cli.c: This introduces a new dialplan function, DEVSTATE,
	  which allows you to do some pretty cool things. First, you can
	  get the device state of anything in the dialplan: NoOp(SIP/mypeer
	  has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234
	  has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows
	  you to create custom device states so you can control phone lamps
	  directly from the dialplan.
	  Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten =>
	  mycustomlamp,hint,Custom:mycustomlamp

	* /, channels/chan_sip.c: Merged revisions 54204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) |
	  5 lines If we fail to create the SIP socket, then return -1 from
	  reload_config() so that load_module() will return
	  AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
	  spammed with error messages every time chan_sip tries to send a
	  message. ........

	* /, channels/chan_sip.c: Merged revisions 54235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) |
	  2 lines Remove a couple of leftover debug messages ........

	* include/asterisk/devicestate.h, /: Merged revisions 54218 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) |
	  3 lines Fix the documentation on the return values from device
	  state provider registration and deletion. ........

	* main/asterisk.c: Use spaces instead of tabs in the help text for
	  a CLI command

	* main/asterisk.c: Simplify WELCOME_MESSAGE to be a single function
	  call instead of one for each line.

	* include/asterisk/cli.h, main/asterisk.c, main/cli.c: - Constify
	  the format string passed to ast_cli() - Simplify printing out the
	  warranty and license

	* main/dial.c, /, include/asterisk/dial.h: Merged revisions 54103
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) |
	  2 lines Change ast_set_state_callback() to
	  ast_dial_set_state_callback() ........

	* main/dial.c, /, apps/app_meetme.c, apps/app_page.c,
	  include/asterisk/dial.h: Merged revisions 54066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) |
	  4 lines - Add the ability to register a callback to monitor state
	  changes in an asynchronous dial operation. - Rename the various
	  references to "status" to "state" in the dial API ........

2007-02-12 15:48 +0000 [r54003-54004]  Russell Bryant <russell@digium.com>

	* configs/users.conf.sample, /: Merged revisions 54002 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12
	  Feb 2007) | 2 lines Fix a typo where "vmpassword" should be
	  "vmsecret" ........

	* main/channel.c: Simplify a small bit of logic.

2007-02-12 02:44 +0000 [r53980]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_realtime.c: Formatting fixes

2007-02-11 20:49 +0000 [r53914-53953]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Be careful with debug messages in trunk,
	  they tend to stay around for release....

	* channels/chan_sip.c: Small fix in outbound proxy support.

	* channels/chan_sip.c, configs/sip.conf.sample: Add support for
	  outbound proxy for peers and [general] This replaces the older,
	  broken, implementation where a setting in [general] did not do
	  anything and the [peer] part was broken.

	* main/acl.c: Fix debug handling in acl.c

2007-02-10 09:23 +0000 [r53882-53885]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/chan_h323.c: Merged revisions 53881 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) |
	  1 line Fix VLDTMF reception ........

	* /, apps/app_echo.c: Merged revisions 53880 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53880 | pcadach | 2007-02-10 01:08:55 -0800 (Сбт, 10 Фев 2007) |
	  1 line Much simpler than previous one ;-) ........

	* main/channel.c, /: Merged revisions 53879 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) |
	  1 line Provide correct DTMF duration ........

2007-02-10 06:14 +0000 [r53851]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, configure.ac: Merged revisions 53850 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r53850 | kpfleming | 2007-02-10 00:06:08 -0600 (Sat, 10
	  Feb 2007) | 3 lines don't display the --with-imap message unless
	  --with-imap was specified without a path use '-n' instead of '!
	  -z' for tests ........

2007-02-10 00:42 +0000 [r53784-53819]  Russell Bryant <russell@digium.com>

	* include/asterisk/app.h, include/asterisk/utils.h, main/dial.c, /,
	  apps/app_meetme.c, channels/chan_sip.c, doc/sla.txt (added),
	  include/asterisk/dial.h, configs/sla.conf.sample: Merged
	  revisions 53810 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) |
	  24 lines Merge team/russell/sla_rewrite This is a completely new
	  implementation of the SLA functionality introduced in Asterisk
	  1.4. It is now functional and ready for testing. However, I will
	  be adding some additional features over the next week, as well.
	  For information on how to set this up, see
	  configs/sla.conf.sample and doc/sla.txt. In addition to the
	  changes in app_meetme.c for the SLA implementation itself, this
	  merge brings in various other changes: chan_sip: - Add the
	  ability to indicate HOLD state in NOTIFY messages. - Queue HOLD
	  and UNHOLD control frames even if the channel is not bridged to
	  another channel. linkedlists.h: - Add support for rwlock based
	  linked lists. dial.c: - Add the ability to run ast_dial_start()
	  without a reference channel to inherit information from. ........

	* channels/chan_jingle.c: add another dependency

	* /, apps/app_echo.c: Merged revisions 53783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) |
	  4 lines When the Echo() application receives the digit '#', echo
	  that back as well. Since we already sent the BEGIN frame for that
	  digit, it makes sense to send the END as well. ........

2007-02-09 23:53 +0000 [r53782]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/get_moduleinfo, res/res_config_odbc.c, /,
	  build_tools/get_makeopts, funcs/func_odbc.c, res/res_adsi.c,
	  channels/chan_gtalk.c, apps/app_adsiprog.c, apps/app_voicemail.c:
	  Merged revisions 53779-53781 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007)
	  | 2 lines fix awk scripts to work when both MODULEINFO and
	  MAKEOPTS are present in a source file ........ r53780 | kpfleming
	  | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines add some
	  inter-module dependencies ........ r53781 | kpfleming |
	  2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines another
	  dependency ........

2007-02-09 19:39 +0000 [r53717-53750]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 53749 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2
	  lines Temporarily change musicclass on channel to one specified
	  in Dial so that the 'm' option functions properly. (issue #8969
	  reported by christianbee) ........

	* apps/app_queue.c: Clean up documentation of Queue application.
	  (issue #9022 reported by seanbright)

2007-02-09 16:43 +0000 [r53716]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/imapstorage.txt, /, configure, configure.ac: Merged revisions
	  53715 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007)
	  | 2 lines clarify the fact that voicemail IMAP storage cannot be
	  built against a distro's binary c-client library package (at
	  least not at this time) ........

2007-02-09 01:57 +0000 [r53602-53691]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: I'm crazy so I think I'll change the
	  musiconhold classes linked list to read/write as well!

	* main/manager.c: It is with pleasure that I announce the return of
	  rawman support through the HTTP server. (issue #9013 reported by
	  Jynger)

	* /, apps/app_speech_utils.c: Merged revisions 53601 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb
	  2007) | 2 lines Fix timeout issue when utterance is longer then
	  timeout itself. ........

2007-02-08 17:19 +0000 [r53580]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Rename this instance of "busy limit" to
	  "busy level" as well

2007-02-08 16:41 +0000 [r53577]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: rename busy-limit
	  to busy-level, since it is not a limit actually parse the
	  busy-limit option from sip.conf, instead of ignoring it

2007-02-08 13:50 +0000 [r53531-53533]  Tilghman Lesher <tlesher@digium.com>

	* /, main/loader.c: Merged revisions 53532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53532 | tilghman | 2007-02-08 07:47:54 -0600 (Thu, 08 Feb 2007)
	  | 2 lines Issue 9007 - Mutex not released on early return
	  ........

	* /, apps/app_voicemail.c: Merged revisions 53530 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53530 | tilghman | 2007-02-08 07:40:02 -0600
	  (Thu, 08 Feb 2007) | 10 lines Merged revisions 53529 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08
	  Feb 2007) | 2 lines Issue 9003 - If fullname is empty, quote()
	  passes back "\"" ........ ................

2007-02-07 23:56 +0000 [r53465-53498]  Russell Bryant <russell@digium.com>

	* /, main/db1-ast/Makefile: Merged revisions 53497 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07
	  Feb 2007) | 6 lines When building libdb1.a, put the additional
	  flags needed at the beginning of ASTCFLAGS, instead of at the
	  end. This way, we ensure that we find the local headers first
	  before accidentally trying to use headers that exist in locations
	  specified in the ASTCFLAGS passed from the main Makefile. (issue
	  #8637, ovi) ........

	* /, main/Makefile: Merged revisions 53464 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) |
	  4 lines The clean target actually needs to run "distclean" on
	  editline. This is because we need to make sure that its configure
	  script gets executed again, because the CFLAGS we want to pass to
	  editline may have changed. ........

2007-02-07 17:57 +0000 [r53435]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 53434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2
	  lines We can not reliably do P2P bridging with DTMF passing back
	  with compensation if we need to listen for DTMF frames. (issue
	  #8962 reported by caio1982) ........

2007-02-07 17:46 +0000 [r53431]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 53429 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) |
	  7 lines When parsing the NTP timestamp in a sender report
	  message, you are supposed to take the low 16 bits of the integer
	  part, and the high 16 bits of the fractional part. However, the
	  code here was erroneously taking the low 16 bits of the
	  fractional part. It then shifted the result 16 bits down, so the
	  result was always zero. This fix makes it grab the appropriate
	  high 16 bits, instead. (issue #8991, pointed out by
	  andre_abrantes) ........

2007-02-07 17:06 +0000 [r53359-53400]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_playback.c: Merged revisions 53399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53399 | file | 2007-02-07 12:04:44 -0500 (Wed, 07 Feb 2007) | 2
	  lines Directly load say.conf in load_module instead of calling
	  the reload function. (issue #8946 reported by junky) ........

	* /, channels/chan_iax2.c: Merged revisions 53358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53358 | file | 2007-02-07 10:43:39 -0500 (Wed,
	  07 Feb 2007) | 10 lines Merged revisions 53357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
	  lines Fix a few potential memory leaks with realtime users and
	  peers. (issue #8999 reported by bsmithurst) ........
	  ................

2007-02-07 15:35 +0000 [r53356]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 53355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53355 | tilghman | 2007-02-07 09:33:51 -0600
	  (Wed, 07 Feb 2007) | 10 lines Merged revisions 53354 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07
	  Feb 2007) | 2 lines Issue 7440 - Macro called from Macro from the
	  h extension exits prematurely ........ ................

2007-02-07 09:51 +0000 [r53334]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 53324 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53324 | crichter | 2007-02-07 10:22:44 +0100
	  (Mi, 07 Feb 2007) | 9 lines Merged revisions 52843 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30
	  Jan 2007) | 1 line fixed some possible segfaults. also fixed an
	  very important bug which occurs on high load (when calls are very
	  fast generated) ........ ................

2007-02-07 05:25 +0000 [r53247-53297]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_jabber.c: Merged revisions 53294 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53294 | tilghman | 2007-02-06 23:24:31 -0600 (Tue, 06 Feb 2007)
	  | 2 lines Text fix for jabber reload command (reported by bkruse
	  via IRC) ........

	* main/manager.c, /: Merged revisions 53246 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53246 | tilghman | 2007-02-06 01:00:52 -0600
	  (Tue, 06 Feb 2007) | 10 lines Merged revisions 53245 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06
	  Feb 2007) | 2 lines Issue 8987 - Status could return two
	  responses (mnicholson) ........ ................

2007-02-05 21:55 +0000 [r53200]  Olle Johansson <oej@edvina.net>

	* main/io.c: Doxygen formatting changes

2007-02-05 17:06 +0000 [r53151-53153]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_playback.c: Merged revisions 53152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2
	  lines Ensure say_cfg is NULL when the module is loaded. (issue
	  #8946 reported by junky) ........

	* /, apps/app_playback.c: Merged revisions 53150 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2
	  lines Unregister Playback CLI commands as well as dialplan
	  application. (issue #8946 reported by junky) ........

2007-02-05 00:30 +0000 [r53144]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 53143 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3
	  lines Add some comments on queue system behaviour and how it
	  affects the SIP channel ........

2007-02-03 22:06 +0000 [r53140-53142]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt: Deprecate SetCallerPres application

	* apps/app_setcallerid.c, funcs/func_callerid.c: Add CALLERPRES
	  dialplan function and deprecate SetCallerPres application

	* funcs/func_odbc.c: Fix compiler warnings

2007-02-03 21:06 +0000 [r53139]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 53138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2
	  lines Make SIPDtmfMode application work with recent capability
	  changes, and also fix an RTP stack issue when the auto option was
	  used. (issue #8972 reported by mdu113) ........

2007-02-03 20:46 +0000 [r53137]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, /: Merged revisions 53136 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53136 | russell | 2007-02-03 14:44:20 -0600
	  (Sat, 03 Feb 2007) | 12 lines Merged revisions 53133 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03
	  Feb 2007) | 4 lines set the DIALSTATUS variable to contain
	  "INVALIDARGS" when the dial application exits early because of
	  invalid arguments instead of just leaving it empty. (issue #8975)
	  ........ ................

2007-02-03 10:12 +0000 [r53132]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx: Merged revisions 53131 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) |
	  1 line Remove quote from H.323 vendor string because due to
	  compatibilities with Nortel Meridian CS1000 reported at
	  www.voip-info.org ........

2007-02-02 20:05 +0000 [r53126-53127]  Olle Johansson <oej@edvina.net>

	* doc/queue.txt: Update with info about SIP channels and queues

	* doc/queue.txt (added): Adding a template for documentation on
	  call queues. Please help us add to this! Thanks /OEJ and BJ

2007-02-02 18:21 +0000 [r53111-53125]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Add onHold value to sip show inuse as well.

	* /, main/rtp.c: Merged revisions 53120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2
	  lines Correct a copy/pasted error message line for RTCP. ........

	* /, main/config.c: Merged revisions 53118 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53118 | file | 2007-02-02 10:59:53 -0600 (Fri,
	  02 Feb 2007) | 10 lines Merged revisions 53117 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
	  lines Pass the glob expanded filename to process_text_line so
	  that error messages contain the actual filename, not the original
	  include one. (issue #8959 reported by tzafrir) ........
	  ................

	* Makefile, /: Merged revisions 53114 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53114 | file | 2007-02-02 09:29:35 -0600 (Fri, 02 Feb 2007) | 2
	  lines Add systemname to asterisk.conf generation per recent
	  discussions about it. (issue #8968 reported by blitzrage)
	  ........

	* main/devicestate.c: Clean up ast_device_state. It's pretty now!

	* main/devicestate.c: Switch the devicestate thread to operate the
	  same way as the logging thread. Pops all entries off the list to
	  be processed, resets the list back to a clean state, and
	  processes each entry. The thread won't have to acquire the list
	  lock again until it checks to see if there are more to process.

	* main/devicestate.c: Read/write lockify the devicestate stuff a
	  bit.

2007-02-02 00:26 +0000 [r53110]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Patch based on
	  this patch with small changes for trunk... Merged revisions 53109
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4
	  lines Disable the direct p2p RTP call setup in SIP. You can
	  enable it in sip.conf, but it is now considered experimental
	  until we solve the AST_CONTROL_ANSWER with payload and videocaps
	  stuff. ........

2007-02-01 22:26 +0000 [r53098-53105]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 53104 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53104 | file | 2007-02-01 16:24:32 -0600 (Thu,
	  01 Feb 2007) | 10 lines Merged revisions 53103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
	  lines Copy noncodeccapability over to the joint variable so that
	  telephone-event will get transmitted in the sent INVITE. ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 53097 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53097 | file | 2007-02-01 15:54:28 -0600 (Thu,
	  01 Feb 2007) | 10 lines Merged revisions 53095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
	  lines Don't negotiate RFC2833 when not configured to do so.
	  (issue #8799 reported by mdu113) ........ ................

2007-02-01 21:27 +0000 [r53094]  Russell Bryant <russell@digium.com>

	* /, funcs/func_strings.c: Merged revisions 53093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53093 | russell | 2007-02-01 15:24:52 -0600 (Thu, 01 Feb 2007) |
	  2 lines Fix the FIELDQTY function to not crash. (reported by
	  blitzrage and Corydon on IRC) ........

2007-02-01 21:17 +0000 [r53092]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 53085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4
	  lines - Clean INC_COUNT flag when we decrement call counter - If
	  it's still set at time of dialog destruction, make sure we
	  decrement the device call counter properly before we destroy the
	  dialog ........

2007-02-01 21:12 +0000 [r53087-53089]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 53088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53088 | file | 2007-02-01 15:11:28 -0600 (Thu,
	  01 Feb 2007) | 10 lines Merged revisions 53084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2
	  lines Return previous behavior of having MOH pick up where it was
	  left off. (issue #8672 reported by sinistermidget) ........
	  ................

2007-02-01 20:44 +0000 [r53080-53083]  Olle Johansson <oej@edvina.net>

	* /, apps/app_queue.c: Merged revisions 53081 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53081 | oej | 2007-02-01 21:38:58 +0100 (Thu, 01 Feb 2007) | 2
	  lines Change debug level for state change message that is not
	  really informative when debugging app_queue ........

	* channels/chan_sip.c, configs/sip.conf.sample: Implementing
	  "busy-limit". If you set call limit and busy limit, chan_sip will
	  indicate BUSY for a device that has reached the busy limit and
	  allow calls up to the call limit, allowing for call transfers
	  (that generate a new call). If you only set call limit, chan_sip
	  will not indicate BUSY until that limit is filled. This affects
	  SIP subscriptions, call queues and manager applications.

	* /, channels/chan_sip.c: Merged revisions 53079 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2
	  lines Cleaning up the devicestate callback function ........

2007-02-01 20:14 +0000 [r53076-53078]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 53075 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53075 | tilghman | 2007-02-01 14:09:52 -0600
	  (Thu, 01 Feb 2007) | 10 lines Merged revisions 53074 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01
	  Feb 2007) | 2 lines Bug 8965 - Allow FIELDQTY to work with both
	  variables and dialplan functions ........ ................

2007-02-01 19:34 +0000 [r53073]  Joshua Colp <jcolp@digium.com>

	* /, main/asterisk.c: Merged revisions 53072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53072 | file | 2007-02-01 13:33:33 -0600 (Thu, 01 Feb 2007) | 2
	  lines Add missing 'F' letter to getopt so it magically becomes a
	  valid option. (issue #8960 reported by tzafrir) ........

2007-02-01 19:27 +0000 [r53071]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, funcs/func_strings.c: Merged revisions 53070 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53070 | tilghman | 2007-02-01 13:21:20 -0600
	  (Thu, 01 Feb 2007) | 10 lines Merged revisions 53069 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01
	  Feb 2007) | 2 lines No wonder FIELDQTY doesn't work with
	  functions... the documentation in pbx.c was wrong ........
	  ................

2007-02-01 19:04 +0000 [r53067]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Signal HOLD status to phones that subscribe
	  for status.

2007-02-01 17:42 +0000 [r53065-53066]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 53064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2
	  lines Fix silly logic. We really want to write UDPTL frames out
	  when the call is up. ........

	* main/db1-ast/hash/hash.c: Make trunk compile under dev mode.

2007-02-01 16:42 +0000 [r53063]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Merged revisions 53062 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb
	  2007) | 2 lines Add explanation of port= in combination with
	  defaultip= (thanks jsmith) ........

2007-02-01 14:43 +0000 [r53061]  Russell Bryant <russell@digium.com>

	* apps/app_rpt.c: Remove duplicate calls to pthread_attr_destroy()
	  that I put in yesterday by accident.

2007-02-01 11:16 +0000 [r53058-53059]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/chan_h323.c: Oops -- Merged revisions 53057 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) |
	  1 line chan_h323 is very stable, so let it built by default
	  ........

2007-02-01 00:38 +0000 [r53054]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Formatting changes

2007-02-01 00:24 +0000 [r53051-53053]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 53052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2
	  lines When going on hold have the side that was put on hold
	  reinvite back to Asterisk. When going off hold have the side that
	  was taken off hold reinvited back to the other party. ........

	* /, main/rtp.c: Merged revisions 53050 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2
	  lines Add more frame types to forward in the RTP bridge loops.
	  ........

2007-01-31 21:35 +0000 [r52905-53047]  Russell Bryant <russell@digium.com>

	* main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
	  channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c,
	  main/cdr.c, main/manager.c, pbx/pbx_spool.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
	  pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c: Merged
	  revisions 53046 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53046 | russell | 2007-01-31 15:32:08 -0600
	  (Wed, 31 Jan 2007) | 11 lines Merged revisions 53045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31
	  Jan 2007) | 3 lines Fix a bunch of places where
	  pthread_attr_init() was called, but pthread_attr_destroy() was
	  not. ........ ................

	* /, apps/app_userevent.c: Merged revisions 53042 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53042 | russell | 2007-01-31 12:18:25 -0600 (Wed, 31 Jan 2007) |
	  2 lines Remove an extra \r\n from manager user events. (issue
	  #8955, mnicholson) ........

	* /, main/rtp.c: Merged revisions 53040 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r53040 | russell | 2007-01-31 11:45:05 -0600
	  (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31
	  Jan 2007) | 3 lines Use the proper format string to print
	  unsigned values in the rtp debug output. (issue #8954, wmis)
	  ........ ................

	* /, apps/app_queue.c: Merged revisions 53037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53037 | russell | 2007-01-31 11:39:28 -0600 (Wed, 31 Jan 2007) |
	  3 lines Only changed the paused status in an existing queue
	  member if the paused column exists. ........

	* /, apps/app_queue.c: Merged revisions 53035 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53035 | russell | 2007-01-31 11:34:22 -0600 (Wed, 31 Jan 2007) |
	  4 lines Instead of always creating a realtime queue member as
	  unpaused, read the "paused" column and use that value for the
	  paused status of the member. (issue #8949, jmls) ........

	* /, contrib/init.d/rc.suse.asterisk: Merged revisions 53001 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r53001 | russell | 2007-01-30 17:38:42 -0600 (Tue, 30 Jan 2007) |
	  2 lines Update init script for SuSE 10. (issue #8363, johnlange)
	  ........

	* /, doc/cdrdriver.txt: Merged revisions 52999 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52999 | russell | 2007-01-30 17:30:34 -0600 (Tue, 30 Jan 2007) |
	  2 lines Add documentation for using cdr_pgsql. (issue #8942,
	  lters) ........

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  codecs/codec_gsm.c: Merged revisions 52997 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) |
	  5 lines When we are checking for a system installed version of
	  libgsm, we need to check for gsm.h as well. Furthermore, when
	  checking for this header, it may be located in a gsm/ sub
	  directory, so check for that, as well. (issue #8773) ........

	* /, channels/chan_sip.c: Merged revisions 52952 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) |
	  5 lines Only set the DTMF flag on the rtp structure if the DTMF
	  mode is actually RFC2833, not just that it is not INFO. This
	  makes it get set for inband DTMF as well, which is not valid.
	  (issue #8936) ........

	* /, main/asterisk.c: Merged revisions 52904 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52904 | russell | 2007-01-30 11:19:39 -0600
	  (Tue, 30 Jan 2007) | 17 lines Merged revisions 52903 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30
	  Jan 2007) | 9 lines The SIGHUP handler was implemented to allow
	  admins to send SIGHUP to a running Asterisk process to reload the
	  configuration. However, doing the actual reload in the signal
	  handler itself is a very bad thing to do, because the reload
	  process includes calling non-reentrant functions such as
	  malloc/calloc/etc. If Asterisk is running in the background, then
	  the reload will happen immediately. However, if running in
	  console mode, the reload doesn't work until something is typed at
	  the console. That sort of defeats the purpose, but I don't see an
	  easy way to get around it at this point. ........
	  ................

2007-01-30 15:39 +0000 [r52858-52860]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Use provided variable for name instead of
	  one in the structure since the structure was just allocated and
	  will be NULL. (issue #8938 reported by st41ker)

2007-01-30 09:13 +0000 [r52818-52820]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, res/res_odbc.c: Merged revisions 52808 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52808 | pcadach | 2007-01-30 00:34:26 -0800 (Втр, 30 Янв 2007) |
	  1 line Don't play with free()'d pointers ........

	* /, configure, acinclude.m4: Merged revisions 52807 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r52807 | pcadach | 2007-01-30 00:33:22 -0800 (Втр, 30
	  Янв 2007) | 1 line Handle non-standard OpenH323/PWLib library
	  names ........

2007-01-30 00:16 +0000 [r52764]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 52763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52763 | russell | 2007-01-29 18:15:50 -0600
	  (Mon, 29 Jan 2007) | 13 lines Merged revisions 52762 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29
	  Jan 2007) | 5 lines Fix the extraction of the timestamp from
	  video frames. It was using the mapping for a mini-frame instead
	  of a video-frame, which caused it to get invalid data. (issue
	  #8795, mihai) ........ ................

2007-01-29 23:45 +0000 [r52718]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 52717 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon,
	  29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2
	  lines Now that filename is part of the structure and since it
	  comes before postprocess... we have to add it to our postprocess
	  line. (reported on asterisk-dev by Boris Bakchiev) ........
	  ................

2007-01-29 22:58 +0000 [r52692-52696]  Russell Bryant <russell@digium.com>

	* /, main/Makefile: Merged revisions 52695 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) |
	  2 lines Add a missing quotation mark. This was pointed out by
	  jcmoore on #asterisk-dev. ........

	* main/manager.c, /: Merged revisions 52688 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52688 | russell | 2007-01-29 16:55:41 -0600 (Mon, 29 Jan 2007) |
	  3 lines Remove a recursive lock of the manager session. This was
	  pointed out by zandbelt in issue #8711. ........

2007-01-29 22:13 +0000 [r52680]  Tilghman Lesher <tlesher@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 52679 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52679 | tilghman | 2007-01-29 16:12:12 -0600 (Mon, 29 Jan 2007)
	  | 2 lines Argument number correction ........

2007-01-29 21:37 +0000 [r52646-52648]  Russell Bryant <russell@digium.com>

	* /, main/Makefile: Merged revisions 52647 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) |
	  3 lines ASTLDFLAGS needs to be passed to the editline configure
	  script as LDFLAGS. (issue #8928, zandbelt) ........

	* /, main/rtp.c: Merged revisions 52645 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) |
	  6 lines Fix a problem with packet-to-packet bridging and DTMF
	  mode translation. P2P bridging can only be used when the DTMF
	  modes don't match if the core is monitoring DTMF in both
	  directions. Then, the core will handle the translation.
	  Otherwise, this bridging method can not be used. (issue #8936)
	  ........

2007-01-29 21:03 +0000 [r52635]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Only use locking for bridge information if intense
	  P2P bridging is enabled.

2007-01-29 20:51 +0000 [r52612-52613]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: The changes for trunk are less extensive, but
	  include - changing the actionlock to a rwlock - not locking the
	  session before doing the action callback The crash issue in 8711
	  should not be an issue here. Merged revisions 52611 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r52611 | russell | 2007-01-29 14:39:20 -0600 (Mon, 29
	  Jan 2007) | 10 lines The session lock can not be held while
	  calling action callbacks. If so, then when the WaitEvent callback
	  gets called, then no event can happen because the session can't
	  be locked by another thread. Also, the session needs to be locked
	  in the HTTP callback when it reads out the output string. This
	  fixes the deadlock reported in both 8711 and 8934. Regarding
	  issue 8711, there still may be an issue. If there is a second
	  action requested before the processing of the first action is
	  finished, there could still be some corruption of the output
	  string buffer used to build the result. (issue #8711, #8934)
	  ........

	* apps/app_voicemail.c: Resolve some warnings when not building
	  with IMAP_STORAGE

2007-01-29 20:22 +0000 [r52580-52610]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Change vmstates list to use linked list
	  macros.

	* apps/app_voicemail.c: Code cleanup of IMAP storage support in
	  app_voicemail.

	* /, apps/app_voicemail.c: Merged revisions 52572 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52572 | file | 2007-01-29 13:59:41 -0500 (Mon, 29 Jan 2007) | 2
	  lines Use ast_calloc instead of malloc. ........

2007-01-29 17:49 +0000 [r52524-52525]  Joshua Colp <jcolp@digium.com>

	* CHANGES, main/cli.c: Add core show channels count CLI command.
	  (issue #8932 reported by mr_mehul_shah)

	* /, apps/app_voicemail.c: Merged revisions 52523 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52523 | file | 2007-01-29 12:33:19 -0500 (Mon, 29 Jan 2007) | 2
	  lines Set quota information to 0 when creating a vm_state. (issue
	  #8924 reported by neutrino88) ........

2007-01-29 17:03 +0000 [r52522]  Russell Bryant <russell@digium.com>

	* /, main/jitterbuf.c, include/jitterbuf.h: Merged revisions
	  52494,52506 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) |
	  4 lines Fixed problem with jitterbuf, whereas it would not
	  complain about, and would allow itself to be overfilled (per the
	  max_jitterbuf parameter). Now it rejects any data over and above
	  that size, and complains about it. ........ r52506 | russell |
	  2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines Clean up a
	  few things in the last commit to the adaptive jitterbuffer code.
	  - Specifically indicate to the compiler that the "dropem"
	  variable only needs one but. - Change formatting to conform to
	  coding guidelines. ........

2007-01-28 05:18 +0000 [r52463]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 52462 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r52462 | tilghman | 2007-01-27 23:15:07 -0600 (Sat, 27
	  Jan 2007) | 2 lines Suggested change to fix normal usage of
	  --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
	  list) ........

2007-01-27 02:15 +0000 [r52332-52417]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 52416 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52416 | file | 2007-01-26 21:13:41 -0500 (Fri,
	  26 Jan 2007) | 10 lines Merged revisions 52415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
	  lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
	  follow documentation. (issue #7677 reported by amilcar) ........
	  ................

	* /, channels/chan_iax2.c: Merged revisions 52370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52370 | file | 2007-01-26 19:08:18 -0500 (Fri,
	  26 Jan 2007) | 10 lines Merged revisions 52360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
	  lines Make the last context entry read in the dominant one.
	  (issue #8918 reported by pj) ........ ................

	* /, main/file.c: Merged revisions 52335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52335 | file | 2007-01-26 18:46:47 -0500 (Fri, 26 Jan 2007) | 2
	  lines Fix core show file formats CLI command. ........

	* main/file.c, main/image.c: Convert some more stuff to read/write
	  lists.

2007-01-25 22:49 +0000 [r52168-52308]  Joshua Colp <jcolp@digium.com>

	* CHANGES, main/db.c: Add DBDel and DBDelTree manager commands.
	  (issue #8516 reported by dprado)

	* /, main/jitterbuf.c: Merged revisions 52265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52265 | file | 2007-01-25 14:18:33 -0500 (Thu,
	  25 Jan 2007) | 10 lines Merged revisions 52264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
	  lines Allow dequeueing of frames with negative timestamp by
	  moving jitterbuffer frames check to jb_next. (issue #8546
	  reported by harmen) ........ ................

	* channels/chan_sip.c: Use atomic operation functions for
	  use/ringing/hold manipulation.

	* /, channels/chan_sip.c: Merged revisions 52210 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2
	  lines Drop out variables I accidentally put in. ........

	* /, channels/chan_sip.c: Merged revisions 52208 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2
	  lines Decrement onHold count if we are hung up on and still on
	  hold. (issue #8909 reported by alexh42) ........

	* /, apps/app_mixmonitor.c: Merged revisions 52163 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed,
	  24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2
	  lines Add another note about audio files being played back to
	  each bridged party. (issue #8718 reported by ppyy) ........
	  ................

2007-01-25 01:38 +0000 [r52108-52161]  Russell Bryant <russell@digium.com>

	* configs/users.conf.sample, /, apps/app_voicemail.c: Merged
	  revisions 52160 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52160 | russell | 2007-01-24 19:37:16 -0600 (Wed, 24 Jan 2007) |
	  2 lines By suggestion from kpfleming last week, change
	  "vmpassword" to "vmsecret". ........

	* /, include/asterisk/dial.h: Merged revisions 52107 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24
	  Jan 2007) | 3 lines Fix the formatting of doxygen comments to
	  properly indicate that the comment documents the previous entity,
	  as opposed to the next one. ........

2007-01-24 20:35 +0000 [r52053-52086]  Steve Murphy <murf@digium.com>

	* UPGRADE.txt, apps/app_chanisavail.c: As per bug 8859 (Add option
	  to revert old ChanIsAvail() with 's' option behavior), this
	  update makes the 't' option available, which calls
	  ast_parse_device_state instead of ast_device_state. This option
	  will not dive into the channel driver to find the status of the
	  device (which could be good if sip devicestate isn't returning
	  full status, for various reasons).

	* utils/Makefile, /, utils/check_expr.c: Merged revisions 52052 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r52052 | murf | 2007-01-24 11:26:22 -0700 (Wed,
	  24 Jan 2007) | 9 lines Merged revisions 52002 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
	  line updated check_expr via 8322 (refactoring of expression
	  checking impl); elfring contributed a nice code reorg, I
	  contributed some time to get it working again, better messages
	  ........ ................

2007-01-24 18:23 +0000 [r52025-52050]  Joshua Colp <jcolp@digium.com>

	* main/dial.c (added), /, apps/app_page.c, main/Makefile,
	  include/asterisk/dial.h (added): Merged revisions 52049 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2
	  lines Merge in dialing API and the app_page that uses it. (issue
	  #BE-118) ........

	* /, channels/chan_sip.c: Merged revisions 52016 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2
	  lines Fix changing channel formats when joint capability changes
	  and there are no audio formats... I didn't break it originally!
	  (issue #8535 reported by ivoc) ........

2007-01-24 09:42 +0000 [r51905-51933]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 51931 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3
	  lines Show capabilities *and* preference in general settings in
	  "sip show settings" (reported by Clona/Telio - Thanks!) ........

	* include/asterisk/http.h, main/http.c: Doxygen updates

	* funcs/func_rand.c, funcs/func_base64.c, funcs/func_module.c,
	  funcs/func_md5.c, funcs/func_db.c, funcs/func_version.c,
	  funcs/func_timeout.c, funcs/func_env.c, funcs/func_math.c,
	  funcs/func_strings.c, funcs/func_sha1.c, funcs/func_logic.c,
	  funcs/func_uri.c, funcs/func_global.c, funcs/func_enum.c,
	  funcs/func_groupcount.c, funcs/func_odbc.c, funcs/func_shell.c,
	  funcs/func_channel.c, funcs/func_cdr.c, funcs/func_callerid.c:
	  Doxygen update

	* main/udptl.c: Adding some doxygen for udptl.c

2007-01-24 01:00 +0000 [r51850]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 51848 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51848 | russell | 2007-01-23 18:59:58 -0600
	  (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23
	  Jan 2007) | 6 lines Fix an issue related to synchronization of
	  recordings when using Monitor(). The bug is a miscalculation of
	  the amount to seek the stream for writing to disk when the number
	  of samples coming in and out of a channel do not match up. (issue
	  #8298, #8887, report and patch by guillecabeza, patch files
	  created and testing done by whoiswes) ........ ................

2007-01-24 00:22 +0000 [r51831]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Close file after we do the translation, and map
	  memory for both reading/writing. (issue #8886 reported by
	  cwegener)

2007-01-24 00:21 +0000 [r51830]  Russell Bryant <russell@digium.com>

	* /, apps/app_while.c: Merged revisions 51829 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51829 | russell | 2007-01-23 18:19:55 -0600
	  (Tue, 23 Jan 2007) | 12 lines Merged revisions 51828 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23
	  Jan 2007) | 4 lines Don't set a new value for the END_ variable
	  on the channel before using the old value. If you do, it will
	  lead to accessing a memory address that has been free()'d. (issue
	  #8895, arkadia) ........ ................

2007-01-23 22:59 +0000 [r51801]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, channels/chan_zap.c, /,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_alsa.c,
	  channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c:
	  Merged revisions 51788 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2
	  lines Update channel drivers to use module referencing so that
	  unloading them while in use will not result in crashes. (issue
	  #8897 reported by junky) ........

2007-01-23 22:09 +0000 [r51751-51787]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 51781 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) |
	  6 lines Fix some bugs in process_message(). The manager session
	  lock needs to be held when sending some sort of response, or
	  calling one of the manager action callbacks. This resolves an
	  issue where people using the GUI would get random crashes when
	  they start clicking around a lot. (issue #8711, reported and
	  debugged by zandbelt) ........

	* main/manager.c, /: Merged revisions 51750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) |
	  4 lines When traversing the list of manager actions, the iterator
	  needs to be initialized to the list head *after* locking the
	  list. Also, lock the actions list in one place it is being
	  accessed where it was not being done. ........

2007-01-23 20:36 +0000 [r51684-51717]  Steve Murphy <murf@digium.com>

	* /, res/res_features.c: Merged revisions 51716 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51716 | murf | 2007-01-23 13:32:54 -0700 (Tue, 23 Jan 2007) | 1
	  line this mod from 8593 (dstchannel in cdr is empty when transfer
	  call). ........

	* /, main/callerid.c: Merged revisions 51683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1
	  line via 8748 (callerid.c loses name when returning
	  PRIVATE_NUMBER flag), the user suggested this mod, saying it
	  would allow 'WITHHELD' to appear in the name field, which would
	  be useful ........

2007-01-23 15:36 +0000 [r51659]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8817 - Registry corruption when
	  packet retransmits fail. (tootai, patchy by oej)

2007-01-23 06:56 +0000 [r51623]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/chan_h323.c, channels/Makefile: Merged revisions
	  51615 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) |
	  1 line Do not abort Asterisk startup if h323 configuration file
	  not found (reported by mithraen) ........

2007-01-23 04:45 +0000 [r51463-51592]  Joshua Colp <jcolp@digium.com>

	* doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Make 'H'
	  command do as advertised and add 'E' and 'V' commands to
	  ExternalIVR. (issue #8165 reported by mnicholson)

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add SRV
	  Lookup support on outbound calls to chan_iax2. It's listed in the
	  RFC so we might want to support it and please don't hurt me Marko
	  ... (issue #7812 reported by drorlb)

	* /, channels/chan_sip.c: Merged revisions 51558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2
	  lines Only change audio formats on the channel if we have an
	  audio format to change to. (issue #8535 reported by ivoc)
	  ........

	* /: No more conflicts on properties! svnmerge-block be gone!

	* /, res/res_musiconhold.c: Merged revisions 51513 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51513 | file | 2007-01-22 20:45:04 -0500 (Mon,
	  22 Jan 2007) | 10 lines Merged revisions 51512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan 2007) | 2
	  lines Yield before reading from zaptel timing source under
	  Solaris so that other threads get a chance to do things. (issue
	  #7875 reported by bob) ........ ................

	* main/autoservice.c: Might as well go crazy here too and make the
	  autoservice list read/write.

	* main/pbx.c, main/autoservice.c, main/frame.c, main/say.c,
	  main/jitterbuf.c, main/devicestate.c, main/utils.c, main/enum.c,
	  main/fskmodem.c, main/config.c, main/cli.c, main/io.c,
	  main/channel.c, main/cdr.c, main/abstract_jb.c, main/logger.c,
	  main/callerid.c, main/file.c, main/app.c, main/image.c,
	  main/alaw.c, main/asterisk.c, main/dsp.c: Cosmetic changes. Make
	  main source files better conform to coding guidelines and
	  standards. (issue #8679 reported by johann8384)

	* main/rtp.c: Change RTP protos list to be read/write. Most of the
	  time it's only going to be read so making it use mutex locks was
	  a waste.

	* main/rtp.c: Make the RTP stack better conform to coding
	  guidelines. (issue #8679 reported by johann8384)

2007-01-22 19:42 +0000 [r51413]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 51409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51409 | murf | 2007-01-22 12:28:51 -0700 (Mon, 22 Jan 2007) | 1
	  line This fixes 8836, according to dnatural ........

2007-01-22 19:22 +0000 [r51408]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 51407 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon,
	  22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2
	  lines Move filestream creation to Mixmonitor loop. This will
	  prevent a blank file from being created if no frames ever pass
	  through to be recorded. (issue #7589 reported by steve_mcneil)
	  ........ ................

2007-01-22 19:00 +0000 [r51405]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Remove (to quote Rizzo) "useless" variable.

2007-01-21 03:25 +0000 [r51353]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Fix bug introduced during constification (reported by
	  tzanger via IRC)

2007-01-20 18:27 +0000 [r51352]  Russell Bryant <russell@digium.com>

	* include/asterisk/frame.h: Add a comment that the frame type
	  constants are transmitted directly over IAX2.

2007-01-20 06:54 +0000 [r51349-51351]  Jason Parker <jparker@digium.com>

	* /, configs/say.conf.sample: Merged revisions 51350 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan
	  2007) | 5 lines Fix Italian numeral support in say.conf for
	  "_[2-9]00" case. "2131" would've translated to something along
	  the lines of (pardon my..Italian {or lack thereof})
	  "duecentocentotrentuno", which makes no sense at all. ........

	* /, configs/say.conf.sample: Merged revisions 51348 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan
	  2007) | 8 lines Fix German language support in say.conf Properly
	  support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the
	  same format as zweiundzwanzig (as do all other "_ZX" spoken
	  numerals) Fix support for numbers in the 10,000,000 to 99,999,999
	  range. Add support for numbers in the 100,000,000 to 999,999,999
	  range. ........

2007-01-20 00:13 +0000 [r51314-51344]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 51343 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51343 | russell | 2007-01-19 18:13:06 -0600 (Fri, 19 Jan 2007) |
	  2 lines Remove an unused instance of an unnamed enum. ........

	* /, apps/app_meetme.c: Merged revisions 51341 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51341 | russell | 2007-01-19 16:19:10 -0600 (Fri, 19 Jan 2007) |
	  2 lines Remove another duplicated definition ........

	* /, apps/app_meetme.c: Merged revisions 51339 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51339 | russell | 2007-01-19 15:20:20 -0600 (Fri, 19 Jan 2007) |
	  2 lines Remove a variable that was declared twice. ........

	* /, codecs/gsm/Makefile: Merged revisions 51331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51331 | russell | 2007-01-19 13:30:54 -0600 (Fri, 19 Jan 2007) |
	  3 lines Add a couple more processors that need optimizations
	  excluded. (issue #8637) ........

	* /, channels/chan_gtalk.c: Merged revisions 51328 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19
	  Jan 2007) | 5 lines Fix VLDTMF support in chan_gtalk.
	  AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
	  thing. So, a digit would have been interpreted incorrectly here.
	  Since the channel driver will always have the begin and end
	  callbacks called for a digit, only support the button-down and
	  button-up messages. ........

	* /, .cleancount: Merged revisions 51326 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51326 | russell | 2007-01-19 13:02:55 -0600 (Fri, 19 Jan 2007) |
	  2 lines Bump the cleancount since my last commit changed the
	  channel structure. ........

	* channels/chan_zap.c, channels/chan_local.c, main/frame.c, /,
	  channels/chan_sip.c, channels/chan_agent.c,
	  include/asterisk/channel.h, channels/chan_gtalk.c,
	  channels/chan_iax2.c, channels/chan_oss.c, main/rtp.c,
	  main/channel.c, channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_alsa.c, channels/chan_mgcp.c: Merged revisions
	  51311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) |
	  23 lines Merge the changes from the /team/group/vldtmf_fixup
	  branch. The main bug being addressed here is a problem introduced
	  when two SIP channels using SIP INFO dtmf have their media
	  directly bridged. So, when a DTMF END frame comes into Asterisk
	  from an incoming INFO message, Asterisk would try to emulate a
	  digit of some length by first sending a DTMF BEGIN frame and
	  sending a DTMF END later timed off of incoming audio. However,
	  since there was no audio coming in, the DTMF_END was never
	  generated. This caused DTMF based features to no longer work. To
	  fix this, the core now knows when a channel doesn't care about
	  DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If
	  this is the case, then Asterisk will not emulate a digit of some
	  length, and will instead just pass through the single DTMF END
	  event. Channel drivers also now get passed the length of the
	  digit to their digit_end callback. This improves SIP INFO support
	  even further by enabling us to put the real digit duration in the
	  INFO message instead of a hard coded 250ms. Also, for an incoming
	  INFO message, the duration is read from the frame and passed into
	  the core instead of just getting ignored. (issue #8597, maybe
	  others...) ........

2007-01-19 18:00 +0000 [r51308-51312]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/strings.h: As the comment in the diff says:
	  AST_INLINE_API() is a macro that takes a block of code as an
	  argument. Using preprocessor #directives in the argument is not
	  supported by all compilers, and it is a bit of an obfuscation
	  anyways, so avoid it. As a workaround, define a macro that
	  produces either its argument or nothing, and use that instead of
	  #ifdef/#endif within the argument to AST_INLINE_API().

	* main/rtp.c: in the interest of portability, avoid using %zd when
	  all we need is to print is an integer that fits in 16 bits.

	* channels/chan_iax2.c: sizeof() is compatible with format %d so
	  don't be too picky on printf formats.

	* channels/chan_zap.c: remove variable declaration in the middle of
	  a block

2007-01-19 17:19 +0000 [r51303-51305]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in: Regenerate configure
	  script to reflect recent zaptel changes

	* include/asterisk/zapata.h: Include tonezone.h for linux, too

	* main/asterisk.c: Merged revisions 51302 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51302 | russell | 2007-01-19 10:56:17 -0600
	  (Fri, 19 Jan 2007) | 12 lines Merged revisions 51300 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19
	  Jan 2007) | 4 lines Fix a memory leak on command line tab
	  completion. The container for the matches was freed, but the
	  individual matches themselves were not. (issue #8851, arkadia)
	  ........ ................

2007-01-19 16:51 +0000 [r51297-51301]  Luigi Rizzo <rizzo@icir.org>

	* main/Makefile: forgot to add AST_LIBS += $(BKTR_LIB)

	* main/channel.c: include "asterisk/zapata.h" to get the zaptel
	  headers. this should be the last one left around...

	* channels/chan_zap.c: whoops, fix a cut&paste error...

	* channels/chan_zap.c: slight change to the initialization of a
	  structure, also using '\0' to make it clear we need a (char)0

2007-01-19 16:30 +0000 [r51296]  Russell Bryant <russell@digium.com>

	* main/manager.c: Break out of the config processing loop for
	  manager.conf once the correct user has been found so that 'cat'
	  is non-NULL. This way, users.conf is only checked when necessary.
	  (issue #8852, akohlsmith, committed patch a bit different)

2007-01-19 16:28 +0000 [r51285-51295]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_zap.c: include "asterisk/zapata.h" to get the
	  zaptel headers.

	* codecs/codec_zap.c: include "asterisk/zapata.h" to get the zaptel
	  headers

	* apps/app_meetme.c: include "asterisk/zapata.h" instead of testing
	  for the location of the header files. On passing, add a cast to
	  insure -Werror clean compilation on FreeBSD 6.x, where time_t
	  does not match %ld

	* apps/app_zapbarge.c, apps/app_flash.c, apps/app_zapscan.c,
	  apps/app_zapras.c, res/res_musiconhold.c, channels/chan_iax2.c,
	  apps/app_rpt.c: include "asterisk/zapata.h" instead of looking
	  directly for the zaptel.h and tonezone.h

	* configure.ac: another freebsd-specific check for zaptel
	  compatibility

	* include/asterisk/zapata.h (added): Add a stub file to find the
	  zaptel headers in the right place, rather than repeating the
	  check on every single file. Changes to the individual files are
	  coming. The header file name has been suggested by kevin.
	  Approved by: kpfleming

	* makeopts.in: forgot to add BKTR_INCLUDE and BKTR_LIB in
	  makeopts.in

	* configure.ac: add comments that AC_USE_SYSTEM_EXTENSIONS and
	  AST_PROG_LD do not work on FreeBSD - presumably they depend on
	  some auto* feature that is not installed by default. I am not
	  sure on what is a proper fix. In my local copy i simply comment
	  them out. The AST_PROG_LD is a long standing isse, there were
	  attempts to fix it in the past but probably not enough has been
	  copied to acinclude.m4, and i had forgotten about it because i
	  commented out this call in configure.ac long ago

	* configure.ac: Add check for backtrace support on platforms that
	  do not have it natively. Part of it leaked in in a previous
	  commit.

	* configure.ac: remove a useless (and harmful on some platforms)
	  -lnsl from IKSEMEL_LIB. Actually i am not even sure whether
	  -lgcrypt -lgpg-error are needed.

	* configure.ac: simplify checking for zaptel version and location
	  (for linux, this is functionally equivalent to the previous
	  method; for FreeBSD, it re-adds inspection in $PREFIX/zaptel.h).
	  Please wait to regenerate the "configure" file as i have another
	  few pending changes to configure.ac Not applicable to 1.4 until
	  acinclude.m4 is also updated.

2007-01-19 00:28 +0000 [r51273-51275]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c, /: Merged revisions 51274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007)
	  | 3 lines chan_zap compiles without libpri after committing 7877
	  patch ........

	* channels/chan_zap.c, /: Merged revisions 51272 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51272 | dhubbard | 2007-01-18 17:56:49 -0600
	  (Thu, 18 Jan 2007) | 11 lines Merged revisions 51271 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18
	  Jan 2007) | 3 lines issue 7877: chan_zap module reload does not
	  use default/initialized values on subsequent loads. Reset
	  configuration variables to default values prior to parsing
	  configuration file. ........ ................

2007-01-18 22:56 +0000 [r51266]  Jason Parker <jparker@digium.com>

	* main/pbx.c, /, funcs/func_strings.c, apps/app_voicemail.c: Merged
	  revisions 51265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51265 | qwell | 2007-01-18 16:50:23 -0600 (Thu, 18 Jan 2007) | 4
	  lines Add some more checks for option_debug before
	  ast_log(LOG_DEBUG, ...) calls. Issue 8832, patch(es) by tgrman
	  ........

2007-01-18 21:57 +0000 [r51263]  Russell Bryant <russell@digium.com>

	* Makefile, /, configure, main/Makefile, acinclude.m4, makeopts.in:
	  Merged revisions 51262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) |
	  5 lines Ensure that the locations given to the Asterisk configure
	  script for ncurses, curses, termcap, or tinfo are further passed
	  along to the editline configure script. This fixes some
	  cross-compilation environments. (issue #8637, reported by ovi,
	  patch by me) ........

2007-01-18 21:15 +0000 [r51257]  Tilghman Lesher <tlesher@digium.com>

	* /, main/stdtime/localtime.c: Merged revisions 51256 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51256 | tilghman | 2007-01-18 15:14:24 -0600
	  (Thu, 18 Jan 2007) | 10 lines Merged revisions 51255 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
	  Jan 2007) | 2 lines If a timezone is not specified, assume
	  localtime (instead of gmtime) (Issue #7748) ........
	  ................

2007-01-18 19:19 +0000 [r51252]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 51251 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan
	  2007) | 2 lines Only start timeout once we reach the end of the
	  files to play back. ........

2007-01-18 19:03 +0000 [r51249]  Jason Parker <jparker@digium.com>

	* main/cli.c: Fix filename completion for "module load" and "load"
	  CLI commands. Issue 8846

2007-01-18 18:54 +0000 [r51247]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix trunk version of manager support for
	  users.conf. Now it actually pays attention to the "hasmanager"
	  option. (Thanks to Anthony L. for pointing out that this was
	  broken!)

2007-01-18 18:39 +0000 [r51244]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 51243 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2
	  lines Copy MOH settings when calling a peer so that if they put
	  someone on hold or get put on hold themselves they get the right
	  music class. (issue #8840 reported by mdu113) ........

2007-01-18 18:36 +0000 [r51242]  Jason Parker <jparker@digium.com>

	* main/channel.c, /: Merged revisions 51241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2
	  lines Fix an issue with deprecated commands ........

2007-01-18 17:52 +0000 [r51237]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/vmdb.sql, /: Merged revisions 51236 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51236 | tilghman | 2007-01-18 11:49:41 -0600
	  (Thu, 18 Jan 2007) | 10 lines Merged revisions 51235 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
	  Jan 2007) | 2 lines Document all the fields, including the
	  indication that "uniqueid" should not be renamed. ........
	  ................

2007-01-18 17:33 +0000 [r51234]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 51233 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51233 | russell | 2007-01-18 11:18:43 -0600 (Thu, 18 Jan 2007) |
	  3 lines Make the "hasmanager" option in users.conf actually have
	  an effect. (issue #8740, LnxPrgr3) ........

2007-01-18 06:59 +0000 [r51221]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Update ast_append_ha() usage

2007-01-18 05:24 +0000 [r51212-51215]  Joshua Colp <jcolp@digium.com>

	* apps/app_page.c, CHANGES: Add 's' option to Page application
	  which checks devicestate before dialing. (issue #8673 reported by
	  sunder)

	* /, apps/app_voicemail.c: Merged revisions 51213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51213 | file | 2007-01-17 19:48:55 -0500 (Wed, 17 Jan 2007) | 2
	  lines Build the IMAP remote directory string better and properly.
	  Fix an issue with encoding the GSM voicemail when attaching to
	  the voicemail. (issue #8808 reported by akohlsmith) ........

	* /, main/rtp.c: Merged revisions 51211 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2
	  lines Pass data as well for hold/unhold/vidupdate frames. (issue
	  #8840 reported by mdu113) ........

2007-01-17 23:35 +0000 [r51199-51207]  Russell Bryant <russell@digium.com>

	* /, funcs/func_odbc.c: Merged revisions 51205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51205 | russell | 2007-01-17 17:31:11 -0600 (Wed, 17 Jan 2007) |
	  5 lines Fix some instances where when loading func_odbc, a
	  double-free could occur. Also, remove an unneeded error message.
	  If the failure condition is actually a memory allocation failure,
	  a log message will already be generated automatically. ........

	* channels/chan_zap.c, /: Merged revisions 51204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) |
	  4 lines Instead of dividing the offset by 2 directly, make it
	  more clear that the offset is being scaled by the size of the
	  elements in the buffer. (Inspired by a discussing on the
	  asterisk-dev list about this code) ........

	* /, channels/chan_sip.c: Merged revisions 51198 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51198 | russell | 2007-01-17 15:18:35 -0600
	  (Wed, 17 Jan 2007) | 11 lines Merged revisions 51197 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17
	  Jan 2007) | 3 lines Move the check for a failure of
	  ast_channel_alloc() to before locking the pvt structure again.
	  Otherwise, on a failure, this will cause a deadlock. ........
	  ................

2007-01-17 20:57 +0000 [r51196]  Tilghman Lesher <tlesher@digium.com>

	* /, main/utils.c: Merged revisions 51195 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51195 | tilghman | 2007-01-17 14:56:15 -0600
	  (Wed, 17 Jan 2007) | 12 lines Merged revisions 51194 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17
	  Jan 2007) | 4 lines When ast_strip_quoted was called with a
	  zero-length string, it would treat a NULL as if it were the
	  quoting character (and would thus return the string in memory
	  immediately following the passed-in string). ........
	  ................

2007-01-17 19:43 +0000 [r51193]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Don't hold channel lock while sleeping/waiting
	  for audio stream to get setup. (issue #8834 reported by phsultan)

2007-01-17 17:37 +0000 [r51189]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 51186 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51186 | qwell | 2007-01-17 11:36:53 -0600 (Wed, 17 Jan 2007) | 2
	  lines re-add "password" for realtime voicemail ........

2007-01-17 06:37 +0000 [r51183]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 51182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2
	  lines Return the correct result when directly writing out a
	  packet so that the core doesn't then decide to handle it the
	  regular way again. (issue #8833 reported by rcourtna) ........

2007-01-17 01:30 +0000 [r51177]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 51176 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51176 | kpfleming | 2007-01-16 19:29:12 -0600 (Tue, 16 Jan 2007)
	  | 2 lines a few more coding style cleanups and one bug fix (from
	  AnthonyL) ........

2007-01-17 00:50 +0000 [r51173]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 51172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2
	  lines Move rescheduling of lagrq/pings into the scheduler
	  callback. ........

2007-01-17 00:22 +0000 [r51166-51171]  Jason Parker <jparker@digium.com>

	* /, main/rtp.c: Merged revisions 51170 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4
	  lines Fix issue with dtmf continuation packets when the dtmf
	  digit is 0... Issue 8831 ........

	* contrib/scripts/vmdb.sql, /, apps/app_voicemail.c: Merged
	  revisions 51167 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51167 | qwell | 2007-01-16 16:50:19 -0600 (Tue, 16 Jan 2007) | 6
	  lines Fix an issue with IMAP storage and realtime voicemail. Also
	  update the vmdb sql script for IMAP specific options. Issue 8819,
	  initial patches by bsmithurst (slightly modified by me) ........

	* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 51165 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51165 | qwell | 2007-01-16 16:07:53 -0600 (Tue, 16 Jan 2007) | 2
	  lines change documentation to reflect new procedure in 1.4/trunk
	  ........

2007-01-16 21:52 +0000 [r51160-51163]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
	  51162 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51162 | tilghman | 2007-01-16 15:51:15 -0600
	  (Tue, 16 Jan 2007) | 10 lines Merged revisions 51161 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16
	  Jan 2007) | 2 lines Add documentation walkthrough on getting
	  Postgres to work with voicemail (from Issue 8513) ........
	  ................

	* /, apps/app_voicemail.c: Merged revisions 51159 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51159 | tilghman | 2007-01-16 15:28:39 -0600
	  (Tue, 16 Jan 2007) | 10 lines Merged revisions 51158 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16
	  Jan 2007) | 2 lines Postgres driver doesn't like a NULL pointer
	  when retrieving the length (Bug 8513) ........ ................

2007-01-16 19:01 +0000 [r51155]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: remove pointless DEBUG message (watch those
	  patch merges, people!)

2007-01-16 17:50 +0000 [r51152]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c, CHANGES, configs/features.conf.sample: Add
	  parkedcalltransfers option for res_features. This basically
	  enables/disables DTMF based transfers. If you want to get former
	  behavior you will have to make sure it is enabled.

2007-01-16 17:47 +0000 [r51151]  Matt O'Gorman <mogorman@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 51150 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ........ r51150
	  | mogorman | 2007-01-16 11:46:12 -0600 (Tue, 16 Jan 2007) | 2
	  lines minor things i missed before i get jumped on ........

2007-01-16 17:42 +0000 [r51149]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 51148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51148 | file | 2007-01-16 12:39:50 -0500 (Tue,
	  16 Jan 2007) | 10 lines Merged revisions 51145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
	  lines Return previous behavior. ParkedCalls will be able to do
	  DTMF based transfers again. trunk however will get an option to
	  allow this to be set on/off. (issue #8804 reported by nortex)
	  ........ ................

2007-01-16 17:39 +0000 [r51147]  Jason Parker <jparker@digium.com>

	* /, main/file.c: Merged revisions 51146 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51146 | qwell | 2007-01-16 11:36:53 -0600 (Tue, 16 Jan 2007) | 6
	  lines Display more useful output when streaming files. Include
	  the channel name to which the file is being played. Issue 8828,
	  patch by junky. ........

2007-01-16 17:23 +0000 [r51144]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, configs/phone.conf.sample, CHANGES: Add
	  support for G729 passthrough with Sigma Designs boards. (issue
	  #8829 reported by ywalther)

2007-01-16 08:38 +0000 [r51123]  Tilghman Lesher <tlesher@digium.com>

	* channels/iax2-parser.h, channels/iax2.h, channels/chan_iax2.c,
	  channels/iax2-parser.c: IAX2 remote variables - Bug 7619

2007-01-16 05:56 +0000 [r51090]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, /: Merged revisions 51087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r51087 | file | 2007-01-16 00:55:23 -0500 (Tue,
	  16 Jan 2007) | 10 lines Merged revisions 51085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
	  lines Add none as a valid callgroup/pickupgroup option. I
	  consider it a bug that it would inherit it all the way down and
	  not have any way to reset it to nothing - so that's why it is in
	  1.2. (issue #8296 reported by gkloepfer) ........
	  ................

2007-01-16 01:20 +0000 [r51058-51060]  Russell Bryant <russell@digium.com>

	* configs/osp.conf.sample: Fix a couple of typos in the sample
	  osp.conf.

	* /, main/config.c: Merged revisions 51057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r51057 | russell | 2007-01-15 19:15:44 -0600 (Mon, 15 Jan 2007) |
	  3 lines It is possible for the config pointer to be NULL here, so
	  it needs to be checked before dereferencing it. ........

2007-01-16 00:29 +0000 [r51031]  Matt O'Gorman <mogorman@digium.com>

	* configs/users.conf.sample, /, apps/app_voicemail.c: Patch allows
	  for changing voicemail password in users.conf from voicemail
	  main, written by AnthonyL bug #8436

2007-01-15 23:51 +0000 [r50995]  Russell Bryant <russell@digium.com>

	* /, Makefile.rules: Merged revisions 50994 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50994 | russell | 2007-01-15 17:49:48 -0600 (Mon, 15 Jan 2007) |
	  2 lines Filter out a few CFLAGS that are not valid CXXFLAGS.
	  ........

2007-01-15 21:12 +0000 [r50958]  Matt O'Gorman <mogorman@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 50957 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ................
	  r50957 | mogorman | 2007-01-15 15:08:07 -0600 (Mon, 15 Jan 2007)
	  | 12 lines Merged revisions 50946 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
	  | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
	  lines Solves issue with forwarding voicemails from folders other
	  than inbox. patch by anthonyl. ........ ................

2007-01-15 18:24 +0000 [r50922]  Jason Parker <jparker@digium.com>

	* /: These deprecated functions were removed in trunk on purpose.
	  No need to re-add.

2007-01-15 16:40 +0000 [r50896]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, /: Merged revisions 50895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50895 | file | 2007-01-15 11:36:07 -0500 (Mon, 15 Jan 2007) | 2
	  lines Move event processing into do_message so that it gets
	  executed again when events are tripped. ........

2007-01-15 15:08 +0000 [r50868-50869]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, main/Makefile,
	  configure.ac, Makefile.rules, acinclude.m4, makeopts.in: Merged
	  revisions 50867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007)
	  | 2 lines use the ACX_PTHREAD macro from the Autoconf macro
	  archive for setting up compiler pthreads support... should
	  improve portability to platforms with unusual pthreads
	  requirements ........

	* codecs/g722: ignore dependency files in this directory

2007-01-15 02:28 +0000 [r50847]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_oss.c: Feature: allow soundcard to be used in both
	  modes (autoanswer and not), selectable by how it is called in the
	  dialplan. This allows a speaker system hooked up to the soundcard
	  to be used for both ring notification, as well as paging.

2007-01-14 22:00 +0000 [r50821]  Joshua Colp <jcolp@digium.com>

	* /, main/astmm.c: Merged revisions 50820 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50820 | file | 2007-01-14 16:59:05 -0500 (Sun, 14 Jan 2007) | 2
	  lines Add missing newlines for two memory CLI commands. ........

2007-01-14 05:34 +0000 [r50783-50784]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c: Bug 8803 - Fix crash in API

	* /, main/db1-ast/hash/hsearch.c, main/db1-ast/btree/bt_page.c,
	  main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
	  main/db1-ast/hash/hash.c, main/db1-ast/db/db.c,
	  main/db1-ast/recno/rec_get.c, main/db1-ast/btree/bt_seq.c,
	  main/db1-ast/hash/hash_func.c, main/db1-ast/btree/bt_utils.c,
	  main/db1-ast/recno/rec_seq.c, main/db1-ast/btree/bt_overflow.c,
	  main/db1-ast/btree/bt_search.c, main/db1-ast/btree/bt_conv.c,
	  main/db1-ast/btree/bt_close.c, main/db1-ast/btree/bt_put.c,
	  main/db1-ast/recno/rec_utils.c, main/db1-ast/hash/hash_bigkey.c,
	  main/db1-ast/recno/rec_open.c, main/db1-ast/recno/rec_delete.c,
	  main/db1-ast/hash/hash_buf.c, main/db1-ast/hash/hash_page.c,
	  main/db1-ast/recno/rec_close.c, main/db1-ast/recno/rec_put.c,
	  main/db1-ast/include/ndbm.h, main/db1-ast/btree/bt_debug.c,
	  main/db1-ast/mpool/mpool.c, main/db1-ast/btree/bt_split.c,
	  main/db1-ast/btree/bt_open.c, main/db1-ast/btree/bt_delete.c,
	  main/db1-ast/hash/hash_log2.c: Merged revisions 50782 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r50782 | tilghman | 2007-01-13 23:13:47 -0600
	  (Sat, 13 Jan 2007) | 10 lines Merged revisions 50781 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
	  Jan 2007) | 2 lines Bug 8814 - db should look for its header
	  using a relative path, instead of the system path (Fixes FreeWRT)
	  ........ ................

2007-01-13 16:47 +0000 [r50755]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, build_tools/make_sample_voicemail (added): Merged
	  revisions 50754 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50754 | kpfleming | 2007-01-13 10:45:37 -0600 (Sat, 13 Jan 2007)
	  | 2 lines when building the sample greetings for maibox
	  1234@default during 'make samples', build a greeting for each
	  language and file format the user selected to install with
	  menuselect (reported by Brian Capouch on asterisk-dev) ........

2007-01-13 06:01 +0000 [r50675-50728]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 50727 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2
	  lines Only write a frame out to the channel if one exists. There
	  are cases where one may not and would therefore cause the channel
	  driver to segfault. (issue #8434 reported by slimey) ........

	* channels/chan_sip.c: Get rid of unneeded code, fix a spelling
	  mistake, and use registry state a bit more. (issue #8402 reported
	  by rizzo)

	* configs/iax.conf.sample: Clarify what the trunkmaxsize value is
	  in (bytes).

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Drop
	  trunkrealloc option and just have the maximum size be a
	  configurable option. This is per Kevin's comments on -dev and my
	  own thoughts after I put the previous option in.

	* channels/chan_sip.c: Ensure error variable is set to 0 or else we
	  might get false error messages. (issue #8798 reported by tootai,
	  fix by anthonyl)

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Merge in
	  trunkrealloc option for chan_iax2. (issue #8267 reported by
	  marcodmb, branch by anthonyl)

	* /, res/res_snmp.c: Merged revisions 50674 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50674 | file | 2007-01-12 22:04:55 -0500 (Fri, 12 Jan 2007) | 2
	  lines Only join the snmp thread on an unload if the thread is
	  actually running. (issue #8810 reported by junky) ........

2007-01-12 19:25 +0000 [r50648]  Jason Parker <jparker@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 50647 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50647 | qwell | 2007-01-12 13:24:40 -0600 (Fri, 12 Jan 2007) | 2
	  lines Update documentation to state that you shouldn't use
	  realtime static with voicemail.conf ........

2007-01-12 18:13 +0000 [r50603-50629]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Exit from session loop upon error (ie: they
	  disconnected) and don't do any buffer manipulation in do_message.
	  get_input will handle it.

	* main/manager.c, /: Merged revisions 50602 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50602 | file | 2007-01-12 11:42:33 -0500 (Fri, 12 Jan 2007) | 2
	  lines We need to check for res being 0 in do_message itself,
	  otherwise our headers will get lost. ........

2007-01-12 15:01 +0000 [r50538-50571]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, main/pbx.c, include/asterisk/channel.h: make the
	  automatic post-answer delay happen only when the answer is
	  'automatic' (not done by the Answer() dialplan application)

	* main/pbx.c, /: Merged revisions 50562 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r50562 | kpfleming | 2007-01-12 08:42:24 -0600
	  (Fri, 12 Jan 2007) | 10 lines Merged revisions 50561 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12
	  Jan 2007) | 2 lines minor documentation clarification ........
	  ................

	* main/channel.c: when a channel gets automatically answered by an
	  application, sleep a bit to give the audio path (for VOIP
	  channels) time to be setup

2007-01-11 05:54 +0000 [r50378-50469]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 50468 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2
	  lines Remove check for channel state as it can definitely be
	  something other then ring, and also clean up the code a bit. This
	  should solve the parking issues and maybe some attended transfer
	  issues people have been seeing. ........

	* /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
	  Merged revisions 50466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2
	  lines Add support to see whether NAT was detected (yay symmetric
	  RTP) and also add a check in chan_sip so that if NAT has been
	  detected and the reinvite behind nat option has been turned off,
	  then just do partial bridge. (issue #8655 reported by mnicholson)
	  ........

	* /, apps/app_speech_utils.c: Merged revisions 50433 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan
	  2007) | 2 lines Merge speech-multi branch which adds support for
	  joining multiple sound files together to be played one after
	  another in SpeechBackground. ........

	* /, main/config.c: Merged revisions 50405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50405 | file | 2007-01-10 14:46:29 -0500 (Wed, 10 Jan 2007) | 2
	  lines Fix parsing when using something like ldap settings. (done
	  by anthonyl) ........

	* include/asterisk/strings.h: Return the useless casts that ensure
	  this file is C++ clean. (issue #8602 reported by mikma)

	* /, channels/chan_sip.c: Merged revisions 50377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2
	  lines Fix chan_sip not working issue. Let's not prematurely
	  return 0. (issue #8783 reported by st41ker) ........

2007-01-10 16:47 +0000 [r50347]  Jason Parker <jparker@digium.com>

	* /, cdr/cdr_manager.c: Merged revisions 50346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50346 | qwell | 2007-01-10 10:45:36 -0600 (Wed, 10 Jan 2007) | 4
	  lines Reverse some logic in cdr_manager, which made it fail to
	  load if the config file existed. Issue 8777 ........

2007-01-10 04:56 +0000 [r50267-50302]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 50298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r50298 | file | 2007-01-09 23:55:13 -0500 (Tue,
	  09 Jan 2007) | 10 lines Merged revisions 50295 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
	  lines Add another return value to dial_exec_full that indicates
	  execution is going to continuing at a new
	  extension/context/priority and to just let it slide. (issue #8598
	  reported by jon) ........ ................

	* channels/chan_zap.c: Allow usedistinctiveringdetection and
	  distinctiveringaftercid to be reset during a reload. (issue #8739
	  reported by tzafrir)

	* main/pbx.c, /: Merged revisions 50266 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50266 | file | 2007-01-09 22:51:29 -0500 (Tue, 09 Jan 2007) | 2
	  lines Ensure data's existence before trying to access it. (issue
	  #8774 reported by rcourtna) ........

2007-01-10 02:50 +0000 [r50229-50230]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Covert some spaces to tabs, and put a list
	  of defines in an enum.

	* Makefile, /: Merged revisions 50228 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r50228 | russell | 2007-01-09 21:17:46 -0500
	  (Tue, 09 Jan 2007) | 14 lines Merged revisions 50227 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09
	  Jan 2007) | 6 lines Make the number that represents the major
	  version number a single digit instead of 2. Using two digits
	  makes it an octal number when put into version.h, which breaks
	  the compilation of any out of tree module that checks the version
	  for any version after 1.2.7 (reported by Matteo Brancaleoni on
	  the asterisk-dev mailing list, who gave credit to vihai for
	  pointing it out) ........ ................

2007-01-09 13:45 +0000 [r50152]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 50151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r50151 | tilghman | 2007-01-09 07:40:45 -0600
	  (Tue, 09 Jan 2007) | 12 lines Merged revisions 50150 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09
	  Jan 2007) | 4 lines The advent of realtime has enabled people to
	  use commas in the fullname field. This could cause an issue with
	  sending voicemails, when the field is unquoted. (Issue 8595)
	  ........ ................

2007-01-09 12:25 +0000 [r50132]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Based on the following patch, changed for
	  trunk... Merged revisions 50124 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3
	  lines - handle re-invites properly in sip_hangup() - Add some
	  invitestate status changes just to be sure ........

2007-01-08 23:40 +0000 [r50099]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 50098 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50098 | qwell | 2007-01-08 17:39:12 -0600 (Mon, 08 Jan 2007) | 4
	  lines Fix an issue with voicemail and users.conf, where it
	  wouldn't ever parse a password, since it was using "secret"
	  instead of "password" Issue 8761, reported by and patch
	  suggestion from ssokol. ........

2007-01-08 21:40 +0000 [r50075]  Joshua Colp <jcolp@digium.com>

	* codecs/codec_zap.c: Move channel acquisition to when the
	  translation path is setup, and clean up.

2007-01-08 21:17 +0000 [r50074]  Matt O'Gorman <mogorman@digium.com>

	* /, apps/app_senddtmf.c: Merged revisions 50073 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ........ r50073
	  | mogorman | 2007-01-08 15:11:16 -0600 (Mon, 08 Jan 2007) | 1
	  line we can't unlock a channel if we cant find it. - AnthonyL bug
	  #8741 ........

2007-01-08 20:10 +0000 [r50033-50056]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Make callback declaration match one used in trunk.

	* include/asterisk/lock.h: Change trylock output for what already
	  has the lock from an error to a warning.

	* /, main/rtp.c: Merged revisions 50032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2
	  lines Disable the more intense packet2packet bridging until the
	  bugs can be worked out. ........

2007-01-08 14:31 +0000 [r49931-50007]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 50006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11
	  lines Issue #8677 - Handle failure of T.38 re-invite This is not
	  a fix, but adding an error message to tell the admin that we have
	  a bad configuration. We should not send T.38 re-invites to
	  devices that can't handle it (with the current architecture where
	  you have to hard-code t.38 support per device). To really fix
	  this, we need to figure out a way to tell the incoming call that
	  the re-invite failed, so we can signal failure on that end and go
	  back to the original call. ........

	* /, channels/chan_sip.c: Merged revisions 49983 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3
	  lines Issue #8524, support multiple via header values (tardieu)
	  Thanks! ........

	* main/frame.c, include/asterisk/frame.h, main/rtp.c: Issue #8663 -
	  Add passthrough support for MPEG4 (neutrino88).

	* /, channels/chan_sip.c: Merged revisions 49945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2
	  lines We only need one forward declaration ........

	* /, channels/chan_sip.c: Merged revisions 49925 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2
	  lines Issue 8735: Terminate state when extension is unavailable
	  for subscription ........

2007-01-08 05:13 +0000 [r49891]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 49890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r49890 | file | 2007-01-08 00:11:54 -0500 (Mon,
	  08 Jan 2007) | 10 lines Merged revisions 49889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
	  lines Ensure we use the default refresh value of 60 if the remote
	  server does not send one. (issue #8746 reported by maethor)
	  ........ ................

2007-01-08 03:56 +0000 [r49870]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, configure.ac: Merged revisions 49866 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r49866 | kpfleming | 2007-01-07 21:53:53 -0600 (Sun, 07
	  Jan 2007) | 2 lines since we use AC_PATH_TOOL to find tools, we
	  should use the results it provides for us (reported by Brian
	  Capouch on the asterisk-dev list) ........

2007-01-07 21:46 +0000 [r49832-49835]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_dictate.c: Merged revisions 49834 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r49834 | tilghman | 2007-01-07 15:44:52 -0600
	  (Sun, 07 Jan 2007) | 10 lines Merged revisions 49833 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07
	  Jan 2007) | 2 lines If openstream fails, then we crash (Issue
	  8564) ........ ................

	* /, channels/chan_sip.c: Merged revisions 49831 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007)
	  | 2 lines Second condition was a subset of the first, so hold was
	  never decremented, thus hint stayed stuck (Issue 8747) ........

2007-01-07 19:00 +0000 [r49816]  Joshua Colp <jcolp@digium.com>

	* funcs/func_base64.c, funcs/func_blacklist.c,
	  funcs/func_callerid.c: One const, two const. Let's stick with
	  everything else - one const. Plus older versions of GCC don't
	  support double const either.

2007-01-07 16:21 +0000 [r49784-49801]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c, include/asterisk/config.h,
	  res/res_realtime.c, main/config.c, funcs/func_realtime.c: When
	  calling the Realtime app more than once, unset fields which were
	  previously set are erroneously still set (Bug 6701). After
	  discussion, it was determined this should only be changed in
	  trunk.

	* funcs/func_shell.c, funcs/func_strings.c, funcs/func_cut.c:
	  Modifications to allow the output of SHELL() to be split per line
	  (Issue 8676)

	* funcs/func_shell.c (added): Add function to execute a shell
	  command and return the output (Issue 8676)

	* main/channel.c: Reduce duplication of code (Issue 6542)

2007-01-07 07:43 +0000 [r49769]  Jason Parker <jparker@digium.com>

	* main/indications.c: Fix a segfault when using "countries" that
	  don't have a matching zone.

2007-01-06 00:28 +0000 [r49743]  Jason Parker <jparker@digium.com>

	* main/pbx.c, /, res/res_features.c, pbx/pbx_config.c: Merged
	  revisions 49742 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7
	  lines Save 1 whopping byte of allocated memory! This looks like
	  it may have been a chicken/egg scenario.. You had to call a
	  cleanup func, because everything was allocated. Then since you
	  had to call a cleanup func, you were forced to allocate - ie;
	  strdup(""). ........

2007-01-06 00:13 +0000 [r49727-49741]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs/func_base64.c, funcs/func_rand.c, funcs/func_md5.c,
	  funcs/func_db.c, channels/chan_zap.c, funcs/func_module.c,
	  funcs/func_version.c, funcs/func_timeout.c, funcs/func_env.c,
	  funcs/func_strings.c, funcs/func_math.c, funcs/func_vmcount.c,
	  funcs/func_cut.c, include/asterisk/channel.h, funcs/func_sha1.c,
	  funcs/func_logic.c, funcs/func_uri.c, funcs/func_global.c,
	  funcs/func_realtime.c, funcs/func_enum.c, funcs/func_curl.c,
	  funcs/func_groupcount.c, funcs/func_odbc.c,
	  funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c,
	  funcs/func_callerid.c: finish const-ifying ast_func_read()

	* main/manager.c: probably shouldn't leave the mmap'ed file hanging
	  around in memory

	* /, configure, acinclude.m4: Merged revisions 49714-49715 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49714 | kpfleming | 2007-01-05 17:49:52 -0600 (Fri, 05 Jan 2007)
	  | 2 lines proper fix for r49712 ........ r49715 | kpfleming |
	  2007-01-05 17:51:31 -0600 (Fri, 05 Jan 2007) | 2 lines one more
	  time... ........

	* main/manager.c, include/asterisk/config.h, main/config.c: a
	  little more const-ification

2007-01-05 23:51 +0000 [r49716]  Joshua Colp <jcolp@digium.com>

	* codecs/codec_zap.c: It is possible for framein to get called and
	  no channel be available, so do a check before we increment the
	  count.

2007-01-05 23:41 +0000 [r49711-49713]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, acinclude.m4: Merged revisions 49712 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r49712 | kpfleming | 2007-01-05 17:40:29 -0600 (Fri, 05
	  Jan 2007) | 2 lines if --with-foo=<path> is specific for a
	  configure option, ensure that it is used for header file checking
	  as well ........

	* main/pbx.c, /, channels/chan_sip.c, channels/chan_agent.c,
	  pbx/pbx_dundi.c, include/asterisk/pbx.h, apps/app_queue.c,
	  channels/chan_iax2.c, main/db.c, apps/app_speech_utils.c,
	  include/asterisk/astdb.h, apps/app_voicemail.c: const-ify some
	  more APIs, and fix rev 49710 from branch-1.4 in a better way here

2007-01-05 23:31 +0000 [r49709]  Matt O'Gorman <mogorman@digium.com>

	* codecs/codec_zap.c: no need to spam everyone with show transcoder
	  messages

2007-01-05 23:17 +0000 [r49706]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /, codecs/codec_zap.c: Merged revisions
	  49705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4
	  lines Make codec_zap and chan_zap also depend on zaptel. This
	  fixes an issue (8727) with zaptel being in a different directory,
	  using --with-zaptel. ........

2007-01-05 22:53 +0000 [r49678-49681]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c, /: Merged revisions 49680 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49680 | kpfleming | 2007-01-05 16:52:37 -0600 (Fri, 05 Jan 2007)
	  | 2 lines don't 'consume' the params list before we try to use it
	  again ........

	* main/manager.c: use mmap() to read in the results of the manager
	  action for an HTTP request, instead of reading it into a buffer

	* main/pbx.c, channels/chan_zap.c, /, channels/chan_sip.c,
	  apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
	  utils/astman.c, res/res_jabber.c, include/asterisk/manager.h,
	  channels/chan_iax2.c, apps/app_queue.c, main/config.c,
	  res/res_monitor.c, main/manager.c, include/asterisk/jabber.h,
	  apps/app_senddtmf.c, main/db.c: Merged revisions 49676 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007)
	  | 2 lines reduce stack consumption for AMI and AMI/HTTP requests
	  by nearly 20K in most cases ........

2007-01-05 22:18 +0000 [r49677]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 49675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49675 | file | 2007-01-05 17:14:47 -0500 (Fri, 05 Jan 2007) | 2
	  lines Don't keep repeating the warning over and over when the end
	  of the call is reached. (issue #8724 reported by xrg) ........

2007-01-05 17:10 +0000 [r49578-49637]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_iax2.c: Merged revisions 49636 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r49636 | kpfleming | 2007-01-05 11:09:00 -0600
	  (Fri, 05 Jan 2007) | 10 lines Merged revisions 49635 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05
	  Jan 2007) | 2 lines ensure that threads which are supposed to be
	  detached (because we aren't going to wait on them) are created
	  properly ........ ................

	* main/threadstorage.c: use a rwlock-list for the list of TLS
	  objects

	* /, channels/chan_iax2.c: Merged revisions 49600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007)
	  | 2 lines revert the dynamic_list insertion change... that was
	  not the right thing to do ........

	* /, channels/chan_iax2.c: Merged revisions 49581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007)
	  | 3 lines create the IAX2 processing threads as background
	  threads so they will use smaller stacks when we create a dynamic
	  thread, put it on the dynamic_list right away so we don't lose
	  track of it ........

	* include/asterisk/strings.h: ensure that the proper
	  file/function/line shows up for dynamic string threadstorage
	  objects remove pointless casts

	* include/asterisk/threadstorage.h: yeah... so... compiling before
	  committing seems like it might be a good idea

	* build_tools/cflags.xml, include/asterisk.h, /,
	  main/threadstorage.c (added), main/Makefile,
	  include/asterisk/strings.h, include/asterisk/threadstorage.h,
	  main/asterisk.c: Merged revisions 49553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49553 | kpfleming | 2007-01-04 16:51:01 -0600 (Thu, 04 Jan 2007)
	  | 2 lines add support for tracking thread-local-storage objects
	  that exist via 'threadstorage' CLI commands ........

2007-01-04 23:02 +0000 [r49552-49573]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 49568 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2
	  lines It's possible for the iax2 pvt to disappear, so if it
	  has... don't bother looking for dpentries. ........

	* /, main/config.c: Merged revisions 49551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49551 | file | 2007-01-04 17:28:29 -0500 (Thu, 04 Jan 2007) | 2
	  lines Only free comments and line buffer once we reach the first
	  level. (issue #8678 reported by ssokol, fixed by anthonyl)
	  ........

2007-01-04 21:59 +0000 [r49538]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c, /, channels/iax2-parser.c: Merged revisions 49536
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007)
	  | 2 lines don't mark these allocations as 'cache' allocations
	  when caching has been disabled ........

2007-01-04 21:40 +0000 [r49525]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: It's pretty difficult to pthread_kill a thread
	  that doesn't exist. (issue #8681 reported by bkruse)

2007-01-04 21:06 +0000 [r49524]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/iax2-parser.c: Merged revisions 49523 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04
	  Jan 2007) | 2 lines if we're going to decrement the frame count
	  when we free a frame, we should inrement it when we create one
	  :-) ........

2007-01-04 20:27 +0000 [r49491-49507]  TransNexus OSP Development <support@transnexus.com>

	* doc/osp.txt: 1. Update osp guide.

	* configs/osp.conf.sample: 1. Update osp module configuration file.

2007-01-04 18:32 +0000 [r49466]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/iax2-parser.h, /, channels/chan_iax2.c,
	  channels/iax2-parser.c: Merged revisions 49465 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007)
	  | 2 lines only do IAX2 frame caching for voice and video frames
	  ........

2007-01-04 18:28 +0000 [r49464]  Matt O'Gorman <mogorman@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 49459 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ................
	  r49459 | mogorman | 2007-01-04 12:11:19 -0600 (Thu, 04 Jan 2007)
	  | 10 lines Merged revisions 49447 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
	  | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
	  lines converted a lot of 256 to PATH_MAX and some white space
	  fixes. ........ ................

2007-01-04 18:19 +0000 [r49463]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/Makefile, main/frame.c, /, channels/iax2-parser.c: Merged
	  revisions 49457,49460-49461 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007)
	  | 2 lines make building of codec_gsm against the system GSM
	  library actually work ........ r49460 | kpfleming | 2007-01-04
	  12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines don't define this
	  type either if LOW_MEMORY is enabled ........ r49461 | kpfleming
	  | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines don't do
	  frame header caching in the core if LOW_MEMORY is defined
	  ........

2007-01-04 18:17 +0000 [r49414-49462]  Matt O'Gorman <mogorman@digium.com>

	* /, channels/iax2-parser.c: Merged revisions 49458 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.4 ........
	  r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007)
	  | 2 lines don't do frame caching in LOW_MEMORY mode ........

	* /, apps/app_voicemail.c: Merged revisions 49413 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ................
	  r49413 | mogorman | 2007-01-04 10:50:56 -0600 (Thu, 04 Jan 2007)
	  | 11 lines Merged revisions 49412 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
	  | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
	  lines good catch russell sorry i missed that. fix magic number
	  with proper sizeof ........ ................

2007-01-03 23:41 +0000 [r49356]  Matt O'Gorman <mogorman@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 49355 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ................
	  r49355 | mogorman | 2007-01-03 17:32:03 -0600 (Wed, 03 Jan 2007)
	  | 14 lines Merged revisions 49354 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
	  | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
	  lines When using ODBC_STORAGE VoicemailMain doesn't create the
	  subdirectories for a mailbox such as the INBOX directory. this
	  patch solves that problem, was written by anthony be-125 ........
	  ................

2007-01-03 11:15 +0000 [r49320-49321]  Christian Richter <christian.richter@beronet.com>

	* doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  configs/misdn.conf.sample, channels/misdn/isdn_lib.c,
	  channels/misdn_config.c: Merged revisions 49313 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r49313 | crichter | 2007-01-03 10:06:50 +0100
	  (Mi, 03 Jan 2007) | 41 lines Merged revisions
	  48319,48321,48467,48552,48576,49135,49303 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
	  1 line changed a few debugs to higher debug levels ........
	  r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
	  1 line added the export and import of the MISDN_ADDRESS_COMPLETE
	  Variable to inidcate wether the extension is already completely
	  dialed or if there might come additional digits by information
	  elements. also added some docs for that. ........ r48467 |
	  crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
	  removed FIXUP state. added check for channel allocation conflict
	  when we create a setup while the other site creates a setup on
	  the same channel, besides the check we resolve this conflict.
	  ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
	  Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
	  preselected channel we just accept it, even when we're NT. added
	  some checks for segfaults. ........ r48576 | crichter |
	  2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
	  reject a channel, because it's in use already, we shouldn't
	  process the setup anymore. made the channel allocation a bit
	  easier and more understandable, removed a few unused lines
	  ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
	  Jan 2007) | 1 line added check for channel ranges in the
	  set/empty channel functions. set pmp_l1_check default to no.
	  added misdn restart pid cli command. added cleaning of channel
	  when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
	  2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
	  check for bridging in misdn_call to avoid setting
	  echocancellation when 2 mISDN channels are involved and when
	  bridging is set. That lead to a kernel panic before under
	  different situations, because we switched about 2 times between
	  hardware bridging and echocancelation * readded MISDN_URATE
	  variable which got lost before, this should make app_v110 work
	  again * fixed typo ........ ................

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c,
	  channels/misdn_config.c: Merged revisions 47989 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47989 | crichter | 2006-11-24 16:46:13 +0100
	  (Fr, 24 Nov 2006) | 9 lines Merged revisions 47968 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
	  Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
	  beatufied some logs, changed some loglevels. changed the default
	  value of block_on_alarm ........ ................

2007-01-03 03:28 +0000 [r49283]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, Makefile.rules: Merged revisions 49282 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r49282 | kpfleming | 2007-01-02 21:21:25 -0600 (Tue, 02
	  Jan 2007) | 2 lines various Makefile improvements to get chan_vpb
	  (and any other C++ modules) to build properly ........

2007-01-03 01:21 +0000 [r49260]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 49259 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2
	  lines Check pvt structure presence before passing to
	  send_command. This gets rid of the irritating message about a
	  packet without pvt structure. This happens because the scheduled
	  item is getting cancelled at almost the exact moment it is
	  getting executed. ........

2007-01-02 22:43 +0000 [r49238]  Steve Murphy <murf@digium.com>

	* /, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex,
	  main/ast_expr2.fl: Merged revisions 49237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49237 | murf | 2007-01-02 15:30:53 -0700 (Tue, 02 Jan 2007) | 1
	  line This is a slight modification to Josh's edits for #8579;
	  both files edited were the produced by flex; so the source files
	  need to be changed instead, and the generated files regenerated.
	  ........

2007-01-02 20:02 +0000 [r49214-49215]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Removing propably accidentally added debug
	  messages sent to verbose channel

	* /, channels/chan_sip.c: Merged revisions 49212 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2
	  lines Small cleanup of add_t38sdp - it's always enabled at that
	  point in the code ........

2007-01-02 17:04 +0000 [r49187]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_math.c: Tweak description text to match new
	  functionality (Issue 7959)

2007-01-02 14:01 +0000 [r49166]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 49165 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007)
	  | 2 lines remove comment that is unrelated to this function
	  ........

2007-01-02 13:50 +0000 [r49152]  Olle Johansson <oej@edvina.net>

	* /, configs/features.conf.sample: Update sample config

2007-01-01 23:43 +0000 [r49100-49103]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /, build_tools/menuselect-deps.in,
	  configure, include/asterisk/autoconfig.h.in, configure.ac,
	  codecs/codec_zap.c: Merged revisions 49102 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007)
	  | 2 lines check specifically for VLDTMF and transcoding support
	  in the system's Zaptel installation, and make only the modules
	  that need those features dependent on them (this will allow
	  building the other Zaptel-using parts of Asterisk against older
	  versions of Zaptel or those on other platforms that haven't
	  caught up yet to the Linux version) ........

	* Makefile, sounds/Makefile: GNU make already knows what the
	  current directory is, there is no need to use 'pwd'

	* Makefile, /: Merged revisions 49098-49099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49098 | kpfleming | 2007-01-01 16:08:24 -0600 (Mon, 01 Jan 2007)
	  | 2 lines revert this change until a better solution can be
	  found... 'env -i' was not being used properly, but even when
	  changed to do so, this process fails during cross-compilation
	  because the menuselect build still sees 'CC' as set to the
	  cross-compiler ........ r49099 | kpfleming | 2007-01-01 16:48:03
	  -0600 (Mon, 01 Jan 2007) | 2 lines use a simpler (and portable)
	  method to ensure that menuselect is built as a host binary
	  ........

2007-01-01 20:16 +0000 [r49092-49097]  Olle Johansson <oej@edvina.net>

	* /: Block cleanup of release branch

	* include/asterisk/indications.h: Doxygen documentationification

	* main/manager.c: Fix manager too.

	* main/frame.c, channels/chan_sip.c, include/asterisk/frame.h: -
	  Add error handling to ast_parse_allow_disallow - Use this in
	  chan_sip configuration parsing

	* include/asterisk/acl.h, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/acl.c,
	  channels/chan_iax2.c, channels/chan_mgcp.c: - Implement error
	  handling in ast_append_ha - Use this in chan_sip - Document ha
	  functions in acl.c

2006-12-31 19:15 +0000 [r49089]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: count is no longer used in the iaxq
	  structure really so let's just make this a statically declared
	  linked list.

2006-12-31 09:38 +0000 [r49080-49082]  Olle Johansson <oej@edvina.net>

	* CHANGES: Update CHANGES, make section about SIP. This might be a
	  good way to handle other parts of this file too, as it grows.

	* configs/sip.conf.sample: Added some docs

	* channels/chan_sip.c: Add version number to useragent string -
	  Issue #8700, blanchet - THANKS!

2006-12-31 05:20 +0000 [r49075-49076]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_math.c: Add power and right/left shift functions
	  (Issue 7959)

	* configs/voicemail.conf.sample, UPGRADE.txt, apps/app_voicemail.c:
	  1. Rename 'maxmessage' to 'maxsecs' to differentiate from
	  'maxmsg'. 2. Rename 'minmessage' to 'minsecs' for parity. 3. Make
	  'maxsecs' a per-user option, in addition to global. (Issue #
	  8624)

2006-12-30 18:32 +0000 [r49071-49074]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 49073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49073 | file | 2006-12-30 13:31:17 -0500 (Sat, 30 Dec 2006) | 2
	  lines IAX has been deprecated for quite some time so we had
	  better use IAX2 when creating the dial string for users. (issue
	  #8697 reported by ssokol) ........

	* main/rtp.c: Clarify why we are reading in a frame in the
	  Packet2Packet bridge.

2006-12-30 13:27 +0000 [r49068-49069]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: now that the 'languageprefix' option defaults to
	  'on', and all channels have a default language of 'en', let's
	  install the English sound files into /var/lib/asterisk/sounds/en,
	  just like the other languages

	* main/channel.c: small formatting fix

2006-12-30 05:49 +0000 [r49064-49067]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 49066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2
	  lines If the Packet2Packet bridge is being broken because of a
	  masquerade then attempt to read a frame in so the masquerade
	  actually happens. Otherwise weirdness will occur. (issue #8696
	  reported by kjotte) ........

	* funcs/func_odbc.c: Initialize obj pointers to NULL. Gets rid of
	  two compiler warnings.

	* /, channels/chan_iax2.c: Merged revisions 49063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2
	  lines Initialize the packet queue in load_module instead of just
	  declaring the list with the default value. (issue #8695 reported
	  by ssokol) ........

2006-12-30 00:51 +0000 [r49062]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_ael.c: Merged revisions 49061 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49061 | murf | 2006-12-29 17:40:37 -0700 (Fri, 29 Dec 2006) | 1
	  line A fix for 8661, where the CUT func needed to have comma args
	  converted to vertical bars. I hope this change does little harm.
	  ........

2006-12-29 13:25 +0000 [r49056]  Russell Bryant <russell@digium.com>

	* channels/chan_oss.c: Convert various comments to doxygen format.

2006-12-29 11:02 +0000 [r49054]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Removing extra output

2006-12-29 06:26 +0000 [r49053]  Russell Bryant <russell@digium.com>

	* include/asterisk/smdi.h: Fix a spelling mistake in a comment.

2006-12-29 00:33 +0000 [r49047]  Kevin P. Fleming <kpfleming@digium.com>

	* /, BUGS: Merged revisions 49046 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r49046 | kpfleming | 2006-12-28 18:32:59 -0600
	  (Thu, 28 Dec 2006) | 10 lines Merged revisions 49045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28
	  Dec 2006) | 2 lines location of the bug posting guidelines has
	  changed ........ ................

2006-12-28 20:13 +0000 [r49030]  Tilghman Lesher <tlesher@digium.com>

	* configs/func_odbc.conf.sample, funcs/func_odbc.c,
	  funcs/func_strings.c: Integrate functionality tested on
	  svncommunity users back into trunk

2006-12-28 20:10 +0000 [r49029]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 49028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49028 | kpfleming | 2006-12-28 14:08:59 -0600 (Thu, 28 Dec 2006)
	  | 2 lines new versions of sounds ........

2006-12-28 20:05 +0000 [r49026-49027]  Joshua Colp <jcolp@digium.com>

	* main/http.c: Convert uri_redirects list to read/write locks.

	* /, main/http.c: Merged revisions 49024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49024 | qwell | 2006-12-28 14:52:46 -0500 (Thu, 28 Dec 2006) | 2
	  lines make the uris_lock a rwlock instead of a mutex lock - needs
	  to be forward ported to trunk ........

2006-12-28 17:56 +0000 [r49019]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
	  pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
	  pbx/ael/ael.tab.h, utils/ael_main.c, main/ast_expr2.fl,
	  main/ast_expr2.c: Jason is having problems with the inclusion of
	  <err.h>; it appears to be unnecessary for sucessful builds, so I
	  either removed or commented out the inclusions from all the AEL
	  related code. New outputs from bison/flex are included, etc.

2006-12-27 22:30 +0000 [r49010]  Joshua Colp <jcolp@digium.com>

	* /, main/ast_expr2f.c, pbx/ael/ael_lex.c: Merged revisions 49009
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2
	  lines ast_copy_string is not available when LOW_MEMORY is used
	  and things are being built in the utils directory, so we need to
	  resort to the old method of strncpy. (issue #8579 reported by
	  mottano) ........

2006-12-27 22:14 +0000 [r49007-49008]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
	  main/dnsmgr.c, main/frame.c, main/manager.c, /, main/http.c,
	  main/logger.c, main/enum.c, main/asterisk.c, main/rtp.c,
	  main/term.c: Merged revisions 49006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006)
	  | 2 lines since these variables all have static duration, none of
	  them need initializers (they default to zero anyway) ........

	* codecs/g722: add file to ignore list

2006-12-27 21:27 +0000 [r49004]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Only include include files once (imported
	  from 1.4)

2006-12-27 21:21 +0000 [r48999-49001]  Kevin P. Fleming <kpfleming@digium.com>

	* main/asterisk.c: apparently we need an explicit message to warn
	  people

	* main/file.c, UPGRADE.txt, main/asterisk.c, doc/asterisk-conf.txt:
	  make the 'languageprefix' option default to on, and deprecate
	  turning it off

	* /, main/file.c, include/asterisk/options.h, main/asterisk.c:
	  Merged revisions 48998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006)
	  | 3 lines move extern declaration for this option to a header
	  file where it belongs provide an initial value for
	  'languageprefix' option, instead of relying on randomness to
	  provide a useful value ........

2006-12-27 20:30 +0000 [r48992-48996]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Only set "rfc2833compensate" option once

	* /, channels/chan_sip.c: Only handle T38 options once

	* channels/chan_sip.c: -Remove "localmask" setting (deprecated in
	  earlier version) - Remove "musiconhold" and "musicclass" settings
	  (also deprecated earlier)

2006-12-27 18:34 +0000 [r48989-48990]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 48988 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006)
	  | 2 lines make the option actually match the documentation
	  ........

	* include/asterisk/utils.h, include/asterisk/astmm.h, main/frame.c,
	  /, main/astmm.c, channels/iax2-parser.c: Merged revisions 48987
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006)
	  | 2 lines allow 'show memory' and 'show memory summary' to
	  distinguish memory allocations that were done for caching
	  purposes, so they don't look like memory leaks ........

2006-12-27 18:02 +0000 [r48976-48986]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Be politically
	  correct

	* apps/app_sms.c: From coding guidelines: Comments should explain
	  what the code does, not when something was changed or who changed
	  it. If you have done a larger contribution, make sure that you
	  are added to the CREDITS file.

	* /, channels/chan_sip.c, configs/sip.conf.sample: Add support for
	  buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)

	* /, channels/chan_sip.c: Cleanup of handle_common_options

	* /, channels/chan_sip.c: Reset invitestate when sending new invite

	* /, channels/chan_sip.c: Issue #8600 - bogus SDP Content Length in
	  T.38 re-invite

2006-12-26 05:23 +0000 [r48961-48967]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 48966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48966 | file | 2006-12-26 00:20:08 -0500 (Tue, 26 Dec 2006) | 2
	  lines Get rid of a needless memory allocation and only create a
	  conference structure in find_conf_realtime if data was read from
	  realtime. (issue #8669 reported by robl) ........

	* /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
	  Merged revisions 48964 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2
	  lines Add an API call that initializes an RTP structure. We need
	  this because chan_sip is cheeky and uses a temporary RTP
	  structure for codec purposes, and the API calls that are used
	  rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
	  ........

	* /, configure, configure.ac: Merged revisions 48960 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r48960 | file | 2006-12-25 12:04:48 -0500 (Mon, 25 Dec
	  2006) | 2 lines Clean up autoconf file (gets rid of warnings seen
	  when rebuilding configure) and rebuild configure. ........

2006-12-25 06:42 +0000 [r48958-48959]  Luigi Rizzo <rizzo@icir.org>

	* codecs/g722/g722.h: provide INT16_MIN and INT16_MAX for platforms
	  where they are not defined.

	* main/channel.c, apps/app_read.c, channels/chan_misdn.c,
	  funcs/func_channel.c, include/asterisk/indications.h,
	  apps/app_disa.c, main/app.c, res/snmp/agent.c,
	  contrib/utils/zones2indications.c, include/asterisk/channel.h,
	  res/res_indications.c, main/indications.c: rename the structs
	  struct tone_zone_sound and struct tone_zone defined in
	  indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid
	  conflicts with the structs with the same names defined in
	  tonezone.h Hope i haven't missed any instance.

2006-12-25 05:22 +0000 [r48929-48957]  Russell Bryant <russell@digium.com>

	* /, funcs/func_math.c: Merged revisions 48956 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48956 | russell | 2006-12-25 00:21:20 -0500
	  (Mon, 25 Dec 2006) | 14 lines Merged revisions 48955 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25
	  Dec 2006) | 6 lines Fix an error introduced by copying and
	  pasting the handling of the >= operator for the MATH function. If
	  a single equal sign was used as an operator, the function would
	  treat it is as if it were the >= operator. Now, it properly
	  handles it as an invalid operator. (issue #8665, patch by
	  tempest1) ........ ................

	* funcs/func_callerid.c: Simplify the if statements used to check
	  to see if the argument was "num" or "number". It is not possible
	  to ever reach the second part of this conditional statement.
	  Thanks to my brother, Brett, for pointing this out. :)

	* main/frame.c: Resolve some compiler warnings

	* /, channels/chan_oss.c: Merged revisions 48948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) |
	  3 lines Fix a typo in an error message that indicated that the
	  MGCP channel type could not be registered, instead of the correct
	  type, OSS. ........

	* main/http.c, configs/http.conf.sample: Use spaces as a separator
	  for the redirect option to improve readability

	* /, channels/chan_iax2.c: Merged revisions 48944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48944 | russell | 2006-12-24 02:25:38 -0500
	  (Sun, 24 Dec 2006) | 11 lines Merged revisions 48943 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24
	  Dec 2006) | 3 lines Check for the proper return value on an error
	  in a call to mmap(). This was reported by Andy Wang on the
	  asterisk-dev list. Thanks! ........ ................

	* channels/chan_sip.c: Merged revisions 48940 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48940 | russell | 2006-12-24 01:49:31 -0500
	  (Sun, 24 Dec 2006) | 11 lines Merged revisions 48939 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24
	  Dec 2006) | 3 lines Remove a couple of misplaced dots in log
	  messages. This was reported by Andrea Spadaccini on the
	  asterisk-dev mailing list. ........ ................

	* main/http.c: Simplify the definition of http_uri_redirect such
	  that only one allocation is done for exactly how much memory is
	  needed. This was suggested by Luigi on the asterisk-dev mailing
	  list. Thanks!

	* include/asterisk/http.h, main/http.c, CHANGES,
	  configs/http.conf.sample: - Convert the list of URI handlers to
	  use the linked list macros. While doing this, implementing
	  locking of this list to make it thread-safe. - Add a "redirect"
	  option to http.conf that allows redirecting one URI to another. I
	  was inspired to do this while playing with the Asterisk GUI. I
	  got tired of typing this URL to get to the GUI:
	  http://localhost:8088/asterisk/static/config/cfgadvanced.html So,
	  now I have the following line in http.conf:
	  redirect=/=/asterisk/static/config/cfgadvanced.html Now, I can
	  type the following into my browser and go to the GUI:
	  http://localhost:8088

	* main/manager.c: Remove a debug message. If this is still needed
	  for debugging something, it should be made a LOG_DEBUG message.

2006-12-23 19:55 +0000 [r48928]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/lock.h: We should probably declare the lock...
	  and not just the constructor/deconstructor.

2006-12-23 19:51 +0000 [r48927]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Use the correct function to destroy an
	  rwlock in the destructor for an ast_rwlock_t

2006-12-22 22:34 +0000 [r48871-48907]  Jason Parker <jparker@digium.com>

	* Makefile, /, main/stdtime/localtime.c: Merged revisions 48906 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48906 | qwell | 2006-12-22 16:33:46 -0600 (Fri, 22 Dec 2006) | 2
	  lines Minor fixes for Solaris. ........

	* /, channels/chan_skinny.c: Merged revisions 48888 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec
	  2006) | 2 lines Note to self: Run make before committing...
	  ........

	* /, channels/chan_skinny.c: Merged revisions 48870 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec
	  2006) | 2 lines Fix for issue 7774 - patch by alamantia ........

2006-12-22 10:35 +0000 [r48825-48857]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_sms.c: improve readability of a few macros.

	* apps/app_sms.c: make sms_hexdump() thread safe; restructure and
	  reduce indentation on some blocks.

	* apps/app_sms.c: make isodate thread-safe

	* apps/app_sms.c: - use the standard option parsing routines; -
	  document existing but undocumented parameters to send a message
	  (untested but unchanged; - ad a new option p(N) to set the
	  initial message delay to N ms so this can be adapted from the
	  dialplan to various countries;

2006-12-21 21:57 +0000 [r48785-48817]  Joshua Colp <jcolp@digium.com>

	* main/logger.c: Merge non-blocking logger from my branch. This
	  should improve things under heavy load with lots of CLI/logging
	  output.

	* /, redhat/asterisk.spec: Merged revisions 48783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48783 | file | 2006-12-21 15:26:29 -0500 (Thu,
	  21 Dec 2006) | 10 lines Merged revisions 48782 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
	  lines Add new silence sound files to the spec for Redhat. (issue
	  #8652 reported by alvaro_palma_aste) ........ ................

2006-12-21 20:15 +0000 [r48781]  Steve Murphy <murf@digium.com>

	* codecs/codec_g722.c: This little mod gets rid of that g722
	  compiler warning that breaks builds configured with
	  --enable-dev-mode; the previous commit of 48767 was to merge in
	  changes for bug 6334, unifying the open mode arguments for saner
	  operation.

2006-12-21 19:52 +0000 [r48768]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_sms.c: put generator functions next to each other.

2006-12-21 19:44 +0000 [r48767]  Steve Murphy <murf@digium.com>

	* include/asterisk.h, channels/chan_zap.c, apps/app_meetme.c,
	  apps/app_festival.c, apps/app_dictate.c, apps/app_record.c,
	  res/res_convert.c, channels/chan_iax2.c, res/res_monitor.c,
	  cdr/cdr_sqlite.c, res/res_agi.c, main/file.c, main/app.c,
	  apps/app_sms.c, apps/app_directory.c, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, main/db.c, apps/app_voicemail.c: a quick
	  fix to app_sms.c to get rid of cursed compiler warnings so I can
	  compile under --enable-dev-mode

2006-12-21 19:36 +0000 [r48736-48766]  Luigi Rizzo <rizzo@icir.org>

	* main/channel.c: same as in other places, check that
	  generator->release is not NULL before calling it. This allows
	  generators to set it to NULL when they have nothing to do there.
	  Later, the three copies of the code that releases a generator
	  should be moved to a function.

	* apps/app_sms.c: reduce indentation

	* apps/app_sms.c: restructure a block to reduce nesting

	* apps/app_sms.c: Add a bit of documentation on this code,
	  including pointers to relevant documents and comment on timing
	  issues. Initial merge of the code in
	  http://bugs.digium.com/view.php?id=8586 by Filippo Grassilli
	  (Hyppo) to support the SMS Protocol 2. In this commit i have
	  tried to minimize the diffs, so further code cleanup will come in
	  subsequent commits.

2006-12-21 15:52 +0000 [r48723]  Steve Murphy <murf@digium.com>

	* pbx/pbx_config.c: This small update will generate WARNINGS if
	  there is garbage in your extensions.conf file (liken extem =>
	  instead of exten => !)

2006-12-21 04:05 +0000 [r48680-48709]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/indications.h, main/indications.c: Really clean
	  up indications to use the linkedlists API

	* main/pbx.c: Switch list of global variables to read/write locks.

	* main/pbx.c: Convert alternate dialplan switch list to use
	  read/write locks.

2006-12-21 00:24 +0000 [r48663]  Steve Murphy <murf@digium.com>

	* configs/iax.conf.sample, main/jitterbuf.c, include/jitterbuf.h,
	  CHANGES, channels/chan_iax2.c: As per bug 7978, this version
	  introduces the jittertargetextra option in config files

2006-12-21 00:11 +0000 [r48661-48662]  Matthew Fredrickson <creslin@digium.com>

	* codecs/codec_g722.c: Minor addition giving props to Steve
	  Underwood for his hard work. Thanks again Steve!

	* codecs/Makefile, codecs/g722/Makefile (added),
	  codecs/codec_g722.c (added), codecs/g722/g722_encode.c (added),
	  codecs/g722 (added), build_tools/embed_modules.xml,
	  codecs/g722/g722_decode.c (added), codecs/g722/g722.h (added),
	  codecs/g722_slin_ex.h (added), codecs/slin_g722_ex.h (added): Add
	  codec G.722 support.

2006-12-20 04:32 +0000 [r48638-48639]  Joshua Colp <jcolp@digium.com>

	* apps/app_page.c: Clean up app_page

	* /, apps/app_voicemail.c: Merged revisions 48637 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48637 | file | 2006-12-19 21:56:09 -0500 (Tue, 19 Dec 2006) | 2
	  lines vms doesn't exist on non-IMAP storage builds. ........

2006-12-20 00:13 +0000 [r48598-48599]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_sms.c: more formatting cleanup. Move some code into a
	  function sms_compose1() in preparation for supporting protocol 2
	  as well.

	* apps/app_sms.c: formatting and code cleanup. Still a lot of
	  copy&pasted code here...

2006-12-19 23:05 +0000 [r48591-48597]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48596 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48596 | file | 2006-12-19 18:04:30 -0500 (Tue, 19 Dec 2006) | 2
	  lines Pass 'vms' pointer to record_and_review so it is then
	  passed to the IMAP store file function. (issue #8614 reported by
	  punknow) ........

	* res/snmp/agent.c: Update res_snmp to use new API declaration of
	  pbx_builtin_serialize_variables (issue #8627 reported by
	  johann8384)

	* /, doc/snmp.txt: Merged revisions 48592 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48592 | file | 2006-12-19 17:00:57 -0500 (Tue, 19 Dec 2006) | 2
	  lines find is not the same as bind when it comes to
	  documentation. (issue #8626 reported by johann8384) ........

	* res/res_limit.c: OpenBSD does not have RLIMIT_AS or RLIMIT_VMEM
	  so until someone finds the right rlimit to use then let's not
	  support the -v option on OpenBSD. (issue #8543 reported by jtodd)

2006-12-19 21:32 +0000 [r48588-48589]  Luigi Rizzo <rizzo@icir.org>

	* /: block 48583

	* apps/app_sms.c: start documenting this code. On passing, fix the
	  bogus datalen on outgoing frames just fixed in 1.4 rev.48583

2006-12-19 21:28 +0000 [r48587]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/Makefile: Merged revisions 48586 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006)
	  | 2 lines suppress compiler warnings in this module until it can
	  be improved ........

2006-12-19 16:36 +0000 [r48580-48581]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dial.c: better name for struct dial_localuser.

	* main/cli.c: remove now useless extern declarations.

2006-12-19 14:57 +0000 [r48578]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, /, funcs/func_odbc.c: Merged revisions
	  48577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48577 | kpfleming | 2006-12-19 08:57:09 -0600 (Tue, 19 Dec 2006)
	  | 2 lines use the proper variable type for these unixODBC API
	  calls, eliminating warnings on 64-bit platforms that use the
	  'new' 64-bit types for ODBC API calls ........

2006-12-19 09:58 +0000 [r48573-48575]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dial.c: introduce a temporary variable for tmp->chan to
	  shorten expressions.

	* apps/app_dial.c: stop what i think is a memory leak in case Dial
	  fails to connect to a channel. Before committing to 1.4 i would
	  like some other people to review and test this fix - thanks.

	* apps/app_dial.c: move a large block related to privacy handling
	  to a separate function.

2006-12-19 03:47 +0000 [r48572]  Joshua Colp <jcolp@digium.com>

	* Makefile, /: Merged revisions 48571 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48571 | file | 2006-12-18 22:46:12 -0500 (Mon, 18 Dec 2006) | 2
	  lines Use env -i to start a fresh environment when going to build
	  menuselect. This is more portable then using unset. (issue #8543
	  reported by jtodd) ........

2006-12-18 17:44 +0000 [r48568]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/channel.h: unbreak the macro used for
	  incrementing the frame counters. I don't know when the bug was
	  introduced, but with the typical usage c->fin =
	  FRAMECOUNT_INC(c->fin) the frame counters stay to 0.

2006-12-18 17:30 +0000 [r48565-48567]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Clean up find_idle_thread function and use
	  atomic operations for dynamic thread count.

	* /, channels/chan_iax2.c: Merged revisions 48564 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2
	  lines Put thread into proper list if we abort handling due to an
	  error, and also hold the lock while putting it back into the
	  proper idle list so we don't prematurely get a signal. (issue
	  #8604 reported by arkadia) ........

2006-12-18 16:57 +0000 [r48562-48563]  Jason Parker <jparker@digium.com>

	* configure.ac: ctrl-w != w (nano search) (surprisingly, the fix
	  was ever so slightly different in 1.4)

2006-12-18 16:24 +0000 [r48558-48560]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/strings.h: apply the proposed fix for bug 8602
	  http://bugs.digium.com/view.php?id=8602 (i am not sure if there
	  is still missing cast in front of the alloca() call - being a
	  macro this is probably triggered only when actually used). Add
	  function ast_str_reset() to reinitialize the string to an empty
	  string instead of playing with the internal fields.

	* main/cdr.c, main/pbx.c, apps/app_dumpchan.c,
	  include/asterisk/cdr.h, include/asterisk/pbx.h, apps/app_queue.c,
	  main/cli.c: convert the final clients of ast_build_string to use
	  ast_str_*() Now the only module left using it is chan_sip.c

	* main/logger.c: debugging shows that we always need more than 128
	  bytes for the verbose and logging messages so start with a larger
	  buffer from the beginning.

2006-12-18 11:59 +0000 [r48555]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/Makefile, codecs/gsm/Makefile, utils/astman.c,
	  utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c,
	  codecs/lpc10/Makefile: Merged revisions 48554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006)
	  | 3 lines remove some now-unnecessary explicit includes of
	  autoconfig.h clean up per-file dependencies during 'make clean'
	  ........

2006-12-18 11:28 +0000 [r48550-48553]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: Replace ast_build_string with ast_str_*(). On
	  passing remove presumably duplicate code to generate the message
	  for the manager_hooks: in the previous version, the message was
	  almost the same as the one sent to regular sessions, with the
	  exception of the empty line at the end, and a few (presumably
	  unintentional) differences e.g. timestamps, debugging, and
	  lowercase headers for "event" and "privilege". now we reuse the
	  same message as before.

	* funcs/func_realtime.c: replace ast_build_string() with
	  ast_str_*(). Unless i am very mistaken, function_realtime_read()
	  was broken in that it would always return an empty string
	  (because ast_build_string() advanced the pointer to the end of
	  the string, and there was no reference to the initial value. This
	  commit should fix this problem.

	* apps/app_queue.c: replace ast_build_string() with ast_str_*();
	  simplify __queues_show()

2006-12-17 18:33 +0000 [r48549]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/prep_tarball: Merged revisions 48548 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r48548 | kpfleming | 2006-12-17 12:33:24 -0600 (Sun, 17
	  Dec 2006) | 2 lines need an additional argument here to make the
	  downloads actually occur ........

2006-12-17 12:47 +0000 [r48543]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: define a mask SIP_INSECURE sam as for other
	  sets of flags.

2006-12-16 21:38 +0000 [r48522-48529]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: Merged revisions 48528 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48528 | kpfleming | 2006-12-16 15:34:41 -0600 (Sat, 16 Dec 2006)
	  | 2 lines use m4 quoting for AC_MSG_NOTICE calls, to keep these
	  calls from thinking they have multiple arguments ........

	* /, agi, codecs, utils, main/Makefile, apps,
	  Makefile.moddir_rules, Makefile.rules, cdr, codecs/ilbc, formats,
	  utils/Makefile, agi/Makefile, Makefile, funcs, main/db1-ast,
	  codecs/lpc10, build_tools/mkdep (removed), main, codecs/gsm, res,
	  pbx, channels: Merged revisions 48525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006)
	  | 2 lines simplify dependency tracking system, using the
	  compiler's built-in method for generating them, and only doing
	  dependency tracking if developer mode is enabled via the
	  configure script ........

	* funcs/func_curl.c: update to use trunk's version of the
	  threadstorage API

	* Makefile, include/asterisk.h, /, main/stdtime/localtime.c: Merged
	  revisions 48521 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006)
	  | 2 lines since we really, really have to have autoconfig.h
	  included before all other headers (especially system headers),
	  the Makefile will now force it to happen (this will fix build
	  problems with files like ast_expr2f.c, where we can't control the
	  inclusion order in the file itself) ........

2006-12-16 11:23 +0000 [r48515-48520]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c: forgot this part...

	* main/cli.c: another conversion from ast_build_str to ast_str

	* main/translate.c: convert ast_build_str to ast_str_*

	* include/asterisk/http.h, main/manager.c, main/http.c,
	  include/asterisk/strings.h: replace ast_build_string() with
	  ast_str_*() functions. This makes the code easier to follow and
	  saves some copies to intermediate buffers.

2006-12-16 04:25 +0000 [r48514]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs/func_curl.c, /: Merged revisions 48513 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48513 | kpfleming | 2006-12-15 22:25:09 -0600 (Fri, 15 Dec 2006)
	  | 2 lines instead of initializing the curl library every time the
	  CURL() function is invoked, do it only once per thread (this
	  allows multiple calls to CURL() in the dialplan for a channel to
	  run much more quickly, and also to re-use connections to the
	  server) (thanks to JerJer for frequently complaining about this
	  performance problem) ........

2006-12-16 02:42 +0000 [r48508-48512]  Luigi Rizzo <rizzo@icir.org>

	* res/res_limit.c: prevent a compiler warning

	* main/manager.c, main/logger.c, main/utils.c,
	  include/asterisk/strings.h, main/cli.c: simplify the
	  ast_dynamic_str_*.... routines by renaming them to ast_str ...
	  and putting the struct ast_threadstorage pointer into the struct
	  ast_str. This makes the code a lot more readable. At this point
	  we can use these routines also to replace ast_build_string().

	* include/asterisk/utils.h, main/utils.c,
	  include/asterisk/strings.h, include/asterisk/threadstorage.h:
	  move the dynamic string support in a better place i.e. string.h
	  While doing this, add a bit of documentation, and slightly extend
	  the functionality as follows: + a max_len of -1 means that we
	  take whatever the current size is, and never try to extend the
	  buffer; + add support for alloca()-ted dynamic strings, which is
	  very useful for all cases where we do an ast_build_string() now.
	  Next step is to simplify the interface by using shorter names
	  (e.g. ast_str as a prefix) and removing the _thread variant of
	  the functions by saving the threadstorage reference into the
	  struct ast_str. This can be done by overloading the 'type' field.
	  Finally, I will do my best to remove the convoluted interface
	  that results from trying to support platforms without va_copy().

	* res/res_smdi.c: remove a duplicate include

2006-12-15 19:57 +0000 [r48503-48507]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 48506 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2
	  lines Turn payload_lock into bridge_lock and make it encompass
	  all RTP structure contents that may relate to bridge information,
	  including who we are bridged to. ........

	* /, channels/chan_iax2.c: Merged revisions 48504 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2
	  lines Hold call structure lock in places where a qualify or peer
	  action can destroy it. ........

	* /, channels/chan_iax2.c: Merged revisions 48502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2
	  lines Lock network retransmission queue in all places that it is
	  used. ........

2006-12-15 18:37 +0000 [r48495-48501]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: unbreak the output for http session. Not long ago
	  i replaced lseek() with fseek() but forgot that filr FILE's you
	  need ftell to give you the current position.

	* main/channel.c, include/asterisk/channel.h: remove
	  ast_safe_string_alloc() - it is completely equivalent to
	  asprintf().

	* channels/chan_zap.c: replace ast_safe_string_alloc() with
	  asprintf()

	* channels/chan_features.c: replace ast_safe_string_alloc() with
	  asprintf()

	* include/asterisk/threadstorage.h: small documentation
	  improvements.

2006-12-15 13:36 +0000 [r48485-48491]  Olle Johansson <oej@edvina.net>

	* main/tdd.c, include/asterisk/tdd.h: Doxygen changes

	* /, channels/chan_sip.c: Issue #8592 - treat 504 as congestion
	  (imported from 1.2/1.4)

	* /, channels/chan_sip.c: Update to latest IANA specs

2006-12-15 06:34 +0000 [r48479-48480]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/lock.h: Add support to see what holds the lock
	  when doing trylock.

	* /, channels/chan_iax2.c: Merged revisions 48478 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2
	  lines Use a wakeup variable so that we don't wait on IO
	  indefinitely if packets need to be retransmitted. ........

2006-12-15 04:03 +0000 [r48476-48477]  Luigi Rizzo <rizzo@icir.org>

	* main/channel.c, include/asterisk/channel.h: constify
	  ast_state2str() and note it is not reentrant.

	* main/pbx.c, include/asterisk/channel.h: remove the macro LOAD_OH
	  and expand it inline in the only place where it was used.

2006-12-14 17:39 +0000 [r48462-48473]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/rtp.h, main/rtp.c: Merged revisions 48472 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2
	  lines Payload values on the RTP structure can change AFTER a
	  bridge has started. This comes from the packet handling of the
	  SIP response when indication that it was answered has been sent.
	  Therefore we need to protect this data with a lock when we
	  read/write. (issue #8232 reported by tgrman) ........

	* /, main/rtp.c: Merged revisions 48461 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2
	  lines Remove direct RTCP bridging. I've come to the conclusion
	  that we should handle this through the core and not just forward
	  it on. Should solve a few bugs. ........

2006-12-13 23:08 +0000 [r48458-48459]  Luigi Rizzo <rizzo@icir.org>

	* main/pbx.c: make sure that showdialplan sends only one 'Response:
	  Success ' message even in case of a recursive call.

	* main/pbx.c: clean up function manager_show_dialplan_helper()
	  reducing indentation and normalizing loops. While doing this,
	  remove some unused variables, fix an uninitialized string
	  (idaction), and mark some places where the behaviour is not what
	  we would expect (e.g. an empty context is reported as an error
	  same as a non-existent one). Given that this function is not in
	  1.4, the above can be changed without too many backward
	  compatibility concerns. Not applicable to 1.4 or below.

2006-12-13 21:23 +0000 [r48455]  Matt O'Gorman <mogorman@digium.com>

	* codecs/codec_zap.c: support for deactivating translation paths
	  that are no longer available and more descriptive show transcoder
	  cli command.

2006-12-13 00:56 +0000 [r48433]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: revert check for a zaptel transcoder related
	  definition that was done in the wrong module.

2006-12-12 23:28 +0000 [r48432]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/prep_tarball: Merged revisions 48427 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r48427 | kpfleming | 2006-12-12 17:18:14 -0600 (Tue, 12
	  Dec 2006) | 2 lines when making a release, we can always use wget
	  and we can't run the configure script to find that out...
	  ........

2006-12-12 22:32 +0000 [r48416-48417]  Russell Bryant <russell@digium.com>

	* include/asterisk/app.h, channels/chan_sip.c,
	  include/asterisk/channel.h, include/asterisk/pbx.h: Fix various
	  spelling mistakes in comments.

	* channels/chan_zap.c: Make chan_zap inform you that your version
	  of zaptel is too old instead of just failing to compile. It seems
	  like the proper way to do this would be in the configure script.
	  However, that wouldn't help existing checkouts unless we forced
	  the configure script to be executed after any code was changed.

2006-12-12 19:55 +0000 [r48415]  Matt O'Gorman <mogorman@digium.com>

	* codecs/codec_zap.c: fixed nubb error on my part, transcoder now
	  unlocks and locks correctly, as well as counts in the correct
	  direction.

2006-12-12 10:36 +0000 [r48408-48410]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: properly initialize a malloc'ed buffer

	* main/manager.c: normalize the scanning of "general" options in
	  the config file.

	* main/cli.c: Make sure tab-completion works even when we have
	  typed a fully matching word (e.g. "sip<TAB>"); this is
	  implemented by this one-line change - for (;; dst++, src += n) {
	  + for (;src < argindex; dst++, src += n) { However this code is
	  not exactly trivial to understand, so i am also adding some
	  comments to help figuring out what it does.

2006-12-12 04:14 +0000 [r48402]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48401 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48401 | file | 2006-12-11 23:13:48 -0500 (Mon, 11 Dec 2006) | 2
	  lines Use S_OR in my previous app_voicemail. This is the way it
	  should have been done. ........

2006-12-11 23:02 +0000 [r48397-48400]  Matt O'Gorman <mogorman@digium.com>

	* /, sounds/Makefile: Merged revisions 48399 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ........ r48399
	  | mogorman | 2006-12-11 17:02:10 -0600 (Mon, 11 Dec 2006) | 2
	  lines new sounds package with 100% more silence ........

	* /, apps/app_externalivr.c: Merged revisions 48396 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.4
	  ................ r48396 | mogorman | 2006-12-11 16:11:35 -0600
	  (Mon, 11 Dec 2006) | 12 lines Merged revisions 48394 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.2 ........
	  r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
	  | 4 lines app_externalivr needs a real silence file, and
	  additional changes to add silence files into core instead of
	  extra patch provided by bug 8177 with minor additions. ........
	  ................

2006-12-11 21:35 +0000 [r48392]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48391 | file | 2006-12-11 16:31:23 -0500 (Mon, 11 Dec 2006) | 2
	  lines Return non-existant callerid handling to that which it was
	  before. In 1.4 and trunk callerid can be allocated but not have
	  any contents so we have to use ast_strlen_zero before passing it
	  to the relevant functions. (issue #8567 reported by pabelanger)
	  ........

2006-12-11 21:04 +0000 [r48390]  Matt O'Gorman <mogorman@digium.com>

	* codecs/codec_zap.c: add support for dynamic channel creation and
	  destruction, and show transcoder to show number of channels in
	  use.

2006-12-11 18:11 +0000 [r48389]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: make sure the argument to ast_malloc() is > 0.
	  Long explaination: The behaviour of the underlying malloc(0)
	  differs depending on the operating system. Some return NULL (SysV
	  behaviour); some still allocate a small chunk of memory and
	  return a valid pointer (e.g. traditional BSD); some (e.g. FreeBSD
	  6.x) return a non-null pointer that causes a memory fault if
	  used, even just for reading. Given the above variety, better
	  never call malloc(0).

2006-12-11 17:00 +0000 [r48388]  Steve Murphy <murf@digium.com>

	* main/app.c: This update fixes the problem reported in bug 8551;
	  that ast_app_getdata() behaves differently in trunk for the case
	  of a null prompt.

2006-12-11 05:40 +0000 [r48384]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 48382 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48382 | tilghman | 2006-12-10 23:37:09 -0600 (Sun, 10 Dec 2006)
	  | 2 lines STRFTIME() does not actually require an argument (issue
	  8540) ........

2006-12-11 05:38 +0000 [r48378-48383]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 48381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2
	  lines Merge in my latest RTP changes. Break out RTP and RTCP
	  callback functions so they no longer share a common one. ........

	* /, apps/app_meetme.c: Merged revisions 48379 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48379 | file | 2006-12-11 00:30:01 -0500 (Mon, 11 Dec 2006) | 2
	  lines Use the correct API call to say a device state changed.
	  (Yes, I'm a nub.) ........

	* /, apps/app_meetme.c: Merged revisions 48377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48377 | file | 2006-12-10 23:57:38 -0500 (Sun, 10 Dec 2006) | 2
	  lines Don't access the conference structure after it has been
	  freed. ........

2006-12-11 00:52 +0000 [r48376]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
	  res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
	  apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48375
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48375 | tilghman | 2006-12-10 18:47:21 -0600
	  (Sun, 10 Dec 2006) | 13 lines Merged revisions 48374 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10
	  Dec 2006) | 5 lines When doing a fork() and exec(), two problems
	  existed (Issue 8086): 1) Ignored signals stayed ignored after the
	  exec(). 2) Signals could possibly fire between the fork() and
	  exec(), causing Asterisk signal handlers within the child to
	  execute, which caused nasty race conditions. ........
	  ................

2006-12-10 03:14 +0000 [r48373]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 48372 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat,
	  09 Dec 2006) | 9 lines Merged revisions 48371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
	  line This version applies the patch suggested by stevens in bug
	  7836 (make inbound channel RINGING state consistent with other
	  channels). ........ ................

2006-12-09 16:44 +0000 [r48359-48365]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: convert the thread IO state and type to use
	  enums.

	* /, channels/chan_iax2.c: Merged revisions 48363 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) |
	  8 lines Use locking when accessing the registrations list. This
	  list is not actually used very often, so the likelihood of there
	  being a problem is pretty small, but still possible. For example,
	  if the CLI command to list the registrations was called at the
	  same time that a reload was occurring and the registrations list
	  was getting destroyed and rebuilt, a crash could occur. In
	  passing, go ahead and convert this list to use the linked list
	  macros. ........

	* channels/chan_iax2.c: chan_iax2 has an extremely large function,
	  socket_process(), to handle incoming frames. The function, before
	  this commit, was roughly 1400 lines long. So, I am working on
	  breaking this up into functions so that the code is easier to
	  follow and debug. Also, I will be committing these changes in
	  chunks as I do them to ease tracking down any potentially
	  introduced problems. Break out roughly 150 lines from
	  socket_process() and introduce a new function,
	  socket_process_meta() which handles the parsing of an incoming
	  meta frame. Also, restructure some of this code a bit to reduce
	  the deep nesting that was in this code.

	* channels/chan_iax2.c: - Fix a few spelling mistakes - Use
	  sizeof() to pass an array size to a function - Use a single bit
	  for a variable in the chan_iax2_pvt stuct since that is all it
	  needs. - Add some comments about the iaxs, iaxl, and lastused
	  arrays.

2006-12-07 18:21 +0000 [r48358]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 48357 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48357 | russell | 2006-12-07 13:17:28 -0500
	  (Thu, 07 Dec 2006) | 11 lines Merged revisions 48356 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
	  Dec 2006) | 3 lines Ensure that the file position is not
	  incremented beyond the total number of files available for
	  playback. (issue #8539, ulogic) ........ ................

2006-12-07 16:42 +0000 [r48351]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/http.h, main/manager.c, main/http.c,
	  configs/manager.conf.sample: - Generalize the function
	  ssl_setup() so that the certificate info are passed as an
	  argument. - Update the code in main/http.c to use the new
	  interface (the diff is large but mostly mechanical, due to the
	  name change of several variables); - And since now it is trivial,
	  implement "AMI over TLS", and document the possible options in
	  manager.conf - And since the test client (openssl s_client
	  -connect host:port ) does not generate \r\n as a line terminator,
	  make get_input() also accept just a \n as a line terminator (Mac
	  users: do you also need the \r-only version ?) The option parsing
	  in manager.conf is not very efficient, and needs to be cleaned up
	  and made similar to what we have in http.conf

2006-12-07 16:03 +0000 [r48350]  Steve Murphy <murf@digium.com>

	* main/manager.c, /: Merged revisions
	  47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri,
	  24 Nov 2006) | 6 lines Doxygen update - Document cause codes -
	  Document a bit more on channel variables - global, predefined and
	  local - Fix some doxygen in channel.h. Adding one comment for two
	  definitions does not work. They won't be copied to each.
	  ................ r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri,
	  24 Nov 2006) | 1 line This fix inspired by a patch supplied in
	  bug 8189, which points out problems with the PLC code
	  ................ r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri,
	  24 Nov 2006) | 1 line removed the svnmerge-integrated property
	  from trunk; it's confusing svnmerge in newly created branches
	  ................ r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat,
	  25 Nov 2006) | 5 lines set pointers to NULL after freeing memory
	  to avoid multiple free() probably 1.4/1.2 issue as well if
	  someone can look into that. ................ r48003 | oej |
	  2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines - Adding
	  comment on suspicious memory allocation. Seems like it's never
	  freed, but I don't have a clear understanding of the frame
	  allocation/deallocation, so I just mark this for investigation.
	  (Reported by Ed Guy). We're trying to see if a free() hurts... -
	  Doxygen comments on p2p rtp bridge stuff. I am a bit worried
	  about shortcutting rtcp this way, but will need feedback from
	  rtcp gurus. This should work for video calls too, and possibly
	  UDPTL. ................ r48004 | oej | 2006-11-25 02:48:30 -0700
	  (Sat, 25 Nov 2006) | 2 lines Changing ERROR to lesser level.
	  Imported from 1.2/1.4 ................ r48008 | rizzo |
	  2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines generalize
	  a bit the functions used to create an tcp socket and then run a
	  service on it. The code in manager.c does essentially the same
	  things, so we will be able to reuse the code in here (probably
	  moving it to netsock.c or another appropriate library file).
	  ................ r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat,
	  25 Nov 2006) | 1 line Updates to show linkset command
	  ................ r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat,
	  25 Nov 2006) | 2 lines Add ss7 show linkset command
	  ................ r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat,
	  25 Nov 2006) | 1 line Make sure we don't send a group reset on a
	  group larger than 32 CICs ................ r48012 | mattf |
	  2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line bug fix
	  ................ r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat,
	  25 Nov 2006) | 1 line Make compiler happier ................
	  r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1
	  line Little fix so we use the right message ................
	  r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9
	  lines Merged revisions 48015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1
	  line A little bit of func_cdr documentation upgrade-- no bug#
	  involved, although 8221 may have inspired it. ........
	  ................ r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat,
	  25 Nov 2006) | 9 lines Merged revisions 48017 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1
	  line might as well also document the raw values of the flag vars
	  ........ ................ r48019 | russell | 2006-11-25 23:55:33
	  -0700 (Sat, 25 Nov 2006) | 6 lines - Add some comments on thread
	  storage with a brief explanation of what it is as well as what
	  the motivation is for using it. - Add a comment by the
	  declaration of ast_inet_ntoa() noting that this function is not
	  reentrant, and the result of a previous call to the function is
	  no longer valid after calling it again. ................

2006-12-06 20:46 +0000 [r48332-48338]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: remove duplicated code to start the server
	  threads, use the infrastructure exposed in http.c earlier today.
	  As a bonus, now we can restart the session on a different port
	  just reloading the module. On passing, fix a bug in the handling
	  of 'enabled' in the configuration file - previously, a missing
	  "enabled=" line in manager.conf meant "whatever the state was
	  before" instead of a specific value.

	* main/manager.c: Part of the transformations necessary to add TLS
	  support, as described in
	  http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html
	  In detail, this commit does the following: b) change the function
	  get_input() to use fread() instead of read() to collect the data.
	  One can still do the ast_wait_for_input() on the original
	  descriptor returned by accept(). c) change the function
	  send_string() to work on the FILE *. As a side effect, this
	  change now really guarantees that we don't spend more than
	  "writetimeout" milliseconds on each line sent. d) modify the
	  function action_command() so that it creates a temporary file
	  descriptor to be passed to ast_cli_command(), and then read back
	  the data from the temp file and write it to the output with
	  send_string(). The code is similar to what is done in
	  generic_http_callback() to support AMI-over-HTTP.

2006-12-06 16:54 +0000 [r48327]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Handle multiple 487's correctly

2006-12-06 16:19 +0000 [r48325]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, /: Merged revisions 48323 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48323 | russell | 2006-12-06 11:15:45 -0500
	  (Wed, 06 Dec 2006) | 11 lines Merged revisions 48322 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
	  Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
	  in the sample configuration file. (issue #8526, arkadia) ........
	  ................

2006-12-06 16:17 +0000 [r48324]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/http.h, main/http.c: Make externally visible
	  some generic code useful to create and implement services over
	  tcp and/or tcp-tls. This commit is nothing more than moving
	  structure definitions (and documentation) from main/http.c to
	  include/asterisk/http.h (temporary location until we find a
	  better place), and removing the 'static' qualifier from
	  server_root() and server_start(). The name change (adding the
	  ast_ prefix as a minimum, and then possibly a more meaningful
	  name) is postponed to future commits. Does not apply to other
	  versions of asterisk.

2006-12-06 12:34 +0000 [r48318]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't send Contact in SIP Messages
	  (imported from 1.2/1.4). Reported by Gunnar at Omnitor.

2006-12-06 07:39 +0000 [r48299-48307]  Russell Bryant <russell@digium.com>

	* apps/app_osplookup.c, apps/app_meetme.c, apps/app_queue.c,
	  apps/app_voicemail.c: Resolve some pointer signedness compiler
	  warnings in app_osplookup, and constify a bunch of usage strings
	  for CLI commands.

	* channels/chan_local.c, channels/chan_skinny.c,
	  channels/chan_agent.c, channels/chan_features.c,
	  channels/chan_alsa.c, channels/iax2-provision.c,
	  channels/chan_gtalk.c, channels/chan_oss.c, channels/chan_mgcp.c:
	  Constify a bunch of usage strings for CLI commands.

	* res/res_config_pgsql.c, res/res_limit.c, res/res_agi.c,
	  res/res_crypto.c, res/res_realtime.c, res/res_jabber.c,
	  res/res_odbc.c: Constify a bunch of usage strings for CLI
	  commands.

	* main/channel.c, main/udptl.c, main/frame.c, main/translate.c,
	  main/file.c, pbx/pbx_dundi.c, main/db.c, main/rtp.c: Staticize
	  one, and Constify a bunch of usage strings for CLI commands.

	* channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
	  main/asterisk.c, main/cli.c: Constify a bunch of the usage
	  strings for CLI commands.

	* channels/chan_iax2.c: Instead of creating an unused instance of
	  an unnamed enum, give it a name.

	* include/asterisk/cli.h: Make the "usage" member of the
	  ast_cli_entry struct const to resolve a compiler warning.

2006-12-05 20:46 +0000 [r48282]  Joshua Colp <jcolp@digium.com>

	* configure: Regenerate configure for Qwell's last commit.

2006-12-05 20:44 +0000 [r48280]  Jason Parker <jparker@digium.com>

	* /, configure.ac: Merged revisions 48279 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48279 | qwell | 2006-12-05 14:42:52 -0600 (Tue, 05 Dec 2006) | 4
	  lines Fix curl version number testing to be much more friendly to
	  non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX
	  compliant now.. ........

2006-12-05 20:39 +0000 [r48277]  Olle Johansson <oej@edvina.net>

	* include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c:
	  Doxygen updates

2006-12-05 20:15 +0000 [r48276]  Jason Parker <jparker@digium.com>

	* main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c,
	  main/fskmodem.c: Expand on r48273 (from issue 8506), to translate
	  more of the fskmodem stuff to English. r48273 dealt with the
	  comments and such, this deals with the code itself. (This
	  couldn't have been easily done if it weren't for 48273 - thanks
	  again for that merbanan)

2006-12-05 19:41 +0000 [r48269-48273]  Olle Johansson <oej@edvina.net>

	* include/asterisk/fskmodem.h, main/fskmodem.c: Issue #8506 -
	  translate spanish comments in fskmodem to english (according to
	  bug guidelines) Thanks merbanan!

	* /: Blocking invitestate patch that is already merged to svn
	  trunk.

	* /, configs/sip.conf.sample: Adding docs on t.38

2006-12-05 14:33 +0000 [r48266]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: 1. Change to remove the compiling warning:
	  "app_osplookup.c:2169: warning: initialization discards
	  qualifiers from pointer target type"

2006-12-05 11:09 +0000 [r48258-48259]  Olle Johansson <oej@edvina.net>

	* main/frame.c, include/asterisk/frame.h, main/rtp.c: Well, yes...

	* main/frame.c, include/asterisk/frame.h, main/rtp.c: Reserving
	  flags for coming code (currently in the "videocaps" branch)
	  implementing T.140 support in RTP. T.140/RFC 4351 is TDD over IP
	  - text telephony for hearing impaired. It defines a realtime text
	  chat, much like the old "talk" application in Unix. T.140 is
	  character by character in real time. It's not the same as our
	  current MESSAGE format - that is more like IM, but can be
	  gatewayed to MESSAGE with a text "codec" if needed. More patches
	  will follow, as soon as we've separated this code from the video
	  capabilities functions in the videocaps branch. Code by John
	  Martin, Aupix (disclaimer on file)

2006-12-05 01:46 +0000 [r48253-48255]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48254 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48254 | tilghman | 2006-12-04 19:41:02 -0600 (Mon, 04 Dec 2006)
	  | 2 lines Oops, forgot to release the odbc handle ........

	* /, apps/app_voicemail.c: Merged revisions 48252 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48252 | tilghman | 2006-12-04 19:34:34 -0600
	  (Mon, 04 Dec 2006) | 14 lines Merged revisions 48251 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04
	  Dec 2006) | 6 lines If the recording in the database is too
	  large, it will fail to retrieve with an mmap error. Not too sure
	  why this doesn't happen when we put it in the database, also, but
	  since that doesn't seem to be broken, I'm not going to fix it (at
	  least until someone reports it). Solution is to ask for the file
	  in smaller chunks. (Bug 8385) ........ ................

2006-12-04 21:49 +0000 [r48249]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48248 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48248 | qwell | 2006-12-04 15:48:41 -0600 (Mon, 04 Dec 2006) | 2
	  lines Fix an issue which didn't allow unavail/greet/busy/etc
	  messages from being saved into ODBC (and probably IMAP). ........

2006-12-04 17:55 +0000 [r48229-48231]  Jason Parker <jparker@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 48230 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4
	  lines Add documentation to voicemail.conf.sample for ODBC
	  storage. Issue 8499 - patch by blitzrage. ........

	* /, doc/snmp.txt: Merged revisions 48228 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48228 | qwell | 2006-12-04 11:43:24 -0600 (Mon, 04 Dec 2006) | 4
	  lines Attempt to document some of the dependencies that are
	  needed for net-snmp Issue 8499 - initial patch by blitzrage.
	  ........

2006-12-03 06:35 +0000 [r48224]  Russell Bryant <russell@digium.com>

	* /, sounds/Makefile: Merged revisions 48223 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48223 | russell | 2006-12-03 01:34:14 -0500 (Sun, 03 Dec 2006) |
	  3 lines When "fetch" is in use, instead of "wget", --continue is
	  not a valid option. (issue #8451) ........

2006-12-02 22:03 +0000 [r48200-48220]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Cleaning up handle_response a bit.
	  (Imported from 1.4)

	* .cleancount: Removing two .h files means we need to update
	  cleancount to force make depend again (or ?)

	* channels/chan_sip.c: Send CANCEL to call with early media
	  (PROGRESS INBAND). This is imported from branch "invitestate" and
	  "invitestate-1.4" *** *** *** IF YOU HAVE ISSUES WITH
	  BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN! *** Thank you! ***
	  *** /Olle

	* channels/chan_sip.c: Invitestate updates

	* agi/Makefile: Oops. Something is wrong in the agi directory.
	  Asking for autoconfig.h. I have it disabled locally, but no
	  reason to commit that change.

	* apps/app_sms.c: Doxygenification

	* main/coef_out.h (removed), main/tdd.c, main/callerid.c,
	  main/fskmodem.c, main/coef_in.h (removed): - Code formatting -
	  remove coef_in.h and coef_out.h that was only included as data
	  definitions in fskmodem.c If you understand spanish, please help
	  us translate the comments in fskmodem.c. Thanks!

	* /, channels/chan_sip.c, include/asterisk/rtp.h,
	  configs/sip.conf.sample, main/rtp.c: - Disable RTP timeouts
	  during T.38 transmission - Encapsulate RTP timers to the RTP
	  structure, so we have one set for video and one for audio -
	  Document RTP keepalive configuration option - Cleanup and
	  document the monitor support function to hangup on RTP timeouts -
	  Add RTP keepalive to SIP show settings Imported from 1.4 with
	  modifications for trunk.

2006-12-01 23:39 +0000 [r48194]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c, /: Merged revisions 48193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48193 | kpfleming | 2006-12-01 17:37:28 -0600
	  (Fri, 01 Dec 2006) | 10 lines Merged revisions 48192 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01
	  Dec 2006) | 2 lines if Dial() is going to send music-on-hold to
	  the calling party, it has to send PROGRESS first to ensure that
	  the reverse audio path has been setup first (BE-106) ........
	  ................

2006-12-01 23:20 +0000 [r48191]  Russell Bryant <russell@digium.com>

	* Makefile, /, configure, configure.ac, makeopts.in,
	  sounds/Makefile: Merged revisions 48190 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) |
	  12 lines FreeBSD 6.1 does not include wget by default. However,
	  it has fetch which will work just fine for our purposes of
	  downloading the sounds packages. So, check for both wget and
	  fetch and the configure script and use what was found to download
	  them. If neither one was found, and sound packages are selected
	  that must be downloaded, the install process will print out an
	  informative error message indicating the situation. Also, fix a
	  couple places where "make" was hard coded into some output
	  messages by replacing them with the $(MAKE) variable. (issue
	  #8451, initial patch by pabelanger, with additional modifications
	  by me) ........

2006-12-01 20:49 +0000 [r48188]  Olle Johansson <oej@edvina.net>

	* main/channel.c: Formatting fix

2006-12-01 20:26 +0000 [r48187]  Jason Parker <jparker@digium.com>

	* /, configs/extensions.conf.sample: Merged revisions 48186 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri,
	  01 Dec 2006) | 10 lines Merged revisions 48183 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
	  lines Fix a small typo - issue 8848, reported by pabelanger
	  ........ ................

2006-12-01 19:41 +0000 [r48180]  Tilghman Lesher <tlesher@digium.com>

	* /, main/cli.c: Merged revisions 48179 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006)
	  | 2 lines Double-unlock error (reported by blitzrage on IRC)
	  ........

2006-12-01 18:16 +0000 [r48175-48178]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, configs/sip.conf.sample: - Remove T.38
	  early media, since T.38 requires two way communication (imported
	  from 1.4) - Small fixes to limitonpeer

	* include/asterisk/threadstorage.h: Tiny doxygen improvement

2006-11-30 21:22 +0000 [r48169]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/rtp.h, channels/chan_gtalk.c, main/rtp.c:
	  Merged revisions 48168 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2
	  lines Do not do a partial bridge for Google Talk since we need to
	  handle STUN. (issue #8448 reported by phsultan) ........

2006-11-30 20:55 +0000 [r48164-48167]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue #8319 (imported from 1.2, 1.4) -
	  Increment nonce-count properly (noriyuki)

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
	  include/asterisk/channel.h, include/asterisk/pbx.h: Documentation
	  updates

2006-11-30 20:29 +0000 [r48153-48163]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 48158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48158 | file | 2006-11-30 15:07:55 -0500 (Thu,
	  30 Nov 2006) | 10 lines Merged revisions 48157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
	  lines Only print out debug message if bridged channel is not
	  NULL. (issue #8412 reported by jubilex) ........ ................

	* /, res/res_features.c: Merged revisions 48155 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48155 | file | 2006-11-30 14:05:14 -0500 (Thu,
	  30 Nov 2006) | 10 lines Merged revisions 48154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
	  lines Do not listen for DTMF on the bridge that comes into
	  existence when ParkedCall is executed. This means native bridging
	  can now occur for this. (issue #8406 reported by kebl0155)
	  ........ ................

	* main/cdr.c, /: Merged revisions 48152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48152 | file | 2006-11-30 13:47:40 -0500 (Thu,
	  30 Nov 2006) | 10 lines Merged revisions 48151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
	  lines Print certain CDR messages out at the NOTICE level versus
	  WARNING since they can occur when used with the CDR applications
	  and are perfectly fine. (issue #8367 reported by dartvader)
	  ........ ................

2006-11-30 18:25 +0000 [r48149-48150]  Olle Johansson <oej@edvina.net>

	* main/devicestate.c: Small update

	* agi/Makefile, contrib/asterisk-ng-doxygen, agi/eagi-test.c,
	  main/devicestate.c, agi/eagi-sphinx-test.c: Doxygen updates

2006-11-30 18:20 +0000 [r48144-48148]  Joshua Colp <jcolp@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 48143 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu,
	  30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2
	  lines Document 'port' for SIP peers, came up because of the
	  current mailing list thread. (issue #8450 reported by blitzrage)
	  ........ ................

2006-11-30 17:15 +0000 [r48130-48139]  Olle Johansson <oej@edvina.net>

	* include/asterisk/doxyref.h, main/devicestate.c: Adding some
	  generic docs on extension and device states - watchers and
	  providers

	* doc/manager.txt, /: Add information on status events

	* /, channels/chan_sip.c: Merging patch from 1.2/1.4. I think this
	  was originally spotted by Luigi, but hit me in the back today.

2006-11-30 03:29 +0000 [r48116-48123]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: I am pretty sure that oej only meant to
	  change the variable name in the source, not the configuration
	  option name so let's turn it back to srvlookup instead of
	  global_srvlookup. (issue #8442 reported by jtodd)

	* /, apps/app_voicemail.c: Merged revisions 48115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48115 | file | 2006-11-29 16:05:17 -0500 (Wed, 29 Nov 2006) | 2
	  lines Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420
	  reported by slimey) ........

2006-11-29 20:57 +0000 [r48111-48114]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Clarify some settings for status
	  reports in subscriptions, queues and manager

	* /, configs/sip.conf.sample: Explain RTP timeouts

	* main/rtp.c: Change logging for p2p rtp bridge mode

2006-11-29 17:59 +0000 [r48109-48110]  Russell Bryant <russell@digium.com>

	* include/asterisk/threadstorage.h: - Fix a few spelling mistakes.
	  - Add some more documentation for the
	  ast_dynamic_str_............() function to document the behavior
	  of the function in the case of a partial write. Also, document
	  the return value and note that the function should never need to
	  be called directly.

	* main/utils.c: Go ahead and make this write unconditional. Making
	  it conditional is more work in both the append and non-append
	  modes. Also, always truncating the partial write makes the
	  behavior of the function more consistent, where in any type of
	  write, no partial result is left in the buffer. Thanks for the
	  feedback, luigi

2006-11-29 16:53 +0000 [r48108]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 48107 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48107 | file | 2006-11-29 11:50:33 -0500 (Wed,
	  29 Nov 2006) | 10 lines Merged revisions 48106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
	  lines If the frame was duplicated before writing out then we need
	  to free it. (issue #8429 reported by edguy3) ........
	  ................

2006-11-29 05:08 +0000 [r48103]  Russell Bryant <russell@digium.com>

	* main/utils.c: Remove an XXX command suggesting that this
	  truncation should not be conditional, and also add a more verbose
	  comment explaining why it is only needed in the case of appending
	  to the string for any curious readers that come along in the
	  future.

2006-11-29 04:28 +0000 [r48100-48102]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48101 | file | 2006-11-28 23:26:53 -0500 (Tue, 28 Nov 2006) | 2
	  lines Don't crash if the mailstream was not created. ........

	* sounds/Makefile: Use the proper version of extra sounds. (issue
	  #8441 reported by jtodd)

2006-11-28 23:13 +0000 [r48098-48099]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Add a comment to note near some code that
	  performs a very expensive operation that occurs for every
	  incoming media frame.

	* codecs/codec_zap.c: resolve a couple of compiler warnings

2006-11-28 18:28 +0000 [r48096]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 48095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48095 | qwell | 2006-11-28 12:26:53 -0600 (Tue, 28 Nov 2006) | 2
	  lines Export several more variables in top level Makefile.
	  Inspired by issue 8438. ........

2006-11-28 17:08 +0000 [r48090]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: don't use outputstr in the struct mansession,
	  it's just an extra allocation on a path where we have way too
	  many already. Unfortunately the AMI-over-HTTP requires multiple
	  copies, because we need to generate a header, then the raw output
	  to an intermediate buffer, then convert it to html/xml, and
	  finally copy everything into a malloc'ed buffer because that's
	  what the generic_http_callback interface expects.

2006-11-28 16:59 +0000 [r48089]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 48088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48088 | file | 2006-11-28 11:57:16 -0500 (Tue,
	  28 Nov 2006) | 10 lines Merged revisions 48087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2
	  lines According to the research I have done we never needed to
	  include compiler.h in the first place so let's not! (issue #8430
	  reported by edguy3) ........ ................

2006-11-28 15:53 +0000 [r48062-48086]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: initialize the dynamic string in a sane way.

	* main/utils.c: some simplifications to
	  ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name()
	  I am unsure whether the truncation of the string in case of a
	  failed attempt should be done unconditionally. See the XXX mark.
	  Russel, ideas ?

	* main/manager.c: do not return 500 Internal error if the AMI
	  command provides no output.

	* main/manager.c: mosty comment and documentation cleanup on
	  waitevent.

	* main/manager.c: Move the code to purge stale sessions to a
	  function, to simplify the body of the main loop of the accepting
	  thread. Rename purge_unused() to purge_events() so one knows what
	  the function does.

	* main/manager.c: Various simplifications of the code: + use a
	  wrapper around ast_carefulwrite(), used in two places, to make
	  life easier when we decide to use a different interface to the
	  socket. + put an ast_verbose() message on astman_append on a case
	  that should never happen now that we use a temporary file for
	  AMI-over-HTTP sessions + document and slightly simplify
	  process_events() by removing unnecessary parentheses. + in
	  get_input(), use ast_wait_for_input() instead of poll(). We may
	  want to move to a completely non-blocking

	* main/manager.c: More informative message on invalid commands.

	* main/manager.c: another normalization of AMI vs HTTP
	  identification. Should really define a macro IS_AMI(s) so it is
	  clear what we want to do.

	* main/manager.c: always use managerid to determine whether this is
	  an AMI or HTTP session, and document it.

	* main/http.c: In the previous commit i forgot to set the
	  poll_timeout to -1, causing the http threads to do busy waiting
	  around the socket... Fix the mistake, sorry for the
	  inconvenience!

	* main/http.c: document the support for running a server on TCP/TLS
	  and opening an SSL socket. We are almost ready to make this code
	  available to other modules.

	* main/http.c, configs/http.conf.sample: add a new http.conf
	  option, sslbindaddr. Because https is more secure than http, it
	  usually makes sense to keep this service more open than the one
	  on the unencrypted port.

	* main/http.c: in the helper thread, separate the FILE * creation
	  from the actual function doing work on the socket. This is
	  another generalization to provide a generic mechanism to open
	  TCP/TLS socket with a thread managing the accpet and children
	  threads managing the individual sessions.

	* main/http.c: staticize a global variable and remove an unused
	  field structure.

2006-11-27 18:10 +0000 [r48056]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 48054 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48054 | file | 2006-11-27 13:06:50 -0500 (Mon,
	  27 Nov 2006) | 10 lines Merged revisions 48053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
	  lines Use the proper function to get the new message count
	  instead of always using the filesystem. (issue #8421 reported by
	  slimey) ........ ................

2006-11-27 17:31 +0000 [r48050]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 48049 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48049 | tilghman | 2006-11-27 11:20:37 -0600
	  (Mon, 27 Nov 2006) | 10 lines Merged revisions 48045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
	  Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
	  ........ ................

2006-11-27 15:48 +0000 [r48039-48040]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_spool.c: More fixes for referencing a structure after it
	  has been freed. (issue #8425 reported by arkadia)

	* pbx/pbx_spool.c, /: Merged revisions 48038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r48038 | file | 2006-11-27 10:32:19 -0500 (Mon,
	  27 Nov 2006) | 10 lines Merged revisions 48037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
	  lines Do not reference the freed outgoing structure in the debug
	  message. (issue #8425 reported by arkadia) ........
	  ................

2006-11-27 14:47 +0000 [r48034]  Luigi Rizzo <rizzo@icir.org>

	* funcs/func_cdr.c: remove an extra comma in an initializer
	  Detected by: AST_DEVMODE=yes

2006-11-27 06:59 +0000 [r48032-48033]  Olle Johansson <oej@edvina.net>

	* include/asterisk/doxyref.h, include/asterisk/threadstorage.h:
	  Doxygen updates

	* /, channels/chan_sip.c: Change error message (imported from 1.4)

2006-11-26 06:55 +0000 [r48019]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, include/asterisk/threadstorage.h: - Add
	  some comments on thread storage with a brief explanation of what
	  it is as well as what the motivation is for using it. - Add a
	  comment by the declaration of ast_inet_ntoa() noting that this
	  function is not reentrant, and the result of a previous call to
	  the function is no longer valid after calling it again.

2006-11-26 00:31 +0000 [r48016-48018]  Steve Murphy <murf@digium.com>

	* /, funcs/func_cdr.c: Merged revisions 48017 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1
	  line might as well also document the raw values of the flag vars
	  ........

	* /, funcs/func_cdr.c: Merged revisions 48015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1
	  line A little bit of func_cdr documentation upgrade-- no bug#
	  involved, although 8221 may have inspired it. ........

2006-11-25 21:50 +0000 [r48009-48014]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Little fix so we use the right message

	* channels/chan_zap.c: Make compiler happier

	* channels/chan_zap.c: bug fix

	* channels/chan_zap.c: Make sure we don't send a group reset on a
	  group larger than 32 CICs

	* channels/chan_zap.c: Add ss7 show linkset command

	* channels/chan_zap.c: Updates to show linkset command

2006-11-25 17:37 +0000 [r48008]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: generalize a bit the functions used to create an tcp
	  socket and then run a service on it. The code in manager.c does
	  essentially the same things, so we will be able to reuse the code
	  in here (probably moving it to netsock.c or another appropriate
	  library file).

2006-11-25 09:48 +0000 [r48003-48004]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Changing ERROR to lesser level. Imported
	  from 1.2/1.4

	* main/rtp.c: - Adding comment on suspicious memory allocation.
	  Seems like it's never freed, but I don't have a clear
	  understanding of the frame allocation/deallocation, so I just
	  mark this for investigation. (Reported by Ed Guy). We're trying
	  to see if a free() hurts... - Doxygen comments on p2p rtp bridge
	  stuff. I am a bit worried about shortcutting rtcp this way, but
	  will need feedback from rtcp gurus. This should work for video
	  calls too, and possibly UDPTL.

2006-11-25 09:02 +0000 [r48001]  Luigi Rizzo <rizzo@icir.org>

	* main/channel.c: set pointers to NULL after freeing memory to
	  avoid multiple free() probably 1.4/1.2 issue as well if someone
	  can look into that.

2006-11-24 18:17 +0000 [r47995-47997]  Steve Murphy <murf@digium.com>

	* /: removed the svnmerge-integrated property from trunk; it's
	  confusing svnmerge in newly created branches

	* /, main/translate.c: This fix inspired by a patch supplied in bug
	  8189, which points out problems with the PLC code

2006-11-24 14:00 +0000 [r47986]  Olle Johansson <oej@edvina.net>

	* include/asterisk/doxyref.h, main/pbx.c,
	  include/asterisk/causes.h, include/asterisk/channel.h: Doxygen
	  update - Document cause codes - Document a bit more on channel
	  variables - global, predefined and local - Fix some doxygen in
	  channel.h. Adding one comment for two definitions does not work.
	  They won't be copied to each.

2006-11-23 11:04 +0000 [r47957-47960]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Remove unused memory allocation

	* doc/asterisk-conf.txt: Document new configuration option.

2006-11-22 21:49 +0000 [r47933-47945]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 47944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2
	  lines Video will never reach Packet2Packet bridging and can do
	  more harm then good. ........

	* CHANGES: Clarify a bit more.

	* CHANGES: Need to update the CHANGES file as well for the maxfiles
	  option.

	* main/asterisk.c: Add support to set the maximum number of files
	  open when Asterisk loads using the 'maxfiles' configuration
	  option. (issue #7499 reported by rkarlsba)

2006-11-22 11:28 +0000 [r47923]  Olle Johansson <oej@edvina.net>

	* channels/chan_h323.c: Don't over-deprecate... :-)

2006-11-22 05:49 +0000 [r47912]  Mark Spencer <markster@digium.com>

	* main/manager.c: Restore some sense of security to manager

2006-11-21 17:34 +0000 [r47898]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 47897 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2
	  lines If we have the non standard G726-32 setting turned on we
	  want to return G726-32 to the SDP, not our AAL2 string. (issue
	  #8330 reported by voipgate) ........

2006-11-21 15:25 +0000 [r47893]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Treat 101 as 100, not 183 session
	  progress

2006-11-21 11:53 +0000 [r47880-47881]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dial.c: better fix for the previous bug. In general this
	  code needs a deep revision, because the body of do_forward()
	  deletes/overwrites the output channel without freeing the resouce
	  in some cases, and without notifying the caller. Also, on FreeBSD
	  with MALLOC_OPTIONS set i am seeing various panics (duplicate
	  freee etc.)

	* apps/app_dial.c: do not ast_hangup() on a NULL channel. In the
	  original code this would happen in the case of o->forwards >=
	  AST_MAX_FORWARDS Likely an 1.2/1.4 isse as well - please someone
	  have a look, while I am hunting a few more similar panics now.

2006-11-20 20:04 +0000 [r47866]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 47864-47865 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47864 | tilghman | 2006-11-20 14:01:58 -0600 (Mon, 20 Nov 2006)
	  | 2 lines Oops, merge missed release of odbc object ........
	  ........

2006-11-20 19:52 +0000 [r47851-47861]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 47860 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47860 | file | 2006-11-20 14:51:36 -0500 (Mon,
	  20 Nov 2006) | 10 lines Merged revisions 47859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
	  lines Don't forget to byte swap if we are exiting the smoother
	  feed early. (issue #8287 reported by arturs) ........
	  ................

	* main/rtp.c: Use RTP/RTCP fds on the RTP structure, don't bother
	  storing them.

	* /, main/rtp.c: Merged revisions 47852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2
	  lines Only remove/destroy the RTCP I/O item if it exists.
	  ........

	* apps/app_dial.c, /, apps/app_directed_pickup.c,
	  include/asterisk/channel.h, .cleancount: Merged revisions 47850
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2
	  lines Use a separate variable in the channel structure to store
	  the context that the channel was dialed from. (issue #8382
	  reported by jiddings) ........

2006-11-20 14:08 +0000 [r47847]  Steve Murphy <murf@digium.com>

	* /: Erased the svnmerge-integrated prop from trunk. Please, in
	  your svnmerge-ings, don't let these props leak into the trunk or
	  branches.

2006-11-20 11:46 +0000 [r47844-47846]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Update docs for videosupport

	* /, channels/chan_sip.c: Properly reset schedule items (rizzo)

2006-11-19 04:22 +0000 [r47835-47836]  Steve Murphy <murf@digium.com>

	* UPGRADE.txt: Added a few words to explain the change to AEL
	  concerning Gosub()

	* doc/ael.txt: Added a few words of explanation about macros

2006-11-18 22:14 +0000 [r47822-47834]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: comments-only change: document a bit more when
	  manager events are delivered to the clients.

	* main/cdr.c, res/res_features.c, res/res_realtime.c:
	  ESS-ification. no need to bring this in 1.4, it is just code
	  cleanup

	* include/asterisk/cli.h, main/cli.c: Move this macro from cli.c to
	  cli.h so apps can use it without duplicating the macro or the
	  code: /*! * In many cases we need to print singular or plural *
	  words depending on a count. This macro helps us e.g. * printf("we
	  have %d object%s", n, ESS(n)); */ #define ESS(x) ((x) == 1 ? "" :
	  "s")

	* /, channels/chan_sip.c: Merged revisions 47823 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5
	  lines fix bug 7450 - Parsing fails if From header contains angle
	  brackets (the bug was only in a corner case where the < was right
	  after the opening quote, and the fix is trivial). ........

	* channels/chan_oss.c: prevent the sound thread from consuming all
	  the available CPU doing busy-wait on the output audio device. As
	  it is set now, it tries to push a frame every 10ms, which is
	  still too frequent but avoids deep restructuring of the code
	  (which i should do, though). Note, this is only for ring tones,
	  regular audio coming from the network is still delivered as soon
	  as it is available. Eventually this could well end up in the 1.4
	  branch, but since i am probably the only user of chan_oss there
	  isn't much urgency to do that.

2006-11-17 23:18 +0000 [r47821]  Steve Murphy <murf@digium.com>

	* include/asterisk/file.h, main/channel.c, res/res_features.c,
	  main/file.c, main/app.c, apps/app_directory.c,
	  apps/app_followme.c, apps/app_voicemail.c: This update fulfils
	  the request of bug 7109, which claimed the language arg to
	  ast_stream_and_wait() was redundant. Almost all calls just used
	  chan->language, and seeing how chan is the first argument, this
	  certainly seems redundant. A change of language could just as
	  easily be done by simply changing the channel language before
	  calling.

2006-11-17 22:56 +0000 [r47815-47818]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: remove a debugging message

	* main/cli.c: convert "help" to new style, fix completion of
	  arguments past the first one that i broke earlier today.

	* main/cli.c: standardize "module show [like]"

2006-11-17 21:51 +0000 [r47814]  Jason Parker <jparker@digium.com>

	* configs/voicemail.conf.sample, apps/app_voicemail.c: Add ability
	  to notify an external application/script that the voicemail
	  password was, while also still changing the password
	  "internally". Issue 7371, initial patch by pdunkel, with
	  rewrite/config comments by me. Additional modifications (yay
	  bitmask) by pdunkel.

2006-11-17 21:50 +0000 [r47813]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: describe a bit the patterns that you can have in the
	  commands, and add support for wildcard (spelled as '%'). On
	  passing fix a bug in the expansion code which was hidden and
	  appeared when implementing the wildcard The fix is just the line
	  'src != argindex', in case someone wants to test this on 1.4 -
	  but i would just keep this in trunk.

2006-11-17 20:46 +0000 [r47806]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Add ability to add custom queue log via manager
	  interface. Issue 7806, patch by alexrch, with slight
	  modifications by me.

2006-11-17 18:26 +0000 [r47801]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add some sense of link state. Don't make
	  calls if link is down. Only reset if we're bringing the link up
	  for the first time.

2006-11-17 12:26 +0000 [r47787-47790]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: merge the implemenmtation of "core set debug" and
	  "core set verbose". No externally visible changes.

	* channels/chan_oss.c: remove an unused function

	* channels/chan_oss.c: use the regexp cli support on some of the
	  command

	* include/asterisk/cli.h, main/cli.c: introduce a bit of regexp
	  support in the internal CLI api. Now you can specify a cli
	  command as "console autoanswer [on|off]" which means the on|off
	  argument is optional, or "console {mute|unmute}" which means the
	  mute|unmute argument is mandatory. The blocks in [] or {} do not
	  necessarily need to be at the end of the string. Completions for
	  the variant parts are generated automatically. This should
	  significantly simplify the implementation of the various
	  handlers.

2006-11-17 01:05 +0000 [r47784]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we choose the last channel as the
	  dchannel if it's not E1 (for BRI). (#8077) Thanks Tzafrir.

2006-11-16 23:20 +0000 [r47783]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, /, apps/app_db.c: Merged revisions 47782 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2
	  lines Fix a couple of typos. Initially pointed out by mrobinson.
	  ........

2006-11-16 23:05 +0000 [r47779]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: convert two entries to new style

2006-11-16 23:00 +0000 [r47778]  Kevin P. Fleming <kpfleming@digium.com>

	* /, doc/billing.txt: Merged revisions 47777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47777 | kpfleming | 2006-11-16 17:00:10 -0600
	  (Thu, 16 Nov 2006) | 12 lines update documentation regarding IAX2
	  transfers and CDRs Merged revisions 47776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
	  | 2 lines update clearly wrong documentation regarding cdr_custom
	  ........ ................

2006-11-16 22:51 +0000 [r47775]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Remove the interim variable for range
	  modifications, and set it on the structure directly. Also move
	  the default checking to where it gets set initially. Fixes
	  suggested by file.

2006-11-16 22:44 +0000 [r47772]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: convert some handlers to new style.

2006-11-16 22:32 +0000 [r47771]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: Add ability to
	  modify range for dring matching. Issue #8369, patch by ssuehring,
	  modified slightly by me.

2006-11-16 22:03 +0000 [r47769-47770]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: fix indentation

	* main/cli.c: remove an unused function

2006-11-16 21:13 +0000 [r47763-47765]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 47764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2
	  lines Compare technology using the pointers instead of a straight
	  comparison based on name. (issue #8228 reported by dean bath)
	  ........

2006-11-16 20:10 +0000 [r47759]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, configure.ac: Merged revisions 47758 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r47758 | kpfleming | 2006-11-16 14:09:10 -0600 (Thu, 16
	  Nov 2006) | 2 lines check for pre-1.4 versions of Zaptel and
	  abort the configure script if found with an appropriate error
	  message ........

2006-11-16 19:29 +0000 [r47756]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Make it possible
	  to enable/disable onhold tracking, in order to make life easier
	  for realtime users.

2006-11-16 18:32 +0000 [r47747-47752]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47751 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47751 | file | 2006-11-16 13:29:12 -0500 (Thu,
	  16 Nov 2006) | 10 lines Merged revisions 47750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2
	  lines Because of the way chan_local is written we should be extra
	  careful and make sure our callback functions have a tech_pvt.
	  (issue #8275 reported by mflorell) ........ ................

	* /, apps/app_meetme.c: Merged revisions 47748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47748 | file | 2006-11-16 12:52:48 -0500 (Thu, 16 Nov 2006) | 2
	  lines Don't unreference the SLA object if there is no SLA object
	  in the devicestate callback. (issue #8354 reported by loloski)
	  ........

	* /: Be gone 1.2 props!

2006-11-16 17:15 +0000 [r47734-47746]  Olle Johansson <oej@edvina.net>

	* /: Merging a fix that was already fixed.

	* channels/chan_sip.c: Merging implementation of invite states from
	  my "invitestate" branch for 1.2. This is a bit more clean
	  platform for the handling of BYE/CANCEL than what we had. It
	  might also need to changes in other parts of the code, since we
	  know the state of the INVITE transaction. Please observe that
	  this is is not dialog states at all, this is INVITE transaction
	  states. Hello Michael Proctor, and thank you! :-)

	* /: Block upgrade to UPGRADE

	* /, channels/chan_sip.c, configs/sip.conf.sample: - CANCEL never
	  uses authentication - Add docs on canreinvite

2006-11-16 14:58 +0000 [r47727-47732]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: reduce indentation on a large function.

	* main/cli.c: use atomic instructions to update the inuse counters
	  for CLI entriesC. The lock is not protecting this field. I wonder
	  if the field should be declared 'volatile' as well.

	* main/cli.c: make kevin (and 64 bit machines) happy and remove a
	  cast from char* to int in handling the return values from
	  new-style handlers. On passing, note that
	  main/loader.c::ast_load_resource() always return 0 so errors are
	  not propagated up. I am not sure this is the intended behaviour.

2006-11-16 08:18 +0000 [r47718]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/channel.c, /, funcs/func_channel.c,
	  include/asterisk/channel.h: Merged revisions 44809 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10
	  Окт 2006) | 1 line CHANNEL() function sometime mix parameter and
	  value ........

2006-11-15 22:32 +0000 [r47713]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47712 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47712 | file | 2006-11-15 17:31:17 -0500 (Wed,
	  15 Nov 2006) | 10 lines Merged revisions 47711 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2
	  lines Make sure that the pvt structure exists before trying to do
	  fixup on Local channels. (issue #7937 reported by mada123, fix by
	  alamantia with mods by me) ........ ................

2006-11-15 21:57 +0000 [r47710]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 47709 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47709 | tilghman | 2006-11-15 15:56:55 -0600 (Wed, 15 Nov 2006)
	  | 2 lines Fix ODBC_STORAGE for when context is NULL ........

2006-11-15 21:36 +0000 [r47708]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 47707 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2
	  lines We need to ensure timelimit stuff is included as well so
	  warnings get played. (issue #8050 reported by KNK) ........

2006-11-15 21:21 +0000 [r47706]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Hunting the initreq change for an ACK

2006-11-15 20:59 +0000 [r47703-47704]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: 1. Fix the bug that Asterisk hangs up the
	  calls if the OSP AuthRsp messages without destination protocol
	  infomation. 2. Fix the bug that Asterisk generats wrong dial
	  string (no in
	  IAX2/[username[:password]@]peer[:port][/exten[@context]][/options]
	  format) for IAX. 3. Add support for oh323 channel driver. 4.
	  Re-formate the code.

	* include/asterisk/astosp.h: 1. Re-format the code.

2006-11-15 20:51 +0000 [r47702]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/file.c: Merged revisions 47701 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47701 | kpfleming | 2006-11-15 14:50:06 -0600 (Wed, 15 Nov 2006)
	  | 2 lines don't try to call fclose() if fopen() failed ........

2006-11-15 20:40 +0000 [r47700]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: - Don't reply to ACK - Improve SIP
	  history for debugging (Imported from 1.4)

2006-11-15 20:28 +0000 [r47685-47694]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 47693 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47693 | kpfleming | 2006-11-15 14:27:38 -0600
	  (Wed, 15 Nov 2006) | 12 lines Merged revisions 47677 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15
	  Nov 2006) | 4 lines ensure that message duration is included in
	  email notifications for forwarded messages (BE-96, fix by me
	  after corydon used his clue-bat on me) ensure that duration in
	  the message metadata is updated if prepending is done during
	  forwarding (related to BE-96) remove prototype for API call that
	  does not exist ........ ................

	* /, main/config.c: Merged revisions 47690 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47690 | kpfleming | 2006-11-15 14:01:22 -0600
	  (Wed, 15 Nov 2006) | 20 lines Merged revisions 47686,47688-47689
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006)
	  | 2 lines clear the category's variable tail pointer as well when
	  variables are detached from it ........ r47688 | kpfleming |
	  2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when
	  appending a list of variable to a category, ensure the tail
	  pointer points to the last variable in the list ........ r47689 |
	  kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2
	  lines when re-writing the config file, don't repeat the path if
	  it hasn't changed ........ ................

	* /, main/config.c: Merged revisions 47684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47684 | kpfleming | 2006-11-15 12:43:30 -0600
	  (Wed, 15 Nov 2006) | 10 lines Merged revisions 47682 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15
	  Nov 2006) | 2 lines ouch... don't use printf, use
	  ast_log/ast_verbose ........ ................

2006-11-15 17:40 +0000 [r47662-47669]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: fix indentation

	* main/cli.c: small simplifications and localization of variables.

	* main/cli.c: new-style "core show channels"

	* main/cli.c: more changes to new style of "module load" and
	  "load". Under FreeBSD, the filename_completion used in the above
	  commands does not work. Not sure why, but on passing i note that
	  the function is part of readline and is not reentrant, so it
	  needs to be fixed one way or another.

	* main/cli.c: move another deprecated command to the new style

	* main/cli.c: move "core set debug" to the new style, and remove
	  duplicated code for the deprecated handler. On passing fix a long
	  standing bug in find_cli() which would not find the longest match
	  - this part (trivial, basically just update matchlen on a match)
	  must go in 1.4 and possibly 1.2 as well.

2006-11-15 16:09 +0000 [r47657-47661]  Olle Johansson <oej@edvina.net>

	* /: Messed up earlier, cleaning up...

	* /, channels/chan_sip.c: Send proper SIP error message improperly
	  when we can't allocate dialog (out of file handles is one cause)

	* channels/chan_sip.c: Update doxygen docs to reflect the code...

2006-11-15 15:02 +0000 [r47652-47654]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/cli.h, main/cli.c: one more step cleaning the
	  internal CLI interface: the NEW_CLI macro now supports extra
	  arguments (to deprecate other commands). use this to implement
	  unload and reload, and remove some unused functions. usual
	  completion fixes (as these function accept multiple arguments).
	  The summary is still a bit inconsistent.

	* include/asterisk/cli.h, main/cli.c: update the internal cli api
	  following comments from kevin. This change basically simplifies
	  the interface of the new-style handler removing almost all the
	  tricks used in the previous implementation to achieve backward
	  compatibility (which is still present and guaranteed.)

2006-11-15 04:47 +0000 [r47646]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 47645 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2
	  lines If NAT detection is turned on or already detected then say
	  NAT is active when setting the remote RTP peer when doing early
	  bridging. (issue #8365 reported by marcelbarbulescu) ........

2006-11-15 00:19 +0000 [r47642]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/term.c: Merged revisions 47641 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47641 | kpfleming | 2006-11-14 18:19:05 -0600 (Tue, 14 Nov 2006)
	  | 2 lines more formatting cleanup, and avoid running off the end
	  of the string ........

2006-11-15 00:15 +0000 [r47640]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 47639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2
	  lines Turn notice about unknown RTCP packet type into a debug
	  message instead. ........

2006-11-15 00:06 +0000 [r47636]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/misdn/isdn_lib.c: Merged revisions 47635 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14
	  Nov 2006) | 2 lines silence compiler warning on 64-bit platforms
	  (this variable is an 'int' anyway, comparing it to 'signed long'
	  is not useful) ........

2006-11-14 22:19 +0000 [r47633]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 47632 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47632 | file | 2006-11-14 17:17:16 -0500 (Tue,
	  14 Nov 2006) | 10 lines Merged revisions 47631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
	  lines Update copyright information in the ADSI logo blob.
	  ........ ................

2006-11-14 22:08 +0000 [r47630]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: add missing casts and remove an unused function.

2006-11-14 22:07 +0000 [r47623-47629]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 47628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2
	  lines Only keep the video RTP structure around if 1. Video
	  support is enabled and 2. A video codec is enabled on the dialog
	  ........

	* /, funcs/func_uri.c: Merged revisions 47625 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47625 | file | 2006-11-14 16:30:44 -0500 (Tue, 14 Nov 2006) | 2
	  lines Small documentation clarification for URIENCODE. (issue
	  #8294 reported by salaud) ........

	* apps/app_dial.c: Make local copy of arguments to parse. (issue
	  #8362 reported by homesick)

2006-11-14 18:58 +0000 [r47622]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 47621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47621 | tilghman | 2006-11-14 12:54:40 -0600 (Tue, 14 Nov 2006)
	  | 3 lines Conversion of res_odbc API to include ast_ prefix did
	  not completely transition app_voicemail when ODBC_STORAGE is used
	  (reported on IRC by caio1982, not in bugtracker) ........

2006-11-14 17:00 +0000 [r47619-47620]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: fix completion for "module reload mod_1 mod_2 ... "
	  (should do the same for the other similar commands, "reload",
	  "module unload" and so on.

	* main/cli.c: partly convert to new style set-verbose, with
	  completion fixes

2006-11-14 16:48 +0000 [r47618]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_amd.c: Merged revisions 47617 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2
	  lines Use LOG_DEBUG to print out the indication that app_amd is
	  using default settings instead of using LOG_NOTICE. This stops
	  needless logging of this information under normal circumstances.
	  (issue #8361 reported by Seb7) ........

2006-11-14 16:38 +0000 [r47614-47616]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: replace two deprecated functions with calls to the
	  standard ones, with fixes to argc/argv

	* main/cli.c: remove duplicated implementation for a deprecated
	  function, use the original one instead with appropriate changes
	  in argc/argv. This is not always applicable, but in some simple
	  cases it is.

2006-11-14 16:15 +0000 [r47610-47611]  Olle Johansson <oej@edvina.net>

	* include/asterisk/cli.h: need to check quoting in the doxygen
	  docs...

	* include/asterisk/cli.h: Some improvements to doxygen docs

2006-11-14 16:09 +0000 [r47606-47609]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: new-style for 'core show uptime', include 'complete'
	  support and better error checking

	* main/cli.c: apply previous fix to old-style cli entries as well.

	* main/cli.c: fix "core set debug atleast ", and fix the simple
	  case where a command can have multiple completions, the first
	  ones coming from keywords in previous CLI entries. There is no
	  solution yet for the complex case of N1 completions from a CLI
	  entry, followed by N2 from the next one, and so on, because the
	  _complete() handlers do not tell us how many matches it
	  generates, so we don't know how many to skip when interrogating
	  the other handlers. The only solution is to avoid, as much as
	  possible, multiple CLI entries with the same prefix.

	* include/asterisk/cli.h, main/cli.c: Bring in the improved
	  internal API for the CLI. WATCH OUT: this changes the binary
	  interface (ABI) for modules, so e.g. users of g729 codecs need a
	  rebuilt module (but read below). The new way to write CLI
	  handlers is described in detail in cli.h, and there are a few
	  converted handlers in cli.c, look for NEW_CLI. After converting a
	  couple of commands i am convinced that it is reasonably
	  convenient to use, and it makes it easier to fix the pending CLI
	  issues. On passing, note a bug with the current 'complete'
	  architecture: if a command is a prefix of multiple CLI entries,
	  we miss some of the possible options. As an example, "core set
	  debug" can continue with "channel" from one CLI entry, and "off"
	  or "atleast" from another one. We address this problem in a
	  separate commit (when i have figured out a fix, that is). ABI
	  issues: I asked Kevin if it was ok to make this change and he
	  said yes. While it would have been possible to make the change
	  without breaking the module ABI, the code would have been more
	  convoluted. I am happy to restore the old ABI (while still being
	  able to use the "new style" handlers) if there is demand.

2006-11-14 13:17 +0000 [r47595-47600]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Adding some debug output to trace bug in
	  realtime

	* channels/chan_sip.c: Adding a new debug line for issue #7351 -
	  trying to find where an ACK can overwrite the initreq...

	* /, channels/chan_sip.c: Issue #8272 imported from 1.2/1.4 - Let
	  the peerpoke system destroy it's own packets, please.

	* channels/chan_sip.c: Remove flags not used any more (thanks
	  Luigi)

2006-11-13 22:40 +0000 [r47586-47587]  Matt O'Gorman <mogorman@digium.com>

	* codecs/codec_zap.c: oops no parens

	* main/frame.c, codecs/codec_zap.c: fix bytesize to 5.3kb for g723
	  codec and add support for multimode of tc400p

2006-11-13 21:32 +0000 [r47585]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 47584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47584 | file | 2006-11-13 16:28:57 -0500 (Mon,
	  13 Nov 2006) | 10 lines Merged revisions 47583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
	  lines Initialize global pointers for connection and result to
	  NULL. (issue #8356 reported by james) ........ ................

2006-11-13 20:21 +0000 [r47582]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 47581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47581 | tilghman | 2006-11-13 14:20:01 -0600
	  (Mon, 13 Nov 2006) | 10 lines Merged revisions 47580 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13
	  Nov 2006) | 2 lines Having more than 255 old messages caused
	  corruption in the new/old count ........ ................

2006-11-13 19:20 +0000 [r47579]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Small fix for uncommon scenario.

2006-11-13 19:19 +0000 [r47577-47578]  Steve Murphy <murf@digium.com>

	* /: Blocking 47576 from merging into trunk. Already done in 47577

	* main/config.c: This solves bug 8342, whereby a crash occurs under
	  certain circumstances while reading a config file with comments--
	  a call to CB_ADD shouldn't happen if withcomments is zero

2006-11-13 19:14 +0000 [r47575]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Make chan_h323 build again and make the CLI
	  commands work. (reported on asterisk-dev mailing list by Di-Shi
	  Sun)

2006-11-13 18:24 +0000 [r47568]  Steve Murphy <murf@digium.com>

	* /: blocked 47564 from 1.4 to be merged onto trunk; 47566 already
	  did it

2006-11-13 18:23 +0000 [r47567]  Joshua Colp <jcolp@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add
	  'loose' option to joinempty and leavewhenempty which is almost
	  exactly like 'strict' except it does not count paused queue
	  members as unavailable. (issue #8263 reported by gnarf)

2006-11-13 18:20 +0000 [r47566]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
	  the messed if, but we all forgot to update the regressions. Until
	  now.

2006-11-13 17:55 +0000 [r47556-47560]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Don't play the "entering conference number
	  <insert number here>" prompts if the 'q' option is used. If
	  others believe this should be in 1.2/1.4 then we can put it in,
	  but I'm uncomfortable doing so right now as it is a change of
	  behavior. (issue #8138 reported by tmancill)

	* pbx/pbx_ael.c: Clean up last commit to better conform to
	  standards.

2006-11-13 17:36 +0000 [r47554-47555]  Steve Murphy <murf@digium.com>

	* /: Blocking 47553 from 1.4 to trunk... 47554 is done for it.

	* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
	  found... just confuses users

2006-11-13 17:10 +0000 [r47543-47552]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_sms.c: Merged revisions 47551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47551 | file | 2006-11-13 12:08:07 -0500 (Mon,
	  13 Nov 2006) | 10 lines Merged revisions 47549 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
	  lines When sending an SMS with a user data header properly set
	  the UDH flag in the first byte. (issue #8347 reported by
	  hoffmeis) ........ ................

	* main/cli.c: Return module show to a working state. (issue #8353
	  reported by jserve)

2006-11-13 16:08 +0000 [r47541]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Only produce error message once, don't
	  fill the screen with them... (Testing SIPP thanks to JerJer and
	  Greg)

2006-11-13 14:29 +0000 [r47536-47539]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: merge from astobj2-r47450: use UNLINK to
	  remove a packet from its queue, and related code rearrangement.
	  Approved by: oej This could be made better if we declared struct
	  sip_pvt *dialpg = pkt->owner; at the beginning of the function,
	  and use it throughout the function. I'll let the boss decide :)

	* channels/chan_sip.c: merge from codename-pineapple and astobj2
	  47499: simplify __sip_ack() removing a strcmp for looking up
	  packets. no functional change, only performance, so don't need to
	  merging to earlier branches now. Approved By: oej

	* main/cli.c: stop looking for new entries when we know we are
	  done. there is no functional change, so it is not necessary to
	  bother merging this to 1.4 now.

	* main/cli.c: free memory when unregistering an entry. needs to be
	  merged to 1.4

2006-11-13 05:58 +0000 [r47530]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample: Feature: allow the
	  sanity SQL to be customized per connection class (Issue 6453)

2006-11-13 05:51 +0000 [r47529]  Russell Bryant <russell@digium.com>

	* /, configure, acinclude.m4: Merged revisions 47527 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r47527 | russell | 2006-11-13 00:48:18 -0500 (Mon, 13
	  Nov 2006) | 5 lines AC_PROG_SED is included in autoconf 2.60, but
	  apparently it is not included in 2.59. So, to maintain
	  compatability with 2.59 since it is a small change, copy this
	  macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
	  #8345) ........

2006-11-13 05:48 +0000 [r47524-47528]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: Merged revisions 47526 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47526 | tilghman | 2006-11-12 23:46:18 -0600
	  (Sun, 12 Nov 2006) | 10 lines Merged revisions 47525 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12
	  Nov 2006) | 2 lines If the execute fails a second time, make sure
	  that we don't pass back a stale handle ........ ................

	* channels/chan_zap.c, /: Merged revisions 47523 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47523 | tilghman | 2006-11-12 19:12:01 -0600
	  (Sun, 12 Nov 2006) | 10 lines Merged revisions 47522 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12
	  Nov 2006) | 2 lines Don't play dialtone if the seizing the
	  channel fails (Bug 7754) ........ ................

2006-11-12 20:47 +0000 [r47521]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Part of patch in #7403 to improve tag
	  checking in pedantic mode (stephen_dredge)

2006-11-12 19:22 +0000 [r47520]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: The use of an ifdef to check for FreeBSD is
	  useless here because the two versions of the format string are
	  identical. Also, since each format is only used once, get rid of
	  the use of defines all together. (issue #8344, julieng) In
	  passing, also clean up the formatting a but to get rid of the
	  nesting without the use of braces, as defined in the coding
	  guidelines.

2006-11-12 16:15 +0000 [r47508-47514]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Restore auto-framing (DEA). Imported from
	  1.4

	* /, channels/chan_sip.c: - Support UDPTL as well as udptl in T38
	  sdp.

	* /, channels/chan_sip.c: - Don't hangup because of failed
	  re-invite. Go back to previous state. - Keep RTP running during
	  T.38 session We might improve the code to issue ast_rtp_stop if
	  T.38 re-invite not fails later on in the code, but I don't see
	  many reasons to.

	* /, channels/chan_sip.c: - Add some comments to t.38 code - Remove
	  improper blocking of ptime: in SDP

2006-11-12 06:31 +0000 [r47493-47498]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 47497 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47497 | russell | 2006-11-12 01:23:23 -0500
	  (Sun, 12 Nov 2006) | 12 lines Merged revisions 47496 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12
	  Nov 2006) | 4 lines Only do the check to determine whether the
	  channel calling this function is an IAX2 channel when getting the
	  IP address using the special argument, CURRENTCHANNEL. (issue
	  #8341, jcovert) ........ ................

	* Makefile, /: Merged revisions 47494 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47494 | russell | 2006-11-11 10:31:08 -0500 (Sat, 11 Nov 2006) |
	  6 lines Add the target "menuconfig" as an alias for the
	  "menuselect" target. This is just a favor to users so that if you
	  accidentally type "make menuconfig" instead of "make menuselect",
	  it still works. (inspired by a comment on IRC from wangster
	  calling me an "especially devious asterisk developer" for having
	  it be menuselect instead of menuconfig. :) ) ........

	* /, main/term.c: Merged revisions 47492 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47492 | russell | 2006-11-11 10:18:02 -0500 (Sat, 11 Nov 2006) |
	  2 lines Tweak the formatting of this new function to better
	  conform to coding guidelines. ........

2006-11-11 02:12 +0000 [r47491]  Matt O'Gorman <mogorman@digium.com>

	* main/logger.c, include/asterisk/term.h, main/term.c: safe
	  terminal output is sweet.

2006-11-10 22:06 +0000 [r47478]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we don't use 32bits for a value
	  that only requires 1 bit. Also, fix a compiler warning for one of
	  the SS7 functions.

2006-11-10 21:55 +0000 [r47467-47477]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Add some history and fix some debug
	  output for autodestruct.

	* /, channels/chan_sip.c: Proper fix for adding debug...

	* /, channels/chan_sip.c: Make sure we destroy dialog in case of
	  loop

	* /, channels/chan_sip.c: Cleanup imported from 1.4

2006-11-10 20:05 +0000 [r47459-47465]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: Fine, take this.

	* main/cli.c: A trunk that builds is a productive trunk.

	* pbx/pbx_dundi.c: Hello compiler working, goodbye compiler
	  warning. (fix compiler warning introduced from pbx_dundi
	  optimizations)

	* /, channels/chan_h323.c: Merged revisions 47457 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47457 | file | 2006-11-10 14:36:25 -0500 (Fri, 10 Nov 2006) | 2
	  lines Let's give this a go... ........

2006-11-10 19:35 +0000 [r47456]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add fix for 7321. Ability to hide
	  calleridname from zapata.conf

2006-11-10 19:01 +0000 [r47455]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue 8336- fix support for multipart SDP
	  (imported from 1.2/1.4). (Alphaque)

2006-11-10 17:22 +0000 [r47445]  Luigi Rizzo <rizzo@icir.org>

	* build_tools/prep_moduledeps: manual merge from 1.4: grep -m not
	  available on bsd, use head -1 which works for all

2006-11-10 17:01 +0000 [r47439]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c, channels/chan_iax2.c, channels/chan_mgcp.c,
	  main/cli.c: Merged revisions 47436 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006)
	  | 2 lines Discussion of these CLI changes resulted in more
	  consistency (Bug 8236) ........

2006-11-10 16:55 +0000 [r47438]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 47437 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47437 | file | 2006-11-10 11:53:16 -0500 (Fri, 10 Nov 2006) | 2
	  lines Only split up extension and context if a value exists.
	  (issue #8332 reported by loloski) ........

2006-11-10 16:38 +0000 [r47434-47435]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_queue.c: Merged revisions 47433 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47433 | kpfleming | 2006-11-10 10:36:49 -0600 (Fri, 10 Nov 2006)
	  | 2 lines if adding a queue member is LOG_NOTICE, then removing
	  them should be LOG_NOTICE, not LOG_DEBUG ........

	* /, apps/app_queue.c: Merged revisions 47432 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47432 | kpfleming | 2006-11-10 10:34:04 -0600 (Fri, 10 Nov 2006)
	  | 2 lines reflect addition/removal of dynamic queue members in
	  queue_log, so that people using dialplan replacement for
	  AgentCallbackLogin can still track login/logout (issue #7736,
	  reported/patched by whoiswes but this commit was written by me
	  and covers all three paths for AQM/RQM) ........

2006-11-10 13:14 +0000 [r47415-47419]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Ripping out bad support for 491 replies
	  to INVITE's. Let's try again properly later.

	* /, channels/chan_sip.c: Fix badly defined flag. Thanks fenlander!

	* channels/chan_sip.c: Small simplification and documentation
	  correction.

2006-11-10 04:30 +0000 [r47408-47410]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Various little bits of code cleanup to reduce
	  nesting, remove useless casts, and to remove a duplicated error
	  message after a memory allocation error

	* include/asterisk/app.h, apps/app_read.c, main/app.c: Add the
	  ability to specify multiple prompts to the Read() dialplan
	  application, similar to Background() and Playback(). (issue
	  #7897, jsmith, with some modifications)

2006-11-10 03:45 +0000 [r47399-47406]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_h323.c: Merged revisions 47405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47405 | file | 2006-11-09 22:44:36 -0500 (Thu, 09 Nov 2006) | 2
	  lines Fix building of chan_h323 by completeing some structure
	  definitions. (issue #8327 reported by Mithraen) ........

	* main/pbx.c: This should already be called while locked.

	* /, apps/app_voicemail.c: Merged revisions 47398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47398 | file | 2006-11-09 17:32:30 -0500 (Thu, 09 Nov 2006) | 2
	  lines Do conversion in a more easier to read and working way for
	  \r, \n, and \t. (issue #8324 reported by johnlange) ........

2006-11-09 21:32 +0000 [r47392]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /, build_tools/prep_moduledeps,
	  apps/app_voicemail.c: Merged revisions 47391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) |
	  7 lines Work around an issue that caused menuselect to display a
	  bogus description for app_voicemail and chan_zap. These modules
	  use some preprocessor directives to determine what it will report
	  to Asterisk as its description. However, the way we extract this
	  information from the source files for menuselect is not smart
	  enough to figure this out. (issue #8326, #8328) ........

2006-11-09 17:08 +0000 [r47382]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 47380 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47380 | file | 2006-11-09 11:53:25 -0500 (Thu,
	  09 Nov 2006) | 10 lines Merged revisions 47379 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2
	  lines Don't include compiler.h on kernels 2.6.18 and higher as,
	  well, it's apparently going to be removed. This should make all
	  you FC6 fans happy as your Asterisk will now build without any
	  mods. ........ ................

2006-11-09 16:30 +0000 [r47353-47378]  Russell Bryant <russell@digium.com>

	* /, main/cli.c: Merged revisions 47377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47377 | russell | 2006-11-09 11:28:15 -0500 (Thu, 09 Nov 2006) |
	  2 lines fix tab completion for "core debug channel" and "core no
	  debug channel" ........

	* /, main/cli.c: Merged revisions 47375 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47375 | russell | 2006-11-09 11:24:02 -0500 (Thu, 09 Nov 2006) |
	  3 lines Fix "core show channel". Also, fix tab completion for
	  both "core show channel" and "core show channels". ........

	* /, main/cli.c: Merged revisions 47372 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47372 | russell | 2006-11-09 11:18:33 -0500 (Thu, 09 Nov 2006) |
	  3 lines Fix "core debug channel <whatever>". I guess someone
	  needs to go through and audit every CLI command that changed
	  number of arguments ... ........

	* /, main/cli.c: Merged revisions 47366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47366 | russell | 2006-11-09 10:49:39 -0500 (Thu, 09 Nov 2006) |
	  3 lines Fix another CLI command, "core show uptime" ... (issue
	  #8323, reported by johnlange, fixed by myself) ........

	* /, main/asterisk.c: Merged revisions 47352 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47352 | russell | 2006-11-09 01:31:37 -0500 (Thu, 09 Nov 2006) |
	  3 lines fix "core show version" to reflect the new number of
	  arguments for this CLI command (issue #8316, kshumard) ........

2006-11-09 00:46 +0000 [r47343-47351]  Steve Murphy <murf@digium.com>

	* /: Blocking 47344 from automerging into trunk

	* /: Blocking 47348 from automerging into trunk

	* main/channel.c: This mod via bug 7531

	* channels/chan_skinny.c: committed in behalf of bug 8190

2006-11-08 22:35 +0000 [r47341]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Add Max-Forwards higher in the packet,
	  following recommendations - Fix documentation for
	  sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf
	  !!! - Change doc for a sip_pvt setting

2006-11-08 21:59 +0000 [r47337-47339]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: restore display of G.722 codec

	* /, channels/chan_sip.c: Merged revisions 47333 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006)
	  | 2 lines add simple fix for SDP to report proper sample rate for
	  G.722 media sessions ........

2006-11-08 18:26 +0000 [r47335]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, CHANGES: Display CID matching information when using
	  dialplan show. (issue #8279 reported by caio1982)

2006-11-08 17:06 +0000 [r47325-47332]  Russell Bryant <russell@digium.com>

	* /, utils/streamplayer.c: Merged revisions 47331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47331 | russell | 2006-11-08 12:03:09 -0500 (Wed, 08 Nov 2006) |
	  5 lines I occasionally get email from users that are trying to
	  figure out what this does, or due to some misunderstanding as to
	  what it is supposed to do, can't get it to work. So, I have added
	  some text here to hopefully explain what this application does
	  and does not do. ........

	* /, configure, configure.ac, acinclude.m4: Merged revisions 47327
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47327 | russell | 2006-11-08 11:31:59 -0500 (Wed, 08 Nov 2006) |
	  4 lines Copy the macros from libtool.m4 to our own acinclude.m4
	  such that libtool is no longer required to be installed to be
	  able to generate the configure script. ........

2006-11-08 15:28 +0000 [r47321]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: coding guidelines, coding guidelines, coding
	  guidelines

2006-11-08 13:59 +0000 [r47314-47318]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47271
	  avoid doing p > 0 when p is a pointer; move a lock closer to the
	  place where it is needed Approved By: oej

	* channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47246 Same
	  as for peers and users, replace ASTOBJ_UNREF(r,
	  sip_registry_destroy) with unref_registry(r); Approved By: oej

	* channels/chan_sip.c: merge from team/rizzo/astobj2, rev 47243,
	  47244, 47245: Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with
	  unref_peer(peer); This places the name of the destructor in one
	  place only (where it should be), eliminates the chance of errors
	  in case you specify the wrong destructor, and also lets the
	  compiler do type checking on the argument, again helping with
	  keeping the code clean. Same for users. remove two duplicate
	  definitions. Approved By: oej

	* channels/chan_sip.c: merge rev.47224 from team/rizzo/astobj2:
	  hide dialoglist lock/unlocking in wrapper functions. Approved By:
	  oej

	* channels/chan_sip.c: silence compiler about uninitialized
	  variables. The compiler is wrong, but it has the last word.

2006-11-08 08:01 +0000 [r47313]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)

2006-11-08 07:21 +0000 [r47306]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_jingle.c, channels/chan_gtalk.c: fix compilation.
	  Overall i think the previous change to ast_channel_alloc() to
	  close bug 7506 should have been done by defining an
	  ast_set_callerid_noevent() function that does the setting without
	  generating the event. Lot less code duplication, and easier to
	  handle.

2006-11-08 03:13 +0000 [r47304-47305]  Russell Bryant <russell@digium.com>

	* configure.ac: add a comment about where AC_PROG_LD comes from

	* aclocal.m4 (removed), /: remove aclocal.m4 from the tree since it
	  is just an intermediate file created while generating the
	  configure script.

2006-11-07 23:14 +0000 [r47295-47300]  Luigi Rizzo <rizzo@icir.org>

	* main/asterisk.c: fix "core show profile" parsing. Needs to go in
	  1.4 too, but ENOTIME now

	* apps/app_queue.c: %ld and time_t don't match, so cast the
	  argument to long to ease portability problems

2006-11-07 21:47 +0000 [r47290]  Steve Murphy <murf@digium.com>

	* main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc,
	  channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
	  channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
	  main/channel.c, channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_alsa.c, channels/chan_nbs.c,
	  include/asterisk/stringfields.h, channels/chan_mgcp.c,
	  apps/app_voicemail.c: A fair number of changes for the sake of
	  bug 7506

2006-11-07 20:16 +0000 [r47285-47288]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47287 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov
	  2006) | 2 lines This is not the commit you are looking for...
	  ........

	* channels/chan_local.c, /: Merged revisions 47284 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov
	  2006) | 2 lines Make MOH work as it did before in chan_local,
	  without this then it can go funky when transfers and MOH are
	  involved. (issue #7671 reported by jmls) ........

2006-11-07 18:56 +0000 [r47280]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configs/musiconhold.conf.sample: Merged revisions 47279 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006)
	  | 2 lines clean up sample config, and make native file playback
	  the more obvious default choice ........

2006-11-07 18:50 +0000 [r47278]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: rge overhaul to voicemail imap support.
	  Allows support for more imap servers, also a better
	  implementation of several parts of the original work. patch
	  provided by 8033 with major upgrades. minor differences from 1.4
	  patch do to changes in app_voicemail

2006-11-07 17:33 +0000 [r47269]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Break -> continue to make stuff work...
	  Thanks, Luigi!

2006-11-07 14:25 +0000 [r47257-47259]  Kevin P. Fleming <kpfleming@digium.com>

	* /: remove another broken property merge

	* /: remove properties that shouldn't be merged to this branch

	* /: use editable URL for menuselect, and switch to trunk

2006-11-07 13:26 +0000 [r47251-47252]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: issue #8265 - don't reply to ACK.
	  Imported from 1.2, 1.4

	* include/asterisk/frame.h: Stealing Tilghman's explanation from
	  the -dev list and producing documentation...

2006-11-07 08:34 +0000 [r47242]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c: explain why ast_carefulwrite is written the way it
	  is, and also that it doesn't really work as claimed.

2006-11-07 01:28 +0000 [r47232-47240]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 47239 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r47239 | russell | 2006-11-06 20:25:10 -0500
	  (Mon, 06 Nov 2006) | 13 lines Merged revisions 47238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
	  Nov 2006) | 5 lines If random order is enabled for files mode
	  music on hold, set a random initial position, instead of always
	  starting at the first file, and doing the random operation only
	  when switching to the next file. (bug reported by John Lange on
	  the asterisk-dev mailing list) ........ ................

	* utils/check_expr.c: check for failure after call to calloc()
	  (issue #8295)

2006-11-06 17:27 +0000 [r47230]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: minor change to test live syncing

2006-11-06 17:05 +0000 [r47229]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, utils/astman.c, include/asterisk/manager.h: Add
	  support for manager hooks, so you could fire off manager events
	  over IRC if you were crazy enough. (issue #5161 reported by anthm
	  with mods by moi)

2006-11-05 01:04 +0000 [r47210-47213]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Make pbx_dundi compile again. Sorry. :(

	* configs/zapata.conf.sample: List ss7 with the rest of the valid
	  signalling types. Group SS7 options together and comment them out
	  by default.

2006-11-04 22:16 +0000 [r47209]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't lock dialoglist if monitor runs
	  __sip_destroy. Hmmm. I did not change pbx_dundi and yet it
	  doesn't compile ;-)

2006-11-04 22:08 +0000 [r47206-47207]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: use the AST_MODULE_LOAD_* return codes from
	  load_module()

	* pbx/pbx_dundi.c: simplify a couple of loops

2006-11-04 21:48 +0000 [r47205]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Move IP address selection for media out of
	  add_sdp

2006-11-04 21:44 +0000 [r47204]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Do some minor cleanup to the section of code
	  that sets the EID by getting the mac address for an ethernet
	  interface

2006-11-04 21:17 +0000 [r47200-47203]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make srvlookup global_srvlookup to follow
	  the rest of the code

	* channels/chan_sip.c: Simplify previous patch

	* channels/chan_sip.c, configs/sip.conf.sample: Adding new config
	  option "limitpeersonly" to only apply call limits to the peer
	  side of a type=friend. This is for trying to support BJ in his
	  quest to solve some issues with the queue system and type=friend
	  objects. BJ: Please test!

	* /, channels/chan_sip.c: Importing patch for Invite/replaces from
	  1.4

2006-11-04 18:12 +0000 [r47197-47198]  Russell Bryant <russell@digium.com>

	* /, main/cli.c: Merged revisions 47196 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47196 | russell | 2006-11-04 13:10:22 -0500 (Sat, 04 Nov 2006) |
	  2 lines Fix another bug in "core set debug" ... ........

	* /, main/asterisk.c, main/cli.c: Merged revisions 47195 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47195 | russell | 2006-11-04 12:59:39 -0500 (Sat, 04 Nov 2006) |
	  2 lines Really fix the "core set debug" and "core set verbose"
	  CLI commands. ........

2006-11-04 17:45 +0000 [r47194]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Reverting rev 47093 until we have an
	  agreement on how to implement this, if at all.

2006-11-04 17:40 +0000 [r47193]  Russell Bryant <russell@digium.com>

	* /, main/cli.c: Merged revisions 47192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47192 | russell | 2006-11-04 12:38:24 -0500 (Sat, 04 Nov 2006) |
	  3 lines fix the "atleast" option to the "core set verbose" and
	  "core set debug" CLI commands ........

2006-11-04 11:00 +0000 [r47179-47189]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dial.c: move out another large block to a large
	  function, and document some possibly missing parts in the privacy
	  screening code. Now that it is more streamlined it is easier to
	  see differences in handling the various cases. Have not tested
	  the code in depth.

	* res/res_agi.c: useless cast removal...

	* main/logger.c: remove many unnecessary casts

	* main/app.c: remove a useless cast

	* configs/manager.conf.sample: document the "debug" parameter, and
	  the change manager list -> manager show

	* apps/app_dial.c: fix indentation of a block, and do minor
	  simplifications at the end of another one.

	* apps/app_dial.c: complete previous commit.

	* apps/app_dial.c: move another block into a function. On passing,
	  avoid two null-pointer string dereference while printing messages
	  (which are sometimes not fatal in some platforms, but still
	  wrong). These two lines at least should be merged to 1.4 once i
	  am done with all the changes here.

	* apps/app_dial.c: move a large block into a separate function.
	  Mark with XXX a possible bug in previous code which used the
	  wrong source in case of a forwarded call. the function
	  do_forward() needs to be split further, as the initial part is
	  replicated in another places (with some minor differences, most
	  likely forgotten when updating after the copy).

2006-11-03 23:27 +0000 [r47178]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This fix introduced via bug 8233

2006-11-03 23:24 +0000 [r47160-47177]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_dial.c: another small set of simplifications

	* apps/app_dial.c: change HANDLE_CAUSE into a function.

	* apps/app_dial.c: remove redundant checks

	* apps/app_dial.c: start integrating the simplifications proposed
	  in bug 0005860, as usual a bit at a time to ease locating new
	  bugs or fixes worth merging into other branches. In this commit,
	  introduce a macro, S_REPLACE, that replaces a string possibly
	  freeing the previous value. In one of these places (see the
	  comment marked XXX) the previous code might leak memory - if so,
	  this ought to be merged in 1.4 The macro might be worth putting
	  in one of the global headers (e.g. include/asterisk/strings.h) as
	  the construct is used in a million places in the asterisk code.

2006-11-03 19:15 +0000 [r47146]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: One has to create the path and filename in
	  order to copy a file there. (issue #8278 reported by davebath)

2006-11-03 18:53 +0000 [r47072-47132]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c, include/asterisk/manager.h: add a new
	  cli/manager.conf option "debug" to enable/disable debugging code
	  in the manager. At the moment the debugging code is very
	  lightweight, if the option is enabled manager messages also carry
	  a sequence number and the info where they have been generated
	  e.g. SequenceNumber: 10 File: chan_sip.c Line: 11927 Func:
	  handle_response_register It is not worthwhile having this as a
	  compile time option right now, because the extra work involved at
	  runtime is just checking one variable.

	* channels/chan_zap.c: remove old/useless usecnt stuff

	* channels/chan_vpb.cc: remove old/useless usecnt stuff. I think
	  this module doesn't compile, anyways, because it has not been
	  updated to the new module interface.

	* main/cli.c: Fix "core show channels" and "core show modules". Not
	  sure it applies like this to 1.4 because of deprecate versions of
	  the same command(s).

	* res/res_jabber.c: move variable declarations to the beginning of
	  a block.

	* /: block other changes of mine already applied to trunk.

	* /: block more changes of mine already applied to trunk

	* /: blocking 47107

	* /: blocking 47108

	* channels/chan_sip.c: Save the 'From' header received in a
	  REGISTER message so we can show it e.g. in the Manager interface.
	  This information is available as a callerid (or something like
	  that) during a call, but not when a device is registered but
	  silent. It may be useful to have it available e.g. when
	  developing a user interface/operator panel, to map numbers to
	  names. experimental, so not committed to 1.4

	* channels/chan_jingle.c, channels/chan_gtalk.c: remove useless
	  usecnt stuff

	* channels/chan_phone.c: remove useless usecnt stuff

	* channels/chan_alsa.c: remove useless usecnt stuff

	* channels/chan_agent.c: remove useless usecnt stuff

	* channels/chan_features.c: remove useless usecnt handling

	* channels/chan_skinny.c: remove useless usecnt handling code

2006-11-02 23:55 +0000 [r47052-47054]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, /, channels/chan_skinny.c, res/res_agi.c,
	  channels/chan_h323.c, res/res_jabber.c, main/rtp.c: Merged
	  revisions 47053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006)
	  | 2 lines More changes making the CLI more consistent with
	  "category verb arguments" (continuation of issue 8236) ........

	* main/pbx.c, channels/chan_local.c, main/frame.c,
	  channels/chan_sip.c, /, res/res_features.c, res/res_crypto.c,
	  channels/chan_agent.c, res/res_musiconhold.c, apps/app_queue.c,
	  channels/chan_iax2.c, main/config.c, main/cli.c, main/channel.c,
	  main/manager.c, channels/chan_skinny.c, res/res_agi.c,
	  channels/chan_features.c, main/logger.c, main/file.c,
	  main/http.c, res/res_indications.c, main/image.c, res/res_odbc.c,
	  main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c:
	  Merged revisions 47051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006)
	  | 2 lines Reverse change of "show" to "list" and make several
	  other commands more consistent with "category verb arguments"
	  ........

2006-11-02 21:40 +0000 [r47037]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: Let's make
	  application/function/hint lists read/write lists... just for
	  kicks

2006-11-02 21:34 +0000 [r47035]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Updates to do unblock correctly

2006-11-02 20:24 +0000 [r46999-47021]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Move check for codec translators to an
	  earlier place in the call, so we can fail gracefully (imported
	  from 1.4)

	* /, channels/chan_sip.c: Disable code for not implemented
	  functionality (T38 over RTP/TCP)

2006-11-02 18:34 +0000 [r46991-46994]  Russell Bryant <russell@digium.com>

	* include/asterisk/astobj.h: Sure enough, some of the uses of
	  astobj are doing recursive locking. This doesn't work with
	  rwlocks, so, this is reverted for now.

	* include/asterisk/astobj.h: astobj was already set up to use read
	  and write locks. Now that we have wrappers for them, use them
	  here.

2006-11-02 18:01 +0000 [r46967-46972]  Joshua Colp <jcolp@digium.com>

	* main/translate.c: Convert translation core linked list over to
	  read/write based one, since it spends most of it's time only
	  reading.

	* include/asterisk/linkedlists.h: Add AST_RWLIST_* set of macros
	  which implement linked lists using read/write locks, the actual
	  list manipulation is still done via the old macros.

2006-11-02 17:51 +0000 [r46966]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 46965 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46965 | russell | 2006-11-02 12:49:54 -0500
	  (Thu, 02 Nov 2006) | 11 lines Merged revisions 46964 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
	  Nov 2006) | 3 lines ignore files in a music on hold directory
	  that begin with '.' (issue #8249, cboie) ........
	  ................

2006-11-02 16:51 +0000 [r46940]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/lock.h: Set the AST_RWLOCK_INIT_VALUE to the
	  PTHREAD_RWLOCK_INIT_VALUE if it is available, that way outside
	  stuff can determine whether to use a constructor or deconstructor
	  for initialization instead of using the init value.

2006-11-02 16:50 +0000 [r46939]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Changes to show blocked/unblocked states, as
	  well as in service, out of service state

2006-11-02 16:45 +0000 [r46938]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 46937 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006)
	  | 2 lines don't send INVITE when we have determined that we can't
	  offer any audio formats due to lack of trancoding support (or
	  incorrect configuration) ........

2006-11-02 16:28 +0000 [r46931-46935]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: I'm crazy so I will add this... pthread
	  rwlock wrappers, along with autoconf stuff that detects the
	  presence of the initializer and the ability to set the kind of
	  lock (in our case we rather like writer preferred locks so writer
	  starvation doesn't occur... but on something like Darwin we don't
	  get that)

	* /, channels/chan_sip.c: Merged revisions 46930 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46930 | file | 2006-11-02 11:06:39 -0500 (Thu,
	  02 Nov 2006) | 10 lines Merged revisions 46920 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
	  lines Repeat after me oej: I will at least make sure my code
	  compiles before I commit it. ........ ................

2006-11-02 16:03 +0000 [r46926]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add simple down event support

2006-11-02 15:47 +0000 [r46906]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c:
	  find_free_chan_in_stack: cleanup buggy usage

2006-11-02 15:31 +0000 [r46902]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't overwrite pkt->flags (imported from
	  1.2/1.4)

2006-11-02 14:15 +0000 [r46846-46886]  Russell Bryant <russell@digium.com>

	* main/callerid.c: various whitespace changes to reduce indentation
	  and to better conform to formatting guidelines

	* main/callerid.c: Change the buffer used in callerid_feed() and
	  callerid_feed_jp() to be allocated on the stack using alloca()
	  instead of using malloc() since they are only used locally to
	  these functions.

	* /, main/say.c: Merged revisions 46857 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46857 | russell | 2006-11-01 18:01:48 -0500 (Wed, 01 Nov 2006) |
	  2 lines fix saying one hundred and two hundred in hebrew (issue
	  #7810, eldadran) ........

	* CHANGES: Add a couple of things to the CHANGES file

	* Makefile, /, configure, codecs/gsm/Makefile, configure.ac,
	  build_tools/strip_nonapi, makeopts.in: Merged revisions 46847 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46847 | russell | 2006-11-01 17:51:21 -0500 (Wed, 01 Nov 2006) |
	  3 lines Fixes for cross-compilation on mips (issue #8058,
	  ywalther, with some modifications) ........

	* aclocal.m4, /, build_tools/menuselect-deps.in, configure,
	  build_tools/embed_modules.xml, configure.ac: Merged revisions
	  46845 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46845 | russell | 2006-11-01 17:32:12 -0500 (Wed, 01 Nov 2006) |
	  5 lines Add a check in the configure script to determine whether
	  ld is GNU ld or not. This is needed because module embedding only
	  works for gnu ld. GNU ld is now listed as a dependency for all of
	  the module embedding options in menuselect. (issue #8143)
	  ........

2006-11-01 20:38 +0000 [r46823]  Matt O'Gorman <mogorman@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 46822 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.4 ........
	  r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006)
	  | 2 lines bind address support from bug 8164 ........

2006-11-01 19:48 +0000 [r46801]  Steve Murphy <murf@digium.com>

	* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
	  accept longer strings or mass confusion and a lot of lost time is
	  the result

2006-11-01 18:41 +0000 [r46782]  Joshua Colp <jcolp@digium.com>

	* /, main/Makefile: Merged revisions 46780 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2
	  lines Force poll() emulation for Darwin to always be on. It's too
	  broken to consider being used. This resolves the console issue
	  OSX users have been seeing. I would have liked to autoconf this
	  but I haven't been able to come up with a test case that works.
	  Que sera. ........

2006-11-01 18:40 +0000 [r46779-46781]  Russell Bryant <russell@digium.com>

	* doc/channelvariables.txt, pbx/pbx_dundi.c: Add the ability to
	  pass options to the Dial application when using the DUNDi switch
	  in the dialplan by setting the DUNDIDIALARGS channel variable.
	  (issue #8084, patch by bluecrow76, with small modifications and
	  documentation updates)

	* /, res/res_monitor.c: Merged revisions 46778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46778 | russell | 2006-11-01 13:26:35 -0500
	  (Wed, 01 Nov 2006) | 17 lines Merged revisions 46776 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01
	  Nov 2006) | 9 lines soxmix and Asterisk expect different file
	  extensions for certain formats. This was already handled for the
	  wav49 format. However, it was not handled for ulaw and alaw. I
	  fixed this in such a way that using the alternate extensions for
	  ulaw and alaw will only happen if we know we're calling soxmix,
	  and not a custom script defined using the MONITOR_EXEC variable.
	  The wav49 processing was left alone so that external scripts will
	  see no behavior change. (issue #7550, reported by mnicholson,
	  proposed patch by junky, committed fix is a bit different)
	  ........ ................

2006-11-01 18:26 +0000 [r46777]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 46775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2
	  lines It's another round of chan_iax2 fixes! Should hopefully fix
	  the deadlock issues people have been reporting. IAXtel now has
	  qualify turned on for 800 peers and it is handling it fine.
	  ........

2006-11-01 18:16 +0000 [r46759-46774]  Steve Murphy <murf@digium.com>

	* CHANGES: OOps. forgot to add this to CHANGES

	* main/say.c, apps/app_voicemail.c: This introduces Brazilian
	  Portuguese via 7663

	* main/config.c: Cleanups suggested by Russell.

2006-11-01 17:09 +0000 [r46758]  Luigi Rizzo <rizzo@icir.org>

	* res/res_features.c: move variable declaration in the middle of a
	  block

2006-11-01 16:51 +0000 [r46745]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 46744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46744 | russell | 2006-11-01 11:39:09 -0500 (Wed, 01 Nov 2006) |
	  2 lines Prevent an infinite loop when config processing gets to a
	  jitterbuffer option ........

2006-11-01 00:07 +0000 [r46732]  Matt O'Gorman <mogorman@digium.com>

	* res/res_features.c: change default return extension after parking
	  timeout. 6953 with minor changes.

2006-10-31 22:19 +0000 [r46719]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/translate.c, include/asterisk/translate.h: Merged
	  revisions 46714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46714 | kpfleming | 2006-10-31 15:47:48 -0600 (Tue, 31 Oct 2006)
	  | 2 lines add an API so that translators can activate/deactivate
	  themselves when needed ........

2006-10-31 22:07 +0000 [r46717-46718]  Jason Parker <jparker@digium.com>

	* main/translate.c: Fix "core show translation" output. Issue
	  #8243, patch by Damin.

2006-10-31 18:10 +0000 [r46683-46696]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_iax2.c: remove old/useless usecount handling

	* channels/chan_sip.c: remove old/useless usecount stuff.

	* channels/chan_oss.c: remove old/useless usecount management code.

2006-10-31 15:22 +0000 [r46661]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix the new send text manager command. There is
	  no way this could have worked. - Check the channel name string
	  length to be zero, not non-zero - Check the message string length
	  to be zero, not non-zero - unlock the channel *after* calling
	  sendtext

2006-10-31 13:56 +0000 [r46582-46650]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Set #define for TIMER T1 value

	* channels/chan_sip.c: Cleaning up code

	* funcs/func_enum.c, /, include/asterisk/enum.h, main/enum.c: Issue
	  #80898 - Restoring func_enum (otmar)

	* main/manager.c: Add manager sendtext action. (Issue 6131, ZX81 -
	  thanks!)

	* /, channels/chan_sip.c, configs/sip.conf.sample: Fix rport
	  handling. ...where did the 1.2 properties come from, really?
	  they're back.

	* /, channels/chan_sip.c: - If peer that register fails ACL, fail
	  him - Remove the 1.2 props I've set by mistake earlier

	* /: Block patch that only applies to 1.4

	* main/loader.c: Take two, using find_resource on Kevin's
	  suggestion. Might need better locking support, giving up if we
	  can't get the lock. Right now, using existing locking in
	  find_resource

2006-10-31 06:37 +0000 [r46556-46565]  Russell Bryant <russell@digium.com>

	* apps/app_cdr.c: add author doxygen tag (issue #8241, kshumard)

	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 46563 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46563 | russell | 2006-10-31 01:30:53 -0500 (Tue, 31 Oct 2006) |
	  3 lines Start Asterisk later in the boot process to ensure it
	  starts after stuff like MySQL (issue #8253, Alric) ........

	* /, main/utils.c: Merged revisions 46561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46561 | russell | 2006-10-31 01:19:56 -0500
	  (Tue, 31 Oct 2006) | 11 lines Merged revisions 46560 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31
	  Oct 2006) | 3 lines When handling the case where the hostname is
	  just an IPV4 numeric address, be sure to set the address type.
	  (issue #8247, alexr) ........ ................

	* /, res/res_agi.c: Merged revisions 46558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46558 | russell | 2006-10-31 01:14:13 -0500
	  (Tue, 31 Oct 2006) | 11 lines Merged revisions 46557 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31
	  Oct 2006) | 3 lines fix some copy/paste bugs in the checking of
	  arguments for the "control stream file" AGI command (issue #8255,
	  mnicholson) ........ ................

	* /, main/translate.c: Merged revisions 46554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) |
	  5 lines Add a small tweak to the code that checks to see whether
	  destination formats are translatable based on the source format.
	  If we have already determined that there is no translation path
	  in one direction, don't bother checking the other direction.
	  ........

2006-10-30 23:11 +0000 [r46541]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, utils/astman.c: These changes submitted by moy
	  via bug 6992, to add a Dial 'End' event to asterisk. I include
	  some changes to astman to cover other events that have been
	  added.

2006-10-30 22:27 +0000 [r46529]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/translate.c: Merged revisions 46526 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006)
	  | 3 lines when unregistering a translator, don't rebuild the
	  translation matrix unless needed when filtering formats out of an
	  offer, ensure we check for translation ability in both directions
	  ........

2006-10-30 21:56 +0000 [r46513-46514]  Olle Johansson <oej@edvina.net>

	* funcs/func_module.c: show, list, view, display... whatever.

	* funcs/func_module.c (added), include/asterisk/module.h,
	  main/loader.c: Adding dialplan function IFMODULE, so you can
	  create dialplans that handle various PBX installations and checks
	  if a module is loaded before using it. example
	  IFMODULE(chan_sip3.so) issue #6671 in the bug tracker, finally
	  gone. Thanks to mithraen for keeping it updated.

2006-10-30 21:46 +0000 [r46512]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 46511 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46511 | kpfleming | 2006-10-30 15:46:07 -0600 (Mon, 30 Oct 2006)
	  | 2 lines ensure that items removed from a list are always
	  unlinked from the list (next pointer set to NULL) ........

2006-10-30 21:22 +0000 [r46508-46509]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Update sip list to eventlist format.

	* main/pbx.c, main/manager.c, include/asterisk/manager.h: Issue
	  #3930 - Add manager command for listing dialplan (coded april
	  2005, in bugtracker since)

2006-10-30 21:11 +0000 [r46507]  Joshua Colp <jcolp@digium.com>

	* /, configure, configure.ac: Merged revisions 46506 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r46506 | file | 2006-10-30 16:09:13 -0500 (Mon, 30 Oct
	  2006) | 2 lines Don't explicitly link in crypt as it is not used
	  on some platforms. ........

2006-10-30 19:56 +0000 [r46476-46489]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Change name of
	  "contact" setting to "callback" which better reflects what it is
	  to the person that configures asterisk. That we use it internally
	  in the contact header is a totally different story. Still not
	  convinced this is a good option.

	* channels/chan_sip.c: Globals need the "global_" prefix in
	  chan_sip, and need to be reset to default value at reload.

2006-10-30 18:17 +0000 [r46475]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 46474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46474 | file | 2006-10-30 13:13:07 -0500 (Mon, 30 Oct 2006) | 2
	  lines We need to lock the pvt structure during retransmission as
	  another worker thread may be doing something as well. ........

2006-10-30 18:04 +0000 [r46466]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we give the linkset number, not
	  the offset in the linksets array

2006-10-30 18:02 +0000 [r46461]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Small conversion to ast_channel_unlock

2006-10-30 17:32 +0000 [r46459]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Specify which linkset we're getting the
	  messages from in the message

2006-10-30 16:59 +0000 [r46439]  Olle Johansson <oej@edvina.net>

	* main/rtp.c: In debug mode, recognize that someone is sending
	  zrtp, even though we can't do anything with it yet. Ideally a
	  first step would be a passthrough mode.

2006-10-30 16:50 +0000 [r46436]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Don't make errors when we don't need them

2006-10-30 16:33 +0000 [r46379-46434]  Olle Johansson <oej@edvina.net>

	* include/asterisk/file.h, include/asterisk/doxyref.h, /,
	  channels/chan_sip.c, main/ast_expr2f.c,
	  include/asterisk/module.h, formats/format_ogg_vorbis.c,
	  main/app.c, include/asterisk/channel.h, include/asterisk/lock.h,
	  include/asterisk/frame.h, main/asterisk.c, apps/app_voicemail.c:
	  Issue 8246 Doxygen updates (kshumard) THANK YOU!

	* /: The RTCP patch started in trunk, so don't start all over again
	  :-)

	* main/asterisk.c: Small formatting changes

	* main/rtp.c: Bind RTCP to the same IP as RTP. I currently don't
	  see this as a bug that needs to be fixed in 1.4/1.2 too, but feel
	  free to backport if you see it that way. RTCP now binds to ALL IP
	  addresses on the host, RTP to a specific address.

	* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
	  redirects.

	* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
	  wrong content-type (imported from 1.2, 1.4)

	* /: Block patch from other branch

	* channels/chan_sip.c: Issues related to issue #7828 - segfault
	  with MWI subscriptions and realtime.

	* /, channels/chan_sip.c: - Fix the OUTGOING stuff (merge from 1.4)
	  - Make sure we UNREF authpeer when not needed

	* apps/app_voicemail.c: Spelling fix.

	* channels/chan_sip.c: Documentation update again

	* channels/chan_sip.c: Documentation update (I guess)

	* channels/chan_sip.c: Documentation correction

	* channels/chan_sip.c: maxtime is not needed any more now that we
	  actually set the T1 timer based on the qualify result.

	* /, channels/chan_sip.c: Only accept message once

	* channels/chan_sip.c: Adding documentation inspired by a virtual
	  drink with an anonymous man in New Jersey

	* channels/chan_sip.c: Don't duplicate function if not needed... -
	  removing transmit_reinvite_with_t38_sdp in favour of adding an
	  argument to transmit_reinvite_with_sdp

	* /, channels/chan_sip.c: Merge from 1.4 : Don't send 183
	  reliably...

	* channels/chan_sip.c: - Don't lock the dialoglist during the whole
	  destruction of a single SIP dialog. Only lock when needed - when
	  we remove the dialog from the dialog list If this doesn't lead to
	  severe problems, it might help with some locking issues in
	  1.4/1.2. - Remove the term "interface" as a synonym for a SIP
	  dialog. Sorry, Mark, but no one understands it... ;-)

2006-10-28 16:39 +0000 [r46378]  Joshua Colp <jcolp@digium.com>

	* utils/Makefile, /: Merged revisions 46377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46377 | file | 2006-10-28 12:37:44 -0400 (Sat, 28 Oct 2006) | 2
	  lines Don't build muted on OpenBSD, it is not supported. ........

2006-10-27 19:28 +0000 [r46372]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Let's make sure we hold the mutex lock before
	  we go looking at values in the queue structure that could
	  potentially be changing while we're running.

2006-10-27 19:04 +0000 [r46371]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 46370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46370 | russell | 2006-10-27 14:03:32 -0500 (Fri, 27 Oct 2006) |
	  4 lines move the copy of the default settings to the global
	  settings back out of process_zap, so that they aren't overwritten
	  when process_zap is called multiple times ........

2006-10-27 18:59 +0000 [r46369]  BJ Weschke <bweschke@btwtech.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: * Added
	  option to run macro when a queue member is connected to a caller,
	  see queues.conf.sample for details. * Added QUEUE_VARIABLES
	  function to set queue variables added setqueuevar and
	  setqueueentryvar options for each queue, see queues.conf.sample
	  for details. (#8216, jmls reported and submitted)

2006-10-27 18:31 +0000 [r46368]  Olle Johansson <oej@edvina.net>

	* /, contrib/asterisk-ng-doxygen: raise the pressure on Christian
	  :-)

2006-10-27 17:46 +0000 [r46366]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: First pass at implementation to be able to
	  block and unblock zap channels for use.

2006-10-27 17:45 +0000 [r46365]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Put this patch on hold pending further
	  testing...

2006-10-27 17:42 +0000 [r46359-46364]  Russell Bryant <russell@digium.com>

	* /, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c,
	  main/asterisk.c: Merged revisions 46363 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) |
	  5 lines We should always be using _exit() after a fork() or
	  vfork() instead of exit(). This is because exit() does some extra
	  cleanup which in some implementations of vfork(), for example,
	  can actually modify the state of the parent process, causing very
	  weird bugs or crashes. (issue #7971, Nick Gavrikov) ........

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
	  the ability to customize some of the prompts used within the
	  voicemail application by configuring them in voicemail.conf
	  (issue #7415, patch by fkasumovic, with some fixes and
	  documentation updates by myself)

	* channels/chan_zap.c, /: Merged revisions 46358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) |
	  5 lines Instead of iterating all of the options once to look for
	  jitterbuffer options, and then again for everything else, move
	  the processing of jitterbuffer options into the main loop so that
	  there are no erroneous messages about ignoring unknown options.
	  (issue #8226) ........

2006-10-27 11:18 +0000 [r46354]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h,
	  channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
	  revisions 46351-46353 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r46351 | crichter | 2006-10-27 11:49:20 +0200
	  (Fr, 27 Okt 2006) | 9 lines Merged revisions 46176 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25
	  Okt 2006) | 1 line added nttimeout option to configure wether we
	  disconnect calls on NT timeouts or not during an overlapdial
	  session ........ ................ r46352 | crichter | 2006-10-27
	  11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line fixed not compile
	  issue, which was just introduced ................ r46353 |
	  crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
	  Merged revisions 46350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
	  1 line fixed a bug which caused chan_misdn to try to allocate 2
	  times the same channel on high load, which then caused
	  instability of mISDN. removed a useless function from isdn_lib.c
	  ........ ................

2006-10-26 20:27 +0000 [r46348]  Jason Parker <jparker@digium.com>

	* /, apps/app_page.c: Merged revisions 46347 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2
	  lines Fix small formatting issue, that causes misaligned line
	  ........

2006-10-26 20:22 +0000 [r46346]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Show if the channel is ready for video or
	  T.38 udptl

2006-10-26 18:04 +0000 [r46341]  Jason Parker <jparker@digium.com>

	* contrib/scripts/astgenkey.8: oops - somebody forgot to change
	  this - long ago, probably.

2006-10-26 17:52 +0000 [r46330-46339]  Russell Bryant <russell@digium.com>

	* main/pbx.c, apps/app_osplookup.c, main/manager.c,
	  apps/app_meetme.c, apps/app_festival.c, main/say.c,
	  apps/app_alarmreceiver.c, apps/app_sms.c, apps/app_rpt.c,
	  main/rtp.c, apps/app_voicemail.c: fix various spelling mistakes
	  in comments (issue #8237, jmls)

	* /, main/translate.c: Merged revisions 46329 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) |
	  11 lines - If the source has no audio or no video portion, do not
	  call powerof() to get the format index. - Don't run through the
	  audio and video loops if there is no audio or video portion of
	  the source If 0 is passed to powerof, it will return -1. This
	  value of -1 was then being used as an array index in these loops,
	  which caused a crash on some systems. Other than this issue, this
	  code works as we expected it to. If a format is not in the
	  source, and we have to translation path to it, it is not offered
	  in the list of acceptable destination formats. (fixes issue
	  #8231) ........

2006-10-26 12:47 +0000 [r46308-46319]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: fix a problem that i recently introduced when the
	  manager receives long commands.

	* configs/sip.conf.sample: document the match_auth_username option

2006-10-26 04:19 +0000 [r46299]  Russell Bryant <russell@digium.com>

	* /, doc/backtrace.txt: Merged revisions 46298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46298 | russell | 2006-10-25 23:18:00 -0500 (Wed, 25 Oct 2006) |
	  2 lines update backtrace documentation to reflect changes in 1.4
	  (issue #8230, kshumard) ........

2006-10-26 01:38 +0000 [r46288]  Mark Spencer <markster@digium.com>

	* main/manager.c, main/config.c: Fix comment preservation code
	  (thanks murf!)

2006-10-25 20:21 +0000 [r46259-46277]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Old todo: Don't add Contact headers on
	  BYE and CANCEL.

	* channels/chan_sip.c: First stab at transaction direction fix,
	  this for trunk for testing

	* /, channels/chan_sip.c: Ugly code to try to remove issue
	  discovered by Luigi as well as attack bug #7608

2006-10-25 19:24 +0000 [r46256]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Send CPG when we get a CONTROL_PROGRESS
	  frame and make sure that it sends ACM (not CPG) when we get
	  CONTROL_PROCEEDING.


2006-10-25 19:14 +0000 [r46251]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, configs/zapata.conf.sample: Update changes
	  to do US style point code parsing/formatting (xxx.xxx.xxx)

2006-10-25 19:10 +0000 [r46250]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 46249 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46249 | russell | 2006-10-25 14:08:18 -0500 (Wed, 25 Oct 2006) |
	  2 lines update warning message to include "agi" option (issue
	  #8225, jmls) ........

2006-10-25 17:12 +0000 [r46238]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/sounds.xml, sounds/Makefile: Merged revisions 46237 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46237 | kpfleming | 2006-10-25 12:08:58 -0500 (Wed, 25 Oct 2006)
	  | 2 lines add support for prebuilt G.722 prompts and music on
	  hold files ........

2006-10-25 16:01 +0000 [r46215-46224]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merge from 1.4

	* /: Block change to 1.4 to block change to 1.2... This is
	  confusing, but I think I got it right.

2006-10-25 14:55 +0000 [r46201-46203]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c, main/translate.c,
	  include/asterisk/translate.h: Merged revisions
	  46082-46083,46152-46153 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006)
	  | 2 lines add an API call to allow channel drivers to determine
	  which media formats are compatible (passthrough or transcode)
	  with the format an existing channel is already using ........
	  r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006)
	  | 2 lines ensure that the translation matrix is properly
	  lock-protected every place it is used ........ r46152 | kpfleming
	  | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines if
	  multiple translators are registered for the same source/dest
	  combination, ensure that the lowest-cost one is always inserted
	  earlier in the list ........ r46153 | kpfleming | 2006-10-24
	  19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines code zone experiment:
	  don't offer formats in the outbound INVITE that aren't either
	  passthrough or translatable ........

	* channels/chan_iax2.c: restore bugfix that was reverted by
	  trunk_mtu patch

	* channels/chan_sip.c, /, apps/app_record.c, apps/app_softhangup.c,
	  res/res_adsi.c, main/utils.c, pbx/dundi-parser.c,
	  apps/app_ices.c, apps/app_getcpeid.c, apps/app_queue.c,
	  channels/chan_iax2.c, main/cli.c, main/cdr.c,
	  channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c,
	  channels/chan_h323.c, pbx/pbx_ael.c, channels/chan_alsa.c,
	  pbx/pbx_realtime.c, apps/app_sms.c, channels/chan_nbs.c,
	  main/image.c, main/db.c, channels/chan_mgcp.c, cdr/cdr_custom.c,
	  apps/app_parkandannounce.c, apps/app_voicemail.c: Merged
	  revisions 46200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006)
	  | 2 lines apparently developers are still not aware that they
	  should be use ast_copy_string instead of strncpy... fix up many
	  more users, and fix some bugs in the process ........

2006-10-25 14:26 +0000 [r46199]  Olle Johansson <oej@edvina.net>

	* CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Ok,
	  second attempt...

2006-10-25 14:18 +0000 [r46198]  Luigi Rizzo <rizzo@icir.org>

	* CHANGES: document a couple of recently introduced feature also
	  including the version number where the feature appeared.

2006-10-25 14:14 +0000 [r46183-46197]  Olle Johansson <oej@edvina.net>

	* CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: On the
	  other hand, don't use 1.4 patches for trunk... Sorry.

	* CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Add
	  ability to adapt the IAX trunk packets to the MTU size, to avoid
	  bad audio when the number of channels fill the MTU on a given
	  link. In the future, this needs to be configurable per peer with
	  trunking enabled.

	* channels/chan_sip.c: Adding comments in the source is more
	  persistent than just adding them to the commit message :-)

	* channels/chan_sip.c: Always add doxygen comments to new
	  functions, more lines than one are appreciated really. (Read the
	  coding guidelines). I've worked hard to make chan_sip a better
	  place to code in, let's keep it that way and don't add more stuff
	  without comments. Thank you.

2006-10-25 00:32 +0000 [r46155]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c, /, main/translate.c, formats/format_pcm.c,
	  channels/chan_h323.c, channels/chan_iax2.c,
	  include/asterisk/frame.h, main/rtp.c: Merged revisions 46154 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006)
	  | 2 lines add passthrough and file format support for G.722 16KHz
	  audio (issue #5084, original patch by andrew, updated by
	  mithraen) ........

2006-10-24 20:22 +0000 [r46141]  Mark Spencer <markster@digium.com>

	* res/res_agi.c: Fix FastAGI to not wait for the non-existant pid

2006-10-24 19:33 +0000 [r46131]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 46130 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46130 | file | 2006-10-24 15:29:56 -0400 (Tue, 24 Oct 2006) | 2
	  lines We need to initialize our scheduler pthread condition...
	  yes. ........

2006-10-24 17:14 +0000 [r46104-46120]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: i really think it is safe to commit this version,
	  that simplifies the manager queue handling as described in the
	  comment, and will make a lot easier to make further work on this
	  code.

	* channels/chan_sip.c: correct fix for the bug i previously
	  introduced - the strings are meant to be always initialized,
	  independently from their content.

2006-10-24 05:24 +0000 [r46094]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 46093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46093 | russell | 2006-10-24 01:23:33 -0400 (Tue, 24 Oct 2006) |
	  3 lines Restore the ability to remove the firmware directory
	  without causing the installation to fail (issue #8111) ........

2006-10-24 03:15 +0000 [r46081]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/imapstorage.txt, /: Merged revisions 46080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46080 | kpfleming | 2006-10-23 22:13:08 -0500 (Mon, 23 Oct 2006)
	  | 2 lines simplify and correct voicemail IMAP storage build
	  instructions ........

2006-10-24 03:09 +0000 [r46079]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 46078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006)
	  | 3 lines Pass through a frame if we don't know what it is,
	  rather than trying to pass a NULL, which will segfault a channel
	  driver (Bug 8149) ........

2006-10-24 01:28 +0000 [r46055-46068]  Russell Bryant <russell@digium.com>

	* utils/muted.c, /, utils/ael_main.c: Merged revisions 46067 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46067 | russell | 2006-10-23 21:27:42 -0400 (Mon, 23 Oct 2006) |
	  7 lines In muted.c, check the return value of strdup. In
	  ael_main.c, check the return value of calloc. (issue #8157) In
	  passing fix a few minor bugs in ael_main.c. The last argument to
	  strncpy() was a hard-coded 100, where it should have been 99. I
	  changed this to use sizeof() - 1. ........

	* /, apps/app_meetme.c: Merged revisions 46065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r46065 | russell | 2006-10-23 21:04:14 -0400 (Mon, 23 Oct 2006) |
	  2 lines Fix the descriptions of some of the MeetMeAdmin options
	  (issue #8098, mflorell) ........

	* channels/chan_sip.c: Fix a seg fault on a registration. Line
	  7706, in parse_register_contact, explicitly passes NULL as the
	  "pass" argument to this function.

2006-10-23 21:46 +0000 [r46003-46045]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: Unlike ast_strdup(), ast_strdupa() does not
	  take a NULL pointer as argument, so fix the places where this
	  might happen. This is also a fix that ought to go into 1.4 [The
	  difference between the two functions is a bit confusing, and in
	  asterisk i believe all string handling functions should be able
	  to handl a NULL string as argument, but changing the API in trunk
	  and not in 1.4 would make backporting harder.]

	* channels/chan_sip.c: remove a useless check for ocseq = 0. As
	  discussed on the mailing lists, 0 is a legal value for Cseq, so
	  there is no point to treat it specially.

	* channels/chan_sip.c: get_header() always returns a non-NULL
	  value, so checking for NULL is certainly wrong and usually
	  disables the checks that we want to make instead. This commit
	  fixes a number of the above bugs where the result of get_header()
	  is immediately checked for NULL. This is certainly a candidate
	  for merging into 1.4

	* channels/chan_sip.c: put another duplicated block of code in a
	  function.

	* channels/chan_sip.c: reformat a statement and comment a
	  potentially wrong assignement (altering state on an unvalidated
	  message).

	* channels/chan_sip.c: Remove unnecessary casts from const char *
	  to char *, if necessary by slightly rearranging the code.

	* channels/chan_sip.c: another use for parse_uri(). On passing,
	  remove a wrong comment (that probably I wrote myself!) and
	  introduce a temporary variable to avoid a misleading cast.

2006-10-23 17:08 +0000 [r46000]  Russell Bryant <russell@digium.com>

	* /, res/res_jabber.c: Merged revisions 45999 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45999 | russell | 2006-10-23 13:07:45 -0400 (Mon, 23 Oct 2006) |
	  2 lines don't crash when an incoming message has no "from" (issue
	  #8205, jmls) ........

2006-10-23 16:54 +0000 [r45945-45989]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c: use autodetected support for gethostbyname_r

	* channels/chan_sip.c: + make sure parse_uri never returns NULL
	  pointers - this simplifies its usage. + add another client for
	  parse_uri, in handling Contact: strings (on passing, document the
	  content of the "fullcontact" field); + in register_verify(), mark
	  with XXX what i believe is another misinterpretation on the URI
	  format when '@' is missing. No code changed here, so no fixes
	  applied.

	* channels/chan_sip.c: After reading better the SIP RFC on sip URI
	  (19.1.1) fix parse_uri() to interpret a missing userinfo section
	  as a domain-only URI, and comment a wrong interpretation of the
	  above in check_user_full(). The function has been patched to
	  preserve the existing behaviour (in what admittedly is a corner
	  case, but could be received under attacks). Hopefully the From:
	  based matching will go away soon!

	* channels/chan_sip.c: in function get_also_info(), move argument
	  stripping before splitting around the @, otherwise the
	  refer_to_domain might contain arguments as well, causing
	  failures. I think this is a true bug that ought to be fixed in
	  1.4 as well.

	* channels/chan_sip.c: start putting the URI parsing code in one
	  place, introducing the function parse_uri() that splits a URI in
	  its components. Right now use it only in one place, because the
	  custom parsing that is done here and there sometimes has bugs
	  that i want to figure out first.

	* channels/chan_sip.c: put common code in function terminate_uri()
	  so we need to fix it only in one place.

	* channels/chan_sip.c: More cleanup of check_user_full with no
	  functional change apart from a small (but disabled by default)
	  new option. In detail: + introduce a new value for enum
	  check_auth_result, AUTH_DONT_KNOW, used (read below) when a
	  function does not have a conclusive response. Possibly this is
	  the same as AUTH_NOT_FOUND, but need to check further. + move the
	  large blocks (checking in the users list and in the peers list,
	  respectively) from check_user_full() to separate functions. They
	  return AUTH_DONT_KNOW in case they don't find a match, so the
	  caller know that it has to try the next method. There is still
	  some duplication of code here, but i have not tried yet to remove
	  it. + [new option] a new option in sip.conf, match_auth_username,
	  has been introduced, and disabled by default. If set, and the
	  incoming request carries authentication info, the username to
	  match in the users list is taken from there rather than from the
	  From: field. This change is easy to identify, being made of - one
	  line to declare the variable match_auth_username - a block of 15
	  lines in check_user_full() - one line in sip list settings - two
	  lines for parsing the config file. check_user_full() is now a lot
	  cleaner - basically a sequence of checks that are applied to the
	  request. This will help future work with new matching schemes.

2006-10-23 00:33 +0000 [r45929]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_odbc.c: Merged revisions 45928 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45928 | file | 2006-10-22 20:27:39 -0400 (Sun,
	  22 Oct 2006) | 10 lines Merged revisions 45927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
	  lines Don't leak memory mmmk? ........ ................

2006-10-22 21:57 +0000 [r45917]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 45916 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45916 | crichter | 2006-10-22 23:44:46 +0200
	  (Sun, 22 Oct 2006) | 9 lines Merged revisions 45808 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
	  Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
	  couldn't be initialized it would cause a segfault after 'reload'.
	  Reported by Drew/Matt thx. ........ ................

2006-10-22 21:08 +0000 [r45904-45915]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: more streamlining of check_user_full

	* channels/chan_sip.c: simplify the flow of function
	  check_user_full() A large block needs reindentation now, but we
	  don't do that because it can be moved to a separate function.

	* channels/chan_sip.c: put duplicated code in functions.

2006-10-22 19:34 +0000 [r45893]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in: regenerate the
	  configure script and autoconfig.h.in to reflect recent changes
	  for https support for the built in http server

2006-10-22 19:09 +0000 [r45858-45892]  Luigi Rizzo <rizzo@icir.org>

	* main/Makefile, configure.ac, main/http.c,
	  configs/http.conf.sample: Fix a few issues in the previous
	  (disabled) HTTPS code, and support linux as well (using
	  fopencookie(), which should be available in glibc). Update
	  configure.ac to check for funopen (BSD) and fopencookie(glibc),
	  and while we are at it also for gethostbyname_r (the generated
	  files need to be updated, or you need to run bootstrap.sh
	  yourself). Document the new options in http.conf.sample (names
	  are only tentative, better ones are welcome). At this point we
	  can safely enable the option. Anyone willing to try this on Sun
	  and Apple platforms ?

	* main/http.c: Implement https support. The changes are not large.
	  Most of the diff comes from putting the global variables
	  describing an accept session into a structure, so we can reuse
	  the existing code for running multiple accept threads on
	  different ports. Once this is done, and if your system has the
	  funopen() library function (and ssl, of course), it is just a
	  matter of calling the appropriate functions to set up the ssl
	  connection on the existing socket, and everything works on the
	  secure channel now. At the moment, the code is disabled because i
	  have not implemented yet the autoconf code to detect the presence
	  of funopen(), and add -lssl to main/Makefile if ssl libraries are
	  present. And a bit of documentation on the http.conf arguments,
	  too. If you want to manually enable https support, that is very
	  simple (step 0 1 2 will be eventually detected by ./configure,
	  the rest is something you will have to do anyways). 0. make sure
	  your system has funopen(3). FreeBSD does, linux probably does
	  too, not sure about other systems. 1. uncomment the following
	  line in main/http.c // #define DO_SSL /* comment in/out if you
	  want to support ssl */ 2. add -lssl to AST_LIBS in main/Makefile
	  3. add the following options to http.conf sslenable=yes
	  sslbindport=4433 ; pick one you like sslcert=/tmp/foo.pem ; path
	  to your certificate file. 4. generate a suitable certificate e.g.
	  (example from mini_httpd's Makefile: openssl req -new -x509 -days
	  365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem and here you
	  go: https://localhost:4433/asterisk/manager now works.

	* main/http.c: it is useless and possibly wrong to use ast_cli() to
	  send the reply back to http clients. Use fprintf/fwrite instead,
	  since we are already using a FILE * to read the input. If you
	  wonder why, this is because it makes it trivial to implement
	  https support (as long as your system has funopen()). And this is
	  what i am going to put in with the next few commits...

2006-10-22 04:44 +0000 [r45847]  Joshua Colp <jcolp@digium.com>

	* Makefile, main/Makefile: Let's have build.h created a bit earlier
	  so that func_version can use it and not stop the build on a fresh
	  machine that has never had Asterisk installed on it before...

2006-10-21 20:24 +0000 [r45836]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: the default port number was erroneously stored in
	  host order, and reading from the config file used ntohs instead
	  of htons. this ought to be merged to 1.4 as well.

2006-10-21 18:52 +0000 [r45820]  Joshua Colp <jcolp@digium.com>

	* /, main/loader.c: Merged revisions 45817 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45817 | file | 2006-10-21 14:48:58 -0400 (Sat, 21 Oct 2006) | 2
	  lines Don't use promotion on Darwin because it doesn't seem to
	  work quite right in all cases, this should solve the unresolved
	  symbol issue people have been seeing. ........

2006-10-21 18:50 +0000 [r45819]  Russell Bryant <russell@digium.com>

	* /, res/res_monitor.c: Merged revisions 45818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45818 | russell | 2006-10-21 14:49:46 -0400 (Sat, 21 Oct 2006) |
	  3 lines Add a couple missing unregistrations of manager actions
	  and remove duplicate unregistrations of applications. (issue
	  #8194, jmls) ........

2006-10-20 20:59 +0000 [r45786]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: introduce sip_pvt_lock() and
	  sip_pvt_unlock() wrappers to lock these data structures. This
	  improve readability, and also hides the underlying locking
	  mechanism so it is a lot easier to add diagnostic code, or move
	  the object locks somewhere else, etc. On passing, rename the lock
	  field in sip_pvt to pvt_lock, also for ease of readability.

2006-10-20 19:04 +0000 [r45776]  Joshua Colp <jcolp@digium.com>

	* Makefile, /: Merged revisions 45775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45775 | file | 2006-10-20 15:03:03 -0400 (Fri, 20 Oct 2006) | 2
	  lines Pass DESTDIR and ASTSBINDIR so that the utilities get
	  installed in the proper location (reported on asterisk-dev
	  mailing list) ........

2006-10-20 15:54 +0000 [r45764]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: put the constants for whether methods can
	  create a dialog or not in an enum

2006-10-20 11:24 +0000 [r45753]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: minor comment changes, code rearrangement and
	  field renaming to minimize diffs with future modifications. The
	  current implementation is problematic for the following reasons:
	  + all insertions are O(N) because the event list does not have a
	  tail pointer; + there is only a single lock protecting both
	  session and users queues. + the implementation of the queue
	  itself is not documented. I think i have figured it out, more or
	  less, but am unclear on whether there is proper locking in place
	  The rewrite (which i have working locally) uses a tailq so
	  insertions are O(1), separate locks for the event and session
	  queues, and has a documented implementation so hopefully we can
	  figure out if/where bug exist.

2006-10-20 08:14 +0000 [r45742-45743]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Let's repair the SIP attack shield :-)

	* main/manager.c: Doxygen corrections

2006-10-19 22:06 +0000 [r45712-45724]  Steve Murphy <murf@digium.com>

	* funcs/func_version.c (added): This new function, VERSION(),
	  created via bug report 8176, may help dialplan programmers in the
	  future. In the meantime, they can use the algorithm I outline on
	  the bug report notes; If anyone invents something better, I'd
	  hope they post it

	* utils/astman.c: astman was slightly weirding out over the new
	  Dial and Newcallerid events

2006-10-19 17:26 +0000 [r45696]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: more fixes to comments and very minor code
	  rearrangement.

2006-10-19 17:25 +0000 [r45693-45695]  Joshua Colp <jcolp@digium.com>

	* /, res/res_jabber.c: Merged revisions 45694 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45694 | file | 2006-10-19 13:24:40 -0400 (Thu, 19 Oct 2006) | 2
	  lines Let's remember to unregister JabberStatus too (issue #8184
	  reported by jmls) ........

	* /, apps/app_externalivr.c: Merged revisions 45692 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45692 | file | 2006-10-19 13:19:47 -0400 (Thu,
	  19 Oct 2006) | 10 lines Merged revisions 45691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2
	  lines Respect language selection when seeing if the file exists
	  (issue #8178 reported by mnicholson) ........ ................

2006-10-19 17:07 +0000 [r45690]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: implement proper XML/HTML formatting of multiple
	  messages (e.g. the result of waitevent). Also fix some comments.

2006-10-19 16:06 +0000 [r45679]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 45678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45678 | file | 2006-10-19 12:03:09 -0400 (Thu, 19 Oct 2006) | 2
	  lines If the jitterbuffer is forced on then we can't partially
	  bridge (reported by wangster on #asterisk-dev) ........

2006-10-19 10:05 +0000 [r45648-45668]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: move a large block out of do_monitor() and
	  into a function, to improve readability.

	* channels/chan_sip.c: + move the definition of netlock as it was
	  not related to the comment just above; + decouple the struct
	  definition and variable declaration (iflist);

	* main/manager.c: more documentation of data structure and
	  functions. Of interest: + ast_get_manager_by_name_locked() is now
	  without the ast_ prefix as it is a local function; +
	  unuse_eventqent() renamed to unref_event(), and returns the
	  pointer to the next entry. + marked with XXX a couple of usages
	  of unref_event() because i suspect we are addressing the wrong
	  entry.

2006-10-19 07:17 +0000 [r45647]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Cleaning up... Removing duplicate (again)

2006-10-19 02:16 +0000 [r45634]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, include/asterisk/threadstorage.h: restore
	  freeing of threadstorage objects without custom cleanup functions
	  allow custom threadstorage init functions to return failure use a
	  custom init function for chan_sip's temp_pvt, to improve
	  performance a bit

2006-10-19 01:04 +0000 [r45623-45624]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merge fix to not leak the stringfields of
	  a thread speicif sip_pvt. This also includes the fix not to leak
	  the actual sip_pvt. Merged revisions 45622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45622 | russell | 2006-10-18 20:59:51 -0400 (Wed, 18 Oct 2006) |
	  2 lines Don't leak the actual thread-specific sip_pvt struct
	  ........

	* main/channel.c, main/frame.c, main/manager.c,
	  channels/chan_sip.c, channels/chan_skinny.c, main/logger.c,
	  main/utils.c, channels/iax2-parser.c,
	  include/asterisk/threadstorage.h, main/cli.c: Extend the thread
	  storage API such that a custom initialization function can be
	  called for each thread specific object after they are allocated.
	  Note that there was already the ability to define a custom
	  cleanup function. Also, if the custom cleanup function is used,
	  it *MUST* call free on the thread specific object at the end.
	  There is no way to have this magically done that I can think of
	  because the cleanup function registered with the pthread
	  implementation will only call the function back with a pointer to
	  the thread specific object, not the parent ast_threadstorage
	  object.

2006-10-18 22:40 +0000 [r45611]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: silent warning from a debugging message (which
	  will go away soon, anyways)

2006-10-18 22:19 +0000 [r45610]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c, CHANGES: Just for Nicholson - here's an
	  option, C, to Meetme that will allow it to continue in the
	  dialplan if the person is kicked out. (issue #7994 reported by
	  mnicholson with mods by myself)

2006-10-18 21:41 +0000 [r45597-45599]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: remove trailing whitespace

	* main/manager.c: ouch! remember to unlink temporary files once
	  done with them.

	* main/manager.c: + move output_format variables in the http
	  section of the file; + more comments on struct mansession and
	  global variables; + small improvements to the session matching
	  code so it supports multiple sessions from the same IP

2006-10-18 21:05 +0000 [r45596]  Joshua Colp <jcolp@digium.com>

	* /, main/asterisk.c: Merged revisions 45595 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45595 | file | 2006-10-18 17:03:34 -0400 (Wed, 18 Oct 2006) | 2
	  lines Don't modify things if we are using vfork as this is very
	  bad and may cause unexpected behavior (issue #7970 reported by
	  Nick Gavrikov) ........

2006-10-18 17:53 +0000 [r45572-45583]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: another bunch of comments on the data structures.

	* main/manager.c: despite the large changes, this commit only moves
	  functions around so that functions belonging to the same group
	  are close to each other. At the beginning of each group i have
	  added a bit of documentation to explain what the group does and
	  what is the typical flow - basically, all i have learned by code
	  inspection over the past few days should be documented for you to
	  read. I have not put many doxygen annotations just because i am
	  not sure what are the proper ones. Hopefully some doxygen experts
	  will jump in. Next on the plate: try to figure out how "struct
	  eventqent" are supposed to work.

	* main/manager.c: more comment and formatting fixes, small
	  simplifications to functions get_input() and session_do()

2006-10-18 16:45 +0000 [r45571]  Matt O'Gorman <mogorman@digium.com>

	* main/manager.c: rizzo compile then commit, maybe even run it too
	  ^_^

2006-10-18 15:49 +0000 [r45529-45561]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: comment and cleanup the main thread. On passing,
	  fix a bug: close the socket if the allocation of a structure for
	  the new session fails. (the bugfix is a candidate for 1.4)

	* main/manager.c: create a new (internal, for the time being)
	  function astman_start_ack() to start manager responses that need
	  further lines. This removes a lot of duplicate code from the
	  various handlers that at the moment build an ActionID string
	  themselves. Once settled, the function should move to manager.h
	  so it can be used by other files (chan_agent, chan_iax2,
	  chan_sip, chan_zap, res_jabber and app_queue). I am not totally
	  clear if there is a preferred position for the ActionID: line in
	  a message. Some instances put it at the end, but one would argue
	  that it is preferable to have it at the beginning.

	* main/manager.c: more indentation cleanup from previous commits,
	  and remove the "busy" field from struct mansession as it was not
	  used correctly anyways.

	* main/manager.c: create proper handlers for "Challenge" and
	  "Login" actions, rather than use inline code for them. Things are
	  more readable this way, and also error processing is more
	  consistent.

	* main/manager.c: fix indentation from a commit of a couple of days
	  ago

	* main/manager.c: another batch of simplifications to
	  authenticate() (they are committed a bit at a time so it is
	  easier to revert them in case we find a bug at a later time).

2006-10-18 12:15 +0000 [r45528]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Remove duplicate declarations...

2006-10-18 11:59 +0000 [r45463-45518]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c, configs/manager.conf.sample: remove unused fields
	  and unimplemented options.

	* main/manager.c: first pass as simplifying authenticate(),
	  avoiding whitespace changes

	* main/manager.c: more code simplifications

	* main/manager.c: simplify ast_strings_to_mask

	* main/manager.c: add a comment to remember that a block of code is
	  completely redundant.

	* main/manager.c: + move the enum declaration for output formats
	  near the head of the file, so it can be used from more places; +
	  make the declaration of contenttype[] more robust; + remove the
	  wrappers around __xml_translate(), since they were used only in
	  one place, and rename to xml_translate(). This allows for a bit
	  of simplifications. + document the output produced by the above
	  function.

	* main/manager.c: merge xml_translate() and html_translate() into
	  one function since they do similar things. Add a small form on
	  top of the html output so request like
	  http://foo:8088/asterisk/manager will suggest you what to do.
	  Note: i suspect there is still a bug somewhere in the session
	  matching code, as sometimes you have to login twice in order for
	  the following commands to be recognised. Apart from this, the cli
	  now is basically usable from a web form!

	* main/http.c: introduce uri_decode() so that '+' are translated
	  into ' ' (e.g. browsers do this when they encode input strings
	  from a form).

	* main/http.c: various code simplifications to reduce nesting
	  depth, minor optimizations to avoid extra calls of strlen(), and
	  some variable localization. One feature worth backporting is the
	  move of ast_variables_destroy() to a different place in
	  handle_uri() to avoid leaking memory in case a uri is not found.

2006-10-18 03:03 +0000 [r45453]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 45452 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2
	  lines Don't segfault if you're using a channel driver that
	  doesn't turn RTCP on ........

2006-10-18 02:46 +0000 [r45440-45442]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 45441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) |
	  7 lines Don't attempt to access private data members of the
	  pthread_mutex_t object, because this does not work on all linux
	  systems. Instead, just access the reentrancy field in the
	  ast_mutex_info struct when DEBUG_THREADS is enabled. If
	  DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
	  DEBUG_THREADS on as well. (issue #8139, me) ........

	* configs/sip_notify.conf.sample, /: Merged revisions 45439 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) |
	  2 lines update entry to reboot a snom phone (issue #7850,
	  pnlarsson) ........

2006-10-17 23:06 +0000 [r45426]  Steve Murphy <murf@digium.com>

	* res/res_agi.c: As per bug 6779, this patch is now applied to
	  trunk; while I was at it, I corrected a reference to a CLI
	  command, to follow the new regime.

2006-10-17 22:32 +0000 [r45409-45411]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/prep_tarball (added): Merged revisions 45410 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45410 | kpfleming | 2006-10-17 17:31:54 -0500 (Tue, 17 Oct 2006)
	  | 2 lines add a project-specific script to be used during release
	  preparation ........

	* main/channel.c, /, channels/chan_sip.c, channels/chan_iax2.c,
	  include/asterisk/stringfields.h, main/ast_expr2.c: Merged
	  revisions 45408 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006)
	  | 3 lines optimize the 'quick response' code a bit more... no
	  more malloc() or memset() for each response expand stringfields
	  API a bit to allow reusing the stringfield pool on a structure
	  when needed, and remove some unnecessary code when the structure
	  was being freed ........

2006-10-17 21:09 +0000 [r45379-45398]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Warning be gone!

	* /, channels/chan_sip.c: Merged revisions 45378 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2
	  lines Don't create a "real" pvt structure for requests that
	  shouldn't be able to create one. Instead use a temporary pvt and
	  fill it with enough information so we can send a reply. ........

2006-10-17 19:57 +0000 [r45365]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, doc/channelvariables.txt: Issue #5484
	  (branch sipdiversion) - Support for Diversion header in redirects
	  of calls with 302 redirection. (tinning)

2006-10-17 18:08 +0000 [r45351]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: simplify authority_to_str() using
	  ast_build_string()

2006-10-17 17:54 +0000 [r45335]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7254 - Add support of "423 Interval
	  too brief" to outbound SIP registrations. Thanks, tardieu!

2006-10-17 17:51 +0000 [r45334]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: Improve the XML formatting of responses coming
	  from web interface. Normal responses are sequences of lines of
	  the form "Name: value", with \r\n as line terminators and an
	  empty line as a response terminator. Generi CLI commands,
	  however, do not have such a clean formatting, and the existing
	  code failed to generate valid XML for them. Obviously we can only
	  use heuristics here, and we do the following: - accept either \r
	  or \n as a line terminator, trimming trailing whitespace; - if a
	  line does not have a ":" in it, assume that from this point on we
	  have unformatted data, and use "Opaque-data:" as a name; - if a
	  line does have a ":" in it, the Name field is not always a legal
	  identifier, so replace non-alphanum characters with underscores;
	  All the above is to be refined as we improve the formatting of
	  responses from the CLI. And, all the above ought to go as a
	  comment in the code rather than just in a commit message...

2006-10-17 17:51 +0000 [r45331-45333]  Olle Johansson <oej@edvina.net>

	* /, configs/sip.conf.sample: Update of docs

	* channels/chan_sip.c: - Remove unneeded code that won't be reached
	  now that we kill responses to unkonwn dialogs earlier in the
	  process. - move debug message.

2006-10-17 17:41 +0000 [r45330]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: open a temporary file to receive the output from
	  cli commands invoked through the http interface. It is not
	  terribly efficient but better than no output at all. Todo: use a
	  configurable /tmp directory instead of a hardwired one.

2006-10-17 17:22 +0000 [r45328]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: Merged revisions 45327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45327 | kpfleming | 2006-10-17 12:22:25 -0500
	  (Tue, 17 Oct 2006) | 10 lines Merged revisions 45326 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45326 | kpfleming | 2006-10-17 12:22:01 -0500 (Tue, 17
	  Oct 2006) | 2 lines provide licensing language for IAXy firmware
	  file ........ ................

2006-10-17 17:19 +0000 [r45325]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: document xml_copy_escape() and add an extra
	  function, namely replace non-alphanum chars with underscore. This
	  is useful when building field names in xml formatting.

2006-10-17 16:27 +0000 [r45295-45316]  Olle Johansson <oej@edvina.net>

	* /: ...block this one too... Only applies to 1.4 since the fix for
	  trunk was different.

	* /: Block patch from 1.4 that does not apply here.

	* channels/chan_sip.c: Get rid of the ignore variable that was only
	  partially replaced by the flag.

2006-10-16 20:26 +0000 [r45234-45286]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: In the course of a
	  data this has been turned into an option to ignore replies, then
	  ignore responses and finally I'm just getting rid of the option
	  altogether and making it the default no matter what. C'est la
	  vie!

	* /: Woof.

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  45280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45280 | file | 2006-10-16 16:06:18 -0400 (Mon,
	  16 Oct 2006) | 10 lines Merged revisions 45265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2
	  lines Use responses rather then replies even though they mean the
	  same thing. ........ ................

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  45262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon,
	  16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2
	  lines Add 'ignoreoodreplies' option which will not create a pvt
	  structure on a SIP response but instead basically drop it.
	  ........ ................

	* apps/app_directed_pickup.c: It's new directed pickup! This now
	  features a more sane way of finding the channel to pick up (I
	  snuck it into the tree on Friday... bet you didn't know I'd
	  actually use it eh?). PICKUPMARK now also works in a different
	  way, you should prefix it with _ when setting it so it gets
	  inherited onto the channel(s) created in app_dial as directed
	  pickup will now look for it on the target channel, not the
	  originating channel. (BE-85)

2006-10-16 14:03 +0000 [r45224]  Olle Johansson <oej@edvina.net>

	* CREDITS, /: Update

2006-10-16 14:00 +0000 [r45219]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: + comment some unclear fields of struct
	  mansession; + let some commands (Challenge, Login) be processed
	  even if already authenticated, as it doesn't harm and prevents
	  some incorrect error messages + remove custom code for Logoff -
	  the existing handler was ok. Some indentation fixes may be
	  necessary

2006-10-16 13:20 +0000 [r45194-45209]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: When adding new functions, please add a
	  forward declaration. I *know* it is not required, but it makes
	  navigation easier and will help when splitting up this large
	  source code file. Thank you!

	* /, channels/chan_sip.c: Importing rev 45196 from 1.4 - don't kill
	  dialog for a bad response

	* channels/chan_sip.c: A B2BUA should *not* issue proxy auth.

2006-10-16 11:29 +0000 [r45151-45185]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: + comment some unclear requirements for
	  master_eventq + remove the need for an snprintf in
	  astman_get_header() + fix comment for manager list eventq +
	  localize one variable and minor code simplifications.

	* main/manager.c: protect access to first_action with actionlock.
	  Mark with XXX one place (during command execution) where
	  navigation should be protected with actionlock, but is not
	  because it would block requests for a long time. To solve this
	  properly we need to put reference counts in the struct
	  manager_action. A suboptimal fix is to copy the record on a
	  search and then unlock the list while we work on the copy.

	* main/http.c: comment some functions, and more small code
	  simplifications

	* main/http.c: fix indentation of a large block after changes in
	  previous commit (basically whitespace only).

	* main/http.c: simplify string parsing routines using ast_skip_*()
	  functions.

	* main/http.c: don't forget to close a descriptor on a malloc
	  failure. On passing, small rearrangement of the code to reduce
	  indentation. There is a bit more cleanup planned for this file,
	  so a merge to 1.4 will be done when it is all done.

	* main/http.c: typo: serer -> server

2006-10-14 04:36 +0000 [r45142]  Steve Murphy <murf@digium.com>

	* funcs/func_rand.c: update the doc string for both AEL and
	  extensions.conf users.

2006-10-13 23:03 +0000 [r45126]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/acl.c: Merged revisions 45125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006)
	  | 7 lines
	  ------------------------------------------------------------------------
	  r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006)
	  | 2 lines don't drop the entire permit/deny list when an attempt
	  is made to add an invalid entry (BE-92)
	  ------------------------------------------------------------------------
	  ........

2006-10-13 21:20 +0000 [r45105-45109]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Inherit the context and extension until the
	  channel is answered

	* /, res/res_speech.c: Merged revisions 45106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45106 | file | 2006-10-13 17:06:09 -0400 (Fri, 13 Oct 2006) | 2
	  lines Clear the quiet flag too since we are restarting a
	  recognition again (reported on -dev by Stephan Edelman) ........

	* /, res/res_speech.c: Merged revisions 45104 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45104 | file | 2006-10-13 17:01:13 -0400 (Fri, 13 Oct 2006) | 2
	  lines Check return value from engine in case of failure (ie: out
	  of licenses) (reported on -dev mailing list) ........

2006-10-13 19:24 +0000 [r45089]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 45088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r45088 | crichter | 2006-10-13 21:19:46 +0200 (Fr, 13
	  Okt 2006) | 1 line avoiding warning, fixing potential bug
	  ........

2006-10-13 18:45 +0000 [r45080]  Joshua Colp <jcolp@digium.com>

	* codecs/lpc10/median.c, codecs/lpc10/encode.c,
	  codecs/lpc10/ivfilt.c, /, codecs/lpc10/bsynz.c,
	  codecs/lpc10/prepro.c, codecs/lpc10/invert.c,
	  codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
	  codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
	  codecs/lpc10/pitsyn.c, codecs/lpc10/difmag.c,
	  codecs/lpc10/voicin.c, codecs/lpc10/synths.c,
	  codecs/lpc10/preemp.c, codecs/lpc10/hp100.c,
	  codecs/lpc10/lpfilt.c, codecs/lpc10/rcchk.c,
	  codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
	  codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
	  codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
	  codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
	  codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
	  codecs/lpc10/analys.c, codecs/lpc10/onset.c,
	  codecs/lpc10/energy.c, codecs/lpc10/lpcdec.c,
	  codecs/lpc10/deemp.c: Merged revisions 45079 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45079 | file | 2006-10-13 14:42:49 -0400 (Fri, 13 Oct 2006) | 2
	  lines And file said... let the compiler warnings STOP! ........

2006-10-13 18:08 +0000 [r45078]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest17 (added),
	  pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest17 (added),
	  pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Correction for bug
	  8128 in trunk

2006-10-13 17:06 +0000 [r45052-45067]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 45066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45066 | file | 2006-10-13 13:05:02 -0400 (Fri,
	  13 Oct 2006) | 10 lines Merged revisions 45060 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45060 | file | 2006-10-13 13:01:22 -0400 (Fri, 13 Oct 2006) | 2
	  lines Turn on volume adjustment if it needs to be on (issue #8136
	  reported by mnicholson) ........ ................

	* /, apps/app_playback.c: Merged revisions 45051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45051 | file | 2006-10-13 12:20:58 -0400 (Fri, 13 Oct 2006) | 2
	  lines Move say.conf existence check to do_say function since it
	  is called from multiple places (issue #8144 reported by kshumard)
	  ........

2006-10-13 16:20 +0000 [r45050]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 45049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45049 | kpfleming | 2006-10-13 11:19:35 -0500
	  (Fri, 13 Oct 2006) | 10 lines Merged revisions 45048 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45048 | kpfleming | 2006-10-13 11:18:08 -0500 (Fri, 13
	  Oct 2006) | 2 lines when sending a call to a peer, use the proper
	  socket if we have multiple bindings (reported on asterisk-dev)
	  ........ ................

2006-10-13 16:02 +0000 [r45032-45047]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 45040 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45040 | file | 2006-10-13 12:01:17 -0400 (Fri, 13 Oct 2006) | 2
	  lines Complete merging in RPID screen changes (issue #8101
	  reported by hristo, patch by oej in revision 44757) ........

	* main/dnsmgr.c, /: Merged revisions 45031 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45031 | file | 2006-10-13 11:53:22 -0400 (Fri,
	  13 Oct 2006) | 10 lines Merged revisions 45030 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45030 | file | 2006-10-13 11:49:53 -0400 (Fri, 13 Oct 2006) | 2
	  lines Pass the right value to usleep for sleeping, and always add
	  the background refresh item back into the scheduler if enabled
	  since it is deleted during reload. (issue #8142 reported by
	  p_lindheimer) ........ ................

2006-10-13 15:47 +0000 [r45029]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: Merged revisions 45027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r45027 | kpfleming | 2006-10-13 10:41:14 -0500 (Fri, 13 Oct 2006)
	  | 2 lines use a configure script test for PMTU discovery control
	  instead of just assuming it's available on Linux ........

2006-10-13 15:42 +0000 [r45028]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 45026 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r45026 | crichter | 2006-10-13 16:45:39 +0200
	  (Fr, 13 Okt 2006) | 9 lines Merged revisions 45020 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45020 | crichter | 2006-10-13 15:11:13 +0200 (Fr, 13
	  Okt 2006) | 1 line fixed some echocandisable issues when bridged.
	  this caused a kernel panic sometimes..also some minor formatting
	  fixes ........ ................

2006-10-13 11:18 +0000 [r45009-45010]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: Try to avoid the use of 'z' modifier in
	  cases where it is not necessary - rather, cast the argument to
	  int. In this case, the string is in a UDP packet and as such
	  limited to 64k so its length can be safely represented in an int
	  without truncation (besides, this is just a debugging message!)

	* channels/chan_sip.c: arguments to auth_headers() needed to be
	  swapped here. To avoid the same mistake in the future (due to
	  slightly confusing variable names), add a comment. On passing,
	  remove a redundant initialization.

2006-10-13 08:23 +0000 [r45000]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/isdn_msg_parser.c: Merged revisions 44994 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44994 | crichter | 2006-10-13 09:52:41 +0200
	  (Fr, 13 Okt 2006) | 9 lines Merged revisions 44993 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44993 | crichter | 2006-10-13 09:40:07 +0200 (Fr, 13
	  Okt 2006) | 1 line fixed issue, that the hangupcause got a wrong
	  isdn cause at RELEASE_COMPLETE ........ ................

2006-10-12 20:41 +0000 [r44983]  Matt O'Gorman <mogorman@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 44982 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.4 ........
	  r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006)
	  | 2 lines fix for bug 7764. ........

2006-10-12 19:16 +0000 [r44957-44973]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: eliminate compiler warning

	* /, channels/chan_sip.c: Merged revisions 44971 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44971 | kpfleming | 2006-10-12 14:14:24 -0500 (Thu, 12 Oct 2006)
	  | 2 lines we can only send one 'a=ptime' attribute per media
	  session, not one for each format ........

	* include/asterisk/utils.h, /, channels/chan_sip.c, main/utils.c,
	  main/netsock.c: Merged revisions 44956 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44956 | kpfleming | 2006-10-12 13:38:51 -0500
	  (Thu, 12 Oct 2006) | 10 lines Merged revisions 44955 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12
	  Oct 2006) | 2 lines ensure that IAX2 and SIP sockets allow UDP
	  fragmentation when running on Linux (thanks to Brian Candler on
	  the asterisk-dev list for the tip) ........ ................

2006-10-12 16:57 +0000 [r44944-44946]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 44945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44945 | russell | 2006-10-12 12:56:32 -0400 (Thu, 12 Oct 2006) |
	  2 lines fix a silly typo in a comment that I saw while reading
	  the commit list ........

	* pbx/pbx_dundi.c: put flags in an enum and remove a couple of
	  unused defines

2006-10-12 16:11 +0000 [r44943]  Joshua Colp <jcolp@digium.com>

	* Makefile, /: Merged revisions 44942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44942 | file | 2006-10-12 12:08:50 -0400 (Thu, 12 Oct 2006) | 2
	  lines Pass off AUDIO_LIBS so muted can link on OSX (issue #8135
	  reported by ssokol) ........

2006-10-12 15:12 +0000 [r44933]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: + move [almost] all instances of
	  WWW-Authenticate/Proxy-Authenticate and friends in a function,
	  auth_headers(), which is used to simplify the interface of
	  do_{proxy|register}_auth(). + use PROXY_AUTH = 407, WWW_AUTH =
	  401 as values for enum sip_auth_type; No functional change, only
	  code cleanup.

2006-10-12 13:04 +0000 [r44922]  Nadi Sarrar <ns@beronet.com>

	* main/manager.c, /: Merged revisions 44921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44921 | nadi | 2006-10-12 14:55:25 +0200 (Do, 12 Okt 2006) | 2
	  lines append_event must be called while holding the session lock
	  ........

2006-10-12 10:26 +0000 [r44912]  Russell Bryant <russell@digium.com>

	* /, res/res_jabber.c: Merged revisions 44911 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44911 | russell | 2006-10-12 06:24:36 -0400 (Thu, 12 Oct 2006) |
	  2 lines change some debug output to use LOG_DEBUG instead of
	  verbose output ........

2006-10-11 23:36 +0000 [r44900-44901]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: reduce indentation of two large blocks

	* channels/chan_sip.c: operator != also works between booleans...

2006-10-11 16:57 +0000 [r44889]  Jason Parker <jparker@digium.com>

	* /, main/db1-ast/Makefile: Merged revisions 44888 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r44888 | qwell | 2006-10-11 11:57:06 -0500 (Wed, 11 Oct
	  2006) | 3 lines These are already set by the parent Makefile..
	  There is no need to have this here (it doesn't actually work
	  anyways). ........

2006-10-11 13:45 +0000 [r44876-44877]  Russell Bryant <russell@digium.com>

	* doc/linkedlists.txt (removed): Remove doc/linkedlists.txt as it
	  is no longer needed. The top of the file reads: As of 2004-12-23,
	  this documentation is no longer maintained. The doxygen
	  documentation generated from linkedlists.h should be referred to
	  in its place, as it is more complete and better maintained.

	* channels/chan_sip.c: Revert Luigi's accidental commit of his
	  local changes when debugging some SIP authentication issues. This
	  was committed in revision 44844, where the commit message was
	  just "small formatting cleanup", so I am pretty sure he didn't
	  mean to commit this part.

2006-10-11 13:21 +0000 [r44844-44875]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: remove duplicate prototypes. Have not
	  checked if there are more.

	* channels/chan_sip.c: simplify and comment
	  handle_response_peerpoke()

	* channels/chan_sip.c: fix indentation of a function after previous
	  commit (whitespace-only change)

	* channels/chan_sip.c: handle_response_peerpoke() does not need to
	  return anything. (Reindentation in the next commit.)

	* channels/chan_sip.c: small formatting cleanup

2006-10-11 08:45 +0000 [r44840-44843]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 44563 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44563 | crichter | 2006-10-06 14:53:41 +0200
	  (Fr, 06 Okt 2006) | 9 lines Merged revisions 44460 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44460 | crichter | 2006-10-05 12:02:38 +0200 (Do, 05
	  Okt 2006) | 1 line fixed segfault which happens during
	  hold/transfer action ........ ................

	* channels/chan_misdn.c, /: Merged revisions 44562 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44562 | crichter | 2006-10-06 14:52:01 +0200
	  (Fr, 06 Okt 2006) | 9 lines Merged revisions 44335 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44335 | crichter | 2006-10-04 17:26:59 +0200 (Mi, 04
	  Okt 2006) | 1 line if INFORMATION Message come with keypad
	  instead of called party number, we just use the keypad as called
	  party number. ........ ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
	  revisions 44561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44561 | crichter | 2006-10-06 14:50:25 +0200
	  (Fr, 06 Okt 2006) | 9 lines Merged revisions 44334 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04
	  Okt 2006) | 1 line added the option 'reject_cause' to make it
	  possible to set the RELEASE_COMPLETE - cause on the 3. incoming
	  PMP channel, which is automatically rejected because chan_misdn
	  does not support that kind of callwaiting. Therefore chan_misdn
	  supports now 3 incoming channels on a PMP BRI Port.
	  misdn_lib_get_free_bc now gets the info if the requested channel
	  is incoming or outgoing to make the 3. channel possible ........
	  ................

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions 44559 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44559 | crichter | 2006-10-06 12:44:34 +0200
	  (Fr, 06 Okt 2006) | 9 lines Merged revisions 44149 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44149 | crichter | 2006-10-02 15:28:14 +0200 (Mo, 02
	  Okt 2006) | 1 line fixed the hold/retrieve/transfer issues,
	  removed a useless bc field, added setting of frame.delivery
	  fields, some minor code cleanups ........ ................

2006-10-10 20:52 +0000 [r44831]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_rpt.c: More whitespace fixes

2006-10-10 17:23 +0000 [r44820]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44819 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44819 | file | 2006-10-10 13:21:44 -0400 (Tue, 10 Oct 2006) | 2
	  lines Move some stuff around so that a NOTIFY dialog won't hang
	  around until the end of the world under certain circumstances
	  ........

2006-10-10 16:46 +0000 [r44810]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_logic.c: Merged revisions 44808 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44808 | tilghman | 2006-10-10 11:42:19 -0500 (Tue, 10 Oct 2006)
	  | 2 lines Lost of a bit of logic when this was simplified between
	  1.2 and 1.4 (Bug 8117) ........

2006-10-10 16:31 +0000 [r44789-44807]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44806 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44806 | file | 2006-10-10 12:30:00 -0400 (Tue, 10 Oct 2006) | 2
	  lines Bail out if we have no refer structure and we get a refer
	  response ........

	* /, channels/chan_sip.c: Merged revisions 44788 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44788 | file | 2006-10-10 11:23:14 -0400 (Tue, 10 Oct 2006) | 2
	  lines Only set DTMF information if an RTP structure exists
	  ........

2006-10-10 14:54 +0000 [r44787]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 44786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44786 | crichter | 2006-10-10 15:50:26 +0200
	  (Di, 10 Okt 2006) | 9 lines Merged revisions 44785 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44785 | crichter | 2006-10-10 15:34:33 +0200 (Di, 10
	  Okt 2006) | 1 line (re)added support of dynamical enabling hdlc
	  on bchannels ........ ................

2006-10-10 08:08 +0000 [r44770-44774]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: clarify the use of the standard SIP port
	  number, 5060, and rename the old DEFAULT_SIP_PORT as
	  STANDARD_SIP_PORT to make it clear that this is not something we
	  can change, unlike other defaults.

	* channels/chan_sip.c: improve formatting of SIP packets when
	  dumped to the verbose output stream, so it is easier to find them
	  in the log.

2006-10-09 18:23 +0000 [r44768]  Joshua Colp <jcolp@digium.com>

	* funcs/func_timeout.c: Timeout values are in seconds (issue #7122
	  reported by jmls)

2006-10-09 16:15 +0000 [r44765]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 44764 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct
	  2006) | 4 lines Fix a problem where phones that go "missing"
	  never got unregistered. Issue #8067, reported by pj, patch by
	  Anthony LaMantia (with minor whitespace modifications) ........

2006-10-09 15:52 +0000 [r44762-44763]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 44759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44759 | file | 2006-10-09 11:41:28 -0400 (Mon, 09 Oct 2006) | 2
	  lines Properly avoid a collision with iax2_hangup (issue #8115
	  reported by vazir) ........

2006-10-09 11:20 +0000 [r44753]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Being pedantic... "media" is easier to
	  understand than "data" in the function name... :-)

2006-10-09 09:04 +0000 [r44745-44752]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: slightly restructure sipsock_read() removing
	  a "goto"

	* channels/chan_sip.c: use S_OR in one place

	* channels/chan_sip.c: update_call_counter(): indentation fixes and
	  small simplifications at the top of the function.

	* channels/chan_sip.c: localize some variables and reduce nesting
	  depth (indentation will be fixed by a separate commit).

	* channels/chan_sip.c: small simplification to initreqprep()

	* channels/chan_sip.c: Simplify function parse_request() using a
	  single loop instead of two very similar ones.

	* channels/chan_sip.c: do not dereference p if we know it is NULL.

2006-10-07 20:42 +0000 [r44697-44731]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fix some debug output for setsockopt for TOS

	* channels/chan_sip.c: - move definition of global_autoframing to
	  the same place as other globals - set initial value at
	  load/reload - Add questionmarks for someone to fill in for
	  doxygen

	* channels/chan_sip.c: Add/change doxygen and comments

	* configs/sip.conf.sample: Recommend using "sip reload" since it's
	  much easier to learn and remember.

	* channels/chan_sip.c: Explain usage of DEFAULT_SIP_PORT

	* channels/chan_sip.c: Do *NOT* use DEFAULT_SIP_PORT in these
	  comparisions, since users may change that, but the protocol
	  clearly states that if we DO NOT mention a port it is 5060.
	  DEFAULT_SIP_PORT is whatever we default to listen to. I believe
	  it's the third time I revert a patch like this.

2006-10-07 14:48 +0000 [r44685-44686]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged
	  revisions 44684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44684 | pcadach | 2006-10-07 20:39:34 +0600 (Сбт, 07 Окт 2006) |
	  1 line Propagate caller's transfer capability too ........

	* include/asterisk/callerid.h, main/callerid.c, CHANGES,
	  funcs/func_callerid.c: Extend CALLERID() function for "pres" and
	  "ton" values

2006-10-07 12:50 +0000 [r44641-44675]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: slightly restructure the code that computes
	  the channel's name

	* channels/chan_sip.c: put repeated code to set nat mode in a
	  function.

	* channels/chan_sip.c: put common code in a function to avoid
	  repetitions.

	* channels/chan_sip.c: remove hardwired usage of 5060, use
	  DEFAULT_SIP_PORT instead

	* channels/chan_sip.c: improve and document function
	  get_in_brackets(), introducing a helper function
	  find_closing_quote() of more general use.

	* channels/chan_sip.c: when possible, use ast_set2_flags instead of
	  ast_set/ast_clr . Also mark XXX some dubious places.

2006-10-06 21:29 +0000 [r44632]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 44631 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44631 | kpfleming | 2006-10-06 16:28:03 -0500 (Fri, 06 Oct 2006)
	  | 2 lines ensure that mutex locks inside list heads are
	  initialized properly on platforms that require constructor
	  initialization (issue #8029, patch from timrobbins) ........

2006-10-06 21:10 +0000 [r44630]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 44628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2
	  lines Remove the seqno check for RFC2833, the handler is smart
	  enough to not need it. ........

2006-10-06 21:04 +0000 [r44616-44626]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: basically fix indentation of a large function
	  after previous simplifications. On passing, use a single exit
	  point. (once done with the cleanup i will merge the changes into
	  1.4, if applicable)

	* main/manager.c: s cannot be null here, so remove the useless test
	  and error-handling block.

	* main/manager.c: simplify logic in preparation to reduce
	  indentation

2006-10-06 18:47 +0000 [r44606]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 44605 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2
	  lines When the sequence number rolls over then reset the recorded
	  sequence number for DTMF (issue #8106 reported by bungalow)
	  ........

2006-10-06 17:27 +0000 [r44595]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_rpt.c: Massive cleanup of the rpt code, updating to
	  current coding guidelines

2006-10-06 16:56 +0000 [r44582]  Joshua Colp <jcolp@digium.com>

	* /, main/file.c: Merged revisions 44581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44581 | file | 2006-10-06 12:53:48 -0400 (Fri,
	  06 Oct 2006) | 10 lines Merged revisions 44580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r44580 | file | 2006-10-06 12:52:14 -0400 (Fri, 06 Oct 2006) | 2
	  lines Even more frames to treat as though the remote side
	  disappeared (issue #8097 reported by eldadran) ........
	  ................

2006-10-06 16:43 +0000 [r44566-44579]  Luigi Rizzo <rizzo@icir.org>

	* configs/sip.conf.sample: document a bit the use of templates.
	  They are highly convenient for writing configuration files,
	  especially if you have many similar entries, or want to switch
	  quickly between different configurations without having to
	  comment/uncomment large sections of the files.

	* configs/sip.conf.sample: document the "contact" option a bit
	  better.

	* res/res_limit.c: help old bsd-system which don't have RLIMIT_AS
	  and use RLIMIT_VMEM for virtual memory limits.

	* main/manager.c, main/http.c: make sure sockets are blocking when
	  they should be blocking.

	* channels/chan_sip.c, configs/sip.conf.sample: Two things: 1.
	  slightly rearrange/simplify the parsing of the argument in
	  sip_register. This brings in a patch that has been in Mantis
	  (5834) for ages, and is the larger part of the commit; 2.
	  implement the "contact" option for peers, similar to the one in
	  users.conf: If you put a "contact" option with a non-empty
	  argument (e.g. contact=123) in a peer section, asterisk will
	  register with the provider as if you had a register=
	  username:secret@host/contact line in the general section. The
	  latter is a very small is a new feature so i am not putting it in
	  the 1.4 branch, although the "contact" option in user.conf is
	  already in the 1.4 branch and so it wouldn't be too strange to
	  merge it. Note that the implementation of "contact" is much
	  simpler than the one in 5834, and limited to a few lines in
	  build_peer().

2006-10-06 09:01 +0000 [r44545]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Remove deprecated "incominglimit" config
	  option

2006-10-06 06:43 +0000 [r44537]  Luigi Rizzo <rizzo@icir.org>

	* configs/sip.conf.sample: update example commands to match current
	  syntax (does not apply to 1.4)

2006-10-06 02:24 +0000 [r44527]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in: regenerate the
	  configure script to reflect the latest changes done by Luigi
	  Rizzo

2006-10-05 20:13 +0000 [r44503-44516]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Fix indenting a bit (issue #8082 reported
	  by selsky)

	* /, main/file.c: Merged revisions 44502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44502 | file | 2006-10-05 15:57:16 -0400 (Thu,
	  05 Oct 2006) | 10 lines Merged revisions 44501 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r44501 | file | 2006-10-05 15:55:41 -0400 (Thu, 05 Oct 2006) | 2
	  lines Treat busy control frames as hangup in the file streaming
	  core (issue #8097 reported by eldadran) ........ ................

2006-10-05 18:29 +0000 [r44489]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: These mods fix a problem pointed out by dgartang,
	  where in certain situations, the target of a goto cannot be
	  found, even right under your nose. This is because the current
	  context is not updated properly, and rather than waste time and
	  find why and where the context should have been updated, I just
	  use my newly added 'dad' ptrs, and pop until I have either the
	  context or extension, and use that instead.

2006-10-05 18:03 +0000 [r44487]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44486 | file | 2006-10-05 14:01:51 -0400 (Thu, 05 Oct 2006) | 2
	  lines One more T.38 fix! Don't leave a reinvite hanging by a
	  thread if the other side is already setup with T.38 ........

2006-10-05 16:11 +0000 [r44477]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/app.c: Merged revisions 44476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44476 | kpfleming | 2006-10-05 11:10:01 -0500 (Thu, 05 Oct 2006)
	  | 3 lines don't segfault when an argument without a close
	  parenthesis is found stop parsing as soon as that situation
	  occurs ........

2006-10-05 15:42 +0000 [r44467]  Luigi Rizzo <rizzo@icir.org>

	* configure.ac, acinclude.m4: Basically, this commit only
	  simplifies configure.ac and makes the mechanism more flexible,
	  but otherwise should not affect your build even if you regenerate
	  the "configure" script. (Most likely you need to run bootstrap.sh
	  as you really need to re-run autoheader for reasons that i do not
	  completely understand). If you don't regenerate "configure", of
	  course you will see no difference. In detail: - restructure the
	  check for mandatory modules to remove some redundant code blocks;
	  - extend the AST_EXT_LIB_CHECK so that it can used also for
	  checking headers; - define the AST_C_DEFINE_CHECK macro to test
	  for #defined symbols; - for the two above macros, add a last
	  argument that getscopied into HAVE_$1_VERSION so the source can
	  adapt to different versions of the same libraries/header/etc -
	  document the above; - document a problem that existed before and
	  i did not manage to solve: the 'description' argument to
	  AC_DEFINE does not substiture shell variables so you will not see
	  the actual values in the comments (in autoconfig.h)..

2006-10-05 02:43 +0000 [r44451]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2
	  lines Don't totally bail out if T.38 was negotiated ........

2006-10-05 01:43 +0000 [r44437]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/Makefile, /: Merged revisions 44436 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44436 | kpfleming | 2006-10-04 20:42:06 -0500 (Wed, 04 Oct 2006)
	  | 2 lines this change was correct, the old version is no longer
	  needed ........

2006-10-05 01:40 +0000 [r44435]  Steve Murphy <murf@digium.com>

	* main/pbx.c, apps/app_read.c, apps/app_waitforring.c, CHANGES,
	  apps/app_speech_utils.c: As per ToDo list, I have made it so that
	  Wait(), WaitExten(), Congestion(), Busy(), Read(), WaitForRing(),
	  will now either actually handle a floating point argument as
	  advertised, or has been upgraded to accept a floating point
	  [timeout] arg.

2006-10-05 01:30 +0000 [r44434]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44433 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44433 | kpfleming | 2006-10-04 20:30:05 -0500
	  (Wed, 04 Oct 2006) | 10 lines Merged revisions 44432 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04
	  Oct 2006) | 2 lines fix Polycom presence notification again
	  ........ ................

2006-10-04 23:52 +0000 [r44408-44423]  Luigi Rizzo <rizzo@icir.org>

	* configure.ac: simplify checks for OSS using AST_EXT_LIB_CHECK;
	  remove two repeated blocks using better logic.

	* acinclude.m4: small formatting fix

	* acinclude.m4: when only checking headers, do not set $1_LIB. Also
	  PBX_$1=0 is the default so we don't need to set it explicitly.

	* acinclude.m4: document, and extend a bit the macro
	  AST_EXT_LIB_CHECK so that it can be used in more places in
	  configure.ac

	* configure.ac: restore proper CPPFLAGS and LDFLAGS for FreeBSD,
	  until a better solution is found. Please do not commit the
	  regenerated "configure" file yet, as there are some more
	  simplifications to be applied to configure.ac and acinclude.m4 in
	  the next few days. For the same reason, i am postponing the
	  commit to the 1.4 branch until the above changes are complete.

	* utils/Makefile: correct libraries for astman, at least so i
	  think...

	* Makefile: put linker flags in ASTLDFLAGS where they belong

2006-10-04 21:20 +0000 [r44379-44394]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44393 | kpfleming | 2006-10-04 16:17:30 -0500
	  (Wed, 04 Oct 2006) | 11 lines Merged revisions 44392 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04
	  Oct 2006) | 3 lines remove workaround for old Polycom firmware
	  SUBSCRIBE requests add workaround for new Polycom firmware
	  SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)
	  ........ ................

	* include/asterisk.h, /, main/utils.c: Merged revisions 44390 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44390 | kpfleming | 2006-10-04 16:04:21 -0500 (Wed, 04 Oct 2006)
	  | 2 lines make LOW_MEMORY builds actually work ........

	* include/asterisk/utils.h, main/autoservice.c, main/dnsmgr.c,
	  channels/chan_zap.c, res/res_snmp.c, /, apps/app_meetme.c,
	  channels/chan_sip.c, main/utils.c, main/devicestate.c,
	  res/res_musiconhold.c, res/res_jabber.c, apps/app_queue.c,
	  channels/chan_iax2.c, channels/chan_oss.c, main/cdr.c,
	  channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c,
	  res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c,
	  main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c,
	  apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c:
	  Merged revisions 44378 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006)
	  | 4 lines update thread creation code a bit reduce standard
	  thread stack size slightly to allow the pthreads library to
	  allocate the stack+data and not overflow a power-of-2 allocation
	  in the kernel and waste memory/address space add a new stack size
	  for 'background' threads (those that don't handle PBX calls) when
	  LOW_MEMORY is defined ........

2006-10-04 19:33 +0000 [r44336-44377]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.tab.c,
	  pbx/ael/ael.y, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test16
	  (added), pbx/ael/ael-test/ael-test16/extensions.ael: These
	  changes resolve the problems in bug 8090, where there's a crash
	  compiling an empty context

	* configs/muted.conf.sample: I've been meaning to add some
	  explanation about muted... here it is

	* configs/manager.conf.sample: CLI reverbification update to this
	  config file

	* apps/app_macro.c: Added a warning to the documentation for Macro
	  in response to bug 7776

2006-10-04 00:26 +0000 [r44323]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, include/asterisk.h, /, main/asterisk.c, main/loader.c,
	  main/term.c: Merged revisions 44322 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44322 | kpfleming | 2006-10-03 19:25:44 -0500 (Tue, 03 Oct 2006)
	  | 3 lines ensure that local include files are always used avoid a
	  duplicate function name (term_init()) ........

2006-10-03 22:36 +0000 [r44313]  Matt O'Gorman <mogorman@digium.com>

	* /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions
	  44312 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44312
	  | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2
	  lines fix issue with dialing client without resource. ........

2006-10-03 20:19 +0000 [r44299]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_queue.c: Merged revisions 44298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44298 | kpfleming | 2006-10-03 15:18:29 -0500
	  (Tue, 03 Oct 2006) | 10 lines Merged revisions 44296 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44296 | kpfleming | 2006-10-03 15:14:13 -0500 (Tue, 03
	  Oct 2006) | 2 lines fix a logic error in my previous fix to the
	  queue reload code ........ ................

2006-10-03 20:17 +0000 [r44297]  Joshua Colp <jcolp@digium.com>

	* CHANGES, apps/app_queue.c: Strat becomes Strategy based on
	  feedback from two nameless fellows

2006-10-03 18:47 +0000 [r44287]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx: Merged revisions 44283,44286 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44283 | pcadach | 2006-10-04 00:30:48 +0600 (Срд, 04 Окт 2006) |
	  1 line Fix preparation of type and presentation of calling number
	  ........ r44286 | pcadach | 2006-10-04 00:42:20 +0600 (Срд, 04
	  Окт 2006) | 1 line Change default presentation indicator to "user
	  provided not screened" if octet 3a missed in CallingPartyNumber
	  IE ........

2006-10-03 18:37 +0000 [r44273-44285]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 44284 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44284 | file | 2006-10-03 14:35:55 -0400 (Tue, 03 Oct 2006) | 2
	  lines Use VideoSupport instead so it is considered a valid XML
	  attribute name. (issue #8075 reported by renemendoza) ........

	* CHANGES, apps/app_queue.c: Add 'Strat' manager field to
	  QueueParams event. (issue #7704 reported by renemendoza)

	* main/channel.c, CHANGES: Add Masquerade manager event which trips
	  when a masquerade happens (issue #7840 reported by moy)

2006-10-03 16:42 +0000 [r44263]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
	  pbx/ael/ael-test/ref.ael-ntest10,
	  pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
	  pbx/ael/ael-test/ref.ael-vtest13: These changes correspond to the
	  changes to app_stack's Gosub() application

2006-10-03 16:15 +0000 [r44262]  Joshua Colp <jcolp@digium.com>

	* UPGRADE.txt: First entry! Tell people about the callerid changes
	  with manager.

2006-10-03 15:53 +0000 [r44253]  Matt O'Gorman <mogorman@digium.com>

	* main/udptl.c, funcs/func_rand.c, main/say.c, apps/app_record.c,
	  apps/app_test.c, funcs/func_strings.c, apps/app_alarmreceiver.c,
	  apps/app_ices.c, channels/chan_iax2.c, main/loader.c,
	  res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c,
	  apps/app_zapras.c, main/http.c, channels/chan_alsa.c,
	  apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, main/sched.c,
	  apps/app_dial.c, main/pbx.c, channels/chan_agent.c,
	  apps/app_disa.c, channels/iax2-provision.c,
	  apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c,
	  channels/chan_misdn.c, apps/app_zapbarge.c,
	  channels/chan_features.c, apps/app_macro.c, apps/app_voicemail.c,
	  apps/app_meetme.c, res/res_musiconhold.c, channels/chan_gtalk.c,
	  res/res_jabber.c, main/enum.c, cdr/cdr_csv.c, main/channel.c,
	  channels/chan_phone.c, apps/app_osplookup.c, main/manager.c,
	  apps/app_mp3.c, res/res_agi.c, main/logger.c, main/app.c,
	  main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c,
	  res/res_config_pgsql.c, channels/chan_zap.c, funcs/func_db.c,
	  channels/chan_sip.c, apps/app_festival.c,
	  apps/app_waitforsilence.c, res/res_adsi.c, res/res_crypto.c,
	  apps/app_queue.c, main/rtp.c, cdr/cdr_tds.c,
	  channels/chan_jingle.c, apps/app_directed_pickup.c, main/file.c,
	  pbx/pbx_dundi.c, channels/chan_nbs.c, main/dsp.c: bug #8076 check
	  option_debug before printing to debug channel. patch provided in
	  bugnote, with minor changes.

2006-10-03 15:50 +0000 [r44252]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Okay, I can't use ast_app_separate_args for
	  that... and add some debugging for murf...

2006-10-03 15:48 +0000 [r44250-44251]  Luigi Rizzo <rizzo@icir.org>

	* configure.ac: comment the fact that autoconf2.59 is ok to process
	  this file, but we want to use 2.60 in case the generated
	  "configure" file must me committed back to the repository, so we
	  keep differences to a minimum.

	* bootstrap.sh: simplify this file

2006-10-03 00:07 +0000 [r44241]  Matt O'Gorman <mogorman@digium.com>

	* include/asterisk/jabber.h, res/res_jabber.c: 44240 same as but
	  without the removing of chan_jingle and such, as I hope to finish
	  jingle support for 1.6

2006-10-02 22:02 +0000 [r44231]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Use the standard parsing routines

2006-10-02 20:58 +0000 [r44200-44218]  Joshua Colp <jcolp@digium.com>

	* configs/queues.conf.sample, doc/channelvariables.txt, CHANGES,
	  apps/app_queue.c: Expand setinterfacevar option to also set a
	  variable, MEMBERNAME, which contains the member's name. (issue
	  #8046 reported by jmls)

	* apps/app_dial.c, main/channel.c, apps/app_meetme.c,
	  res/res_features.c, apps/app_dumpchan.c, CHANGES,
	  apps/app_queue.c: Make callerid fields in Manager events more
	  consistent. CallerIDNum for number and CallerIDName for name.
	  (issue #7976 reported by suhler)

	* /, channels/chan_sip.c: Merged revisions 44215 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44215 | file | 2006-10-02 16:11:02 -0400 (Mon,
	  02 Oct 2006) | 10 lines Merged revisions 44213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2
	  lines Change the fd on the I/O context in case it changed during
	  the reload, which is indeed possible. (issue #7943 reported by
	  eclubb) ........ ................

	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 44199 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44199 | file | 2006-10-02 15:41:39 -0400 (Mon,
	  02 Oct 2006) | 10 lines Merged revisions 44198 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r44198 | file | 2006-10-02 15:39:59 -0400 (Mon, 02 Oct 2006) | 2
	  lines We should be using $AST_SBIN instead of hardcoding the path
	  for the error message (issue #7942 reported by eclubb) ........
	  ................

2006-10-02 19:01 +0000 [r44187-44196]  Paul Cadach <paul@odt.east.telecom.kz>

	* configs/users.conf.sample, /, pbx/pbx_config.c: Merged revisions
	  44186 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) |
	  1 line Missed part of userconf functionality for chan_h323
	  ........

	* /, doc/realtime.txt: Merged revisions 44167 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44167 | pcadach | 2006-10-02 23:16:37 +0600 (Пнд, 02 Окт 2006) |
	  1 line Typo fix ........

	* /, channels/chan_h323.c: Merged revisions 44166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44166 | pcadach | 2006-10-02 23:15:11 +0600 (Пнд, 02 Окт 2006) |
	  1 line Optimization of oh323_indicate(): less locks - less
	  problems, plus single exit point ........

2006-10-02 17:54 +0000 [r44153-44172]  Joshua Colp <jcolp@digium.com>

	* main/logger.c, CHANGES, configs/logger.conf.sample: Add option to
	  logger to rename log files with timestamp (issue #8020 reported
	  by jmls)

	* /, main/io.c: Merged revisions 44169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44169 | file | 2006-10-02 13:25:13 -0400 (Mon,
	  02 Oct 2006) | 10 lines Merged revisions 44168 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r44168 | file | 2006-10-02 13:22:27 -0400 (Mon, 02 Oct 2006) | 2
	  lines Shrink when current_ioc is unused. It is set to -1 when
	  unused, not 0. (issue #7941 reported by eclubb) ........
	  ................

	* res/res_monitor.c: Get rid of the IS_NULL_STRING macro and use
	  ast_strlen_zero instead (issue #8070 reported by wrmem)

2006-10-02 16:00 +0000 [r44152]  Kevin P. Fleming <kpfleming@digium.com>

	* main/asterisk.c: clean up formatting and conformance to code
	  guidelines revert Mark's change that caused a memory leak
	  (cap_set_proc() does not free the capability structure so we
	  always need to call cap_free())

2006-10-02 15:40 +0000 [r44150]  Joshua Colp <jcolp@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option
	  'keepstats' which will keep queue statistics during a reload.
	  (issue #7908 reported by jmls)

2006-10-02 04:17 +0000 [r44148]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: It makes more sense that in GosubIf that the
	  two labels might have different arguments.

2006-10-02 02:38 +0000 [r44145-44147]  Mark Spencer <markster@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Don't use channel when
	  you don't mean a channel

	* main/asterisk.c: Uhm yah, not sure who committed this into
	  trunk... Anyway, I think this is what was intended...

2006-10-01 19:40 +0000 [r44136]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/chan_h323.c: Merged revisions 44135 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44135 | pcadach | 2006-10-02 01:32:24 +0600 (Пнд, 02 Окт 2006) |
	  1 line Do not simulate any audio tones if we got PROGRESS message
	  ........

2006-10-01 18:30 +0000 [r44112-44126]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 44125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44125 | russell | 2006-10-01 14:30:06 -0400 (Sun, 01 Oct 2006) |
	  6 lines Fix a problem that cuased AST_DATA_DIR in defaults.h to
	  be empty. The cause is that since ASTDATADIR is explicitly
	  exported using "export ASTDATADIR" at the top of the Makefile,
	  make no longer considers the variable "undefined", so the
	  Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
	  #8063, reported by akohlsmith, fixed by me) ........

	* /, configs/queues.conf.sample: Merged revisions 44111 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r44111 | russell | 2006-10-01 11:20:12 -0400
	  (Sun, 01 Oct 2006) | 11 lines Merged revisions 44110 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01
	  Oct 2006) | 3 lines Fix the name of the "eventmemberstatus"
	  option in the sample queues.conf (issue #8065, adamg) ........
	  ................

2006-10-01 05:37 +0000 [r44100]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_zapateller.c: Janitor for Zapateller: convert to use
	  argument macros

2006-09-30 19:23 +0000 [r44091]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, main/rtp.c: Merged revisions 44090 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) |
	  1 line Allow one-way RTP streams (device->Asterisk) ........

2006-09-30 16:37 +0000 [r44081]  Luigi Rizzo <rizzo@icir.org>

	* Makefile, main/Makefile, codecs/lpc10/Makefile: merge compile
	  fixes from 44080: - with AST_DEVMODE, building codecs/lpc10 fails
	  because of lots of warnings, and the configure step in editline
	  fails as well. Fix this by removing the -Werror in these steps. -
	  on FreeBSD (but probably on other platforms as well), the final
	  link of asterisk fails because AST_LIBS was not exported to the
	  subdirs Makefiles. Add a proper fix in the top-level Makefile (a
	  possible alternative way is to add "export AST_LIBS" near the
	  beginning of the file). With this fix, i believe that some of the
	  platform-specific conditionals in main/Makefile are redundant
	  (because they should be already dealt with in the top level
	  Makefile) but i don't have a platform to check.

2006-09-30 16:15 +0000 [r44069-44079]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/chan_sip.c: Merged revisions 44078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) |
	  6 lines Fix issue #7928 correctly. Next is a comment of previous
	  fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan
	  with a small fix by me, myself or I. Thanks, Philippe! (This was
	  caused by my changes to the transaction handling) ........

	* /, channels/chan_sip.c: Merged revisions 44068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) |
	  14 lines Found some buggy SIP clients (phones Planet VIP-153T
	  firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK
	  not on OK message only (when remote party answers) but on RINGING
	  message too, so when we send 200 OK message, we get unidentified
	  ACK message (because INVITE acknowledged on RINGING message
	  already), so 200 OK retransmits within its retransmission
	  interval then call gets dropped. If someone else knows how to
	  provide workaround for such cases, please, fix it in correct way.
	  Thanks to ssh from #asteriskru for provide access to his box to
	  study and fix this case. ........

2006-09-29 22:52 +0000 [r44056-44058]  Kevin P. Fleming <kpfleming@digium.com>

	* /, agi, utils: Merged revisions 44057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44057 | kpfleming | 2006-09-29 17:51:53 -0500 (Fri, 29 Sep 2006)
	  | 2 lines ignore temporary files made by the Makefiles during a
	  build ........

	* /, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules,
	  Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile,
	  agi/Makefile, codecs/Makefile, utils/Makefile, configure,
	  build_tools/embed_modules.xml, codecs/ilbc/Makefile,
	  codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions
	  44055 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44055 | kpfleming | 2006-09-29 17:47:40 -0500 (Fri, 29 Sep 2006)
	  | 2 lines fix a few build system bugs, and convert Makefiles to
	  be compatible with GNU make 3.80 ........

2006-09-29 22:36 +0000 [r44054]  Jason Parker <jparker@digium.com>

	* /, main/asterisk.c, main/cli.c: Merged revisions 44053 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44053 | qwell | 2006-09-29 15:35:09 -0700 (Fri, 29 Sep 2006) | 3
	  lines Fix a bug with the removal of 'atleast' argument to 'core
	  verbose' and 'core debug'. Add that argument back in. ........

2006-09-29 21:13 +0000 [r44044]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx: Merged revisions 44034,44042-44043
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) |
	  1 line Fake display name by called number on incoming calls
	  (until passing connected number/connected name is not
	  implemented) ........ r44042 | pcadach | 2006-09-30 03:05:43
	  +0600 (Сбт, 30 Сен 2006) | 1 line Set TON/PRESENTATION
	  information more carefully when no CallingNumber IE available
	  ........ r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30
	  Сен 2006) | 1 line Compile first, please ........

2006-09-29 20:16 +0000 [r44033]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Remove locking conflict

2006-09-29 19:16 +0000 [r44024-44025]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Merged
	  revisions 44022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44022 | pcadach | 2006-09-30 01:06:55 +0600 (Сбт, 30 Сен 2006) |
	  3 lines Properly pass TON/PRESENTATION information - original
	  H323Connection::SendSignalSetup() destroys Q.931 fields. ........

2006-09-29 18:54 +0000 [r44013]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, codecs/Makefile, utils/Makefile, /, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile,
	  codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules,
	  Makefile.rules, pbx/Makefile, channels/Makefile,
	  main/db1-ast/Makefile: Merged revisions
	  43996-43997,44008,44011-44012 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006)
	  | 2 lines another cross-compile fix ........ r43997 | kpfleming |
	  2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines support
	  --without-curl in configure script ........ r44008 | kpfleming |
	  2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines don't
	  abuse CFLAGS and LDFLAGS for build of Asterisk components,
	  because they are also then used for non-Asterisk components (like
	  menuselect); use our own variables instead ........ r44011 |
	  kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2
	  lines missed one conversion to ASTCFLAGS ........ r44012 |
	  kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2
	  lines yet another place where we were not using the correct
	  CFLAGS by default ........

2006-09-29 18:35 +0000 [r44010]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged
	  revisions 44009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r44009 | pcadach | 2006-09-30 00:30:34 +0600 (Сбт, 30 Сен 2006) |
	  1 line Pass TON/PRESENTATION information too ........

2006-09-29 16:38 +0000 [r43979-43994]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 43993 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43993 | kpfleming | 2006-09-29 11:38:27 -0500 (Fri, 29 Sep 2006)
	  | 2 lines a couple more environment settings that can't leak into
	  the menuselect build ........

	* /, main/cli.c: Merged revisions 43978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43978 | kpfleming | 2006-09-29 08:45:40 -0500 (Fri, 29 Sep 2006)
	  | 2 lines proper fix for ast_group_t change ........

2006-09-29 01:36 +0000 [r43954-43961]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: One must remember to unlock their list...
	  thanks to Qwell for letting me into his box

	* main/pbx.c: Cache the application pointer so we don't have to
	  needlessly search for it over and over. This should yield a
	  suitable performance increase.

2006-09-28 22:43 +0000 [r43953]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 43952 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43952 | kpfleming | 2006-09-28 17:42:18 -0500 (Thu, 28
	  Sep 2006) | 2 lines eliminate compiler warning when
	  DEBUG_CHANNEL_LOCKS is enabled and users of this header file
	  don't also include channel.h ........

2006-09-28 20:13 +0000 [r43945]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 43944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43944 | qwell | 2006-09-28 13:11:22 -0700 (Thu, 28 Sep 2006) | 4
	  lines Fix incorrect argument order for member names, on persisted
	  members. Issue 8047, patch by jmls. ........

2006-09-28 18:09 +0000 [r43934]  Joshua Colp <jcolp@digium.com>

	* main/udptl.c, main/frame.c, /, channels/chan_sip.c,
	  funcs/func_timeout.c, apps/app_festival.c,
	  apps/app_alarmreceiver.c, channels/iax2-provision.c,
	  res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c,
	  res/res_monitor.c, apps/app_playback.c,
	  include/asterisk/logger.h, res/res_smdi.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c:
	  Merged revisions 43933 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2
	  lines Put in missing \ns on the end of ast_logs (issue #7936
	  reported by wojtekka) ........

2006-09-28 17:38 +0000 [r43921]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_queue.c: Merged revisions 43919 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43919 | kpfleming | 2006-09-28 12:35:42 -0500
	  (Thu, 28 Sep 2006) | 10 lines Merged revisions 43916 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28
	  Sep 2006) | 2 lines fix buggy (and overly complex) loop used
	  during reload of app_queue for static member list updating
	  ........ ................

2006-09-28 17:36 +0000 [r43920]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx: Merged revisions 43918 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43918 | pcadach | 2006-09-28 23:34:19 +0600 (Чтв, 28 Сен 2006) |
	  1 line Extend call establishment timeout ........

2006-09-28 17:32 +0000 [r43912-43917]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 43915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2
	  lines Make sure the pvt exists before accessing it again as it
	  may have gone away (issue #7562 reported by Seb7 and issue #7939
	  reported by sorg) ........

	* /, main/cli.c: Merged revisions 43913 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43913 | file | 2006-09-28 13:14:07 -0400 (Thu, 28 Sep 2006) | 2
	  lines Warning be gone! ........

	* channels/chan_sip.c: Add jitterbuffer information to sip list
	  settings (issue #7945 reported by sergee)

2006-09-28 16:54 +0000 [r43902]  BJ Weschke <bweschke@btwtech.com>

	* /, apps/app_queue.c: Merged revisions 43899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43899 | bweschke | 2006-09-28 12:41:05 -0400
	  (Thu, 28 Sep 2006) | 11 lines Merged revisions 43897 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28
	  Sep 2006) | 3 lines app_queue is comparing the device names
	  incorrectly while checking their statuses. It's internal list of
	  interfaces includes the dial string, while the argument passed to
	  this function does not have the dial string (/n for a local
	  channel). This causes it to ignore the device state changes
	  because it thinks it belongs to none of its members. (#8040
	  reported and patch by tim_ringenbach) ........ ................

2006-09-28 16:43 +0000 [r43900]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/cli.c: Merged revisions 43898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43898 | kpfleming | 2006-09-28 11:38:25 -0500
	  (Thu, 28 Sep 2006) | 10 lines Merged revisions 43895 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28
	  Sep 2006) | 2 lines eliminate compiler warning introduced by
	  recent changes ........ ................

2006-09-28 16:19 +0000 [r43894]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 43893 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43893 | file | 2006-09-28 12:17:36 -0400 (Thu,
	  28 Sep 2006) | 10 lines Merged revisions 43891 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2
	  lines Stop the stream after waitstream returns so that our
	  formats get restored. (issue #7370 reported by kryptolus)
	  ........ ................

2006-09-28 16:01 +0000 [r43888]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx: Merged revisions 43877 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43877 | pcadach | 2006-09-28 21:56:21 +0600 (Чтв, 28 Сен 2006) |
	  1 line Fix compiler warning ........

2006-09-28 15:32 +0000 [r43865-43875]  BJ Weschke <bweschke@btwtech.com>

	* /, apps/app_queue.c: Merged revisions 43873 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43873 | bweschke | 2006-09-28 11:29:21 -0400
	  (Thu, 28 Sep 2006) | 11 lines Merged revisions 43871 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28
	  Sep 2006) | 3 lines Fix race condion crash with get_member_status
	  (#7864 - tim_ringenbach reported and patched) ........
	  ................

	* /, apps/app_queue.c: Merged revisions 43864 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43864 | bweschke | 2006-09-28 09:24:10 -0400 (Thu, 28 Sep 2006)
	  | 3 lines Autopause not working for queue members. (#8042 - jmls
	  reported and patch) ........

2006-09-28 13:02 +0000 [r43863]  Paul Cadach <paul@odt.east.telecom.kz>

	* /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h,
	  include/asterisk/compiler.h: Merged revisions 43861-43862 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) |
	  1 line Put attribute tag at correct place ........ r43862 |
	  pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
	  Force remote side to start media on outgoing PROGRESS message
	  ........

2006-09-28 11:32 +0000 [r43854-43855]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions 43852 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43852 | crichter | 2006-09-28 13:03:05 +0200
	  (Do, 28 Sep 2006) | 9 lines Merged revisions 43764 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27
	  Sep 2006) | 1 line fixed a bug which led to chan_list zombies,
	  when the call could not be properly established in misdn_call.
	  also removed the ACK_HDLC stuff which is not really needed.
	  ........ ................

	* channels/chan_misdn.c, /, channels/Makefile: Merged revisions
	  43775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) |
	  1 line removed the chan_misdn versioning, since asterisk has it's
	  own ........

2006-09-28 11:12 +0000 [r43845-43853]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/cisco-h225.h, /, channels/h323/ast_h323.cxx,
	  main/file.c, channels/h323/cisco-h225.asn,
	  channels/h323/cisco-h225.cxx: Merged revisions
	  43635,43843-43844,43846 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) |
	  1 line Fix ASN1 description of non-standard Cisco extensions
	  ........ r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28
	  Сен 2006) | 1 line Don't treat unknown control frames as voice
	  ........ r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28
	  Сен 2006) | 1 line Don't warn on HOLD/UNHOLD control frames
	  ........ r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28
	  Сен 2006) | 1 line Do not open transmit channel until TCS is
	  received ........

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, CHANGES, channels/h323/chan_h323.h,
	  configs/h323.conf.sample: Handle HOLD/RETRIEVE notifications

2006-09-27 22:01 +0000 [r43827-43836]  Joshua Colp <jcolp@digium.com>

	* CHANGES: Update CHANGES to reflect libcap capability that was
	  added.

	* configure, main/Makefile, configure.ac, makeopts.in,
	  doc/security.txt, main/asterisk.c: Add ability to set high ToS
	  bits as non-root on Linux using libcap (issue #7047 reported by
	  maddison)

	* apps/app_voicemail.c: Finish up last commit

	* apps/app_voicemail.c: Do the directory walk dance instead of
	  repeated stat calls as it seems to be faster (issue #7507
	  reported by Corydon76)

2006-09-27 20:27 +0000 [r43817]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 43816 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43816 | tilghman | 2006-09-27 15:21:54 -0500
	  (Wed, 27 Sep 2006) | 10 lines Merged revisions 43815 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27
	  Sep 2006) | 2 lines Avoid inability to lock directory log message
	  by creating the directory ahead of time. (Issue 7631) ........
	  ................

2006-09-27 20:03 +0000 [r43804-43814]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Add BACKGROUNDSTATUS to Background() Issue #7835,
	  original patch by bcnit - redone by me.

	* main/pbx.c, /, apps/app_playback.c: Merged revisions 43803 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4
	  lines Fix an issue with PLAYBACKSTATUS not being set under
	  certain circumstances. Fix a minor issue, to make it use the
	  filenames that were parsed, instead of the entire argument
	  string. Fix Background() to return -1 like Playback(), if no args
	  are specified. ........

2006-09-27 19:39 +0000 [r43792-43802]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: I *think* this is the last list in
	  chan_iax2

	* /, main/rtp.c: Merged revisions 43798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2
	  lines Compensate for out of order packets better if RFC2833
	  compensation is turned on. ........

	* /, channels/chan_iax2.c: Merged revisions 43783 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2
	  lines Get rid of two functions from a time now past (we THINK
	  these are from pre-recursive lock time) that may be contributing
	  to two open issues on the bug tracker (7562/7939) and that has
	  the potential to just make bad things happen if the timing is
	  right. ........

2006-09-27 17:00 +0000 [r43785]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Fix some little things

2006-09-27 16:57 +0000 [r43780]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, res/res_features.c: Merged revisions 43779 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43779 | russell | 2006-09-27 12:55:49 -0400
	  (Wed, 27 Sep 2006) | 50 lines Merged revisions 43778 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27
	  Sep 2006) | 42 lines Fix a problem that occurred if a user
	  entered a digit that matched a bridge feature that was configured
	  using multiple digits, and the digit that was pressed timed out
	  in the feature digit timeout period. For example, if blind
	  transfer is configured as '##', and a user presses just '#'. In
	  this situation, the call would lock up and no longer pass any
	  frames. (issue #7977 reported by festr, and issue #7982 reported
	  by michaels and valuable input provided by mneuhauser and kuj.
	  Fixed by me, with testing help and peer review from Joshua Colp).
	  There are a couple of issues involved in this fix: 1) When
	  ast_generic_bridge determines that there has been a timeout, it
	  returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
	  this result, it calls ast_generic_bridge over again with the same
	  timestamp for the next event. This results in an endless loop of
	  nothing until the call is terminated. This is resolved by simply
	  changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
	  sees a timeout. 2) I also changed ast_channel_bridge such that if
	  in the process of calculating the time until the next event, it
	  knows a timeout has already occured, to immediately return
	  AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
	  anyway. 3) In the process of testing the previous two changes, I
	  ran into a problem in res_features where ast_channel_bridge would
	  return because it determined that there was a timeout. However,
	  ast_bridge_call in res_features would then determine by its own
	  calculation that there was still 1 ms before the timeout really
	  occurs. It would then proceed, and since the bridge broke out and
	  did *not* return a frame, it interpreted this as the call was
	  over and hung up the channels. The reason for this was because
	  ast_bridge_call in res_features and ast_channel_bridge in
	  channel.c were using different times for their calculations.
	  channel.c uses the start_time on the bridge config, which is the
	  time that the feature digit was recieved. However, res_features
	  had another time, 'start', which was set right before calling
	  ast_channel_bridge. 'start' will always be slightly after
	  start_time in the bridge config, and sometimes enough to round up
	  to one ms. This is fixed by making ast_bridge_call use the same
	  time as ast_channel_bridge for the timeout calculation. ........
	  ................

2006-09-27 16:49 +0000 [r43777]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add CLI block and unblock circuit commands
	  for SS7.

2006-09-27 16:25 +0000 [r43776]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 43774 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2
	  lines Make rfc2833compensate a global option. ........

2006-09-27 12:32 +0000 [r43763]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Use ast_strdupa() instead of strdup(),
	  thanks to sergee

2006-09-27 04:37 +0000 [r43754-43757]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: remote an unused buffer in mm_login()
	  (issue #8038, selsky) In passing, I have cleaned up some
	  formatting to better comply with our guidelines. I have also
	  changed one place to use S_OR(), and a couple of places to use
	  ast_strlen_zero() as appropriate.

2006-09-27 03:45 +0000 [r43740-43747]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ael-test11/extensions.ael,
	  pbx/ael/ael-test/ref.ael-test6, CHANGES,
	  pbx/ael/ael-test/ael-test3/extensions.ael,
	  pbx/ael/ael-test/ref.ael-test7,
	  pbx/ael/ael-test/ael-test5/extensions.ael,
	  pbx/ael/ael-test/ref.ael-vtest13: This commits the changes to AEL
	  to use the gosub-with-args from Tilghman to perform macro calls.
	  This results in substantially smaller stack footprint, which
	  allows macro call depths in excess of 100,000 levels, rather than
	  the limit of 7 calls deep, which the Macro app is subject to.

	* /, configs/extensions.ael.sample: Merged revisions 43739 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1
	  line This change to extensions.ael was to fix bug 8031; the
	  install scripts are causing it to be copied to
	  /etc/asterisk/extensions.ael, and because it is a fairly direct
	  conversion of the original extensions.conf, the macro and context
	  names clash with the existing extensions.conf. So, I put an ael-
	  in front of all macros and contexts, and checked every goto and
	  macro call. Also, this file compiles under aelparse. ........

2006-09-27 01:39 +0000 [r43733]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Clean up code and convert last two things
	  (firmware/dialplan cache) to linked list macros.

2006-09-26 22:18 +0000 [r43721-43727]  Jason Parker <jparker@digium.com>

	* apps/app_meetme.c: Fire a manager event when a meetme is
	  started/stopped. Issue #7891, patch by suhler.

	* apps/app_queue.c: Add QueueSummary manager action. Gives "at a
	  glance" information about a single queue, or all queues. Issue
	  #8035, patch by rgollent, slightly modified (formatting) by me.

2006-09-26 21:01 +0000 [r43715]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.c: Merged revisions 43710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43710 | russell | 2006-09-26 16:56:42 -0400
	  (Tue, 26 Sep 2006) | 17 lines (This was actually BE-65) Merged
	  revisions 43708 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) |
	  7 lines Back in revision 4798, this message was changed from
	  using ast_cli() to directly calling write(). During this change,
	  checking if this was a remote console was removed. This caused
	  this message about using "exit" or "quit" to exit an Asterisk
	  console to come up in times where it did not make sense. This
	  change restores the check to see if this is a remote console
	  before printing the message. (fixes BE-4) ........
	  ................

2006-09-26 20:51 +0000 [r43709]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c, include/asterisk/channel.h, .cleancount,
	  main/cli.c: Merged revisions 43707 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43707 | file | 2006-09-26 16:47:26 -0400 (Tue,
	  26 Sep 2006) | 10 lines Merged revisions 43705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2
	  lines Use proper type to represent the group variable (issue
	  #8025 reported by makoto) ........ ................

2006-09-26 20:30 +0000 [r43702]  Jason Parker <jparker@digium.com>

	* CHANGES: update CHANGES file to reflect codec support in
	  chan_skinny

2006-09-26 20:26 +0000 [r43701]  Russell Bryant <russell@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 43700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43700 | russell | 2006-09-26 16:24:39 -0400
	  (Tue, 26 Sep 2006) | 14 lines Merged revisions 43699 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26
	  Sep 2006) | 6 lines When parsing the sections of voicemail.conf
	  that contain mailbox definitions, don't introduce a length limit
	  on the definition by using a 256 byte temporary storage buffer.
	  Instead, make the temporary buffer just as big as it needs to be
	  to hold the entire mailbox definition. (fixes BE-68) ........
	  ................

2006-09-26 20:20 +0000 [r43696-43698]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 43697 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep
	  2006) | 2 lines Strip options off the argument passed for
	  devicestate in chan_local. (issue #8034 reported by pcardozo)
	  ........

	* main/channel.c, /, main/slinfactory.c, apps/app_chanspy.c: Merged
	  revisions 43695 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2
	  lines Slight overhaul of the whisper support. 1. We need to
	  duplicate the frame from ast_translate 2. We need to ensure we
	  always have signed linear coming in for signed linear combining.
	  3. We need to ensure we are always feeding signed linear out. 4.
	  Properly store and restore write format when beeping on the
	  channel we are whispering on. 5. Properly discontinue the stream
	  on the channel for the beep. (issue #8019 reported by
	  timkelly1980) ........

2006-09-26 19:37 +0000 [r43677-43687]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: start a CHANGES file for trunk... no need to force
	  people to have to review commit logs after branching

	* /, sounds/Makefile: Merged revisions 43676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43676 | kpfleming | 2006-09-26 13:34:27 -0500 (Tue, 26 Sep 2006)
	  | 2 lines update to use 1.4.3 core sounds, with corrected
	  beep/beeperr/tt-monkeys files ........

2006-09-26 18:10 +0000 [r43675]  Jason Parker <jparker@digium.com>

	* main/frame.c, /, doc/rtp-packetization.txt: Merged revisions
	  43674 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43674 | qwell | 2006-09-26 11:08:51 -0700 (Tue, 26 Sep 2006) | 4
	  lines Issue #8015, patch by Dan Austin. Maximum values were
	  incorrect, which is why this is being put in 1.4 ........

2006-09-26 17:25 +0000 [r43667]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Gosub arguments (Issue 7780)

2006-09-26 17:09 +0000 [r43666]  Jason Parker <jparker@digium.com>

	* main/logger.c, configs/logger.conf.sample: Add optional
	  queue_log_name config option for logger.conf, to change the name
	  of the queue_log file. Issue #7363, patch by Steve Davies,
	  slightly modified by me.

2006-09-26 16:56 +0000 [r43658-43659]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: MailboxExists should be a dialplan
	  function, not an application (Issue 7503)

	* res/res_limit.c: These three are not defined on all platforms
	  that we support

2006-09-26 15:35 +0000 [r43651]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 43650 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep
	  2006) | 11 lines Add proper codec support to chan_skinny. Works
	  with at least ulaw, alaw, and g729a. This is technically a "new
	  feature", but there are justifications for it. I found a bug with
	  the recent rtp packetization changes, which caused the media
	  setup to fail under certain circumstances, particularly when
	  using allow=all, or having no allow= statements (globally or on
	  the device). I could have either removed the rtp packetization
	  features, or I could add proper codec support (which, without, I
	  think most people would consider to be a bug anyways). ........

2006-09-25 22:09 +0000 [r43641-43643]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 43642 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43642 | tilghman | 2006-09-25 17:07:44 -0500 (Mon, 25 Sep 2006)
	  | 2 lines Should have moved these lines up in the merge, instead
	  of removing them ........

	* /, apps/app_voicemail.c: Merged revisions 43640 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43640 | tilghman | 2006-09-25 17:04:47 -0500
	  (Mon, 25 Sep 2006) | 12 lines Merged revisions 43634 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43634 | tilghman | 2006-09-25 16:14:41 -0500 (Mon, 25
	  Sep 2006) | 4 lines Two bugs when forwarding voicemail (Issue
	  7824): 1) delete=yes was ignored 2) maxmessages was ignored
	  ........ ................

2006-09-25 20:30 +0000 [r43627]  Paul Cadach <paul@odt.east.telecom.kz>

	* /: Block revision 43626 from 1.4 tree - already here

2006-09-25 15:24 +0000 [r43617]  Jason Parker <jparker@digium.com>

	* /, sounds/Makefile: Merged revisions 43616 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43616 | qwell | 2006-09-25 08:23:31 -0700 (Mon, 25 Sep 2006) | 4
	  lines One more fix for sounds installation - this time for
	  portability. Reported to asterisk-dev mailing list. ........

2006-09-25 14:49 +0000 [r43604]  Steve Murphy <murf@digium.com>

	* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
	  crashing if trying to play an OGG moh file.

2006-09-25 09:03 +0000 [r43571-43597]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
	  channels/chan_h323.c, channels/h323/ast_h323.h,
	  channels/h323/chan_h323.h, configs/h323.conf.sample: Support for
	  negotiation and receiption of Cisco's RTP DTMF

	* channels/h323/ast_h323.cxx: Disable fastStart if requested by
	  remote side

	* /: Block revision 43582

	* channels/chan_h323.c, configs/h323.conf.sample: Specify RFC2833
	  payload on dtmfmode option rather than dtmfcodec option
	  (deprecated)

	* channels/h323/ast_h323.cxx, channels/chan_h323.c: DTMF mode is
	  bitmask, not valued field

	* channels/h323/caps_h323.cxx, channels/h323/caps_h323.h: Define
	  Cisco RTP capability

	* channels/h323/caps_h323.cxx: Specify non-standard data
	  independedly on OpenH323's codec name (it can be easily changed)

	* channels/chan_h323.c, channels/h323/chan_h323.h: Define DTMF
	  payload types

2006-09-24 15:01 +0000 [r43554-43565]  Russell Bryant <russell@digium.com>

	* /, channels/iax2-provision.c: Merged revisions 43564 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24
	  Sep 2006) | 5 lines Fix a CLI command registration issue where an
	  erroneous message claiming that "iax2 show provisioning" was
	  already registered. This was because this command was registering
	  itself as both the command, as well as the command it is
	  deprecating. (issue #8022, reported by bjweeks, fixed by myself)
	  ........

	* /, channels/chan_iax2.c: Merged revisions 43553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43553 | russell | 2006-09-24 09:53:35 -0400
	  (Sun, 24 Sep 2006) | 12 lines Merged revisions 43552 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24
	  Sep 2006) | 4 lines Check to see if the channel that is
	  activating the IAXPEER function is actually an IAX2 channel
	  before proceeding to process it to avoid crashing. (issue #8017,
	  reported by admott, fixed by myself) ........ ................

2006-09-24 12:15 +0000 [r43539-43546]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/rtp.c: Small Cisco's RTP DTMF update

	* channels/chan_h323.c: Avoid possible deadlock on channel
	  destruction

	* main/rtp.c: Correct behavior on Cisco's DTMF

2006-09-22 23:46 +0000 [r43525-43526]  Kevin P. Fleming <kpfleming@digium.com>

	* /: file forgot one :-)

	* Makefile, /: Merged revisions 43524 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43524 | kpfleming | 2006-09-22 18:44:47 -0500 (Fri, 22 Sep 2006)
	  | 2 lines don't output the 'build complete' message when the
	  target being run is already going to do an installation ........

2006-09-22 23:34 +0000 [r43522]  Joshua Colp <jcolp@digium.com>

	* /: You see nothing...

2006-09-22 22:13 +0000 [r43519]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 43518 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep
	  2006) | 4 lines Allow chan_skinny.so to be unloaded properly.
	  Remove reload support, since it doesn't actually...work. ........

2006-09-22 21:34 +0000 [r43506-43507]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This commits a change to return
	  MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
	  goes well for bug 8004

	* pbx/pbx_ael.c: As per bug 8004, we now return
	  AST_MODULE_LOAD_DECLINE when we can't read extensions.ael

2006-09-22 20:33 +0000 [r43495-43500]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Move from h.323 to h323 command prefix

	* channels/chan_h323.c: Fix compilation warnings

	* channels/h323/compat_h323.h: Use own factory for our
	  OpalMediaFormats too

	* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h: Fix our
	  capability's factory

2006-09-22 17:26 +0000 [r43493]  Jason Parker <jparker@digium.com>

	* /, main/cli.c: Merged revisions 43492 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43492 | qwell | 2006-09-22 10:25:05 -0700 (Fri, 22 Sep 2006) | 2
	  lines Make sure we explicitly set the CLI command to not be
	  deprecated, if it isn't. ........

2006-09-22 16:43 +0000 [r43488-43490]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 43489 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43489 | kpfleming | 2006-09-22 11:42:46 -0500 (Fri, 22 Sep 2006)
	  | 2 lines use rebuilt extra sounds ........

	* main/channel.c, /: Merged revisions 43486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006)
	  | 2 lines all the Linux systems I have don't use '__m_count' for
	  this field, so I don't know where this came from... ........

2006-09-22 15:50 +0000 [r43483-43485]  Russell Bryant <russell@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 43482 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43482 | russell | 2006-09-22 11:42:44 -0400 (Fri, 22
	  Sep 2006) | 3 lines return AST_MODULE_LOAD_DECLIDE if mISDN could
	  not be configured (issue #8006, Mithraen) ........

2006-09-22 14:58 +0000 [r43479-43480]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/threadstorage.h: compatibility fix: use
	  "attribute_XXX" instead of *__attribute__ ((XXX)) so we can
	  handle compiler/os dependencies in our compiler.h

	* channels/chan_sip.c: style fix: move variable declaration at the
	  beginning of the block.

2006-09-22 14:04 +0000 [r43478]  Russell Bryant <russell@digium.com>

	* main/frame.c, /: Merged revisions 43477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43477 | russell | 2006-09-22 10:02:58 -0400 (Fri, 22 Sep 2006) |
	  3 lines Suppress a compiler warning about the use of a
	  potentially uninitialized variable. It couldn't actually happen,
	  though. ........

2006-09-22 04:54 +0000 [r43472]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/caps_h323.cxx: Add missing include

2006-09-22 03:09 +0000 [r43470]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 43469 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep
	  2006) | 4 lines First shot at unload_module in chan_skinny.. More
	  to come. ........

2006-09-21 23:55 +0000 [r43467]  Matt O'Gorman <mogorman@digium.com>

	* /, include/asterisk/jabber.h, channels/chan_gtalk.c,
	  res/res_jabber.c: Merged revisions 43466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006)
	  | 2 lines updates for better compontent support ........

2006-09-21 23:29 +0000 [r43463-43465]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
	  43464 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006)
	  | 2 lines Twould help if we actually documented how the new
	  features in res_odbc actually work. (Oops) ........

	* res/res_limit.c (added): Set process limits without restarting
	  Asterisk

2006-09-21 22:53 +0000 [r43461]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Oh look more changes,
	  but these are my own! (Clean up module load functions)

2006-09-21 22:44 +0000 [r43460]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Suppress compiler warnings

2006-09-21 22:32 +0000 [r43459]  Joshua Colp <jcolp@digium.com>

	* channels/chan_alsa.c: Clean up chan_alsa load module function
	  (issue #8000 reported by Mithraen)

2006-09-21 22:23 +0000 [r43458]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/acl.h, doc/ip-tos.txt, channels/chan_sip.c,
	  doc/mp3.txt, doc/ael.txt, doc/channelvariables.txt, main/acl.c:
	  And some deprecated APIs and modifications to documentation

2006-09-21 22:23 +0000 [r43455-43457]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_oss.c: Merged revisions 43456 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43456 | file | 2006-09-21 18:21:40 -0400 (Thu, 21 Sep 2006) | 2
	  lines Some more clean up in the load function for chan_oss (issue
	  #8002 reported by Mithraen with minor mods by moi) ........

	* /, channels/chan_mgcp.c: Merged revisions 43454 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2
	  lines Clean up chan_mgcp's module load function (issue #8001
	  reported by Mithraen with mods by moi) ........

2006-09-21 21:59 +0000 [r43452]  Tilghman Lesher <tlesher@digium.com>

	* doc/ip-tos.txt, channels/chan_local.c, channels/chan_sip.c,
	  res/res_features.c, channels/chan_agent.c, res/res_convert.c,
	  res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c,
	  channels/chan_oss.c, channels/chan_skinny.c,
	  channels/chan_features.c, res/res_agi.c, channels/chan_h323.c,
	  channels/chan_alsa.c, apps/app_settransfercapability.c (removed),
	  res/res_indications.c, pbx/pbx_config.c, res/res_odbc.c,
	  channels/chan_mgcp.c: Lots more removal of deprecated things

2006-09-21 21:22 +0000 [r43451]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/Makefile, build_tools/strip_nonapi (added): Merged
	  revisions 43450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43450 | kpfleming | 2006-09-21 16:21:29 -0500 (Thu, 21 Sep 2006)
	  | 2 lines add another attempt to strip non-API symbols from the
	  final binary... script will need to be extended to work on
	  non-Linux systems ........

2006-09-21 21:17 +0000 [r43442-43449]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, main/pbx.c, main/frame.c, main/translate.c,
	  apps/app_queue.c, main/config.c, main/rtp.c,
	  apps/app_setcdruserfield.c (removed), main/cli.c, main/channel.c,
	  main/manager.c, main/file.c, main/http.c, main/logger.c,
	  main/astmm.c, main/image.c, main/asterisk.c: Remove deprecated
	  CLI apps from the core

	* /, apps/app_url.c: Merged revisions 43445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43445 | tilghman | 2006-09-21 15:22:43 -0500 (Thu, 21 Sep 2006)
	  | 2 lines Fix documentation to reflect how Url() really works
	  ........

	* apps/app_setcallerid.c, apps/app_voicemail.c: More removal of
	  deprecated stuff

	* main/pbx.c, main/manager.c, UPGRADE.txt: Remove 1.4 changes from
	  UPGRADE.txt, remove deprecated callerid field, remove deprecated
	  SetGlobalVar app

	* /, apps/app_rpt.c: Merged revisions 43441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43441 | tilghman | 2006-09-21 14:43:32 -0500 (Thu, 21 Sep 2006)
	  | 2 lines Oops, missed the merge breakage ........

2006-09-21 19:42 +0000 [r43440]  Kevin P. Fleming <kpfleming@digium.com>

	* makeopts.in: fix this so chan_zap links properly again

2006-09-21 19:35 +0000 [r43439]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_language.c (removed), funcs/func_moh.c (removed),
	  apps/app_lookupcidname.c (removed), funcs/func_md5.c,
	  apps/app_hasnewvoicemail.c (removed), funcs/func_blacklist.c
	  (added), apps/app_random.c (removed), funcs/func_vmcount.c
	  (added), res/res_realtime.c (added), apps/app_lookupblacklist.c
	  (removed), apps/app_realtime.c (removed), apps/app_queue.c:
	  Remove deprecated apps and funcs

2006-09-21 19:27 +0000 [r43437]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, main/channel.c, /, channels/chan_sip.c,
	  include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c:
	  SS7 marked the start of an open season for trunk again but here's
	  something minor - abstract early bridging into the technology so
	  that we don't always assume they use RTP and try it.

2006-09-21 19:22 +0000 [r43436]  Kevin P. Fleming <kpfleming@digium.com>

	* configure: regenerated at PCadach's request

2006-09-21 19:18 +0000 [r43429-43434]  Paul Cadach <paul@odt.east.telecom.kz>

	* acinclude.m4: Check for 64-bit OpenH323/PWLib versions too,
	  thanks to Mithraen (please, re-build configure script)

	* channels/h323/caps_h323.cxx: Declare our own media formats to not
	  rely on OpenH323 configuration

	* channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
	  channels/chan_h323.c, channels/h323/caps_h323.h: Introduce Cisco
	  G.726-32 capability (g726aal2 form)

2006-09-21 18:42 +0000 [r43427-43428]  Matthew Fredrickson <creslin@digium.com>

	* configure: Update configure

	* channels/chan_zap.c, build_tools/menuselect-deps.in,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
	  configs/zapata.conf.sample: Merge in SS7 changes.... need to
	  still cleanup zapata.conf

2006-09-21 17:06 +0000 [r43411-43423]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_rpt.c: Merged revisions 43422 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43422 | tilghman | 2006-09-21 12:04:40 -0500
	  (Thu, 21 Sep 2006) | 10 lines Merged revisions 43420 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43420 | tilghman | 2006-09-21 12:01:48 -0500 (Thu, 21
	  Sep 2006) | 2 lines Whitespace change... really just an excuse to
	  test repotools ........ ................

	* /: Last merge should not have brought in the 1.2 props

	* /, configure, configure.ac, cdr/cdr_tds.c: Merged revisions 43410
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r43410 | tilghman | 2006-09-21 11:31:59 -0500
	  (Thu, 21 Sep 2006) | 10 lines Merged revisions 43409 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r43409 | tilghman | 2006-09-21 11:18:19 -0500 (Thu, 21
	  Sep 2006) | 2 lines TDS 0.64 updates ........ ................

2006-09-21 16:09 +0000 [r43403-43406]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/Makefile: Merged revisions 43405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r43405 | kpfleming | 2006-09-21 11:08:03 -0500 (Thu, 21 Sep 2006)
	  | 2 lines remove this change... it requires binutils 2.17
	  ........

	* /: remove extraneous property

2006-09-20 23:20 +0000 [r43397]  Jason Parker <jparker@digium.com>

	* /, build_tools/make_version: Merged revisions 43396 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r43396 | qwell | 2006-09-20 16:19:25 -0700 (Wed, 20 Sep
	  2006) | 2 lines fix minor typo in the way version is handled
	  ........

2006-09-20 23:02 +0000 [r43393]  Kevin P. Fleming <kpfleming@digium.com>

	* /: this has been manually merged

2006-09-20  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta1 released.