aboutsummaryrefslogtreecommitdiffstats
path: root/ChangeLog
blob: 7cf94eee42b1f70156bcfe599d37d89399870367 (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
2008-10-03 - Russell Bryant <russell@digium.com>

	* Asterisk 1.6.1-beta1 Released

2008-10-02 19:31 +0000 [r145960-145964]  Russell Bryant <russell@digium.com>

	* CHANGES, /: Merged revisions 145962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145962 |
	  russell | 2008-10-02 14:30:45 -0500 (Thu, 02 Oct 2008) | 2 lines
	  The 'P' command for ExternalIVR was also added in 1.6.0 ........

	* CHANGES, /: Merged revisions 145959 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145959 |
	  russell | 2008-10-02 14:27:37 -0500 (Thu, 02 Oct 2008) | 2 lines
	  TCP support for ExternalIVR went in to 1.6.1, not 1.6.0 ........

2008-10-02 15:30 +0000 [r145781]  Sean Bright <sean.bright@gmail.com>

	* /, configure, configure.ac: Merged revisions 145771 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r145771 | seanbright | 2008-10-02 11:28:48 -0400 (Thu, 02 Oct
	  2008) | 1 line This is much cleaner, methinks. ........

2008-10-02 15:19 +0000 [r145754]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, /: Merged revisions 145752 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r145752 | tilghman | 2008-10-02 10:17:16 -0500 (Thu, 02 Oct 2008)
	  | 10 lines Merged revisions 145751 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008)
	  | 3 lines Some sanity checks that may have led to prior crashes,
	  found by codefreeze-lap (murf) on IRC. Also some cleanup of
	  incorrectly-used constants. ........ ................

2008-10-01 23:54 +0000 [r145694]  Sean Bright <sean.bright@gmail.com>

	* /, configure, configure.ac: Merged revisions 145692 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r145692 | seanbright | 2008-10-01 19:48:16 -0400 (Wed, 01 Oct
	  2008) | 7 lines Try a test compile using the GMime library. Some
	  distros install gmime-config in the base package instead of the
	  -devel package. Now we print a notice and disable GMime support
	  instead of bombing during the main compilation. (closes issue
	  #13583) Reported by: arkadia ........

2008-10-01 22:24 +0000 [r145557-145609]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: Merged revisions 145606 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145606 |
	  mmichelson | 2008-10-01 17:23:50 -0500 (Wed, 01 Oct 2008) | 11
	  lines Okay, this should really do it now. While I did manage to
	  fix blind transfers with my last commit here, I also caused an
	  unwanted side-effect. That is, only the first priority of the 'h'
	  extension would be executed when a blind transfer occurred
	  instead of all priorities. Essentially, my last commit corrected
	  the return value of ast_bridge_call. However, the implementation
	  still was not 100% correct. Now it is. ........

	* /, main/features.c: Merged revisions 145579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145579 |
	  mmichelson | 2008-10-01 16:33:11 -0500 (Wed, 01 Oct 2008) | 4
	  lines if (!(x) == 0) is the same as if (x). ........

	* /, main/features.c: Merged revisions 145553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145553 |
	  mmichelson | 2008-10-01 16:06:26 -0500 (Wed, 01 Oct 2008) | 13
	  lines The logic surrounding the return value of
	  ast_spawn_extension within ast_bridge_call was reversed. This
	  problem was observed when a blind transfer placed from the callee
	  channel of a test call failed. While the problem I am solving
	  here is exactly the same as what was reported in issue #13584,
	  the difference is that this fix I am applying is trunk-only.
	  Issue #13584 was reported against the 1.4 branch, and my tests of
	  1.4's blind transfers appear to work fine. ........

2008-10-01 17:33 +0000 [r145517]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145487
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r145487 | lmadsen | 2008-10-01 13:26:20 -0400
	  (Wed, 01 Oct 2008) | 12 lines Merged revisions 145479 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008)
	  | 6 lines Update the realtime_pgsql.sql script to create the
	  setinterfacevar column. (closes issue #13549) Reported by: fiddur
	  ........ ................

2008-10-01 15:45 +0000 [r145430]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_sms.c: Merged revisions 145428 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145428 |
	  tilghman | 2008-10-01 10:44:06 -0500 (Wed, 01 Oct 2008) | 7 lines
	  Initializing buffer prevents a segfault when arguments are
	  incomplete. (closes issue #13471) Reported by: alecdavis Patches:
	  20080916__bug13471.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: alecdavis ........

2008-09-30 22:26 +0000 [r145262]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 145249 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r145249 |
	  jpeeler | 2008-09-30 17:21:19 -0500 (Tue, 30 Sep 2008) | 6 lines
	  (closes issue #13337) Reported by: pj Tested by: pj Set transport
	  to SIP_TRANSPORT_UDP mode if not specified which fixes calls to
	  get_transport returning UNKNOWN. ........

2008-09-27 16:49 +0000 [r144993]  Kevin P. Fleming <kpfleming@digium.com>

	* main/ast_expr2.c, Makefile, agi/Makefile, utils/Makefile,
	  include/asterisk/astmm.h, main/ast_expr2f.c, pbx/pbx_ael.c,
	  utils/ael_main.c, main/astmm.c, main/stdtime/localtime.c,
	  utils/extconf.c, main/ast_expr2.fl, include/asterisk.h, /,
	  main/Makefile, main/ast_expr2.y, Makefile.moddir_rules,
	  utils/astman.c: Merged revisions 144949-144951 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r144949 | kpfleming | 2008-09-27 10:52:56 -0500 (Sat, 27 Sep
	  2008) | 17 lines Merged revisions 144924-144925 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep
	  2008) | 6 lines improve header inclusion process in a few small
	  ways: - it is no longer necessary to forcibly include
	  asterisk/autoconfig.h; every module already includes asterisk.h
	  as its first header (even before system headers), which serves
	  the same purpose - astmm.h is now included by asterisk.h when
	  needed, instead of being forced by the Makefile; this means
	  external modules will build properly against installed headers
	  with MALLOC_DEBUG enabled - simplify the usage of some of these
	  headers in the AEL-related stuff in the utils directory ........
	  r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep
	  2008) | 2 lines fix some minor issues with rev 144924 ........
	  ................ r144950 | kpfleming | 2008-09-27 11:10:33 -0500
	  (Sat, 27 Sep 2008) | 2 lines fix bugs caused by r144949 when
	  MALLOC_DEBUG is defined ................ r144951 | kpfleming |
	  2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) | 1 line remove
	  incorrect comment ................

2008-09-27 01:08 +0000 [r144881]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_queue.c, channels/chan_dahdi.c, /: Merged revisions
	  144879 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144879 |
	  mvanbaak | 2008-09-27 02:49:24 +0200 (Sat, 27 Sep 2008) | 2 lines
	  fix a couple of CLI commands that did not have a help
	  description. ........

2008-09-26 23:16 +0000 [r144832]  Joshua Colp <jcolp@digium.com>

	* /, configs/rtp.conf.sample: Merged revisions 144829 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r144829 | file | 2008-09-26 20:12:13 -0300 (Fri, 26 Sep 2008) | 2
	  lines Update documentation to include default setting. This is
	  for you jtodd! ........

2008-09-26 18:09 +0000 [r144484-144684]  Steve Murphy <murf@digium.com>

	* /, pbx/pbx_lua.c: Merged revisions 144681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144681 |
	  murf | 2008-09-26 12:02:06 -0600 (Fri, 26 Sep 2008) | 14 lines
	  (closes issue #13564) Reported by: mnicholson Patches:
	  pbx_lua9.diff uploaded by mnicholson (license 96) Many thanks to
	  Matt for his upgrade to the lua dialplan option! the Description
	  from the bug: This patch adds a stack trace to errors encountered
	  while executing lua extensions. The patch also handles out of
	  memory errors reported by lua. ........

	* main/pbx.c, /: Merged revisions 144678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r144678 | murf | 2008-09-26 11:50:35 -0600 (Fri, 26 Sep 2008) |
	  20 lines Merged revisions 144677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) |
	  12 lines (closes issue #13563) Reported by: mnicholson Patches:
	  found1.diff uploaded by mnicholson (license 96) This patch was
	  mainly meant to apply to trunk and 1.6.x, but I'm applying it to
	  1.4 also, which should be a perfectly harmless fix to the vast
	  majority of users who are not using external switches, but the
	  few who might be affected will not have to go to the pain of
	  filing a bug report. ........ ................

	* utils/build-extensions-conf.lua (removed), /: Merged revisions
	  144635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144635 |
	  murf | 2008-09-26 10:51:30 -0600 (Fri, 26 Sep 2008) | 1 line Matt
	  suggests we remove utils/build-extensions-conf.lua, as per bug
	  12961, it is no longer necessary. ........

	* channels/chan_oss.c, apps/app_playback.c, main/pbx.c, /,
	  funcs/func_cut.c: Merged revisions 144569 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144569 |
	  murf | 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) | 14 lines
	  (closes issue #13557) Reported by: nickpeirson The user attached
	  a patch, but the license is not yet recorded. I took the liberty
	  of finding and replacing ALL index() calls with strchr() calls,
	  and that involves more than just main/pbx.c; chan_oss,
	  app_playback, func_cut also had calls to index(), and I changed
	  them out. 1.4 had no references to index() at all. ........

	* /, pbx/pbx_lua.c: Merged revisions 144563 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144563 |
	  murf | 2008-09-25 15:54:11 -0600 (Thu, 25 Sep 2008) | 7 lines
	  (closes issue #13559) Reported by: mnicholson Patches:
	  pbx_lua8.diff uploaded by mnicholson (license 96) ........

	* include/asterisk/hashtab.h, /, pbx/pbx_lua.c,
	  configs/extensions.lua.sample: Merged revisions 144523 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r144523 | murf | 2008-09-25 15:18:12 -0600 (Thu, 25 Sep
	  2008) | 13 lines I added a little verbage to hashtab for the
	  hashtab_destroy func. It was pretty sparsely documented. This
	  update fleshes out the pbx_lua module, to add the switch
	  statements to the extensions in the extensions.lua file, as well
	  as removing them when the module is unloaded. Many thanks to Matt
	  Nicholson for his fine contribution! ........

	* /, pbx/pbx_lua.c: Merged revisions 144482 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144482 |
	  murf | 2008-09-25 11:51:11 -0600 (Thu, 25 Sep 2008) | 14 lines
	  (closes issue #13558) Reported by: mnicholson Considering that
	  the example extensions.lua used nothing but ["12345"] notation,
	  and that the resulting error message: [Sep 24 17:01:16]
	  ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua extension:
	  attempt to call a nil value is not very informative as to the
	  nature of the problem, I think this bug fix is a big win!
	  ........

2008-09-23 23:36 +0000 [r144151]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 144149 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144149 |
	  mmichelson | 2008-09-23 18:33:33 -0500 (Tue, 23 Sep 2008) | 3
	  lines Fix a conflict in flag values ........

2008-09-23 17:00 +0000 [r144069]  Steve Murphy <murf@digium.com>

	* /, main/features.c: Merged revisions 144067 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r144067 | murf | 2008-09-23 10:52:32 -0600 (Tue, 23 Sep 2008) |
	  37 lines Merged revisions 144066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) |
	  29 lines (closes issue #13489) Reported by: DougUDI Tested by:
	  murf (closes issue #13490) Reported by: seanbright Tested by:
	  murf (closes issue #13467) Reported by: edantie Tested by: murf,
	  edantie, DougUDI This crash happens because we are unsafely
	  handling old pointers. The channel whose cdr is being handled,
	  has been hung up and destroyed already. I reorganized the code a
	  bit, and tried not to lose the fork-cdr-chain concepts of the
	  previous code. I now verify that the 'previous' channel (the
	  channel we had when the bridge was started), still exists, by
	  looking it up by name in the channel list. I also do not try to
	  reset the CDR's of channels involved in bridges. Testing shows it
	  solves the crash problem, and should not negatively impact
	  previous fixes involving CDR's generated during/after blind
	  transfers. (The reason we need to reset the CDR's on the
	  "beginning" channels in the first place). ........
	  ................

2008-09-23 15:39 +0000 [r144027]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 144025 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r144025 |
	  mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16
	  lines When a promiscuous redirect contained both a user and host
	  portion in the Contact URI and specifies a transport, the parsing
	  done in parse_moved_contact resulted in a malformed URI. This
	  commit fixes the parsing so that a proper Dial string may be
	  formed when the forwarded call is placed. (closes issue #13523)
	  Reported by: mattdarnell Patches: 13523v2.patch uploaded by
	  putnopvut (license 60) Tested by: mattdarnell ........

2008-09-22 22:52 +0000 [r143906]  Sean Bright <sean.bright@gmail.com>

	* /, formats/format_pcm.c: Merged revisions 143904 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r143904 | seanbright | 2008-09-22 18:50:07 -0400
	  (Mon, 22 Sep 2008) | 16 lines Merged revisions 143903 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep
	  2008) | 8 lines Use the advertised header size in .au files
	  instead of just assuming they are 24 bytes (the minimum). (closes
	  issue #13450) Reported by: jamessan Patches: pcm-header.diff
	  uploaded by jamessan (license 246) ........ ................

2008-09-21 10:06 +0000 [r143839-143845]  Michiel van Baak <michiel@vanbaak.info>

	* /, doc/tex/privacy.tex: Merged revisions 143843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r143843 |
	  mvanbaak | 2008-09-21 11:53:01 +0200 (Sun, 21 Sep 2008) | 3 lines
	  fix privacymanager example so it shows how to use the
	  PRIVACYMRGSTATUS variable ........

	* /, doc/tex/privacy.tex: Merged revisions 143840 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r143840 |
	  mvanbaak | 2008-09-21 11:31:54 +0200 (Sun, 21 Sep 2008) | 3 lines
	  document the new context argument for privacymanager so people
	  can do pattern matching on the input ........

	* /, doc/tex/privacy.tex: Merged revisions 143837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r143837 |
	  mvanbaak | 2008-09-21 11:27:08 +0200 (Sun, 21 Sep 2008) | 2 lines
	  fix privacy documentation. We no longer do priority jumping +101
	  ........

2008-09-20 00:55 +0000 [r143739]  Sean Bright <sean.bright@gmail.com>

	* /, contrib/scripts/vmail.cgi: Merged revisions 143737 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r143737 | seanbright | 2008-09-19 20:52:20 -0400
	  (Fri, 19 Sep 2008) | 17 lines Merged revisions 143736 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep
	  2008) | 9 lines Make vmail.cgi work with mailboxes defined in
	  users.conf, too. (closes issue #13187) Reported by: netvoice
	  Patches: 20080911__bug13187.diff.txt uploaded by Corydon76
	  (license 14) (Slightly modified to take alchamist's comments on
	  mantis into account) Tested by: msales, alchamist, seanbright
	  ........ ................

2008-09-19 15:49 +0000 [r143611]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 143609 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r143609 | mmichelson | 2008-09-19 10:43:28 -0500 (Fri, 19 Sep
	  2008) | 11 lines We should only unsubscribe to the device state
	  event subscription if we have previously subscribed. Otherwise a
	  segfault will occur. (closes issue #13476) Reported by: jonnt
	  Patches: 13476.patch uploaded by putnopvut (license 60) Tested
	  by: jonnt ........

2008-09-18 23:55 +0000 [r143561]  Steve Murphy <murf@digium.com>

	* /, channels/chan_sip.c: Merged revisions 143559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r143559 | murf | 2008-09-18 17:41:33 -0600 (Thu, 18 Sep 2008) | 9
	  lines Merged revisions 143534 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1
	  line A micro-fix, in sip_park_thread, where d is freed before the
	  func is done using it. ........ ................

2008-09-17 20:59 +0000 [r143407]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 143405 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r143405 | tilghman | 2008-09-17 15:57:58 -0500
	  (Wed, 17 Sep 2008) | 13 lines Merged revisions 143404 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17 Sep 2008)
	  | 6 lines When callerid is blank, we want to use "unknown caller"
	  in those cases, too. (closes issue #13486) Reported by: tomo1657
	  Patches: 20080917__bug13486.diff.txt uploaded by Corydon76
	  (license 14) ........ ................

2008-09-17 18:30 +0000 [r143349]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, /: Merged revisions 143340 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r143340 | mmichelson | 2008-09-17 13:26:35 -0500 (Wed, 17 Sep
	  2008) | 14 lines Merged revisions 143337 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep
	  2008) | 6 lines Allow for "G.729" if offered in an SDP even
	  though it is not RFC 3551 compliant. Some Cisco switches will
	  send this in an SDP, and it doesn't hurt to be able to accept
	  this. ........ ................

2008-09-15 21:33 +0000 [r143143]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 143141 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r143141 | tilghman | 2008-09-15 16:31:36 -0500
	  (Mon, 15 Sep 2008) | 13 lines Merged revisions 143140 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 Sep 2008)
	  | 6 lines Set the raw formats at the same time as the other
	  formats. (closes issue #13240) Reported by: jvandal Patches:
	  20080813__bug13240.diff.txt uploaded by Corydon76 (license 14)
	  ........ ................

2008-09-14 22:24 +0000 [r143086]  Michiel van Baak <michiel@vanbaak.info>

	* /, channels/chan_skinny.c: Merged revisions 143082 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r143082 | mvanbaak | 2008-09-15 00:16:34 +0200 (Mon, 15 Sep 2008)
	  | 11 lines plug a couple of memleaks in chan_skinny. (closes
	  issue #13452) Reported by: pj Patches: memleak5.diff uploaded by
	  wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak (closes
	  issue #13294) Reported by: pj ........

2008-09-13 13:58 +0000 [r143033]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c, channels/iax2-parser.c, apps/app_dial.c, /:
	  Merged revisions 143031 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r143031 |
	  tilghman | 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) | 8 lines
	  Repair IAXVAR implementation so that it works again (regression?)
	  (closes issue #13354) Reported by: adomjan Patches:
	  20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
	  20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license
	  14) Tested by: Corydon76, adomjan ........

2008-09-12 22:25 +0000 [r142935]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c, /: Merged revisions 142929 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r142929 | jpeeler | 2008-09-12 17:24:13 -0500
	  (Fri, 12 Sep 2008) | 14 lines Merged revisions 142927 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 Sep 2008)
	  | 6 lines (closes issue #12965) Reported by: rlsutton2 Prevents
	  local channels from playing MOH at each other which was causing
	  ast_generic_bridge to loop much faster. ........ ................

2008-09-12 20:52 +0000 [r142743-142868]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  142866 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008)
	  | 18 lines Merged revisions 142865 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008)
	  | 11 lines Create rules for disallowing contacts at certain
	  addresses, which may improve the security of various
	  installations. As this does not change any default behavior, it
	  is not classified as a direct security fix for anything within
	  Asterisk, but may help PBX admins better secure their SIP
	  servers. (closes issue #11776) Reported by: ibc Patches:
	  20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76, blitzrage ........ ................

	* /, main/app.c: Merged revisions 142748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r142748 |
	  tilghman | 2008-09-12 11:54:44 -0500 (Fri, 12 Sep 2008) | 3 lines
	  When checking for an encoded character, make sure the string
	  isn't blank, first. (Closes issue #13470) ........

	* apps/app_voicemail.c, /: Merged revisions 142745 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r142745 | tilghman | 2008-09-12 11:38:55 -0500
	  (Fri, 12 Sep 2008) | 12 lines Merged revisions 142744 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12 Sep 2008)
	  | 4 lines Missing merge from 1.2 fixes errant exit on DTMF, only
	  when language is Italian (cf commit 34242) (Closes issue #7353)
	  ........ ................

	* /, main/file.c: Merged revisions 142741 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142741 | tilghman | 2008-09-12 11:29:01 -0500 (Fri, 12 Sep 2008)
	  | 12 lines Merged revisions 142740 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008)
	  | 4 lines Don't return a free'd pointer, when a file cannot be
	  opened. (closes issue #13462) Reported by: wackysalut ........
	  ................

2008-09-12 05:03 +0000 [r142632-142678]  Steve Murphy <murf@digium.com>

	* apps/app_queue.c, apps/app_dial.c, main/pbx.c, /,
	  main/features.c, include/asterisk/channel.h: Merged revisions
	  142676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) |
	  40 lines Merged revisions 142675 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) |
	  29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is
	  a "second attempt" to restore the previous "endbeforeh" behavior
	  in 1.4 and up. In order to capture information concerning all the
	  legs of transfers in all their infinite combinations, I was
	  forced to this particular solution by a chain of logical
	  necessities, the first being that I was not allowed to rewrite
	  the CDR mechanism from the ground up! This change basically
	  leaves the original machinery alone, which allows IVR and local
	  channel type situations to generate CDR's as normal, but a
	  channel flag can be set to suppress the normal running of the h
	  exten. That flag would be set by the code that runs the h exten
	  from the ast_bridge_call routine, to prevent the h exten from
	  being run twice. Also, a flag in the ast_bridge_config struct
	  passed into ast_bridge_call can be used to suppress the running
	  of the h exten in that routine. This would happen, for instance,
	  if you use the 'g' option in the Dial app. Running this routine
	  'early' allows not only the CDR() func to be used in the h
	  extension for reading CDR variables, but also allows them to be
	  modified before the CDR is posted to the backends. While I dearly
	  hope that this patch overcomes all problems, and introduces no
	  new problems, reality suggests that surely someone will have
	  problems. In this case, please re-open 13251 (or 13289), and
	  we'll see if we can't fix any remaining issues. ** trunk note:
	  some code to suppress the h exten being run from app_queue was
	  added; for the 'continue' option available only in trunk/1.6.x.
	  ........ ................

	* /, main/features.c: Merged revisions 142576 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142576 | murf | 2008-09-11 17:12:53 -0600 (Thu, 11 Sep 2008) |
	  28 lines Merged revisions 142575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) |
	  20 lines (closes issue #13364) Reported by: mdu113 Well,
	  fundamentally, the problems revealed in 13364 are because of the
	  ForkCDR call that is done before the dial. When the bridge is in
	  place, it's dealing with the first (and wrong) cdr in the list.
	  So, I wrote a little func to zip down to the first non-locked cdr
	  in the chain, and thru-out the ast_bridge_call, these results are
	  used instead of raw chan->cdr and peer->cdr pointers. This
	  shouldn't affect anyone who isn't forking cdrs before a dial, and
	  should correct the cdr's of those that do. So, this change ends
	  up correcting the dstchannel and userfield; the disposition was
	  fixed by a previous patch, it was OK coming into this problem.
	  ........ ................

2008-09-10 22:18 +0000 [r142478]  Steve Murphy <murf@digium.com>

	* /, main/features.c: Merged revisions 142475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142475 | murf | 2008-09-10 16:11:27 -0600 (Wed, 10 Sep 2008) |
	  38 lines Merged revisions 142474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) |
	  30 lines (closes issue #12318) Reported by: krtorio I made a
	  small change to the code that handles local channel situations.
	  In that code, I copy the answer time from the peer cdr, to the
	  bridge_cdr, but I wasn't also copying the disposition from the
	  peer cdr. So, Now I copy the disposition, and I've tested against
	  these cases: 1. phone 1 never answers the phone; no cdr is
	  generated at all. this should show up as a manager command
	  failure or something. 2. phone 2 never answers. CDR is generated,
	  says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4.
	  phone 2 answers: CDR is generated, times are correct; disposition
	  is ANSWERED, which is correct. The start time is the time that
	  the manager dialed the first phone. The answer time is the time
	  the second phone picks up. I purposely left the cid and src
	  fields blank; since this call really originates from the manager,
	  there is no 'easy' data to put in these fields. If you feel
	  strongly that these fields should be filled in, re-open this bug
	  and I'll dig further. ........ ................

2008-09-10 19:14 +0000 [r142419]  Sean Bright <sean.bright@gmail.com>

	* /, configure, acinclude.m4: Merged revisions 142417 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r142417 | seanbright | 2008-09-10 15:09:03 -0400
	  (Wed, 10 Sep 2008) | 17 lines Merged revisions 142416 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, 10 Sep
	  2008) | 9 lines Fix detection of PWLIB and OpenH323 version when
	  spacing in the headers isn't consistent. (closes issue #13426)
	  Reported by: bamby Patches: detect_openh323.diff uploaded by
	  bamby (license 430) (Modified by me to use sed instead of tr)
	  ........ ................

2008-09-10 16:57 +0000 [r142361]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile, /: Merged revisions 142359 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142359 | tilghman | 2008-09-10 11:55:31 -0500 (Wed, 10 Sep 2008)
	  | 10 lines Merged revisions 142358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008)
	  | 2 lines Publish new extra sounds version. ........
	  ................

2008-09-10 16:42 +0000 [r142357]  Russell Bryant <russell@digium.com>

	* main/sched.c, /: Merged revisions 142355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142355 | russell | 2008-09-10 11:41:55 -0500 (Wed, 10 Sep 2008)
	  | 15 lines Merged revisions 142354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008)
	  | 7 lines It is a normal situation that a task gets put in the
	  scheduler that should run as soon as possible. Accept "0" as an
	  acceptable time to run, and also treat negative as "run now", and
	  don't print a debug message about it. (inspired by a message
	  asking about the "request to schedule in the past" debug message
	  on the -dev list) ........ ................

2008-09-09 19:18 +0000 [r142082-142221]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 142219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142219 | mmichelson | 2008-09-09 14:16:30 -0500 (Tue, 09 Sep
	  2008) | 22 lines Merged revisions 142218 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep
	  2008) | 14 lines Make sure that the branch sent in CANCEL
	  requests matches the branch of the INVITE it is cancelling.
	  (closes issue #13381) Reported by: atca_pres Patches:
	  13381v2.patch uploaded by putnopvut (license 60) Tested by:
	  atca_pres (closes issue #13198) Reported by: rickead2000 Tested
	  by: rickead2000 ........ ................

	* apps/app_queue.c: Merging Revision 142090 from 1.6.0.

	* /, channels/chan_sip.c: Merged revisions 142080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r142080 | mmichelson | 2008-09-09 11:20:41 -0500 (Tue, 09 Sep
	  2008) | 29 lines Merged revisions 142079 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep
	  2008) | 21 lines When determining if codecs used by SIP peers
	  allow the media to be natively bridged, use the jointcapability
	  instead of the peercapability. It seems that the intent of using
	  the peercapability was to expand the choice of codecs for the
	  call to increase the chances of being able to native bridge the
	  channels. The problem is that if a codec were settled on for the
	  native bridge and that wasn't a codec that was configured to be
	  used by Asterisk for that peer, then Asterisk would send a
	  REINVITE with no codecs in the SDP which is a bug no matter how
	  you slice it. (closes issue #13076) Reported by: ramonpeek
	  Patches: 13076.patch uploaded by putnopvut (license 60) Tested
	  by: tbelder ........ ................

2008-09-09 15:46 +0000 [r142066]  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 [r141997-142001]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 141998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r141998 |
	  mmichelson | 2008-09-09 07:32:38 -0500 (Tue, 09 Sep 2008) | 7
	  lines Use ast_debug for debug messages. I was wondering why debug
	  messages weren't showing up when I had set the debug level high
	  for just app_queue.c. It's because we were only checking the
	  global option_debug variable instead of using the awesome macro
	  which checks both the global and file-specific value ........

	* 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 01:51 +0000 [r141951-141952]  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 ........ ................

	* 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 22:15 +0000 [r141812-141870]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 141868 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r141868 |
	  mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4
	  lines Um, apparently I didn't actually finish merging before
	  committing. Bad bad bad ........

	* /, channels/chan_sip.c: Merged revisions 141810 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep
	  2008) | 22 lines Merged revisions 141809 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep
	  2008) | 14 lines Fix pedantic mode of chan_sip to only check the
	  remote tag of an endpoint once a dialog has been confirmed. Up
	  until that point, it is possible and legal for the far-end to
	  send provisional responses with a different To: tag each time.
	  With this patch applied, these provisional messages will not
	  cause a matching problem. (closes issue #11536) Reported by: ibc
	  Patches: 11536v2.patch uploaded by putnopvut (license 60)
	  ........ ................

2008-09-08 20:20 +0000 [r141747]  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 17:15 +0000 [r141684]  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 20:23 +0000 [r141572]  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 15:28 +0000 [r141506]  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 21:13 +0000 [r141369]  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 14:25 +0000 [r141117-141159]  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 ........
	  ................

	* res/ael/ael.flex, 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: 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 17:28 +0000 [r141042]  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 [r140888-140977]  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. ........

	* apps/app_voicemail.c, /: Merged revisions 140887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r140887 | mmichelson | 2008-09-03 09:41:54 -0500 (Wed, 03 Sep
	  2008) | 3 lines Fix compilation ........

2008-09-03 14:39 +0000 [r140886]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y,
	  res/ael/ael.tab.h: Merged revisions 140824 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r140824 |
	  murf | 2008-09-03 08:01:27 -0600 (Wed, 03 Sep 2008) | 21 lines In
	  these changes, I have added some explanation of changes to the
	  Set and MSet apps, so people aren't so shocked and surprised when
	  they upgrade from 1.4 to 1.6. Also, for the sake of those
	  upgrading from 1.4 to 1.6 with AEL, I provide automatic support
	  for the "old" way of using Set(), that still does the exact same
	  old thing with quotes and backslashes and so on as 1.4 did, by
	  having AEL compile in the use of MSet() instead of Set(),
	  everywhere it inserts this code. But, if the app_set var is set
	  to 1.6 or higher, it uses the "new", non-evaluative Set(). This
	  only usually happens if the user manually inserts this into the
	  asterisk.conf file, or runs the "make samples" command. (closes
	  issue #13249) Reported by: dimas Patches: ael-MSet.diff uploaded
	  by murf (license 17) Tested by: dimas, murf ........

2008-09-03 14:32 +0000 [r140867]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 140860 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140860 | mmichelson | 2008-09-03 09:31:33 -0500
	  (Wed, 03 Sep 2008) | 17 lines Merged revisions 140850 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed, 03 Sep
	  2008) | 9 lines Fix voicemail forwarding when using ODBC storage.
	  (closes issue #13387) Reported by: moliveras Patches: 13387.patch
	  uploaded by putnopvut (license 60) Tested by: putnopvut,
	  moliveras ........ ................

2008-09-03 13:27 +0000 [r140819]  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 23:51 +0000 [r140755]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 140752 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140752 | mmichelson | 2008-09-02 18:48:25 -0500
	  (Tue, 02 Sep 2008) | 14 lines Merged revisions 140751 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, 02 Sep
	  2008) | 6 lines After adding the context checking to
	  app_voicemail for IMAP storage, I left out a crucial place to
	  copy the context to the vm_state structure. This is the
	  correction. ........ ................

2008-09-02 23:46 +0000 [r140693-140750]  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. ........ ................

2008-09-02 18:18 +0000 [r140608]  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 15:13 +0000 [r140565-140568]  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 ........

	* res/ais/amf.c (removed), res/ais/lck.c (removed), /,
	  res/ais/ckpt.c (removed): Merged revisions 140563 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r140563 | russell | 2008-09-02 10:09:20 -0500 (Tue, 02 Sep 2008)
	  | 3 lines I'm not sure how these files got to trunk (probably my
	  fault), but they should not be here ........

2008-08-29 17:55 +0000 [r140492]  Jeff Peeler <jpeeler@digium.com>

	* CHANGES, /, main/features.c: Merged revisions 140491 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r140491 | jpeeler | 2008-08-29 12:53:32 -0500 (Fri, 29 Aug 2008)
	  | 2 lines Added the option s to the Park application which will
	  silence the announcement of the parking space number. Also, fixes
	  the bug of just clearing the flags instead of actually parsing
	  the arguments to Park. ........

2008-08-29 17:48 +0000 [r140420-140490]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, channels/chan_iax2.c, main/config.c,
	  main/manager.c, res/ais/lck.c, /, channels/chan_sip.c,
	  funcs/func_dialgroup.c, res/res_timing_pthread.c,
	  main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
	  channels/chan_console.c, main/taskprocessor.c: Merged revisions
	  140489 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140489 | mmichelson | 2008-08-29 12:47:17 -0500 (Fri, 29 Aug
	  2008) | 30 lines Merged revisions 140488 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug
	  2008) | 22 lines After working on the ao2_containers branch, I
	  noticed something a bit strange. In all cases where we provide a
	  callback function to ao2_container_alloc, the callback function
	  would only return 0 or CMP_MATCH. After inspecting the
	  ao2_callback() code carefully, I found that if you're only
	  looking for one specific item, then you should return CMP_MATCH |
	  CMP_STOP. Otherwise, astobj2 will continue traversing the current
	  bucket until the end searching for more matches. In cases like
	  chan_iax2 where in 1.4, all the peers are shoved into a single
	  bucket, this makes for potentially terrible performance since the
	  entire bucket will be traversed even if the peer is one of the
	  first ones come across in the bucket. All the changes I have made
	  were for cases where the callback function defined was passed to
	  ao2_container_alloc so that calls to ao2_find could find a unique
	  instance of whatever object was being stored in the container.
	  ........ ................

	* /, main/file.c: Merged revisions 140433 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r140433 |
	  mmichelson | 2008-08-29 11:24:37 -0500 (Fri, 29 Aug 2008) | 10
	  lines Allow for video files to be opened as well as audio files.
	  (closes issue #13372) Reported by: epicac Patches: 13372.patch
	  uploaded by putnopvut (license 60) Tested by: epicac ........

	* apps/app_voicemail.c, /: Merged revisions 140422 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140422 | mmichelson | 2008-08-29 11:06:09 -0500
	  (Fri, 29 Aug 2008) | 20 lines Merged revisions 140421 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, 29 Aug
	  2008) | 12 lines Add context checking when retrieving a vm_state.
	  This was causing a problem for people who had identically named
	  mailboxes in separate voicemail contexts. This commit affects
	  IMAP storage only. (closes issue #13194) Reported by: moliveras
	  Patches: 13194.patch uploaded by putnopvut (license 60) Tested
	  by: putnopvut, moliveras ........ ................

	* /, channels/chan_sip.c: Merged revisions 140418 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140418 | mmichelson | 2008-08-29 10:32:02 -0500 (Fri, 29 Aug
	  2008) | 18 lines Merged revisions 140417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug
	  2008) | 10 lines Fix SIP's parsing so that if a port is specified
	  in a string to Dial(), it is not ignored. (closes issue #13355)
	  Reported by: acunningham Patches: 13355v2.patch uploaded by
	  putnopvut (license 60) Tested by: acunningham ........
	  ................

2008-08-27 20:14 +0000 [r140303]  Mark Michelson <mmichelson@digium.com>

	* /, 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 18:50 +0000 [r140206]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 140205 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ 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. ........ ................

2008-08-26 18:12 +0000 [r140055-140171]  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.
	  ........

	* /, channels/chan_sip.c: Merged revisions 140061 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r140061 | russell | 2008-08-26 11:10:06 -0500 (Tue, 26 Aug 2008)
	  | 14 lines Merged revisions 140060 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008)
	  | 6 lines Fix some bogus scheduler usage in chan_sip. This code
	  used the return value of a completely unrelated function to
	  determine whether the scheduler should be run or not. This would
	  have caused the scheduler to not run in cases where it should
	  have. Also, leave a note about another scheduler issue that needs
	  to be addressed at some point. ........ ................

	* channels/chan_iax2.c, /: Merged revisions 140053 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r140053 | russell | 2008-08-26 10:29:25 -0500
	  (Tue, 26 Aug 2008) | 23 lines Merged revisions 140051 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008)
	  | 15 lines Fix a race condition with the IAX scheduler thread. A
	  lock and condition are used here to allow newly scheduled tasks
	  to wake up the scheduler just in case the new task needs to run
	  sooner than the current wakeup time when the thread is sleeping.
	  However, there was a race condition such that a newly scheduled
	  task would not properly wake up the scheduler or affect the wake
	  up period. The order of execution would have been: 1) Scheduler
	  thread determines wake up time of N ms. 2) Another thread
	  schedules a task and signals the condition, with an execution
	  time of < N ms. 3) Scheduler thread locks and goes to sleep for N
	  ms. By moving the sleep time determination to inside the critical
	  section, this possibility is avoided. ........ ................

2008-08-25 21:49 +0000 [r139929]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, /: Merged revisions 139928 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r139928 | jpeeler | 2008-08-25 16:48:51 -0500 (Mon, 25 Aug 2008)
	  | 11 lines Merged revisions 139927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008)
	  | 3 lines Fix a typo I made. Lesson learned, apply the patch if
	  one exists. ........ ................

2008-08-25 21:34 +0000 [r139919]  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 21:13 +0000 [r139874]  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 16:05 +0000 [r139778]  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 16:36 +0000 [r139709]  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 22:37 +0000 [r139629-139674]  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 21:58 +0000 [r139623-139625]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, /: Merged revisions 139624 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  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)
	  ........ ................

	* /, main/features.c: Merged revisions 139622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r139622 |
	  jpeeler | 2008-08-22 16:52:20 -0500 (Fri, 22 Aug 2008) | 1 line
	  remove extra comma typo ........

2008-08-22 20:21 +0000 [r139459-139565]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 139563 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r139563 |
	  mmichelson | 2008-08-22 15:20:58 -0500 (Fri, 22 Aug 2008) | 15
	  lines The -1 return value from incomplete or improper headers for
	  the SipNotify manager command was causing the current manager
	  session to become disconnected. Change the return value to 0 for
	  these cases. Also change a test for a NULL pointer to be
	  ast_strlen_zero instead. (closes issue #13351) Reported by:
	  Laureano Patches: sipnotify_action_fix.patch uploaded by Laureano
	  (license 265) ........

	* /, main/features.c: Merged revisions 139558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r139558 |
	  mmichelson | 2008-08-22 15:02:35 -0500 (Fri, 22 Aug 2008) | 9
	  lines Add missing unique id to ParkedCallGiveUp and
	  ParkedCallTimeOut manager events (closes issue #13358) Reported
	  by: srt Patches: 13358_parking_events.diff uploaded by srt
	  (license 378) ........

	* 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 23:44 +0000 [r139399]  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) ........ ................

2008-08-20 22:19 +0000 [r139217]  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 22:07 +0000 [r139212]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 139210 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r139210 |
	  qwell | 2008-08-20 17:06:40 -0500 (Wed, 20 Aug 2008) | 7 lines
	  Fix output of sipshowpeer manager response. (closes issue #13346)
	  Reported by: srt Patches:
	  13346_malformed_sip_show_peer_response.diff uploaded by srt
	  (license 378) ........

2008-08-20 17:34 +0000 [r139106]  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 [r138888-139018]  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
	  ........ ................

	* /, channels/chan_agent.c: Merged revisions 138943 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r138943 | mmichelson | 2008-08-19 18:19:40 -0500
	  (Tue, 19 Aug 2008) | 19 lines Merged revisions 138942 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug
	  2008) | 11 lines Reset agent_pvt variables back to the values in
	  agents.conf (from what the corresponding channel variables were
	  set to) when the agent logs out. (closes issue #13098) Reported
	  by: davidw Patches:
	  20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
	  bbryant (license 36) Tested by: davidw ........ ................

	* apps/app_chanspy.c, /: Merged revisions 138887 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r138887 | mmichelson | 2008-08-19 13:52:04 -0500 (Tue, 19 Aug
	  2008) | 31 lines Merged revisions 138886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug
	  2008) | 23 lines Add a lock and unlock prior to the destruction
	  of the chanspy_ds lock to ensure that no other threads still have
	  it locked. While this should not happen under normal
	  circumstances, it appears that if the spyer and spyee hang up at
	  nearly the same time, the following may occur. 1.
	  ast_channel_free is called on the spyee's channel. 2. The chanspy
	  datastore is removed from the spyee's channel in
	  ast_channel_free. 3. In the spyer's thread, the spyer attempts to
	  remove and destroy the datastore from the spyee channel, but the
	  datastore has already been removed in step 2, so the spyer
	  continues in the code. 4. The spyee's thread continues and calls
	  the datastore's destroy callback, chanspy_ds_destroy. This
	  involves locking the chanspy_ds. 5. Now the spyer attempts to
	  destroy the chanspy_ds lock. The problem is that in step 4, the
	  spyee has locked this lock, meaning that the spyer is attempting
	  to destroy a lock which is currently locked by another thread.
	  The backtrace provided in issue #12969 supports the idea that
	  this is possible (and has even occurred). This commit does not
	  close the issue, but should help in preventing one type of crash
	  associated with the use of app_chanspy. ........ ................

2008-08-19 17:01 +0000 [r138853-138855]  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 00:17 +0000 [r138777-138782]  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 20:24 +0000 [r138689-138698]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
	  138694 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138694 |
	  mmichelson | 2008-08-18 15:23:11 -0500 (Mon, 18 Aug 2008) | 10
	  lines Change the queue timeout priority logic into less ugly and
	  confusing code pieces. Clarify the logic within
	  queues.conf.sample. (closes issue #12690) Reported by: atis
	  Patches: queue_timeoutpriority.patch uploaded by atis (license
	  242) ........

	* 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 15:55 +0000 [r138633]  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 [r138520]  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 14:27 +0000 [r138444-138498]  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 13:08 +0000 [r138411-138414]  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) ........

	* main/pbx.c, /: Merged revisions 138409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138409 |
	  tilghman | 2008-08-16 07:52:06 -0500 (Sat, 16 Aug 2008) | 3 lines
	  Also make sure hinting won't crash on reload. (Closes issue
	  #13312) ........

2008-08-16 01:14 +0000 [r138359-138363]  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 22:56 +0000 [r138208-138261]  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 ........
	  ................

	* /: Merged revisions 138207 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ ........

2008-08-15 20:28 +0000 [r138184]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 138155 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r138155 | jpeeler | 2008-08-15 15:12:19 -0500 (Fri, 15 Aug 2008)
	  | 1 line rename all zfd instances in chan_dahdi to dfd to match
	  1.4 (left over from DAHDI transition) ........

2008-08-15 20:21 +0000 [r138158]  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 19:37 +0000 [r138026-138150]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 138148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138148 |
	  tilghman | 2008-08-15 14:36:11 -0500 (Fri, 15 Aug 2008) | 2 lines
	  Change free to ast_free_ptr, too ........

	* main/pbx.c, /: Merged revisions 138124 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138124 |
	  tilghman | 2008-08-15 14:22:48 -0500 (Fri, 15 Aug 2008) | 4 lines
	  e->data can be NULL, so use the safe version of ast_strdup()
	  (closes issue #13312) Reported by: pj ........

	* /, channels/chan_sip.c: Merged revisions 138086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r138086 |
	  tilghman | 2008-08-15 13:02:15 -0500 (Fri, 15 Aug 2008) | 2 lines
	  regseconds is actually stored as the epoch time, not registration
	  length ........

	* /, 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 [r137989]  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 18:50 +0000 [r137935]  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 18:15 +0000 [r137904]  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 15:39 +0000 [r137815]  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 15:35 +0000 [r137813]  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 15:06 +0000 [r137782]  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 02:08 +0000 [r137648-137683]  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 ........
	  ................

	* /: configure for merging from trunk

	* / (added): now that 1.6.0 has reached the 'release candidate'
	  stage, it's time to branch 1.6.1

2008-08-13 23:00 +0000 [r137627-137640]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/prep_tarball: make this script actually work

	* /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: 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 21:08 +0000 [r137496-137532]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: 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: Add FAXMODE variable with what fax transport was
	  used. (closes issue #13252) Patches: v1-13252.patch uploaded by
	  dimas (license 88)

2008-08-13 17:36 +0000 [r137456]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Convert deprecated routines to the new names.
	  (closes issue #13297) Reported by: snuffy Patches:
	  bug13297_20080814.diff uploaded by snuffy (license 35)

2008-08-13 14:41 +0000 [r137403-137406]  Sean Bright <sean.bright@gmail.com>

	* /, doc/tex/cdrdriver.tex: 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: Use the ast_vasprintf macro instead of vasprintf
	  directly.

2008-08-12 19:48 +0000 [r137299-137301]  Russell Bryant <russell@digium.com>

	* doc/tex/asterisk.tex: Grammar hax from Qwell

	* doc/tex/asterisk.tex: Note that developer documentation belongs
	  in doxygen, and not integrated with the user manual stuff in
	  doc/tex/.

2008-08-11 16:14 +0000 [r137239]  Russell Bryant <russell@digium.com>

	* Makefile: Make PRINT_DIR work as advertised.

2008-08-11 14:25 +0000 [r137203]  Sean Bright <sean.bright@gmail.com>

	* UPGRADE.txt, cdr/cdr_tds.c: 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 00:25 +0000 [r137150]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_odbc.c: 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 ........

2008-08-10 21:10 +0000 [r137028-137112]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/channel.h: Fix this again so we can compile with
	  shadow warnings enabled and IMAP chosen in voicemail.

	* main/udptl.c, main/say.c, main/taskprocessor.c, main/sched.c:
	  That's all, folks. Not going to update the Makefile until
	  res_jabber is converted (snuffy, you there? :))

	* main/channel.c, main/pbx.c, main/frame.c, main/logger.c,
	  apps/app_queue.c, main/indications.c, main/asterisk.c,
	  main/rtp.c, apps/app_voicemail.c, main/cli.c: Another batch of
	  files from RSW. The remaining apps and a few more files from
	  main/

	* main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_delete.c,
	  main/jitterbuf.c, main/acl.c, main/db1-ast/recno/rec_put.c,
	  main/astobj2.c, main/config.c, main/rtp.c, main/channel.c,
	  main/cdr.c, main/manager.c, main/tdd.c, main/features.c,
	  main/abstract_jb.c, main/file.c, main/http.c, main/callerid.c,
	  main/app.c, main/event.c, main/audiohook.c,
	  main/db1-ast/btree/bt_delete.c, main/asterisk.c: Another big
	  chunk of changes from the RSW branch. Bunch of stuff from main/

	* apps/app_dial.c, apps/app_dahdibarge.c, apps/app_meetme.c,
	  apps/app_festival.c, apps/app_record.c, apps/app_dahdiscan.c,
	  apps/app_disa.c, apps/app_waituntil.c, apps/app_playback.c,
	  apps/app_forkcdr.c, apps/app_osplookup.c, apps/app_minivm.c,
	  apps/app_macro.c, apps/app_sms.c, apps/app_directory.c,
	  apps/app_rpt.c, apps/app_while.c, apps/app_adsiprog.c: More RSW
	  merges. Everything from apps/ except for the big offenders
	  app_voicemail and app_queue.

	* res/res_config_pgsql.c, res/res_smdi.c, res/res_timing_pthread.c,
	  res/res_adsi.c, res/res_agi.c, res/res_phoneprov.c,
	  res/ael/ael_lex.c, res/res_musiconhold.c, res/ael/ael.flex,
	  res/res_config_ldap.c, res/res_odbc.c: All of the res/ stuff
	  (other than res_jabber) from the RSW branch.

2008-08-09 15:26 +0000 [r136947]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/compat.h, include/asterisk/astobj2.h: 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 14:12 +0000 [r136888-136917]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_unistim.c, channels/chan_sip.c,
	  channels/chan_skinny.c, codecs/codec_dahdi.c,
	  channels/chan_iax2.c, channels/xpmr/xpmr.c,
	  channels/iax2-parser.c, channels/chan_mgcp.c: More RSW merges.
	  This should do it for the channels/ dir.

	* channels/chan_dahdi.c: Biggest offender? chan_dahdi.c! More RSW
	  merging.

	* channels/chan_jingle.c, channels/chan_phone.c,
	  channels/chan_agent.c, channels/chan_features.c,
	  channels/chan_alsa.c, channels/chan_console.c: Merge more changes
	  from the resolve-shadow-warnings branch (henceforth known as RSW
	  since i am too lazy to keep typing it all out). This time a few
	  of the channels.

2008-08-09 01:15 +0000 [r136859]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: 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 18:19 +0000 [r136819]  Sean Bright <sean.bright@gmail.com>

	* configure, configure.ac, makeopts.in: Bring in the configure and
	  makeopts jazz for -Wshadow, but don't add it to the Makefile yet.

2008-08-08 15:58 +0000 [r136787]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_dahdi.c: use ARRAY_LEN

2008-08-08 15:31 +0000 [r136784]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix compilation for ODBC voicemail

2008-08-08 02:34 +0000 [r136751]  Tilghman Lesher <tlesher@digium.com>

	* /: Removing bad properties

2008-08-08 00:48 +0000 [r136746]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /, pbx/ael/ael-test/ref.ael-ntest10,
	  include/asterisk/ael_structs.h, 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 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. ........

2008-08-07 23:39 +0000 [r136715-136722]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Remove one last batch of debug messages

	* apps/app_voicemail.c: Fix build for non-IMAP storage and get rid
	  of some debug messages. Thanks to eliel for alerting me. No
	  thanks to buildbot.

	* /, apps/app_voicemail.c: 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 21:19 +0000 [r136679]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: show correct called party id and also
	  store this to the 'placed calls' list once the call is connected.
	  (closes issue #13180) Reported by: pj Patches:
	  2008080700_skinny_calledpartyid.diff uploaded by mvanbaak
	  (license 7) Tested by: mvanbaak, pj

2008-08-07 20:54 +0000 [r136676]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Updating codec_dahdi to the new transcoder
	  interface.

2008-08-07 20:25 +0000 [r136631-136660]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Bump a LOG_NOTICE message to LOG_DEBUG since it
	  appears once for every bridged call

	* main/pbx.c: Don't allow Answer() to accept a negative argument.
	  Negative argument means an infinite delay and we don't want that.

	* main/channel.c: 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: 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 19:01 +0000 [r136594]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /, configs/misdn.conf.sample,
	  channels/misdn_config.c: 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 [r136504-136542]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/doxyref.h, /: Merged revisions 136541 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ ........

	* apps/app_jack.c: stop using deprecated API call

2008-08-07 16:55 +0000 [r136489]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: 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
	  ........

2008-08-07 16:29 +0000 [r136477]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: fix some format strings to actually compile
	  without errors

2008-08-07 15:16 +0000 [r136408]  Sean Bright <sean.bright@gmail.com>

	* codecs/Makefile, utils/muted.c, utils/astman.c, utils/smsq.c,
	  codecs/codec_dahdi.c, formats/msgsm.h, utils/extconf.c,
	  utils/frame.c: More merges from resolve-shadow warnings: utils/
	  codecs/ and a change I missed from formats/

2008-08-07 15:10 +0000 [r136406]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_tds.c: Fix runtime symbol error

2008-08-07 14:36 +0000 [r136298-136402]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/callerid.h, include/asterisk/strings.h: Merge in
	  a few more changes. This time the include/ directory.

	* funcs/func_config.c, funcs/func_timeout.c, funcs/func_odbc.c,
	  funcs/func_strings.c: Continue merging in changes from
	  resolve-shadow-warnings. funcs/ this time.

	* cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
	  cdr/cdr_tds.c, cdr/cdr_csv.c: More from the
	  resolve-shadow-warnings branch. This time the cdr/ directory.

	* formats/format_pcm.c, pbx/pbx_dundi.c, formats/msgsm.h,
	  pbx/dundi-parser.c, pbx/pbx_config.c: Start moving in changes
	  from my resolve-shadow-warnings branch. Going to do this in
	  pieces so the diffs are a little bit smaller and more reviewable.
	  pbx/ and formats/ first.

2008-08-06 21:22 +0000 [r136245]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/taskprocessor.c: move taskprocessor CLI commands into the
	  core namespace

2008-08-06 20:15 +0000 [r136112-136191]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/init.d/rc.redhat.asterisk: 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 ........

	* apps/app_meetme.c: Janitor ast_str project (closes issue #13058)
	  Reported by: pputman Patches: app_meetme_aststr2.patch uploaded
	  by pputman (license 81)

	* funcs/func_dialgroup.c: 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 15:59 +0000 [r136063]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_skinny.c, main/rtp.c: 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 14:51 +0000 [r136034]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Use a dynamic buffer for rendered SQL, instead
	  of hardcoding 2048 bytes. Also, switch to using RWLISTs for the
	  linked list of queries.

2008-08-06 13:34 +0000 [r136005]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: - 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 03:55 +0000 [r135900-135950]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: 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 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/utils.h, /, include/asterisk/threadstorage.h:
	  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 00:30 +0000 [r135851]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /, main/abstract_jb.c, main/fixedjitterbuf.h,
	  include/asterisk/abstract_jb.h: 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 23:45 +0000 [r135821]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c,
	  include/asterisk/cdr.h: 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 21:37 +0000 [r135717-135748]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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). ........

	* include/asterisk/config.h, UPGRADE.txt, CHANGES, main/config.c:
	  Add '+=' append operator to configuration files.

2008-08-05 17:05 +0000 [r135680-135681]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/datastore.h, include/asterisk/channel.h:
	  datastore inheritance is a channel feature, so move this
	  definition back

	* apps/app_dial.c, funcs/func_speex.c, main/pbx.c, main/Makefile,
	  funcs/func_lock.c, pbx/pbx_lua.c, include/asterisk/channel.h,
	  apps/app_queue.c, channels/chan_iax2.c,
	  include/asterisk/manager.h, funcs/func_global.c,
	  apps/app_speech_utils.c, main/channel.c, funcs/func_enum.c,
	  main/manager.c, res/res_smdi.c, funcs/func_odbc.c,
	  funcs/func_volume.c, res/res_agi.c, include/asterisk/datastore.h
	  (added), pbx/pbx_dundi.c, main/audiohook.c, apps/app_chanspy.c,
	  apps/app_stack.c, main/datastore.c (added): make datastore
	  creation and destruction a generic API since it is not really
	  channel related, and add the ability to add/find/remove
	  datastores to manager sessions

2008-08-05 15:30 +0000 [r135648]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/make_version: Always output a version string, even
	  when we can't figure out what we are. (Closes issue #13223)

2008-08-05 13:26 +0000 [r135598]  Sean Bright <sean.bright@gmail.com>

	* /, main/cli.c: 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 [r135537]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample, /: 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 [r135476-135485]  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): Rename Mandrake
	  scripts to Mandriva (Closes issue #13221)

	* contrib/init.d/rc.mandrake.asterisk, /: 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 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) ........

	* include/asterisk/http.h, main/http.c, res/res_http_post.c: HTTP
	  module memory leaks (closes issue #13230) Reported by: eliel
	  Patches: res_http_post_leak.patch uploaded by eliel (license 64)

2008-08-04 16:28 +0000 [r135439-135474]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample, /: 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: 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.

2008-08-03 16:14 +0000 [r135405]  Sean Bright <sean.bright@gmail.com>

	* build_tools/cflags.xml, doc/hoard.txt (added),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
	  CHANGES, makeopts.in: Merge in changes that allow Asterisk to be
	  built against the Hoard memory allocator. See doc/hoard.txt for
	  more details.

2008-08-03 00:03 +0000 [r135332-135373]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: whitespace fixes only.

	* channels/chan_skinny.c: Dont coredump on register of
	  non-configured devices (closes issue #13224) Reported by:
	  mvanbaak Patches: noncon.diff uploaded by wedhorn (license 30)
	  with whitespace fixes by me Tested by: wedhorn, mvanbaak

	* channels/chan_skinny.c: make this work again, and not segfault on
	  device registration

2008-08-02 13:21 +0000 [r135302]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_skinny.c: --enable-dev-mode is your friend :-)

2008-08-02 12:29 +0000 [r135300]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: pass device instead of session to
	  transmit_ functions. (closes issue #10396) Reported by: wedhorn
	  Patches: transmit3a.diff uploaded by wedhorn (license 30) Tested
	  by: wedhorn, mvanbaak

2008-08-02 04:51 +0000 [r135265]  Steve Murphy <murf@digium.com>

	* main/pbx.c, main/features.c: (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.

2008-08-01 21:56 +0000 [r135235]  Terry Wilson <twilson@digium.com>

	* main/http.c, res/res_http_post.c: Fix mime parsing by re-adding
	  support for passing headers to callback functions

2008-08-01 19:29 +0000 [r135197]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_mgcp.c: 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 18:16 +0000 [r135158]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, channels/iax2.h, CHANGES,
	  channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from
	  team/bbryant/keyrotation This set of changes enhances IAX2
	  encryption support by adding key rotation to provide enhanced
	  security. The key used for encryption is rotated right after the
	  call gets set up, and then again every few minutes. This was
	  discussed at the last AstriDevCon. For interoperability with
	  older versions of Asterisk, there is an option that disables key
	  rotation. (closes issue #13018) Reported by: bbryant Patches:
	  07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
	  Tested by: russell, bbryant

2008-08-01 17:09 +0000 [r135126-135128]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Picky, picky, buildbot

	* channels/chan_sip.c, configs/sip.conf.sample: 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 14:42 +0000 [r135067-135068]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: IMAP-specific items must go in IMAP_STORAGE
	  defines...

	* configs/voicemail.conf.sample, apps/app_voicemail.c: 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)

2008-08-01 12:17 +0000 [r135056-135061]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/safe_asterisk: Make safe_asterisk work on
	  dash/sh/bash etc. (closes issue #13111) Reported by: pabelanger
	  Patches: 2008071901_issue13111_safe_asterisk.diff uploaded by
	  mvanbaak (license 7) Tested by: mvanbaak, pabelanger

	* /, apps/app_ices.c: 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 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 22:28 +0000 [r135016]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/http.c: 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 21:53 +0000 [r134925-134977]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c: Switch command order, to meet with
	  current specs

	* res/res_config_pgsql.c: Increase column size beyond the minimum
	  required, since PostgreSQL won't let us modify existing columns.

2008-07-31 19:48 +0000 [r134922]  Steve Murphy <murf@digium.com>

	* /, main/features.c: 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.
	  ........ and as to 1.4 to trunk; have I expressed my feelings
	  about code shifting from one file to another? Good.

2008-07-31 19:43 +0000 [r134919]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c: Two errors: 1) If a function returns
	  SQLITE_LOCKED, no recovery is possible. 2) An error message can
	  be allocated, even when no error is signalled. (closes issue
	  #13109) Reported by: gknispel_proformatique

2008-07-31 19:39 +0000 [r134916-134917]  Russell Bryant <russell@digium.com>

	* /, apps/app_ices.c: 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 ........

	* channels/iax2-parser.c: fix the potential use of an uninitialized
	  variable

2008-07-31 19:03 +0000 [r134867]  Tilghman Lesher <tlesher@digium.com>

	* channels/iax2-parser.c: Optimize frame cache by realloc'ing the
	  smallest frame when the cache is full. This ensures that we don't
	  just keep a cache of tiny frames, continually doing an alloc/free
	  for each data frame, thus negating the point of having a cache.

2008-07-31 16:50 +0000 [r134803-134815]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.c: 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 ........

	* doc/tex/app-sms.tex, doc/tex/queuelog.tex: Fix some tex errors
	  (closes issue #13211) Reported by: eliel Patches:
	  fixtexerrors.patch uploaded by eliel (license 64)

2008-07-31 16:05 +0000 [r134759]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 22:38 +0000 [r134703]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/sched.h, main/sched.c: Oops, wrong define

2008-07-30 22:04 +0000 [r134653]  Steve Murphy <murf@digium.com>

	* /: blocking 134652 from trunk because this problem only applies
	  to 1.4

2008-07-30 21:40 +0000 [r134650]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: 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 21:05 +0000 [r134598]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.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 20:38 +0000 [r134596]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: 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 20:24 +0000 [r134556]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix the parsing of the "reason" parameter in
	  the Diversion: header. (closes issue #13195) Reported by:
	  woodsfsg

2008-07-30 19:55 +0000 [r134541]  Russell Bryant <russell@digium.com>

	* funcs/func_curl.c, /: 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 19:48 +0000 [r134481-134538]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: 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: 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 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 [r134476]  Mark Michelson <mmichelson@digium.com>

	* /, main/app.c: 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 17:36 +0000 [r134401-134443]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Document adaptive capabilities

	* res/res_config_sqlite.c: Add adaptive capabilities to the sqlite
	  realtime driver (closes issue #13097) Reported by:
	  gknispel_proformatique Patches: 20080730__bug13097.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: Corydon76

	* configs/iax.conf.sample, configs/chan_dahdi.conf.sample,
	  channels/chan_sip.c, main/features.c, configs/sip.conf.sample,
	  configs/skinny.conf.sample, CHANGES: Move implementation of an
	  attended-transfer-complete sound from one channel driver into a
	  common place for multiple channel drivers. (closes issue #13152)
	  Reported by: caio1982 Patches: atxfer_complete_sound3.diff
	  uploaded by caio1982 (license 22)

2008-07-30 15:32 +0000 [r134355]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: 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
	  ........

2008-07-30 15:30 +0000 [r134312-134353]  Tilghman Lesher <tlesher@digium.com>

	* doc/tex/cliprompt.tex, main/asterisk.c: Add %u and %g to the
	  ASTERISK_PROMPT settings, for username and group, respectively.
	  Also, take the opportunity to clean up the CLI prompt generation
	  code. (closes issue #13175) Reported by: eliel Patches:
	  cliprompt.patch uploaded by eliel (license 64)

	* Makefile: Minor changes to reduce packaging changes made by the
	  Fedora maintainer. (closes issue #12974) Reported by: jcollie
	  Patches: 0001-Don-t-override-duplicate-optimization-flags.patch
	  uploaded by jcollie (license 412)

2008-07-29 22:22 +0000 [r134260]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dahdibarge.c, channels/chan_dahdi.c, /,
	  apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c,
	  apps/app_rpt.c: build against the now-typedef-free dahdi/user.h,
	  and remove some #ifdefs for features that will always be present
	  in DAHDI

2008-07-29 21:23 +0000 [r134253]  Brett Bryant <bbryant@digium.com>

	* main/http.c: Fix deadlock when unloading res_http_post because
	  the uris lock was still locked.

2008-07-28 22:07 +0000 [r134162-134163]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: 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 ........

	* /: Restore properties mistakenly removed (broke merging)

2008-07-28 19:53 +0000 [r134125]  Mark Michelson <mmichelson@digium.com>

	* configure, main/Makefile, configure.ac, CHANGES: 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 [r134088]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt, apps/app_image.c, CHANGES: Change SendImage() to
	  output a more consistent status variable. (closes issue #13134)
	  Reported by: eliel Patches: app_image.c.patch uploaded by eliel
	  (license 64) UPGRADE.patch uploaded by eliel (license 64)

2008-07-28 16:42 +0000 [r134086]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/cflags.xml, main/channel.c, apps/app_dahdibarge.c,
	  channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  doc/ss7.txt, apps/app_dahdiscan.c, apps/app_dahdiras.c,
	  doc/osp.txt, pbx/pbx_config.c, apps/app_parkandannounce.c,
	  main/loader.c, sample.call, contrib/scripts/autosupport: remove
	  remaining Zaptel references in various places

2008-07-28 16:00 +0000 [r134050]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_chanspy.c,
	  main/asterisk.c: merging the zap_and_dahdi_trunk branch up to
	  trunk

2008-07-27 21:12 +0000 [r134005]  Russell Bryant <russell@digium.com>

	* funcs/func_config.c, /: Add a missing unlock within error
	  handling (closes issue #13176) Reported by: pj

2008-07-26 15:16 +0000 [r133941-133946]  Russell Bryant <russell@digium.com>

	* main/devicestate.c: actually use the cache_cache argument

	* main/devicestate.c: 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.

	* main/devicestate.c: Re-work comment about how device state
	  changes are processed to be a bit more clear

	* main/devicestate.c: Remove the code that decided when device
	  state changes should be cached or not. It is no longer needed.

2008-07-25 22:08 +0000 [r133860-133904]  Tilghman Lesher <tlesher@digium.com>

	* doc/lang/hebrew.ods, apps/app_voicemail.c: Hebrew syntax for
	  voicemail prompts (closes issue #13155) Reported by:
	  greenfieldtech Patches: app_voicemail.c.patch uploaded by
	  greenfieldtech (license 369) hebrew.ods uploaded by
	  greenfieldtech (license 369)

	* contrib/scripts/asterisk.ldif: Update version (closes issue
	  #13163) Reported by: suretec Patches: asterisk.ldif uploaded by
	  suretec (license 70)

	* main/channel.c, channels/chan_dahdi.c,
	  include/asterisk/devicestate.h, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_agent.c,
	  main/devicestate.c, channels/chan_iax2.c: Deprecate
	  *_device_state_* APIs in favor of *_devstate_* APIs

2008-07-25 20:56 +0000 [r133819]  Kevin P. Fleming <kpfleming@digium.com>

	* main/logger.c: minor change to test automerge

2008-07-25 19:12 +0000 [r133770]  Brandon Kruse <bkruse@digium.com>

	* main/manager.c: Revert tilghman and pari's code changes, as we do
	  NOT need to uri_decode in manager. (if I sent
	  core%20show%20channels from a telnet session, it should be
	  interpreted literally, however, if I send that from an http
	  session, it should be decoded, which is the behaivor now)

2008-07-25 17:24 +0000 [r133665]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /, channels/chan_agent.c, main/devicestate.c:
	  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 17:21 +0000 [r133651]  Brandon Kruse <bkruse@digium.com>

	* main/http.c: Committing a fix that was introduced a long time ago
	  (does not affect 1.4), where you would pass a pointer to the end
	  of a character array, and ast_uri_decode would do no good.

2008-07-25 15:00 +0000 [r133575-133579]  Russell Bryant <russell@digium.com>

	* /, LICENSE: 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 ........
	  ................

	* include/asterisk/doxyref.h, main/asterisk.c: Modify the main page
	  of the doxygen documentation to link to a new page dedicated to
	  Asterisk licensing information. The licensing page includes the
	  Asterisk license, as well as a (not yet complete) list of 3rd
	  party libraries that may be used, as well as what license we
	  receive them under. Help filling out this list in the format that
	  I have started in doxyref.h would be much appreciated. :)

2008-07-25 14:40 +0000 [r133570-133573]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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. ........

	* apps/app_queue.c: Add a missing unlock. Pointed out by Atis
	  Lezdins in #asterisk-dev

2008-07-25 13:01 +0000 [r133566-133568]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Minor coding guidelines tweaks ... - Use
	  ast_strlen_zero in one place - check for successful string
	  comparison the way most of Asterisk code does it

	* main/devicestate.c: When the ast_device_state() function is
	  called to retrieve device state, and the code checks to see if
	  there is a cached state available, use the aggregate cached state
	  across all servers, and not just the local state.

2008-07-24 21:27 +0000 [r133509]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: 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 20:40 +0000 [r133486]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: 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 19:53 +0000 [r133448]  Mark Michelson <mmichelson@digium.com>

	* main/logger.c: 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 [r133391-133400]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, contrib/scripts/asterisk.logrotate: Build the logrotate
	  script according to paths (Closes issue #13147)

	* Makefile: Optionally install logrotate file (Closes issue #13148)

2008-07-23 22:03 +0000 [r133299]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (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 [r133296]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c, /: 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 20:33 +0000 [r133197]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: 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 19:48 +0000 [r133041-133171]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c, include/asterisk/options.h,
	  main/asterisk.c: 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 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 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 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 17:20 +0000 [r132977-132981]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: 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): Add logrotate script
	  for Asterisk (closes issue #13085) Reported by: pabelanger
	  Patches: logrotate uploaded by pabelanger (license 224)

2008-07-23 16:38 +0000 [r132964-132966]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c: use correct function name... please
	  compile with --enable-dev-mode

	* /, main/utils.c, include/asterisk/stringfields.h: 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 12:07 +0000 [r132883]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/isdn_lib.c: 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. ........

2008-07-23 11:40 +0000 [r132827]  Kevin P. Fleming <kpfleming@digium.com>

	* /: remove bogus property that is breaking automerges

2008-07-23 08:13 +0000 [r132823]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: 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 22:17 +0000 [r132795]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 22:14 +0000 [r132791]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: correct fix made in r132777... the code
	  *did* compile in dev-mode, as long as libpri was installed and
	  enabled

2008-07-22 21:53 +0000 [r132778]  Tilghman Lesher <tlesher@digium.com>

	* configs/iax.conf.sample, /, channels/chan_iax2.c: 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 21:52 +0000 [r132777]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c: Get chan_dahdi to compile in devmode

2008-07-22 21:21 +0000 [r132705-132721]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: 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 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 ........

2008-07-22 20:46 +0000 [r132703]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c, doc/sip-retransmit.txt (added): 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 19:59 +0000 [r132573-132643]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
	  configure, include/asterisk/autoconfig.h.in, configure.ac: 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 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 22:49 +0000 [r132514-132572]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: Add autocompletion to "iax2 set debug
	  peer". (closes issue #13129) Reported by: eliel Patches:
	  chan_iax2.c.patch uploaded by eliel (license 64)

	* configs/gtalk.conf.sample, configs/jingle.conf.sample,
	  configs/manager.conf.sample, configs/features.conf.sample: 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 21:00 +0000 [r132510-132511]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/fskmodem.h (added), main/fskmodem.c (added):
	  (Step 2 of 2)

	* 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), main/fskmodem.c (removed): 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 20:54 +0000 [r132466-132508]  Brett Bryant <bbryant@digium.com>

	* apps/app_sendtext.c: 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_iax2.c: Add "iax2 set debug peer" command and
	  remove deprecated iax2 debug commands that conflicted with adding
	  new features to the newer debug commaands. (closes issue #13103)
	  Reported by: mvanbaak Patches:
	  2008071901__issue13103_iax2_set_debug_peer.diff uploaded by
	  mvanbaak (license 7) Tested by: bbryant, mvanbaak

	* channels/chan_sip.c: 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: 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 15:33 +0000 [r132425]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: make buffers config option
	  (chan_dahdi.conf) parsing safer and added logging in case of
	  failure

2008-07-21 14:47 +0000 [r132388-132390]  Russell Bryant <russell@digium.com>

	* 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,
	  apps/app_jack.c: 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.

	* codecs/codec_resample.c, apps/app_jack.c: Enable higher quality
	  resampling, as it doesn't have a noticeable performance impact on
	  my machine ..

2008-07-19 16:46 +0000 [r132312]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: 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 10:46 +0000 [r132277]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_config_sqlite.c: 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 22:19 +0000 [r132242]  Brett Bryant <bbryant@digium.com>

	* main/manager.c: 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 20:57 +0000 [r132203-132206]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Russell pointed out that using ast_strdupa()
	  within a loop like this is probably not a good idea, as we might
	  run out of stack space. Therefore, changing this over to use the
	  ast_str infrastructure for buffers is probably a good idea.

	* main/manager.c: Fix trunk devmode

2008-07-18 20:14 +0000 [r132169]  Pari Nannapaneni <paripurnachand@digium.com>

	* main/manager.c: updateconfig is not uri decoding variables,values
	  from the GET url

2008-07-18 19:09 +0000 [r132109-132113]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c: 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) ........

	* /, main/config.c: 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 18:50 +0000 [r132108]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: 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 17:55 +0000 [r132050]  Brett Bryant <bbryant@digium.com>

	* main/hashtab.c, cdr/cdr_radius.c: 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 17:10 +0000 [r131982-131989]  Tilghman Lesher <tlesher@digium.com>

	* /, main/sched.c: 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 ........

	* /, include/asterisk/sched.h, main/sched.c: 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 ........

	* /, include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c:
	  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 [r131923]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/Makefile, include/asterisk/dlfcn-compat.h (removed),
	  main/dlfcn.c (removed), main/loader.c: 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 15:38 +0000 [r131916]  Brett Bryant <bbryant@digium.com>

	* /, main/features.c: 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 22:40 +0000 [r131868]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: 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 21:26 +0000 [r131824]  Mark Michelson <mmichelson@digium.com>

	* apps/app_senddtmf.c: 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 20:37 +0000 [r131753-131791]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: 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: 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 18:14 +0000 [r131717]  Brett Bryant <bbryant@digium.com>

	* main/features.c: 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 15:45 +0000 [r131681]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c: Fix memory leak. (Closes issue #13096)
	  Reported by gknispel_proformatique

2008-07-17 14:46 +0000 [r131643]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Instead of attempting to pass through
	  AST_EVENT_DEVICE_STATE, use DEVICE_STATE_CHANGE instead.
	  DEVICE_STATE is a state change on one server, and
	  DEVICE_STATE_CHANGE is the "real" state of that device across all
	  servers sharing state. This would have only been a problem with
	  distributed device state.

2008-07-17 14:00 +0000 [r131606]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, UPGRADE.txt, CHANGES: Change several 'core' commands
	  to be 'dialplan' commands (with appropriate deprecation, of
	  course) (closes issue #13016) Reported by: caio1982 Patches:
	  dialplan_globals6.diff uploaded by caio1982 (license 22)

2008-07-16 23:53 +0000 [r131570]  Steve Murphy <murf@digium.com>

	* include/asterisk/lock.h: (closes issue #13089) Reported by: murf
	  Most of this bug was already fixed by Tilghman before I opened
	  it; Many thanks to Tilghman for his fix in svn version 125794.
	  That fix cleared up some of the fields in the lock_info. This
	  commit changes the address that is stored for the lock in the
	  lock_info struct, so that it is the same as that passed into the
	  locking macros. This makes searching for a lock_info (as in
	  log_show_lock()) by its lock addr possible. The lock_addr field
	  is infinitely more useful if it is the same as what is 'publicly'
	  available outside the lock_info code. Many thanks to kpfleming,
	  putnopvut, and Russell for their invaluable insights earlier
	  today.

2008-07-16 22:28 +0000 [r131445-131529]  Brett Bryant <bbryant@digium.com>

	* apps/app_rpt.c: Janitor project: convert free to ast_free (closes
	  issue #13082) Reported by: eliel Patches: app_rpt.c.patch
	  uploaded by eliel (license 64)

	* /, channels/chan_iax2.c: 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)
	  ........

	* funcs/func_sysinfo.c: Fixes sysinfo operator issue also fixed
	  elsewhere in r131445. (issue #13057)

	* main/asterisk.c: 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 [r131422]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 [r131300-131375]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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: 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 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 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 17:59 +0000 [r131243]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /: 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 15:08 +0000 [r131207]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Add missing terminator to
	  ast_event_subscribe to fix a crash. (from rev 131206 in the 1.6.0
	  branch)

2008-07-16 00:52 +0000 [r131166]  Tilghman Lesher <tlesher@digium.com>

	* main/logger.c: Fix rotate strategy (Closes issue #13086)

2008-07-15 23:36 +0000 [r131129]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (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 18:46 +0000 [r131072]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: 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 18:25 +0000 [r131044]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, main/manager.c, /, channels/chan_sip.c,
	  apps/app_voicemail.c: 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 18:14 +0000 [r131015]  Brett Bryant <bbryant@digium.com>

	* apps/app_queue.c: Fix memory leak in app_queue when a device
	  state is changed but it isn't a member of any queue. (closes
	  issue #13073) Reported by: eliel Patches: app_queue.c.patch
	  uploaded by eliel (license 64)

2008-07-15 17:49 +0000 [r131013]  Michiel van Baak <michiel@vanbaak.info>

	* main/cdr.c, /: 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 16:20 +0000 [r130890-130951]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Additional
	  option for videosupport (always) that disables the optimization
	  to fail to setup video RTP if the two endpoints will not support
	  it. This assists with call files and certain transfers to ensure
	  that if two video phones are ever connected, they will always
	  share a video feed.

	* /, channels/chan_iax2.c: 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 22:22 +0000 [r130794-130854]  Mark Michelson <mmichelson@digium.com>

	* main/asterisk.c: 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 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 17:21 +0000 [r130744]  Michiel van Baak <michiel@vanbaak.info>

	* main/dnsmgr.c, /: 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 16:50 +0000 [r130732-130733]  Luigi Rizzo <rizzo@icir.org>

	* channels/vgrabbers.c: free memory used by the x11 grabber when
	  closing it.

	* channels/console_video.c: use
	  ast_pthread_create_detached_background() instead of redoing it
	  with separate calls

2008-07-14 15:44 +0000 [r130697]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_unistim.c, channels/h323/ast_h323.cxx,
	  include/asterisk/module.h, channels/misdn/isdn_lib.c: Swap
	  "static" and "const", so that "static" appears at the beginning
	  of each declaration (suppresses a warning). (closes issue #13070)
	  Reported by: gknispel_proformatique Patches:
	  asterisk_trunk_const_static.patch uploaded by gknispel (license
	  261)

2008-07-14 10:39 +0000 [r130635]  Russell Bryant <russell@digium.com>

	* /, main/audiohook.c: 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 23:14 +0000 [r130574-130578]  Michiel van Baak <michiel@vanbaak.info>

	* doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: 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.

	* channels/chan_skinny.c: Convert chan_skinny's open-coded linked
	  lists to the list macros (closes issue #12956) Reported by: DEA
	  Patches: chan_skinny-linkedlists-v2.txt uploaded by DEA (license
	  3) Tested by: DEA, mvanbaak

	* main/manager.c, /: 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 17:58 +0000 [r130515]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 14:58 +0000 [r130479]  Michiel van Baak <michiel@vanbaak.info>

	* doc/tex/asterisk.tex: restore ASTERISKVERSION marker to
	  asterisk.tex. This got lost in commit 97634

2008-07-13 02:34 +0000 [r130444]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_agent.c: Unlock list before returning

2008-07-11 22:23 +0000 [r130320]  Kevin P. Fleming <kpfleming@digium.com>

	* /: not needed here

2008-07-11 22:03 +0000 [r130296-130297]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (closes issue #13041) Reported by: eliel OK, now the
	  context registrar slot is strdup'd. It is freed on destruction. I
	  don't see the need to do this with all the structs' registrar
	  fields, but if some wild case proves they should also be handled
	  this way, then we can put in the extra work at that time.

	* res/res_odbc.c: a small change to make things compile

2008-07-11 21:36 +0000 [r130293]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Support new TRANSPORT definitions in
	  libss7

2008-07-11 20:03 +0000 [r130237]  Mark Michelson <mmichelson@digium.com>

	* /, main/audiohook.c: 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 19:56 +0000 [r130230-130234]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Don't copy on NULL.

	* include/asterisk/res_odbc.h, res/res_odbc.c: Add some debug code
	  and add a missing release

	* channels/chan_dahdi.c, utils/astman.c: Fix trunk breakage

2008-07-11 19:14 +0000 [r130174]  Mark Michelson <mmichelson@digium.com>

	* /, main/audiohook.c: 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 18:52 +0000 [r130170]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 18:32 +0000 [r130167]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: Missed one. Formatting only.

2008-07-11 18:24 +0000 [r130145]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (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.

2008-07-11 18:09 +0000 [r130129]  Brett Bryant <bbryant@digium.com>

	* 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, 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, apps/app_milliwatt.c,
	  main/asterisk.c, main/dsp.c: 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 17:29 +0000 [r130126]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_agent.c: 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 [r130040-130044]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/ss7.txt, contrib/utils/zones2indications.c, CHANGES: 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 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) ........

2008-07-11 14:22 +0000 [r129985-129987]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/astobj.h: 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 ... ........

	* /: remove space in property value

2008-07-11 14:16 +0000 [r129916-129968]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/astmm.c: 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 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 00:55 +0000 [r129864]  Sean Bright <sean.bright@gmail.com>

	* res/res_config_pgsql.c, res/res_config_ldap.c: Fix some usages of
	  snprintf, and clarify a couple variable names.

2008-07-10 22:06 +0000 [r129758-129804]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 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 20:56 +0000 [r129738]  Terry Wilson <twilson@digium.com>

	* Makefile: Move phoneprov config files to be installed with 'make
	  samples' so changes aren't inadvertently lost on a 'make install'

2008-07-10 20:33 +0000 [r129734]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removed the fn2 field from the vm_state
	  structure. fn2 was used in three functions. In every case, it was
	  initialized in the function it was used in. This meant there was
	  no need to have it in a malloc'd structure just taking up space.
	  Furthermore two of the functions it was used in were completely
	  unnecessary since fn2 was set to exactly the same value as the
	  vm_state's fn string. fn2 was a char array sized at PATH_MAX. On
	  my system, PATH_MAX is 4096. This equates to a 4K memory savings
	  per vm_state allocated. Since there is a vm_state malloc'd for
	  every voicemail user on the system, this could potentially add up
	  nicely if there are lots of users. In addition, a vm_state is
	  allocated on the stack each time a caller calls the VoiceMailMain
	  application, meaning that there is a significant stack savings
	  with this patch too. Of course, a single vm_state struct still
	  takes up approximately 20K on my system (when using IMAP storage.
	  Without IMAP storage, there would be about another 300 bytes
	  fewer usage), even with this removal. Further optimizations are
	  probably possible, but most likely not as easy as this one.

2008-07-10 19:13 +0000 [r129684]  Brett Bryant <bbryant@digium.com>

	* apps/app_queue.c: 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 18:19 +0000 [r129638-129642]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: A couple more minor text changes

	* channels/chan_sip.c: Remove extraneous \n. Pointed out by eliel
	  on #asterisk-dev.

2008-07-10 16:21 +0000 [r129581]  Michiel van Baak <michiel@vanbaak.info>

	* main/features.c: Remove deprecated 'show parkedcalls' CLI command
	  (closes issue #13038) Reported by: eliel Patches:
	  finish.deprecate.patch uploaded by eliel (license 64)

2008-07-10 16:12 +0000 [r129569]  Russell Bryant <russell@digium.com>

	* /, sample.call: 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 13:54 +0000 [r129503]  Sean Bright <sean.bright@gmail.com>

	* main/editline: Update svn:ignore

2008-07-09 19:40 +0000 [r129437]  Mark Michelson <mmichelson@digium.com>

	* /, main/rtp.c: 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 15:57 +0000 [r129399]  Matthew Fredrickson <creslin@digium.com>

	* main/pbx.c: Add Proceeding() application (#13025)

2008-07-09 13:44 +0000 [r129344]  Sean Bright <sean.bright@gmail.com>

	* main/editline/makelist.in (added), /, main/editline/configure,
	  main/editline/Makefile.in, main/editline/configure.in,
	  main/editline/makelist (removed): 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. ........

2008-07-09 03:39 +0000 [r129307]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, main/manager.c, res/res_agi.c, pbx/pbx_realtime.c,
	  include/asterisk/channel.h, include/asterisk/pbx.h, main/cli.c:
	  Code wasn't ready to be merged - see -dev list discussion

2008-07-08 22:56 +0000 [r129270]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix compilation error when IMAP storage is
	  enabled

2008-07-08 21:00 +0000 [r129156]  Brett Bryant <bbryant@digium.com>

	* main/dnsmgr.c, main/srv.c, main/dns.c: 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 20:30 +0000 [r129048-129152]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, /, channels/chan_sip.c,
	  include/asterisk/causes.h: 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 ........

	* main/asterisk.c: Reduce length of time that 'asterisk -rx' waits.
	  (closes issue #13001) Reported by: eliel Patches:
	  20080708__bug13001.diff.txt uploaded by Corydon76 (license 14)
	  20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64)
	  Tested by: Corydon76, eliel

	* /, channels/chan_iax2.c: 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 16:40 +0000 [r129045]  Brett Bryant <bbryant@digium.com>

	* 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, main/rtp.c, main/channel.c,
	  channels/chan_dahdi.c, main/manager.c, formats/format_pcm.c,
	  main/callerid.c, main/logger.c, apps/app_parkandannounce.c,
	  apps/app_adsiprog.c: Janitor project to convert sizeof to
	  ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982
	  Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22)

2008-07-08 14:17 +0000 [r129006]  Russell Bryant <russell@digium.com>

	* apps/app_fax.c: 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 10:02 +0000 [r128927-128951]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 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 00:02 +0000 [r128830-128857]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: 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 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 22:42 +0000 [r128796]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 20:50 +0000 [r128738]  Sean Bright <sean.bright@gmail.com>

	* /, channels/chan_iax2.c: 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 20:30 +0000 [r128599-128733]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Crap

	* apps/app_voicemail.c: 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

	* apps/app_voicemail.c: Get app_voicemail compiling when IMAP
	  storage is used. Brought up by reporter on issue #12987

	* /, channels/chan_iax2.c: 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) ........

	* doc/tex/extensions.tex, configs/extensions.conf.sample: Update a
	  few instances of "extensions reload" to "dialplan reload" in the
	  documentation. Patch provided by caio1982 (license 22)

2008-07-07 11:53 +0000 [r128564]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: As pointed out on the -dev list, actually
	  use the result of find_peer() so that a peer reference is not
	  leaked.

2008-07-06 20:19 +0000 [r128274-128525]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: - Adding alias
	  "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and
	  "tlsbindaddr". Note: I don't think we can start properly without
	  UDP port open, that needs to be tested. - Removing "bindport"
	  from configuration example, not needed to mention this any more I
	  suggest we deprecate "bindaddr" and "bindport" in trunk (for
	  1.6.1)

	* channels/chan_sip.c, configs/sip.conf.sample: - 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: - Remove unused variable "expiry" - Set
	  global_outboundproxy.force at reload.

	* channels/chan_sip.c: More doxygen comments.

	* channels/chan_sip.c: - Formatting changes - Doxygen changes -
	  Replacing a doxygen description that was copied from another
	  function

	* channels/chan_sip.c: Adding note about incorrect manager
	  registration...

	* doc/realtimetext.txt (added): Adding documentation on the T.140
	  support in Asterisk. This is a function that we're the reference
	  implementation on now. :-)

	* channels/chan_sip.c: Remove comments that doesn't make sense. The
	  deprecation of type=user will come at a later stage, as indicated
	  by previous commit message

	* channels/chan_sip.c: Fix severe problem with my previous commit
	  of "kill-the-user". Russell saw a problem with this code, but not
	  the correct problem. Thanks, anyway! ;-)

	* main/pbx.c, main/manager.c, pbx/pbx_realtime.c,
	  include/asterisk/pbx.h: Changing name of global api call to ast_*
	  My mistake, pointed out by Russell.

	* channels/chan_sip.c: Disabling code used by dumpdb with #ifdef,
	  since I believe we might use it sometime in the future, but also
	  want to avoid compiler warnings now.

	* channels/chan_sip.c: Removing the CLI dumpdb command (see
	  asterisk-dev discussion and decision)

	* channels/chan_sip.c: Adding a few reminders

	* channels/chan_sip.c: 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: Adding TCP and TLS to "sip show settings".
	  TLS needs to have one configuration per configured domain at some
	  point.

	* channels/chan_sip.c: Add some comments...

	* channels/chan_sip.c: Set tls setting to default in reload_config.

2008-07-05 21:20 +0000 [r128254]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_sip.c: fix compiling of chan_sip.c

2008-07-05 21:11 +0000 [r128247]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: I like it when the tree is not broken.

2008-07-05 20:59 +0000 [r128201-128242]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: KILL THE USER! Actually, kill the in-memory
	  structure for type=user and start using the sip_peer structure
	  for every object. Have only one in-memory list and use them
	  different ways depending on type=user, type=peer and type=friend
	  - like always. The idea with this first patch is that
	  configurations should work as before. Some additional features
	  for realtime peers. By adding a type= field, you can now have
	  multiple functions. Let's test this for a while. Won't be
	  integrated in 1.6.0, only in trunk, for testing. There's propably
	  more to clean up and simplify here. Help is welcome and
	  encouraged!

	* main/pbx.c, main/manager.c, res/res_agi.c, pbx/pbx_realtime.c,
	  include/asterisk/channel.h, include/asterisk/pbx.h, main/cli.c:
	  Implement flags for AGI in the channel structure so taht "show
	  channels" and AMI commands can display that a channel is under
	  control of an AGI. Work inspired by work at customer site, but
	  paid for by Edvina AB

	* configs/sip.conf.sample: Make TCP disabled by default (it's
	  considered experimental)

	* configs/sip.conf.sample: Reformatting the config sample

	* channels/chan_sip.c: Stop cli command completion with tabs

2008-07-05 19:52 +0000 [r128198]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Make this actually evaluate how it was intended to
	  be.

2008-07-05 19:27 +0000 [r128197]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: Add new
	  SIP cli command "sip show channelstats" that displays some QoS
	  data (if we have RTCP reports and not use the p2p rtp bridge). I
	  could not find a way to detect us using the p2p bridge, which
	  would be nice.

2008-07-05 15:17 +0000 [r128160]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: 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 03:39 +0000 [r128122-128125]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: It would help if we actually parsed the
	  ss7_explicitacm option in the config file...

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: 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 16:41 +0000 [r128027-128082]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Fullcontact needs more than 20 characters,
	  even for the simplest case

	* main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: 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 ........

2008-07-04 14:36 +0000 [r127995]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - reorganize SIP extensions alphabetically,
	  to make it easier to synch with the IANA list - add a few new
	  registered and well-known extension names

2008-07-03 22:47 +0000 [r127931-127934]  Brett Bryant <bbryant@digium.com>

	* channels/iax2-parser.c: Fix one more file that got changed.

	* channels/iax2.h, channels/chan_iax2.c: Remove commit that somehow
	  got mergeed into trunk.

	* channels/iax2.h, channels/chan_iax2.c, channels/iax2-parser.c:
	  Update these files with transfer code.

2008-07-03 22:23 +0000 [r127903]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /, apps/Makefile, main/editline/np/vis.c: 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 20:59 +0000 [r127831-127857]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Make change proposed by andrew53 on
	  bugtracker

	* apps/app_chanspy.c: Thanks to a suggestion from seanbright, print
	  a warning if the attachment of the whisper or barge audiohooks
	  fails.

	* apps/app_chanspy.c: Fix build

	* apps/app_chanspy.c: Fix a crash when attempting to spy on an
	  unbridged channel. (closes issue #12986) Reported by: andrew53

2008-07-03 17:16 +0000 [r127793]  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 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 16:48 +0000 [r127779-127791]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure we stop session timers as soon as
	  we start hanging up an active call. May fix issue 12919.

	* channels/chan_sip.c: 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 16:22 +0000 [r127767]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, configure.ac, acinclude.m4: some minor fixes found
	  while working on issue #12911 (and block the rev from 1.4 since
	  the equivalent is already here)

2008-07-03 14:34 +0000 [r127720]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Added a
	  new option, "timeoutpriority" to queues.conf. A detailed
	  explanation of the change may be found in
	  configs/queues.conf.sample (closes issue #12690) Reported by:
	  atis

2008-07-03 09:59 +0000 [r127685]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fix bad formatting in a very confusing
	  function. Who added the sipdb sql output? It's mixing peers and
	  users in a strange way and should really not be a CLI command,
	  since it's not meant for human output. It should be done with an
	  app connecting to manager.

2008-07-02 22:17 +0000 [r127622]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Oops

2008-07-02 22:16 +0000 [r127621]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: Update transport= in sip so that the option
	  is not broken from a recent commit.

2008-07-02 21:27 +0000 [r127609]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_unistim.c, include/asterisk/app.h, main/manager.c,
	  channels/chan_sip.c, main/app.c, channels/chan_iax2.c,
	  apps/app_voicemail.c: Keep ast_app_inboxcount API compatible with
	  1.6.0.

2008-07-02 21:09 +0000 [r127566]  Mark Michelson <mmichelson@digium.com>

	* doc/janitor-projects.txt: Add a janitor project to use ARRAY_LEN
	  instead of in-line sizeof() and division.

2008-07-02 20:52 +0000 [r127564]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Fix some crashlike bugs because flag could
	  be NULL in play_record_review(). (Closes issue #12892) Reported
	  by: jaroth Patch originally by jaroth, fixed by me.

2008-07-02 20:49 +0000 [r127558-127562]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: 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 ........

	* configs/agents.conf.sample, channels/chan_agent.c, CHANGES: The
	  ackcall and endcall options in agents.conf now have supplemental
	  options acceptdtmf and enddtmf. These allow for the DTMF pressed
	  to be configurable instead of being hardcoded to '#' and '*'.
	  (AST-86)

2008-07-02 20:28 +0000 [r127545]  Terry Wilson <twilson@digium.com>

	* include/asterisk/http.h, main/http.c: Expose the prefix variable
	  so that it can be used by modules depending on http support

2008-07-02 18:31 +0000 [r127466]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Solaris fix (closes issue #12949) Reported by: snuffy
	  Patches: bug_12949.diff uploaded by snuffy (license 35)

2008-07-02 17:27 +0000 [r127434]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: Fix to sip_parse_host so that it passes the
	  correct information to sip_registry.

2008-07-02 14:50 +0000 [r127401]  Russell Bryant <russell@digium.com>

	* include/asterisk/logger.h, include/asterisk/devicestate.h,
	  include/asterisk/astobj2.h, include/asterisk/timing.h,
	  include/asterisk/strings.h, include/asterisk/dnsmgr.h,
	  include/asterisk/threadstorage.h, include/asterisk/slinfactory.h,
	  main/libresample/include/libresample.h, include/asterisk/time.h:
	  Fix a bunch of places where \arg was used instead of \param.
	  Using \arg to document arguments seems logical, and does work,
	  but is not the best thing to use. \arg in doxygen is simply for
	  creating non-nested unordered lists. \param is the correct tag to
	  use to document function parameters, and will come out better in
	  the generated documentation.

2008-07-02 14:30 +0000 [r127398]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c: Fix a bug I noticed while doing the previous merge

2008-07-02 12:08 +0000 [r127363]  Russell Bryant <russell@digium.com>

	* doc/CODING-GUIDELINES: 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 12:06 +0000 [r127330-127362]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_video.c: plug another panic when the gui cannot
	  be started. We can still send video, just don't try to use what
	  is not available.

	* channels/console_video.c: prevent a segfault when trying to start
	  the gui without any specific configuration in oss.conf (reported
	  by Klaus Darillion on the -video mailing list).

2008-07-02 02:48 +0000 [r127297]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: 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 23:38 +0000 [r127245]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: 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 21:43 +0000 [r127210]  Russell Bryant <russell@digium.com>

	* funcs/func_devstate.c: Add a \todo

2008-07-01 21:21 +0000 [r127169]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Add AMI events for start/stop of MOH
	  (closes issue #12909) Reported by: chris-mac Patches:
	  res_musiconhold-event.patch uploaded by chris-mac (license 506)

2008-07-01 21:16 +0000 [r127157]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: 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

2008-07-01 21:03 +0000 [r127154]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Add a configuration
	  option so the global outboundproxy can use tcptls without it
	  being defined by each sip user.

2008-07-01 20:51 +0000 [r127152]  Jason Parker <jparker@digium.com>

	* Makefile: 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 20:28 +0000 [r127143]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, /, channels/chan_iax2.c: 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) ........

2008-07-01 19:53 +0000 [r127113]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c: change the process of inserting a delay into the
	  ast_answer() path so that we don't tell the calling channel that
	  it has been answered unutil after the delay; for a single-thread
	  call this won't matter all, but for a dual-thread call (using
	  chan_local) this may fix the problem in issue 12924

2008-07-01 19:20 +0000 [r127074]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 17:22 +0000 [r127017]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_ais.c, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, res/Makefile,
	  res/ais/ais.h, makeopts.in: make the AIS checking a little more
	  generic, and have a more useful configure script command line
	  option for OpenAIS

2008-07-01 16:52 +0000 [r127000]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 16:28 +0000 [r126991]  Luigi Rizzo <rizzo@icir.org>

	* images/kpad2.jpg: even uglier gui with more buttons

2008-07-01 16:16 +0000 [r126960]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c, include/asterisk/channel.h, apps/app_queue.c:
	  another minor ast_channel memory size decrease... for nearly all
	  channels, 'dialcontext' is only going to be set once during the
	  channel's lifetime, so make it a string field instead of a char
	  array

2008-07-01 16:14 +0000 [r126959]  Luigi Rizzo <rizzo@icir.org>

	* doc/video.txt, doc/video_console.txt (added): add documentation
	  on video console support

2008-07-01 15:03 +0000 [r126845-126903]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 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 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) ........

2008-07-01 12:29 +0000 [r126835]  Luigi Rizzo <rizzo@icir.org>

	* main/logger.c: use %p to print a pointer

2008-07-01 11:58 +0000 [r126755-126790]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 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 [r126675]  Jeff Peeler <jpeeler@digium.com>

	* configs/chan_dahdi.conf.sample (added),
	  configs/zapata.conf.sample (removed): rename zapata.conf.sample
	  to chan_dahdi.conf.sample

2008-06-30 20:25 +0000 [r126637]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Add support to see MTP2 down events when
	  the link layer drops in SS7

2008-06-30 16:07 +0000 [r126574]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: 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 15:45 +0000 [r126571-126572]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c,
	  channels/chan_oss.c, channels/console_video.h: implement the
	  'freeze' function for incoming frames; fix a bug which caused a
	  crash when a videodevice was specified after startgui=1 in the
	  config file. This also involves a slightly different method to
	  determine if the gui is active or not.

	* apps/app_voicemail.c: fix an uninitialized variable

2008-06-30 13:03 +0000 [r126517]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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)
	  ........

2008-06-30 12:49 +0000 [r126515]  Russell Bryant <russell@digium.com>

	* doc/CODING-GUIDELINES: a few minor updates and typo fixes

2008-06-30 11:57 +0000 [r126513]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/freetds.tex, cdr/cdr_tds.c: 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)

2008-06-29 21:17 +0000 [r126448-126480]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c, channels/console_video.c,
	  channels/console_board.c: import the recent additions for video
	  console into trunk, giving you support for up to 9 video sources
	  (e.g. webcams, or X11 grabbers, etc.) active at once, displaying
	  thumbnails for each of them in the main GUI window, and with the
	  ability to switch between them on the fly during a conversation.
	  The code also implements a 'Picture in Picture' feature, allowing
	  you to select any source as primary or secondary, and move the
	  PiP window by just dragging it with the mouse. The window looks
	  like this:
	  ________________________________________________________________
	  | ______ ______ ______ ______ ______ ______ ______ | | | tn.1 | |
	  tn.2 | | tn.3 | | tn.4 | | tn.5 | | tn.6 | | tn.7 | | | |______|
	  |______| |______| |______| |______| |______| |______| | | ______
	  ______ ______ ______ ______ ______ ______ | | |______| |______|
	  |______| |______| |______| |______| |______| | |
	  _________________ __________________ _________________ | | | | |
	  | | | | | | | | | | | | | | | | | | | | | | remote video | | | |
	  local video | | | | | | | | ______ | | | | | | keypad | | | PIP
	  || | | | | | | | |______|| | | |_________________| | |
	  |_________________| | | | | | | | | | | |__________________| |
	  |________________________________________________________________|

	* channels/console_gui.c, channels/console_board.c,
	  channels/console_video.h: fix wrong argument in checking
	  boundaries for a rectangle some whitespace fixes

2008-06-29 16:19 +0000 [r126356]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac, pbx/pbx_lua.c, pbx/Makefile,
	  pbx/pbx_gtkconsole.c: 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 add searching for OpenAIS libraries in /usr/lib/openais
	  if a path is not specified; not sure if this is really the
	  optimal solution, but it works make the compiler shut up about
	  some ignored function results in pbx_gtkconsole; this module is
	  badly coded anyway

2008-06-29 13:20 +0000 [r126312-126319]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c: This was bogus, need to find a better way.

	* cdr/cdr_tds.c: While we're at it, escape all the columns in our
	  TDS queries as well. Double quotes seems to be more standard than
	  square brackets (Sybase and SQL Server both support them).

2008-06-29 13:02 +0000 [r126308-126311]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_oss.c: implement a 'toggle' option for 'console
	  mute' and 'console unmute'

	* channels/console_video.h: add some defines and fields in
	  preparation for the import of the video source switching support

	* channels/vgrabbers.c: accept any name starting with X11 for X11
	  grabbers - this lets you have multiple active instances of this
	  grabber; require v4l device names to start with '/dev/' -
	  prevents some useless attempt to open a file as a device.

	* channels/vcodecs.c, channels/console_video.c: make this compile
	  after ast_frame's data field changed to a union

2008-06-29 12:06 +0000 [r126226-126274]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_pgsql.c: Quote column names when inserting CDRs into
	  postgres to avoid conflicts with reserved words. (closes issue
	  #12947) Reported by: panolex

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  UPGRADE.txt, cdr/cdr_tds.c: 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

2008-06-28 15:54 +0000 [r126152-126187]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/channel.h: yay for airplane ride
	  optimizations... sort the fields in ast_channel by alignment
	  requirements, saving 36 bytes per instance on a 64-bit platform

	* Makefile: fix silly syntax error

	* Makefile: add message when no UI for menuselect is present

	* Makefile: use batch-mode (no user interface) menuselect for
	  --check-deps operations move automatic user interface selection
	  for menuselect to this Makefile

2008-06-27 23:29 +0000 [r126115]  Sean Bright <sean.bright@gmail.com>

	* main/cdr.c: Pretty up the 'cdr show status' output.

2008-06-27 22:10 +0000 [r126021-126057]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: 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 ........

	* contrib/scripts/dbsep.cgi: Separate multiple items encoded into a
	  single field with ';'

2008-06-27 19:19 +0000 [r125988]  Russell Bryant <russell@digium.com>

	* doc/siptls.txt: Fix a typo. Someone on IRC copied this literally
	  and then wondered why it wasn't working. :)

2008-06-27 19:05 +0000 [r125980-125984]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Revert this part of the fix. We'll fix it
	  in libss7

	* channels/chan_dahdi.c: Obviously somebody didn't compile with
	  libss7 support when doing the DAHDI conversion.

	* channels/chan_dahdi.c: Add support for new commands to
	  block/unblock all CICs on a linkset

2008-06-27 17:35 +0000 [r125947]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: Small error in the function that converts
	  peer transports to a string.

2008-06-27 17:02 +0000 [r125895]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Document DLA_UNLOCK and DLA_LOCK

2008-06-27 16:28 +0000 [r125891]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: 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 16:23 +0000 [r125855-125880]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/lock.h: Optimization suggested by Russell to
	  cache the value of pthread_self() so that it isn't evaluated
	  every time through the loop.

	* apps/app_queue.c: Remove debug message

	* apps/app_queue.c: Ensure the thread-safety of the monexec
	  variable in app_queue. Thanks to Russell for pointing out the
	  problem

2008-06-27 16:00 +0000 [r125853]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: 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 14:14 +0000 [r125799]  Mark Michelson <mmichelson@digium.com>

	* utils/Makefile: Remove an unneeded target from the Makefile

2008-06-27 14:08 +0000 [r125741-125796]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Push relatively unused compiler options down the list,
	  keeping the popular options at the top.

	* /, main/utils.c, include/asterisk/lock.h: 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 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 07:28 +0000 [r125703]  Philippe Sultan <philippe.sultan@gmail.com>

	* include/asterisk/jabber.h, res/res_jabber.c: Fix a compile time
	  error that occurs if OpenSSL is not installed. Reported by Noel
	  Morais on the users mailing list

2008-06-27 00:22 +0000 [r125647-125666]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make this compile with dev-mode on

	* apps/app_queue.c: 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

	* apps/app_dial.c, CHANGES: Improve consistency between app_dial
	  and app_queue with regards to how language is handled between two
	  channels whose native language is different. Prior to this patch,
	  app_dial would have the callee inherit the caller's language, and
	  app_queue would not. After this patch, app_dial no longer has the
	  language inheritance capability. This seems to make the most
	  sense since it seems more natural for a person to hear files
	  played back in his/her native language instead of the language of
	  the person on the far end of the call. See the CHANGES file for
	  hints on how to keep the previous behavior of app_dial if
	  desired. (closes issue #12489) Reported by: bcnit

2008-06-26 23:18 +0000 [r125593-125596]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: remove block of commented code to set
	  __ourip This is now handled in skinny_register and load_config.
	  part two of chan_skinny cleanup

	* channels/chan_skinny.c: remove paging device from chan_skinny.
	  This has never been used, and noone could give us info about what
	  it is used for.

2008-06-26 23:06 +0000 [r125591]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a really stupid mistake

2008-06-26 23:04 +0000 [r125589]  Jason Parker <jparker@digium.com>

	* /, main/utils.c: 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 23:01 +0000 [r125586]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 22:49 +0000 [r125583]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astcli: Don't hang if the command is blank

2008-06-26 20:57 +0000 [r125477]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 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 17:40 +0000 [r125386-125438]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Don't play "your message has been saved"
	  twice. (closes issue #12893) Reported by: jaroth Patches:
	  duplicate_saved.patch uploaded by jaroth (license 50)

	* 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,
	  main/translate.c, codecs/codec_g726.c, codecs/codec_gsm.c,
	  codecs/codec_resample.c, codecs/codec_ulaw.c,
	  codecs/codec_ilbc.c, include/asterisk/translate.h: Convert casts
	  to unions, to fix alignment issues on Solaris (closes issue
	  #12932) Reported by: snuffy Patches: bug_12932_20080627.diff
	  uploaded by snuffy (license 35)

2008-06-26 16:54 +0000 [r125385]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 15:50 +0000 [r125333]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: 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) ........

2008-06-26 15:37 +0000 [r125332]  Russell Bryant <russell@digium.com>

	* main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
	  include/asterisk/timing.h, main/timing.c: - add get_max_rate
	  timing API call - change ast_settimeout() to honor max rate in
	  edge cases of file playback (this will make some warning messages
	  go away at the end of playing back a file)

2008-06-26 12:09 +0000 [r125279]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_musiconhold.c: fix compile failure found by buildbot (go,
	  buildbot!)

2008-06-26 11:02 +0000 [r125191-125277]  Tilghman Lesher <tlesher@digium.com>

	* /, main/rtp.c: 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 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: 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 23:05 +0000 [r125138]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dahdibarge.c, /, apps/app_meetme.c, main/Makefile,
	  apps/app_dahdiscan.c, apps/app_dahdiras.c, configure.ac,
	  res/res_timing_dahdi.c, include/asterisk/dahdi.h (removed),
	  res/res_musiconhold.c, main/channel.c, channels/chan_dahdi.c,
	  apps/app_flash.c, configure, codecs/codec_dahdi.c,
	  apps/app_rpt.c, main/asterisk.c: 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 22:40 +0000 [r125133-125135]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/lock.h: Fix indentation

	* include/asterisk/lock.h: Fix a bug in the rwlock tracking.
	  ast_rwlock_unlock did not take into account that multiple threads
	  could hold the same rdlock at the same time. As such, it expected
	  that when a thread released a lock that it must have been the
	  last to acquire the lock as well. Erroneous error messages would
	  be sent to the console stating that a thread was attempting to
	  unlock a lock it did not own. Now all threads are examined to be
	  sure that the message is only printed when it is supposed to be
	  printed.

2008-06-25 19:37 +0000 [r125096]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: implement transfer functionality in
	  chan_skinny (closes issue #9939) Reported by: wedhorn Patches:
	  transfer_v6.diff uploaded by wedhorn (license 30)
	  chan_skinny-transfer-trunk-v10.txt uploaded by DEA (license 3)
	  chan_skinny-transfer-trunk-v12.txt uploaded by mvanbaak (license
	  7) Tested by: DEA, wedhorn, mvanbaak

2008-06-25 16:00 +0000 [r124912-125055]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c, funcs/func_curl.c, res/res_curl.c (added):
	  Separate the global initialization routines for cURL into its own
	  separate module.

	* channels/chan_dahdi.c, channels/chan_local.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  include/asterisk/lock.h, channels/chan_iax2.c: More expansion of
	  the deadlock avoidance macro, including a macro to do locking of
	  the channel lock

	* channels/chan_dahdi.c, /, include/asterisk/lock.h: 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 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 17:50 +0000 [r124870-124872]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: 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: Code simplification

2008-06-24 11:02 +0000 [r124835]  Sean Bright <sean.bright@gmail.com>

	* UPGRADE.txt, CHANGES: Update CHANGES and UPGRADE.txt per
	  kpfleming's mail to #asterisk-dev.

2008-06-24 02:16 +0000 [r124798]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c: fix a memory leak. (inspired by, and
	  potentially fixes issue #12917)

2008-06-23 15:24 +0000 [r124707]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/taskprocessor.c: make solaris happy...pointed out by
	  snuff-home on IRC

2008-06-22 17:36 +0000 [r124596-124669]  Sean Bright <sean.bright@gmail.com>

	* configs/meetme.conf.sample: Revert my change to the sample meetme
	  conf file as it was incorrect.

	* configs/meetme.conf.sample: Fix a comment in meetme.conf.sample
	  per jmls via #asterisk-dev (And this time, do it in the correct
	  repository :-))

	* apps/app_rpt.c: Let app_rpt compile.

2008-06-22 02:58 +0000 [r124541]  Steve Murphy <murf@digium.com>

	* apps/app_forkcdr.c, /: 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 12:53 +0000 [r124396-124505]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_ldap.c: 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 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 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 21:43 +0000 [r124392-124393]  Jeff Gehlbach <jeffg@opennms.org>

	* /: (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 ........ ................

	* doc/asterisk-mib.txt, doc/digium-mib.txt: 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 20:17 +0000 [r124316]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: 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 16:30 +0000 [r124243-124278]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/doxyref.h, main/ast_expr2f.c, main/ast_expr2.fl:
	  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: 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 22:59 +0000 [r124183]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: 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 22:34 +0000 [r124180]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix attachment behavior when using IMAP
	  storage for voicemails 1. Filenames had an extra "msg" in the
	  attachment name 2. The attachment was being saved twice (closes
	  issue #12894) Reported by: jaroth Patches: imap_attach.patch
	  uploaded by jaroth (license 50)

2008-06-19 20:48 +0000 [r124127]  Michiel van Baak <michiel@vanbaak.info>

	* doc/CODING-GUIDELINES, channels/chan_sip.c, apps/app_minivm.c,
	  main/logger.c, pbx/pbx_realtime.c, res/res_realtime.c,
	  res/res_musiconhold.c, apps/app_directory.c, apps/app_queue.c,
	  channels/chan_iax2.c, include/asterisk/compiler.h,
	  apps/app_voicemail.c, funcs/func_realtime.c: Older versions of
	  GNU gcc do not allow 'NULL' as sentinel. They want (char *)NULL
	  as sentinel. An example is OpenBSD (confirmed on 4.3) that ships
	  with gcc 3.3.4 This commit introduces a contstant SENTINEL which
	  is declared as: #define SENTINEL ((char *)NULL) All places I
	  could test compile on my openbsd system are converted. Update
	  CODING-GUIDELINES to tell about this constant.

2008-06-19 20:35 +0000 [r124125]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Oops

2008-06-19 20:30 +0000 [r124121]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: 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 20:25 +0000 [r124102]  Tilghman Lesher <tlesher@digium.com>

	* main/netsock.c: Make OpenBSD compile again (reported by mvanbaak
	  via IRC -dev)

2008-06-19 19:48 +0000 [r124064]  Brett Bryant <bbryant@digium.com>

	* main/utils.c: Add errors that report any locks held by threads
	  when they are being closed.

2008-06-19 19:22 +0000 [r124049]  Tilghman Lesher <tlesher@digium.com>

	* configs/users.conf.sample, CHANGES, pbx/pbx_config.c: Allow
	  alternative extensions to be specified for a user. (closes issue
	  #12830) Reported by: jcollie Patches:
	  astertisk-trunk-121496-alternate-extensions.patch uploaded by
	  jcollie (license 412)

2008-06-19 18:57 +0000 [r124024]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: Fix bug in sip registration that sets the
	  default port to 5060 for tls.

2008-06-19 18:30 +0000 [r124023]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c, main/timing.c: - Make
	  res_timing_pthread allow a max rate of 100/sec instead of 50/sec
	  - change the "timing test" CLI command to let you specify a
	  timing rate to test

2008-06-19 17:55 +0000 [r123870-123988]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/logger.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/sched.h, include/asterisk/compiler.h: Detect if
	  the installed gcc version supports the warn_unused_result
	  attribute. Reported by mvanbaak via IRC -dev.

	* res/res_config_ldap.c: 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 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 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 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 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 ........

2008-06-19 15:55 +0000 [r123867]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Forwarding non-urgent IMAP messages could
	  inadvertently cause the messages to be marked urgent. This fixes
	  that issue. (closes issue #12895) Reported by: jaroth Patches:
	  urgent_forwarding.patch uploaded by jaroth (license 50)

2008-06-19 15:52 +0000 [r123865]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_externalivr.c: Missing comma (closes issue #12891)
	  Reported by: chris-mac

2008-06-19 14:28 +0000 [r123828-123830]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/queuelog.tex: Update the queuelog.tex documentation as
	  well.

	* apps/app_queue.c: Include original position in TRANSFER entries
	  written to queue_log. (closes issue #12888) Reported by: slavon
	  Patches: app_queue_transfer_patch_trunk.diff uploaded by slavon
	  (license 288)

2008-06-18 22:17 +0000 [r123715-123770]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: 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 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 20:07 +0000 [r123692]  Brett Bryant <bbryant@digium.com>

	* main/tcptls.c: Fix a crash in tcp and tls connections related to
	  reference counts.

2008-06-18 15:08 +0000 [r123650-123652]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 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)

	* apps/app_queue.c: Silly pointers. This fixes a memory corruption
	  I introduced with the attended transfer logging.

2008-06-18 13:09 +0000 [r123648]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c: Channel lock janitor -- add locks around
	  retrieval of channel variables (closes issue #12840) Reported by:
	  pputman Patches: app_dial_threadsafe3.patch uploaded by pputman
	  (license 81)

2008-06-18 00:33 +0000 [r123609]  Sean Bright <sean.bright@gmail.com>

	* res/res_agi.c: Whitespace only

2008-06-17 22:24 +0000 [r123546-123575]  Brett Bryant <bbryant@digium.com>

	* main/astobj2.c: Revert a previous regression in astobj2.c from
	  merging a branch.

	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
	  apps/app_externalivr.c, include/asterisk/tcptls.h,
	  main/astobj2.c: 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 21:42 +0000 [r123544]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_talkdetect.c: Add an option, specifying maximum analysis
	  time for talk detection. (closes issue #12149) Reported by:
	  davevg Patches: app_talkdetect.c.diff uploaded by davevg (license
	  209) (Plus a few additional cleanups by moi)

2008-06-17 21:33 +0000 [r123456-123541]  Mark Michelson <mmichelson@digium.com>

	* main/astobj2.c: Put quotes around "test"

	* main/astobj2.c: _ys pointed out in #asterisk-bugs that he was
	  experiencing a memory leak when running the astobj2 test CLI
	  command. After searching, it appears the leak was in the command
	  handler itself. Each object was allocated (recount = 1) and then
	  linked into a container (refounct = 2). Then at the end of the
	  function, the container was unreffed, causing all the objects to
	  have their refcount decremented by one, leaving the refcount for
	  all objects allocated in that function at 1. I've now added an
	  extra unref to the mix so that the refcount equals zero when the
	  container is unreffed.

	* /, channels/chan_sip.c: 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 ........

	* main/astobj2.c: Add the same fix from revision 123271 to
	  container_destruct_debug.

2008-06-17 20:17 +0000 [r123446-123448]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c, CHANGES: Changes to list peers and users in
	  alpha. order, as per a reasonable request in 12494. Due to
	  changes in trunk to use the astobj2 i/f in the sip channel
	  driver, the order of the entries in the config file was lost,
	  thus the output was in a random order, but no longer.

	* cdr/cdr_tds.c: This solves a crash in the cdr_tds module on 'stop
	  gracefully', for situations where 'settings' is not set to a
	  pointer

2008-06-17 19:00 +0000 [r123393]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c: Fix the check against the max supported
	  rate

2008-06-17 18:57 +0000 [r123358-123392]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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) ........

	* main/pbx.c: If we don't match registrar when destroying a
	  context, it can cause a crash. (closes issue #12835) Reported by:
	  ys Patches: pbx.c.diff uploaded by ys (license 281)

2008-06-17 18:09 +0000 [r123275-123334]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 ........

	* /, apps/app_queue.c: 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 [r123272]  Russell Bryant <russell@digium.com>

	* /, main/astobj2.c: 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 23:05 +0000 [r123238]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix some (more) variables that were
	  forgotten to be renamed, related to 117658

2008-06-16 21:42 +0000 [r123203]  Doug Bailey <dbailey@digium.com>

	* include/asterisk/callerid.h, channels/chan_dahdi.c,
	  main/callerid.c: Clean up code that handles fsk mwi message
	  generation by pulling it from do_monitor and creating its own
	  thread. Added RP-AS mwi message generation using patches from
	  meneault as a basis. (closes issue #8587) Reported by: meneault
	  Tested by: meneault

2008-06-16 21:31 +0000 [r123201]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Oopsie, breakage

2008-06-16 21:15 +0000 [r123166]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix some variables that were forgotten to
	  be renamed, related to 117658

2008-06-16 20:43 +0000 [r123165]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/pbx.c, main/features.c,
	  include/asterisk/pbx.h, apps/app_queue.c, apps/app_stack.c:
	  (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 20:02 +0000 [r123115]  Chris Tooley <chris@tooley.com>

	* apps/app_externalivr.c: Changes response to the ExternalIVR() P
	  command from pipe delimited to comma delimited. closes issue
	  #12804

2008-06-16 19:57 +0000 [r123111-123114]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_iax2.c, channels/chan_mgcp.c: 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 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 17:33 +0000 [r123009-123076]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_tds.c: Last commit for a bit, minor cleanups and move the
	  lock initialization.

	* cdr/cdr_tds.c: Convert to use stringfields. Still some more work
	  to do on config load/reload.

	* cdr/cdr_tds.c: Remove some unused variables

	* cdr/cdr_tds.c: Coding guidelines stuff only.

2008-06-16 13:31 +0000 [r122923-122977]  Russell Bryant <russell@digium.com>

	* configs/modules.conf.sample: Note that only one timing interface
	  should get loaded.

	* res/res_timing_pthread.c (added): Merge res_timing_pthread. This
	  is a timing interface for Asterisk that does not require DAHDI.
	  It's called "pthread" because it uses a pthread API call in the
	  timing thread for sleeping and ensuring we wake up at an
	  appropriate time. I wasn't sure what else to call it. :) The
	  timing API requires a file descriptor that can be polled on. So,
	  when you open a timer, this module creates a pipe and returns the
	  read end of the pipe. There is a background thread that wakes up
	  every 10ms and checks to see if any of the currently open timers
	  need a 'tick' and writes to the appropriate pipe.

	* include/asterisk/_private.h, main/asterisk.c, main/timing.c: Add
	  a "timing test" CLI command. It opens a timer and configures it
	  for 50 ticks per second, and then counts to see how many ticks it
	  actually gets in a second.

	* main/channel.c, include/asterisk/timing.h, main/timing.c: - Fix a
	  typo in a timing API call - Convert the last part of channel.c
	  over to use the timing API. This would not have made a difference
	  when using the dahdi timing module. I noticed it when trying to
	  use another timing source. Oops. :)

2008-06-16 12:32 +0000 [r122870-122920]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 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) ........

2008-06-16 03:33 +0000 [r122834]  Sean Bright <sean.bright@gmail.com>

	* apps/app_fax.c (added): Resurrected app_fax

2008-06-15 15:21 +0000 [r122802]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, funcs/func_channel.c, UPGRADE.txt,
	  channels/chan_iax2.c: Add some more IAX2-specific information
	  about the channel to the CHANNEL() function and begin the
	  transition from SIPCHANINFO() to just using CHANNEL(). (closes
	  issue #12856) Reported by: mostyn Patches:
	  iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
	  (with some additional cleanup by me)

2008-06-13 22:52 +0000 [r122716-122766]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/config.h: Document the input for
	  ast_realtime_require_field()

	* res/res_config_pgsql.c: Properly detect the size of char/varchar
	  fields

2008-06-13 21:45 +0000 [r122714]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c, /: 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 [r122557]  Tilghman Lesher <tlesher@digium.com>

	* main/dial.c: 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 12:53 +0000 [r122523-122526]  Russell Bryant <russell@digium.com>

	* res/res_timing_dahdi.c: Do not allow res_timing_dahdi to be
	  unloaded. We can re-enable this once we add automatic use count
	  handling for timing modules.

	* main/channel.c, res/res_timing_dahdi.c (added), main/file.c,
	  include/asterisk/timing.h, include/asterisk/channel.h,
	  channels/chan_iax2.c, main/asterisk.c, main/timing.c: Merge
	  changes from timing branch - Convert chan_iax2 to use the timing
	  API - Convert usage of timing in the core to use the timing API
	  instead of using DAHDI directly - Make a change to the timing API
	  to add the set_rate() function - change the timing core to use a
	  rwlock - merge a timing implementation, res_timing_dahdi Basic
	  testing was successful using res_timing_dahdi

2008-06-13 11:20 +0000 [r122493]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Implement call parking in chan_skinny.
	  (closes issue #11342) Reported by: DEA Patches:
	  chan_skinny-park.txt uploaded by DEA (license 3)
	  chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7)
	  Tested by: DEA, mvanbaak

2008-06-12 23:58 +0000 [r122461]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a segfault by not trying to store a stack
	  address for long-term use. Instead use the heap. I can't believe
	  this never happened *once* in my developer branch when I was
	  testing.

2008-06-12 23:08 +0000 [r122433]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c, apps/app_parkandannounce.c: (closes issue
	  0012193) Reported by: davidw Patch by: Corydon76, modified by me
	  to work properly with ParkAndAnnounce app

2008-06-12 21:23 +0000 [r122399]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Recommitting revision 122228, which was
	  accidentally reverted as a result of commit 122234.

2008-06-12 20:38 +0000 [r122371]  Russell Bryant <russell@digium.com>

	* include/asterisk/timing.h: Complete the documentation for the
	  timing API.

2008-06-12 18:53 +0000 [r122312]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 18:23 +0000 [r122262]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 17:49 +0000 [r122243-122244]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix pseudo channel allocation errors on
	  startup when using SS7. (from mattf r121914, moving from chan_zap
	  to chan_dahdi)

	* channels/chan_dahdi.c: Make sure we hangup any calls we have and
	  NULL out the ss7call value when we get a reset circuit message.
	  Fixes crash bug. (from mattf r121857, moving from chan_zap to
	  chan_dahdi)

2008-06-12 17:38 +0000 [r122241]  Russell Bryant <russell@digium.com>

	* include/asterisk/network.h: Get default entity ID determination
	  working on Linux again (closes issue #12839)

2008-06-12 17:30 +0000 [r122240]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/timing.h: clarify documentation on how timer
	  intervals should be specified

2008-06-12 17:27 +0000 [r122234]  Jeff Peeler <jpeeler@digium.com>

	* README, apps/app_dahdibarge.c (added),
	  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,
	  include/asterisk/doxyref.h, channels/chan_dahdi.c (added),
	  configure, apps/app_zapscan.c (removed), main/features.c,
	  doc/tex/backtrace.tex, 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),
	  configs/queues.conf.sample, configure.ac,
	  include/asterisk/channel.h, doc/tex/configuration.tex,
	  configs/zapata.conf.sample, Makefile, apps/app_zapbarge.c
	  (removed), doc/janitor-projects.txt, configs/vpb.conf.sample,
	  doc/sms.txt, codecs/codec_dahdi.c (added),
	  contrib/scripts/loadtest.tcl, 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, res/res_musiconhold.c,
	  configs/extensions.ael.sample, include/asterisk/dahdi.h (added),
	  contrib/init.d/rc.mandrake.zaptel, codecs/codec_zap.c (removed),
	  configs/meetme.conf.sample, 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/channelvariables.tex, doc/tex/ael.tex, apps/app_queue.c,
	  doc/tex/enum.tex, apps/app_getcpeid.c, doc/tex/security.tex,
	  configs/sla.conf.sample, include/asterisk/zapata.h (removed),
	  build_tools/menuselect-deps.in, doc/tex/privacy.tex,
	  apps/app_flash.c, doc/osp.txt, main/file.c,
	  contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in,
	  doc/asterisk.sgml, configs/extensions.conf.sample: 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.

2008-06-12 17:14 +0000 [r122232]  Russell Bryant <russell@digium.com>

	* channels/misdn/isdn_lib.c: Make this build under dev mode

2008-06-12 16:25 +0000 [r122228]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merging the work done in the
	  queue-log-atxfer branch. The net result of this work is that
	  attended transfers made by queue members will now show up in the
	  queue_log as a TRANSFER message instead of COMPLETECALLER as it
	  had been. As far as the details go, I created a datastore which
	  is attached to the calling channel just prior to when the caller
	  is bridged with the queue member. If the calling channel is
	  masqueraded, then during the "fixup" portion, the TRANSFER will
	  be logged and the datastore will be removed.

2008-06-12 15:26 +0000 [r122131-122174]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_meetme.c: 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 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 14:56 +0000 [r122091-122128]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c, /, CHANGES: 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 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! ........

2008-06-12 14:21 +0000 [r122062]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile, include/asterisk/timing.h (added), main/timing.c
	  (added): add infrastructure so that timing source can be a
	  loadable module... next steps are to convert channel.c and
	  chan_iax2.c to use this new API, and to move all the
	  DAHDI-specific timing source code into a new res_timing_dahdi
	  module

2008-06-12 14:06 +0000 [r122047]  Russell Bryant <russell@digium.com>

	* main/netsock.c: Don't log not being able to set a default EID.
	  Most people don't care, and those that do can check their setup
	  using CLI commands. (closes issue #12839)

2008-06-11 21:38 +0000 [r121955]  Terry Wilson <twilson@digium.com>

	* main/features.c: Initialize parkingtime to DEFAULT_PARK_TIME
	  instead of 0

2008-06-11 18:53 +0000 [r121914]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Fix pseudo channel allocation errors on
	  startup when using SS7

2008-06-11 18:19 +0000 [r121867]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /, channels/chan_agent.c, main/abstract_jb.c,
	  main/sched.c: 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 17:50 +0000 [r121857]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: 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 17:44 +0000 [r121855]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/realtime_pgsql.sql, UPGRADE.txt,
	  include/asterisk/cdr.h: Expand CDR uniqueid field to 150 chars,
	  to account for maximum systemname. (Closes issue #12831)

2008-06-11 16:11 +0000 [r121805]  Jeff Peeler <jpeeler@digium.com>

	* /, doc/backtrace.txt: 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 ........

2008-06-11 11:52 +0000 [r121770]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/isdn_lib.c: 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. ........

2008-06-10 21:51 +0000 [r121716]  Russell Bryant <russell@digium.com>

	* doc/distributed_devstate.txt: don't refer to asterisk-events, as
	  that implies that the code was checked out from a branch

2008-06-10 21:14 +0000 [r121683]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, res/res_config_odbc.c,
	  res/res_odbc.c: Move the table cache routines to res_odbc, so
	  they can be used from other places (app_voicemail, for example).
	  (Related to bug #11678)

2008-06-10 19:52 +0000 [r121649]  Mark Michelson <mmichelson@digium.com>

	* main/event.c: Add an additional sanity check in case an event is
	  passed between Asterisk boxes with mismatched ie_maps.

2008-06-10 19:03 +0000 [r121599]  Donny Kavanagh <donnyk@gmail.com>

	* codecs/codec_ilbc.c: Revision 117802 changed frame.data to
	  frame.data.ptr however codec_ilbc.c was not updated. This
	  resolves that oversight.

2008-06-10 18:35 +0000 [r121597]  Sean Bright <sean.bright@gmail.com>

	* /, contrib/scripts/get_ilbc_source.sh: 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 15:12 +0000 [r121555-121559]  Russell Bryant <russell@digium.com>

	* main/pbx.c, res/res_ais.c (added), res/ais/clm.c,
	  doc/distributed_devstate.txt (added), res/ais/evt.c, res/ais
	  (added), main/devicestate.c, res/Makefile, res/ais/ais.h,
	  configs/ais.conf.sample (added), CHANGES, apps/app_queue.c: Merge
	  another big set of changes from team/russell/events This commit
	  merges in the rest of the code needed to support distributed
	  device state. There are two main parts to this commit. Core
	  changes: - The device state handling in the core has been updated
	  to understand device state across a cluster of Asterisk servers.
	  Every time the state of a device changes, it looks at all of the
	  device states on each node, and determines the aggregate device
	  state. That resulting device state is what is provided to modules
	  in Asterisk that take actions based on the state of a device. New
	  module, res_ais: - A module has been written to facilitate the
	  communication of events between nodes in a cluster of Asterisk
	  servers. This module uses the SAForum AIS (Service Availability
	  Forum Application Interface Specification) CLM and EVT services
	  (Cluster Management and Event) to handle this task. This module
	  currently supports sharing Voicemail MWI (Message Waiting
	  Indication) and device state events between servers. It has been
	  tested with openais, though other implementations of the spec do
	  exist. For more information on testing distributed device state,
	  see the following doc: - doc/distributed_devstate.txt

	* include/asterisk/event.h, include/asterisk/event_defs.h,
	  main/event.c: Merge some more changes from team/russell/events
	  This commit pulls in a batch of improvements and additions to the
	  event API. Changes include: - the ability to dynamically build a
	  subscription. This is useful if you're building a subscription
	  based on something you receive from the network, or from options
	  in a configuration file. - Add tables of event types and IE types
	  and the corresponding string representation for implementing text
	  based protocols that use these events, for showing events on the
	  CLI, reading configuration that references event information,
	  among other things. - Add a table that maps IE types and the
	  corresponding payload type. - an API call to get the total size
	  of an event - an API call to get all events from the cache that
	  match a subscription - a new IE payload type, raw, which I used
	  for transporting the Entity ID in my code for handling
	  distributed device state. - Code improvements to reduce code
	  duplication - Include the Entity ID of the server that originated
	  the event in every event - an additional event type,
	  DEVICE_STATE_CHANGE, to help facilitate distributed device state.
	  DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE
	  is the aggregate device state change across all servers.

2008-06-10 14:11 +0000 [r121503]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue where session timer headers were
	  present when they should not have been. (closes issue #12706)
	  Reported by: falves11 Patches: chan_sip.c.diff uploaded by rjain
	  (license 226) Tested by: falves11

2008-06-10 14:06 +0000 [r121501]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c:
	  Merge another change from team/russell/events This commit breaks
	  out some logic from pbx.c into a simple API. The hint processing
	  code had logic for taking the state from multiple devices and
	  turning that into the state for a single extension. So, I broke
	  this out and made an API that lets you take multiple device
	  states and determine the aggregate device state. I needed this
	  for some core device state changes to support distributed device
	  state.

2008-06-10 13:36 +0000 [r121444-121496]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 121495 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4
	  lines If we are destroying a dialog only set the MWI dialog
	  pointer on the related peer to NULL if it is the dialog currently
	  being destroyed. (closes issue #12828) Reported by: ramonpeek
	  ........

	* main/channel.c, /: 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 12:50 +0000 [r121401-121441]  Russell Bryant <russell@digium.com>

	* configs/dundi.conf.sample: Update dundi.conf to indicate that the
	  asterisk.conf entityid option can be used to set the entityid
	  used in DUNDi, as well.

	* include/asterisk/utils.h, main/pbx.c, include/asterisk/dundi.h,
	  doc/tex/channelvariables.tex, pbx/pbx_dundi.c,
	  pbx/dundi-parser.c, main/asterisk.c, main/netsock.c,
	  doc/tex/asterisk-conf.tex, pbx/dundi-parser.h: Merge another
	  change from team/russell/events ... DUNDi uses a concept called
	  the Entity ID for unique server identifiers. I have pulled out
	  the handling of EIDs and made it something available to all of
	  Asterisk. There is now a global Entity ID that can be used for
	  other purposes as well, such as code providing distributed device
	  state, which is why I did this. The global Entity ID is set
	  automatically, just like it was done in DUNDi, but it can also be
	  set in asterisk.conf. DUNDi will now use this global EID unless
	  one is specified in dundi.conf. The current EID for the system
	  can be seen in the "core show settings" CLI command. It is also
	  available in the dialplan via the ENTITYID variable.

	* channels/chan_iax2.c: Bump up the debug level of a couple of
	  messages

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  Merge a couple of configure script checks in from
	  team/russell/events. This adds the checks for the CLM and EVT
	  services from the SAForum AIS. I'm going to work on merging in
	  changes from this branch in pieces.

	* main/taskprocessor.c: Properly initialize the cli_ping condition
	  and lock

	* main/taskprocessor.c: Change system header includes to be like
	  how it is done in other files

2008-06-09 22:51 +0000 [r121367]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c, res/res_config_pgsql.c,
	  res/res_config_odbc.c, apps/app_meetme.c, channels/chan_sip.c,
	  include/asterisk/config.h, main/utils.c, apps/app_queue.c,
	  channels/chan_iax2.c, apps/app_voicemail.c: Expand RQ_INTEGER
	  type out to multiple types, one for each precision

2008-06-09 22:42 +0000 [r121365]  Terry Wilson <twilson@digium.com>

	* main/taskprocessor.c: Initialize the lock and destroy lock and
	  cond in the destructor (thanks, mmichelson)

2008-06-09 19:33 +0000 [r121334]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Add storage of the useragent in the realtime
	  database. (Closes AST-38)

2008-06-09 16:55 +0000 [r121282-121286]  Russell Bryant <russell@digium.com>

	* main/dsp.c: arbitrary formatting change to test mantis change
	  (closes issue #12824)

	* main/channel.c: arbitrary formatting change to test a mantis
	  change (closes issue #12824)

	* main/channel.c: Minor formatting change to test a mantis change
	  ... (closes issue #12824)

	* main/channel.c, /: 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 16:35 +0000 [r121279]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: 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 15:08 +0000 [r121230]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: 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 11:40 +0000 [r121197]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_privacy.c, CHANGES: add a new argument to PrivacyManager
	  to specify a context where the entered phone number is checked.
	  You can now define a set of extensions/exten patterns that
	  describe valid phone numbers. PrivacyManager will check that
	  context for a match with the given phone number. This way you get
	  better control. For example people blindly hitting 10 digits just
	  to get past privacymanager Example line in extensions.conf: exten
	  => incoming,n,PrivacyManager(3,10,,route-outgoing)

2008-06-08 01:41 +0000 [r121131-121163]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_console.c: 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: 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 14:18 +0000 [r121079]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /, channels/chan_agent.c: 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 20:24 +0000 [r121010-121042]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, CHANGES: Added a facility for sending
	  arbitrary SIP notify commands from AMI. (closes issue #12562)
	  Reported by: michael-fig Patches: 20080515__bug12562.diff.txt
	  uploaded by Corydon76 (license 14)

	* main/pbx.c: 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 18:30 +0000 [r120906-120960]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: 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 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, main/features.c: 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 17:34 +0000 [r120904]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_exec.c: 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 21:34 +0000 [r120828]  Steve Murphy <murf@digium.com>

	* main/pbx.c: a small fix for a crash that occurs when compiling
	  AEL with global vars

2008-06-05 19:07 +0000 [r120789]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c, include/asterisk/res_odbc.h,
	  res/res_config_pgsql.c, res/res_config_odbc.c, apps/app_meetme.c,
	  channels/chan_sip.c, include/asterisk/config.h,
	  contrib/scripts/dbsep.cgi, apps/app_queue.c,
	  channels/chan_iax2.c, main/config.c,
	  configs/res_pgsql.conf.sample, apps/app_voicemail.c: Merge the
	  adaptive realtime branch, which will make adding new required
	  fields to realtime less painful in the future.

2008-06-05 18:03 +0000 [r120733-120734]  Russell Bryant <russell@digium.com>

	* UPGRADE-1.2.txt: revert 120733, wrong branch

	* UPGRADE-1.2.txt: Update file names

2008-06-05 17:02 +0000 [r120676]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: 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 16:41 +0000 [r120635-120673]  Brett Bryant <bbryant@digium.com>

	* CHANGES: Update CHANGES file for the things done in revision
	  120635.

	* channels/chan_sip.c, funcs/func_channel.c,
	  include/asterisk/rtp.h, main/rtp.c: This patch adds more detailed
	  statistics for RTP channels, and provides an API call to access
	  it, including maximums, minimums, standard deviatinos, and normal
	  deviations. Currently this is implemented for chan_sip, but could
	  be added to the func_channel_read callbacks for the CHANNEL
	  function for any channel that uses RTP. (closes issue #10590)
	  Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz
	  (license 219) rtp_c.diff uploaded by gasparz (license 219)
	  rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff
	  uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff
	  uploaded by sergee (license 138) Tested by: jsmith, gasparz,
	  snuffy, marsosa, chappell, sergee

2008-06-05 15:58 +0000 [r120567-120602]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c, apps/app_stack.c, main/loader.c: 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: Add info on the [compat] section of asterisk.conf.

2008-06-04 22:07 +0000 [r120514]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 20:34 +0000 [r120426-120477]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: MSet doesn't necessarily need chan to be set

	* channels/chan_zap.c, /: Merged revisions 120425 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008)
	  | 6 lines If we fail to setup the PRI request channel, don't
	  continue, exit with an error. (closes issue #11989) Reported by:
	  Corydon76 Patches: 20080213__zap_memleak.diff.txt uploaded by
	  Corydon76 (license 14) ........

2008-06-04 15:38 +0000 [r120337]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: We like tabs.

2008-06-04 14:12 +0000 [r120286]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 13:33 +0000 [r120283]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: 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 23:17 +0000 [r120227-120230]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_channel.c: Add a function, CHANNELS(), which retrieves
	  a list of all active channels. (closes issue #11330) Reported by:
	  rain Patches: func_channel-channel_list_function.diff uploaded by
	  rain (license 327) (with some additional changes by me, mostly to
	  meet coding guidelines)

	* pbx/pbx_loopback.c, /: 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 22:17 +0000 [r120174]  Jeff Peeler <jpeeler@digium.com>

	* /, main/config.c: 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 22:05 +0000 [r120171]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, main/pbx.c, res/res_agi.c, pbx/pbx_realtime.c,
	  configs/pbx_realtime.conf (removed), include/asterisk/options.h,
	  main/asterisk.c: 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 [r120169]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 21:22 +0000 [r120166]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, apps/app_queue.c: Adding two new queue log events. The
	  ADDMEMBER event is logged when a dynamic realtime queue member is
	  added to the queue, and the REMOVEMEMBER event is logged when a
	  dynamic realtime member is removed. Since no calling channel is
	  associated with these events the string "REALTIME" is placed
	  where the channel's unique id is normally placed. (closes issue
	  #12774) Reported by: atis Patches: queue_log_rt_members.patch
	  uploaded by atis (license 242)

2008-06-03 19:48 +0000 [r120064-120129]  Russell Bryant <russell@digium.com>

	* apps/app_channelredirect.c, apps/app_disa.c,
	  apps/app_chanisavail.c: Use proper return values for a few
	  application modules

	* include/asterisk/lock.h: fix build for non debug threads

	* main/channel.c, main/utils.c, include/asterisk/lock.h,
	  utils/ael_main.c, utils/conf2ael.c: Add lock tracking for
	  rwlocks. Previously, lock.h only had the ability to hold tracking
	  information for mutexes. Now, the "core show locks" output will
	  output information about who is holding a rwlock when a thread is
	  waiting on it. (closes issue #11279) Reported by: ys Patches:
	  trunk_lock_utils.v8.diff uploaded by ys (license 281)

2008-06-03 16:19 +0000 [r120012]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: 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 15:49 +0000 [r119930-119998]  Steve Murphy <murf@digium.com>

	* 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, 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-vtest13: 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 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 14:47 +0000 [r119927]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 13:29 +0000 [r119744-119892]  Russell Bryant <russell@digium.com>

	* main/logger.c: 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 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): 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: 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: Add
	  a configure script check for spandsp

	* main/manager.c, /: 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 14:35 +0000 [r119741]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c:
	  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 12:30 +0000 [r119688]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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. ........

2008-06-02 09:35 +0000 [r119586-119637]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/isdn_lib.c: 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/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: 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). ........

2008-06-02 01:08 +0000 [r119531-119534]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 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 21:06 +0000 [r119479]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_dial.c, /: 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 21:51 +0000 [r119423]  Russell Bryant <russell@digium.com>

	* main/utils.c: Fix a minor merge issue that caused a function to
	  not get compiled in with DEBUG_THREADS like it was supposed to

2008-05-30 21:23 +0000 [r119419]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: 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 19:47 +0000 [r119355]  Joshua Colp <jcolp@digium.com>

	* main/autoservice.c, /: 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 16:47 +0000 [r119302]  Michiel van Baak <michiel@vanbaak.info>

	* 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,
	  contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk:
	  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 [r119296-119299]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Suppress warning about pbx structure already
	  existing

	* apps/app_dial.c, include/asterisk/agi.h, CHANGES,
	  apps/app_stack.c: 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 12:59 +0000 [r119239]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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. ........
	  ................

2008-05-30 11:26 +0000 [r119207]  Olle Johansson <oej@edvina.net>

	* include/asterisk/frame.h: Prefer T140 with REDundance before T140
	  without.

2008-05-29 22:28 +0000 [r119157]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: 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 21:30 +0000 [r119126]  Brett Bryant <bbryant@digium.com>

	* include/asterisk/logger.h, main/logger.c, main/asterisk.c: Adds
	  support for changing logger settingss on remote consoles with a
	  new command "logger set level". i.e. "logger set level debug off"
	  (closes issue #10891)

2008-05-29 20:26 +0000 [r119074]  Steve Murphy <murf@digium.com>

	* main/taskprocessor.c: Had to move the ASTERISK_FILE_VERSION decl
	  to just after the include of "asterisk.h" or you get undefined
	  variable errors when you are compiling under the influence of
	  MTX_PROFILE

2008-05-29 20:25 +0000 [r119072]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c, /: 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 19:10 +0000 [r119015]  Michiel van Baak <michiel@vanbaak.info>

	* main/features.c: Make sure the nrfds and nefds are reset to NULL
	  before we enter manage_parkinglot. This will get rid of CLI
	  warnings like: __ast_read: Exception flag set on
	  'SIP/<NUMBER>-<ID>', but no exception handler (closes issue
	  #12748) Reported by: nreinartz Patches:
	  asterisk-multiparking_initialize_filedescr_sets-0.0.1.patch
	  uploaded by nreinartz (license 452)

2008-05-29 19:05 +0000 [r118959-119013]  Russell Bryant <russell@digium.com>

	* /, apps/app_milliwatt.c: 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 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 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 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 17:39 +0000 [r118955-118957]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/lock.h: 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 ........

	* channels/chan_zap.c, /, channels/chan_agent.c,
	  channels/chan_alsa.c, main/utils.c, include/asterisk/lock.h,
	  channels/chan_iax2.c, channels/chan_mgcp.c: 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. ........

2008-05-29 12:12 +0000 [r118911]  Sean Bright <sean.bright@gmail.com>

	* utils/check_expr.c: Avoid build warning when execinfo.h isn't
	  available. (closes issue #12751) Reported by: ys Patches:
	  check_expr.diff uploaded by ys (license 281)

2008-05-29 01:29 +0000 [r118880]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c, /: 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 22:05 +0000 [r118790-118824]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: formatting changes. A lot of whitespace
	  issues have been resolved in this commit Also some doc updates,
	  but that's only 6 lines

	* channels/chan_phone.c, channels/DialTone.h (removed),
	  channels/chan_phone.h (added): rename DialTone.h to chan_phone.h
	  because chan_phone.c is the only file using it

2008-05-28 19:56 +0000 [r118783]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Update to the janitor project for making sure
	  to be thread-safe when retrieving the value of a channel
	  variable. This covers app_queue. This commit also incorporates a
	  logical change. Previously, if MixMonitor is to be used to record
	  the call, all the arguments were parsed first. Then the
	  MixMonitor app would be located. Now the order of these
	  operations has been swapped. Now the app is located first so that
	  we only go through the work of parsing the arguments if the app
	  was found. (closes issue #12742) Reported by: snuffy Patches:
	  bug_12742.diff uploaded by snuffy (license 35)

2008-05-28 17:58 +0000 [r118750]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: remove unused astobj.h header file from
	  chan_skinny.c

2008-05-28 16:01 +0000 [r118702]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: Fixes a bug in chan_iax that uses
	  send_command to poke a peer while a channel is unlocked in some
	  cases, and because it can cause seemingly random failures could
	  be related to some bugs in the tracker...

2008-05-28 15:56 +0000 [r118695]  Russell Bryant <russell@digium.com>

	* utils/check_expr.c: Fix a linkage error related to the lock
	  backtrace support

2008-05-28 14:29 +0000 [r118647]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 14:10 +0000 [r118614-118644]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, include/asterisk/jingle.h: 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: Code simplification

2008-05-27 19:45 +0000 [r118562]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: Remove loop from the detection of a
	  sequence number that acknowledges the receiving of a packet that
	  we've kept in memory just incase the packet needs to be
	  retransmitted.

2008-05-27 19:34 +0000 [r118560]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 19:27 +0000 [r118556]  Russell Bryant <russell@digium.com>

	* include/asterisk/compat.h: Add printf format attribute for
	  vasprintf(). (closes issue #12729) Reported by: snuffy Patches:
	  bug_12729.diff uploaded by snuffy (license 35)

2008-05-27 19:21 +0000 [r118554]  Tilghman Lesher <tlesher@digium.com>

	* /, main/cli.c: 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 19:08 +0000 [r118514]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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 [r118466]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: 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 17:33 +0000 [r118417-118419]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_voicemail.c: Zap is now DAHDI, mkay

	* 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 16:43 +0000 [r118371]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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 ........

2008-05-27 15:46 +0000 [r118359]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/queues.conf.sample: 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 [r118328]  Russell Bryant <russell@digium.com>

	* include/asterisk/compat.h: Add printf attribute to asprintf

2008-05-27 13:30 +0000 [r118300-118302]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_ldap.c: 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: 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 16:17 +0000 [r118223-118252]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: 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) ........

	* main/pbx.c, UPGRADE.txt: Change space-zero to now evaluate to
	  false, as is expected by a great many. (Inspired by a post on the
	  -users list)

2008-05-24 01:14 +0000 [r118176-118178]  Jeff Peeler <jpeeler@digium.com>

	* doc/api-1.6.0-changes.odt (added): add document describing API
	  changes from 1.4.0 to 1.6.0

	* main/features.c: Fixes segfault in parking, patch submitted by
	  bmd.

2008-05-23 22:41 +0000 [r118173-118175]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/lock.h: Make sure not to include non-existent
	  headers if they indeed are non-existent

	* include/asterisk/logger.h, utils/refcounter.c, main/logger.c,
	  utils/hashtest.c, main/utils.c, include/asterisk/lock.h,
	  utils/ael_main.c, utils/hashtest2.c, CHANGES, utils/conf2ael.c,
	  utils/check_expr.c: A new feature thanks to the fine folks at
	  Switchvox! If a deadlock is detected, then the typical lock
	  information will be printed along with a backtrace of the stack
	  for the offending threads. Use of this requires compiling with
	  DETECT_DEADLOCKS and having glibc installed. Furthermore, issuing
	  the "core show locks" CLI command will print the normal lock
	  information as well as a backtraces for each lock. This requires
	  that DEBUG_THREADS is enabled and that glibc is installed. All
	  the backtrace features may be disabled by running the configure
	  script with --without-execinfo as an argument

2008-05-23 21:26 +0000 [r118164]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c, /: 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 21:19 +0000 [r118161]  Brett Bryant <bbryant@digium.com>

	* main/manager.c, main/http.c, include/asterisk/manager.h: 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 20:55 +0000 [r118157-118159]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Get rid of warnings for those silly
	  compilers which warn when freeing a const pointer

	* apps/app_voicemail.c: Use a deep copy on strings that come from
	  ast_events. Otherwise it is likely that after the event is freed,
	  we no longer refer to valid memory. (closes issue #12712)
	  Reported by: tomo1657 Patches: 12712.patch uploaded by putnopvut
	  (license 60) Tested by: tomo1657

2008-05-23 18:09 +0000 [r118129]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Protect the object from changing while the 'odbc
	  show' CLI command is running (Closes issue #12704)

2008-05-23 17:12 +0000 [r118101]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_chanisavail.c, CHANGES: add option 'a' to chanisavail.
	  If you give chanisavail a list of channels, it will only return
	  the first available channel. When this option is set, it will
	  return all the available channels from the given list. (closes
	  issue #12248) Reported by: dagmoller Patches:
	  app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license
	  436) - major changes by me because russellb pointed out some
	  buffer overflows and codeguideline issues. Converted it all to
	  the ast_str_* api Tested by: dagmoller, mvanbaak

2008-05-23 13:00 +0000 [r118053]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/cli.txt (added): 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 [r118049]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, /, main/utils.c: 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 10:33 +0000 [r118020]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c, res/res_jabber.c: - 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 [r117988]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_usbradio.c: Split the compile flags out and wire up
	  some dependencies

2008-05-22 21:42 +0000 [r117983-117986]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Add a
	  compatibility option for upgrading realtime extensions

	* channels/chan_vpb.cc: Fix trunk breakage

2008-05-22 20:01 +0000 [r117950]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_usbradio.c, apps/app_rpt.c: A couple more places
	  the frame data change was missed.

2008-05-22 18:54 +0000 [r117900]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: 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 17:50 +0000 [r117834-117870]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_nbs.c: one more place I forgot

	* channels/chan_console.c: chan_console fixes because of
	  ast_frame.data => ast_frame.data.ptr

2008-05-22 17:10 +0000 [r117828]  Jason Parker <jparker@digium.com>

	* funcs/func_speex.c, codecs/codec_speex.c,
	  formats/format_ogg_vorbis.c, apps/app_jack.c: Fix a few places
	  where frame data was used directly.

2008-05-22 17:08 +0000 [r117802-117825]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_misdn.c: oops

	* channels/chan_misdn.c: forgot chan_misdn

	* main/udptl.c, channels/chan_local.c, main/frame.c,
	  codecs/codec_adpcm.c, apps/app_test.c, apps/app_alarmreceiver.c,
	  formats/format_sln16.c, formats/format_wav_gsm.c,
	  apps/app_ices.c, channels/chan_iax2.c, main/indications.c,
	  channels/chan_skinny.c, formats/format_pcm.c, apps/app_zapscan.c,
	  main/features.c, channels/chan_alsa.c, formats/format_h263.c,
	  apps/app_externalivr.c, formats/format_jpeg.c,
	  apps/app_milliwatt.c, formats/format_gsm.c, apps/app_dial.c,
	  codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c,
	  apps/app_disa.c, include/asterisk/channel.h,
	  channels/iax2-parser.c, apps/app_speech_utils.c,
	  channels/chan_misdn.c, apps/app_zapbarge.c, main/audiohook.c,
	  apps/app_chanspy.c, formats/format_g726.c,
	  channels/chan_unistim.c, apps/app_meetme.c, formats/format_sln.c,
	  codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c,
	  apps/app_followme.c, codecs/codec_zap.c, formats/format_ilbc.c,
	  main/channel.c, channels/chan_phone.c, res/res_agi.c,
	  apps/app_mp3.c, main/app.c, codecs/codec_resample.c,
	  formats/format_h264.c, include/asterisk/frame.h,
	  channels/chan_mgcp.c, codecs/codec_lpc10.c, apps/app_nbscat.c,
	  codecs/codec_a_mu.c, channels/chan_zap.c, channels/chan_sip.c,
	  apps/app_festival.c, codecs/codec_alaw.c, main/slinfactory.c,
	  main/translate.c, res/res_adsi.c, channels/chan_console.c,
	  apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
	  channels/chan_jingle.c, formats/format_vox.c, main/abstract_jb.c,
	  channels/chan_h323.c, main/file.c, apps/app_sms.c,
	  formats/format_g723.c, codecs/codec_ulaw.c, main/dsp.c,
	  formats/format_g729.c: - revert change to ast_queue_hangup and
	  create ast_queue_hangup_with_cause - make data member of the
	  ast_frame struct a named union instead of a void Recently the
	  ast_queue_hangup function got a new parameter, the hangupcause
	  Feedback came in that this is no good and that instead a new
	  function should be created. This I did. The hangupcause was
	  stored in the seqno member of the ast_frame struct. This is not
	  very elegant, and since there's already a data member that one
	  should be used. Problem is, this member was a void *. Now it's a
	  named union so it can hold a pointer, an uint32 and there's a
	  padding in case someone wants to store another type in there in
	  the future. This commit is so massive, because all ast_frame.data
	  uses have to be altered to ast_frame.data.data Thanks russellb
	  and kpfleming for the feedback. (closes issue #12674) Reported
	  by: mvanbaak

2008-05-22 16:05 +0000 [r117794]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Committing a fix pointed out by Atis Lezdins on
	  the asterisk-dev list. Thanks!

2008-05-22 15:49 +0000 [r117792]  Sean Bright <sean.bright@gmail.com>

	* configs/jabber.conf.sample: Minor text fix. roster -> resource.

2008-05-22 13:40 +0000 [r117756]  Russell Bryant <russell@digium.com>

	* build_tools/make_buildopts_h, main/asterisk.c: 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.

2008-05-22 05:10 +0000 [r117725]  Tilghman Lesher <tlesher@digium.com>

	* doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Enhance
	  ExternalIVR with new options and commands. (closes issue #12705)
	  Reported by: ctooley Patches:
	  new_externalivr_argument_format-v2.diff uploaded by ctooley
	  (license 136) new_externalivr_documentation.diff uploaded by
	  ctooley (license 136) and a few additional fixes by me

2008-05-21 22:34 +0000 [r117693]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/logger.h, utils/refcounter.c, main/logger.c,
	  utils/hashtest.c, utils/ael_main.c, utils/hashtest2.c: This
	  change makes it so that logs will report the correct source of
	  verbose messages. Until this change, all verbose messages in
	  Asterisk's log files reported logger.c as the source of the
	  message.

2008-05-21 21:31 +0000 [r117628-117658]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c, /: 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 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 20:27 +0000 [r117625]  Mark Michelson <mmichelson@digium.com>

	* doc/manager_1_1.txt, apps/app_queue.c: Add a new manager event,
	  AgentRingNoAnswer to app_queue. (closes issue #12591) Reported
	  by: CCHAsteria Patches: app_queue_RNA_event.diff uploaded by
	  CCHAsteria (license 477)

2008-05-21 19:39 +0000 [r117575]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 18:43 +0000 [r117520]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: 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 18:31 +0000 [r117517]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Optimize the update_realtime_member_field
	  function by not having to query the database for the member and
	  instead using a cached uniqueid. Special thanks to atis for
	  creating this and for keeping it up to date with necessary
	  changes (closes issue #11896) Reported by: atis Patches:
	  realtime_uniqueid_v6.patch uploaded by atis (license 242) Tested
	  by: atis

2008-05-21 18:29 +0000 [r117481-117515]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.c: 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 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 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 13:39 +0000 [r117431]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_externalivr.c: On socket-based connections, there is no
	  error FD, so don't try waiting on one. (closes issue #12697)
	  Reported by: ctooley Patches:
	  fix_externalivr_waitfor_nandfds-v3.diff uploaded by ctooley
	  (license 136)

2008-05-21 11:24 +0000 [r117401]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: do not die on SDL_ACTIVEEVENT reporting
	  lost focus.

2008-05-21 02:20 +0000 [r117367]  Mark Michelson <mmichelson@digium.com>

	* main/config.c: 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 01:00 +0000 [r117335]  Steve Murphy <murf@digium.com>

	* utils/ael_main.c: 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!

2008-05-20 18:07 +0000 [r117266-117297]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c: + Implement a variant of astman_get_header() to
	  return the first or last match, and possibly skip empty fields.
	  The function is useful (and used here) when a form submits
	  multiple 'Action' fields to the Manager. This change slightly
	  modifies the current behaviour, but only in the case the user
	  supplies multiple 'Action: ' lines and the first ones are empty,
	  so the change is totally harmless. + Fix style on a couple of "if
	  (displayconnects)" statements; + Expand a bit the 'Manager Test'
	  interface, to make it slightly more user friendly. But also
	  comment that the HTML should not be embedded in the C source.
	  None of this stuff needs to be applied to 1.4.

	* main/http.c: Document the possible presence of multiple variables
	  with the same name in http queries, which might confuse the
	  manager. Replace calls to ast_uri_decode() with a local function
	  that also replaces '+' with ' ', as this is the normal encoding
	  for spaces in http requests. This allows passing cli commands to
	  the manager through the http interface.

	* main/http.c: Reverse the check for Cookie: and remove leftover
	  code implementing the same thing. Add an ast_debug() call to help
	  debugging the url matching.

2008-05-20 16:25 +0000 [r117262-117264]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES, res/res_odbc.c: Increase limit of unshared connections
	  from 1023 to 4.2 billion. (Related to issue #12677)

	* res/res_odbc.c: Revert part of previous fix, and heavily comment
	  the logic for object destruction, for future users. (Closes issue
	  #12677)

2008-05-19 20:45 +0000 [r117212]  Russell Bryant <russell@digium.com>

	* main/channel.c: Minor formatting change to test a mantis change
	  ... (issue #12674)

2008-05-19 20:06 +0000 [r117182]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Change
	  the default for the pridialplan parameter to the far more common
	  case of 'unknown', and better document the use of each parameter.
	  (closes issue #12633) Reported by: tzafrir Patches:
	  pridialplan_unknown_2.diff uploaded by tzafrir (license 46)

2008-05-19 16:53 +0000 [r117133-117136]  Joshua Colp <jcolp@digium.com>

	* res/res_smdi.c, /: 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) ........

	* res/res_odbc.c: Remove a premature mutex destroy (the destruction
	  callback will end up destroying it) and use a callback to purge
	  remaining classes. (closes issue #12677) Reported by: falves11

2008-05-19 16:07 +0000 [r117088]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/utils.h, /: 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 16:03 +0000 [r117085]  Joshua Colp <jcolp@digium.com>

	* main/logger.c: 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

2008-05-19 15:47 +0000 [r117084]  Luigi Rizzo <rizzo@icir.org>

	* channels/console_gui.c: trap potential failures of SDL when
	  SDL_WINDOWID is pointing to a random window. This commit is
	  essentially a workaround for some undesirable behaviour of SDL;
	  we should not be doing this in the application, but in the
	  library.

2008-05-19 15:24 +0000 [r117082]  Joshua Colp <jcolp@digium.com>

	* /, channels/h323/ast_h323.cxx: 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 14:54 +0000 [r117024-117053]  Luigi Rizzo <rizzo@icir.org>

	* configs/oss.conf.sample: fix example configuration for video
	  support in chan_oss

	* channels/console_gui.c: Some fixes to the code to support running
	  on an externally supplied window. SDL (at least recent 1.2.x
	  versions) has the ability to run the graphic output into an
	  externally supplied window, whose ID in the environment variable
	  SDL_WINDOWID. Ideally, applications should run unchanged
	  irrespective of who creates the window. Unfortunately, SDL does
	  not subscribe to mouse, key and resize events on externally
	  supplied windows, so we need to do ask for these events
	  explicitly. On passing, also add some code to handle
	  SDL_ACTIVEEVENT so if the X11 window is killed while we are
	  active, we call "stop now" to terminate the asterisk instance.

	* channels/console_video.c: Allow users to specify 'startgui=1' in
	  oss.conf so that the graphic screen for the video console is
	  activated at startup.

2008-05-19 03:44 +0000 [r116979]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 21:15 +0000 [r116948]  Tilghman Lesher <tlesher@digium.com>

	* utils/astcanary.c: 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 19:58 +0000 [r116919]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove duplicate colon on Reason header
	  (closes issue #12678)

2008-05-17 19:39 +0000 [r116800-116884]  Joshua Colp <jcolp@digium.com>

	* channels/iax2-parser.h, channels/chan_iax2.c: Improve native
	  transfers when a chain of IAX2 connections are in use. (closes
	  issue #7567) Reported by: tjd Patches: bug_7567_update_v2.diff
	  uploaded by snuffy (license 35)

	* channels/chan_sip.c: Try to fix attended transfers.

	* /, channels/chan_skinny.c: 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 20:00 +0000 [r116797]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Try to see if we can make our ringback
	  situation a little better

2008-05-16 17:08 +0000 [r116765]  Sean Bright <sean.bright@gmail.com>

	* channels/xpmr/xpmr.c: Compile under dev-mode, please.

2008-05-16 00:51 +0000 [r116731]  Jim Dixon <telesistant@hotmail.com>

	* channels/chan_usbradio.c, channels/xpmr/xpmr.h,
	  channels/xpmr/sinetabx.h, channels/Makefile,
	  channels/xpmr/xpmr.c, apps/app_rpt.c, channels/xpmr/xpmr_coef.h:
	  Bring all app_rpt and chan_usbradio stuff up to date

2008-05-15 22:05 +0000 [r116694]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/utils.h, include/asterisk/strings.h: Add an
	  extra check in ast_strlen_zero, and make ast_assert() not print
	  the file, line, and function name twice. (Closes issue #12650)

2008-05-15 21:54 +0000 [r116663]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Fixes a problem I was having with two SIP
	  phones using Packet2Packet bridging dropping audio nearly
	  immediately. The problem was that the lock on the SIP dialog was
	  not being unlocked while the bridge was still active. (Related to
	  issue #12566)

2008-05-15 17:58 +0000 [r116631]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_csv.c: Don't unload config on reload, when config has not
	  changed. (Closes issue #12652)

2008-05-15 15:40 +0000 [r116590-116594]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: When counting urgent messages when using
	  IMAP storage, take into account that the urgent messages are not
	  in their own folder but are actually "flagged" messages in the
	  INBOX. (closes issue #12659) Reported by: jaroth Patches:
	  urgentfolder_v2.patch uploaded by jaroth (license 50) Tested by:
	  jaroth

	* UPGRADE.txt, apps/app_voicemail.c: Modify externnotify to take
	  the number of urgent voicemails as a final argument instead of
	  the string "Urgent" (closes issue #12660) Reported by: jaroth
	  Patches: externnotify.patch uploaded by jaroth (license 50)

	* apps/app_voicemail.c: Prevent crashes from occurring due to a
	  strcmp of a NULL pointer. (closes issue #12661) Reported by:
	  jaroth Patches: urgentcompare.patch uploaded by jaroth (license
	  50)

2008-05-15 10:56 +0000 [r116557]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c, funcs/func_timeout.c, main/features.c,
	  apps/app_waituntil.c, main/utils.c, main/taskprocessor.c,
	  main/sched.c: Use casts or intermediate variables to remove a
	  number of platform/compiler-dependent warnings when handing
	  struct timeval fields, both reading and printing them. It is a
	  lost battle to handle the different ways struct timeval is
	  handled on the various platforms and compilers, so try to be
	  pragmatic and go through int/long which are universally
	  supported.

2008-05-14 22:15 +0000 [r116522]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, apps/app_chanspy.c: Adding a new option to Chanspy().
	  The 'd' option allows for the spy to press DTMF digits to switch
	  between spying modes. Pressing 4 activates spy mode, pressing 5
	  activates whisper mode, and pressing 6 activates barge mode. Use
	  of this feature overrides the normal operation of DTMF numbers.
	  This feature is courtesy of Switchvox.

2008-05-14 21:54 +0000 [r116471]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix pedanticness.

2008-05-14 21:40 +0000 [r116469]  Russell Bryant <russell@digium.com>

	* main/channel.c, main/udptl.c, include/asterisk/utils.h, /,
	  channels/chan_agent.c, main/abstract_jb.c,
	  include/asterisk/channel.h, main/rtp.c, main/sched.c: 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 [r116467]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: 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 21:11 +0000 [r116461]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Add a missing context unlock. (closes issue #12649)
	  Reported by: ys Patches: pbx.c.diff uploaded by ys (license 281)

2008-05-14 20:43 +0000 [r116407-116410]  Jason Parker <jparker@digium.com>

	* /, configs/voicemail.conf.sample: 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: 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 [r116350-116353]  Joshua Colp <jcolp@digium.com>

	* /, main/Makefile: 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: Make the ldap version setting work without
	  having both version and protocol set. (closes issue #12613)
	  Reported by: suretec

2008-05-14 16:53 +0000 [r116298]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_externalivr.c: 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 16:52 +0000 [r116297]  Jeff Peeler <jpeeler@digium.com>

	* main/pbx.c, main/features.c: Fixed a few problems with
	  multiparking: call not being parked in the correct parking spot,
	  caller not being notified of parking spot position, and
	  improperly hanging up the call during a transfer due to timing
	  out (not providing the extension in which to transfer).

2008-05-14 14:16 +0000 [r116179-116240]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't add linefeed on received MESSAGE

	* channels/chan_sip.c: Properly declare charset for text messages.

	* CREDITS, main/frame.c, channels/chan_sip.c,
	  include/asterisk/rtp.h, CHANGES, include/asterisk/frame.h,
	  main/rtp.c: Adding spport for T.140 RED - Simple RTP redundancy
	  to prevent packet loss in text stream Work sponsored by Omnitor
	  AB, Stockholm, Sweden (http://www.omnitor.se)

	* /, channels/chan_sip.c: 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 ........

	* main/manager.c, pbx/pbx_spool.c, channels/chan_sip.c, CHANGES,
	  sample.call: Add support for codec settings in originate via call
	  file and manager. This is to enable video and text in originated
	  calls. Development sponsored by Omnitor AB, Sweden.
	  (http://www.omnitor.se)

	* res/res_agi.c: Formatting changes (coding guidelines) while
	  thinking about something else...

	* channels/chan_sip.c: Reformatting

	* channels/chan_sip.c: Adding comments

	* pbx/pbx_spool.c: Doxygen formatting change only

2008-05-14 00:20 +0000 [r116089-116138]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_skinny.c: Undo inadvertent changes to chan_skinny
	  caused by the merging of urgent messaging support. Thanks to
	  Damien Wedhorn for pointing out the problem.

	* main/channel.c, /, include/asterisk/lock.h: 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 21:18 +0000 [r116001-116039]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c, /: 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 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 20:29 +0000 [r115945]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_alsa.c: 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 20:18 +0000 [r115939-115941]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Need to clear calling_party_cat variable
	  after we retrieve it

	* channels/chan_zap.c: Add support for receiving calling party
	  category

2008-05-13 18:38 +0000 [r115886]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: 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 17:42 +0000 [r115847-115850]  Russell Bryant <russell@digium.com>

	* funcs/func_speex.c, apps/app_skel.c, apps/app_jack.c:
	  Re-introduce proper error handling that was removed in recent
	  commits.

	* res/res_smdi.c: Initialize the start time in smdi_msg_wait.
	  Somehow this code got lost in trunk.

2008-05-12 20:34 +0000 [r115813]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/install_prereq (added): Add a script which
	  installs every package needed for a Debian install of Asterisk,
	  and includes possible support (to be contributed) for various
	  other distributions. (closes issue #10523) Reported by: tzafrir
	  Patches: install_prereq_2 uploaded by tzafrir (license 46)

2008-05-12 18:39 +0000 [r115784]  Olle Johansson <oej@edvina.net>

	* main/features.c, doc/tex/channelvariables.tex: Add support for
	  playing an audio file for caller and callee at start and stop of
	  monitoring (one-touch monitor). Keep messages short, since the
	  other party is waiting while one party hear the message...

2008-05-12 17:55 +0000 [r115737]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: 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 16:35 +0000 [r115705]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Correctly document state interface for
	  AddQueueMember. Discovered while looking at issue #12626.

2008-05-12 15:17 +0000 [r115669]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: A small change to fix iax2 native bridging.

2008-05-11 03:23 +0000 [r115598-115600]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Add Zap MTP2 support to chan_zap

	* channels/chan_zap.c: Open up audio channel when we get ACM on SS7
	  event

2008-05-10 14:19 +0000 [r115596]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Ensure that "calldate" is acceptable for a
	  column name.

2008-05-10 03:30 +0000 [r115593-115595]  Claude Patry <cpatry@gmail.com>

	* configs/queues.conf.sample: fix a sample since we now required ,
	  and not | for the arguments separator

	* apps/app_skel.c, apps/app_jack.c: ameliorate load and unload to
	  dont use DECLINED or FAILED, when theres no .conf involved.

	* funcs/func_speex.c: since we unregister, that has not been
	  properly registered, i standardized this.

2008-05-09 22:36 +0000 [r115588-115591]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Remove a debug line

	* channels/chan_unistim.c, include/asterisk/app.h, main/manager.c,
	  channels/chan_sip.c, channels/chan_skinny.c, UPGRADE.txt,
	  main/app.c, CHANGES, channels/chan_iax2.c, apps/app_voicemail.c:
	  Adding support for "urgent" voicemail messages. Messages which
	  are marked "urgent" are considered to be higher priority than
	  other messages and so they will be played before any other
	  messages in a user's mailbox. There are two ways to leave an
	  urgent message. 1. send the 'U' option to VoiceMail(). 2. Set
	  review=yes in voicemail.conf. This will give instructions for a
	  caller to mark a message as urgent after the message has been
	  recorded. I have tested that this works correctly with file and
	  ODBC storage, and James Rothenberger (who wrote initial support
	  for this feature) has tested its use with IMAP storage. (closes
	  issue #11817) Reported by: jaroth Based on branch
	  http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
	  Tested by: putnopvut, jaroth

2008-05-09 20:05 +0000 [r115584-115586]  Brett Bryant <bbryant@digium.com>

	* CHANGES: Update CHANGES file for previous commit of ENUM and
	  TXCIDNAME changes.

	* funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: The
	  following patch adds new options and alters the default behavior
	  of the ENUM* functions. The TXCIDNAME lookup function has also
	  gotten a new paramater. The new options for ENUM* functions
	  include 'u', 's', 'i', and 'd' which return the full uri, trigger
	  isn specific rewriting, look for branches into an infrastructure
	  enum tree, or do a direct dns lookup of a number respectively.
	  The new paramater for TXCIDNAME adds a zone-suffix argument for
	  looking up caller id's in DNS that aren't e164.arpa. This patch
	  is based on the original code from otmar, modified by snuffy, and
	  tested by jtodd, me, and others. (closes issue #8089) Reported
	  by: otmar Patches: 20080508_bug8089-1.diff - original code by
	  otmar (license 480), - revised by snuffy (license 35) Tested by:
	  oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant

2008-05-09 17:28 +0000 [r115582]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Allow a password change to be validated by an external script.
	  (closes issue #12090) Reported by: jaroth Patches:
	  vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7)
	  20080509__bug12090.diff.txt uploaded by Corydon76 (license 14)

2008-05-09 16:36 +0000 [r115580]  Joshua Colp <jcolp@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  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 19:20 +0000 [r115552-115569]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 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 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
	  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 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 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 15:04 +0000 [r115548]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Remove unused code as well as demote an
	  error message to a debug message

2008-05-08 14:41 +0000 [r115537-115546]  Russell Bryant <russell@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: 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
	  ........

	* main/sched.c: Fix up a problem that was introduced into the
	  scheduler when it was converted to use doubly linked lists. The
	  schedule() function had an optimization that had it try to guess
	  which direction would be better for the traversal to insert the
	  task into the scheduler queue. However, if the code chose the
	  path where it traversed the queue in reverse, and the result was
	  that the task should be at the head of the queue, then the code
	  would actually put it at the tail, instead. (Problem found by
	  bbryant, debugged and fixed by bbryant and me)

2008-05-07 20:22 +0000 [r115525-115535]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile: Advance to next sounds release

	* res/res_odbc.c: Don't free the object on destroy, as astobj2
	  takes care of that for you

2008-05-07 18:33 +0000 [r115513-115523]  Russell Bryant <russell@digium.com>

	* res/res_config_ldap.c: Only save a password if a username exists.
	  (closes issue #12600) Reported By: suretec Patch by me

	* res/res_config_ldap.c: 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: Let chan_h323 build in dev mode

	* include/asterisk/dlinkedlists.h (added): re-add dlinkedlists.h to
	  trunk, oops!

	* /, include/asterisk/dlinkedlists.h (removed),
	  channels/chan_iax2.c: 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 [r115509]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: Update typos in description fields
	  (closes issue #12598) Reported by: suretec Patches:
	  asterisk_schema_changes.patch uploaded by suretec (license 70)

2008-05-07 13:41 +0000 [r115507]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove redundant header getting. (closes
	  issue #12597) Reported by: hooi

2008-05-06 20:15 +0000 [r115473]  Mark Michelson <mmichelson@digium.com>

	* utils/refcounter.c: Get refcounter to build with LOW_MEMORY
	  defined

2008-05-06 19:55 +0000 [r115419-115423]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: 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 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 19:32 +0000 [r115416]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: 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 15:14 +0000 [r115344]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Change some NOTICE log messages to debug.

2008-05-06 13:55 +0000 [r115342]  Joshua Colp <jcolp@digium.com>

	* /, configure, configure.ac: 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 23:38 +0000 [r115334-115337]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Merge refcounting of res_odbc

	* /, main/logger.c, main/asterisk.c: 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 22:14 +0000 [r115329]  Mark Michelson <mmichelson@digium.com>

	* main/config.c: #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 22:13 +0000 [r115328]  Joshua Colp <jcolp@digium.com>

	* funcs/func_speex.c, /, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac: 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 22:01 +0000 [r115324]  Russell Bryant <russell@digium.com>

	* main/event.c: Simplify code by using a taskprocessor for
	  dispatching events in the Asterisk core.

2008-05-05 21:43 +0000 [r115321]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 [r115315]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove my rant, since I have now replaced
	  the rant with code.

2008-05-05 20:22 +0000 [r115309-115313]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, /: 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 ........

	* include/asterisk/res_odbc.h, /: 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 [r115305]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: 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 19:42 +0000 [r115301-115302]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt: Note change for ExecIf syntax (caught by jmls on
	  IRC)

	* main/manager.c, CHANGES: Optionally display the value of several
	  variables within the Status command. (Closes issue AST-34)

2008-05-05 13:52 +0000 [r115290]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Document the 'B' option of app_chanspy.
	  (closes issue #12582) Reported by: IgorG Patches:
	  app_chanspy_B_option.diff uploaded by IgorG (license 20)

2008-05-05 10:55 +0000 [r115288]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: clarify wording

2008-05-05 03:25 +0000 [r115286]  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.slackware.asterisk,
	  contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk:
	  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 02:11 +0000 [r115277-115283]  Joshua Colp <jcolp@digium.com>

	* /, configure, acinclude.m4: 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 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 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 04:23 +0000 [r115268-115274]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_voicemail.c: app_voicemail uses a taskprocessor for mwi
	  notification subscriptions

	* main/pbx.c: pbx uses a taskprocessor for device state changes

	* apps/app_queue.c: app_queue uses a taskprocessor for device state
	  changes

	* include/asterisk/taskprocessor.h (added), main/Makefile,
	  main/taskprocessor.c (added), include/asterisk/_private.h,
	  main/asterisk.c: A taskprocessor is an object that has a name, a
	  task queue, and an event processing thread. Modules reference a
	  taskprocessor, push tasks into the taskprocessor as needed, and
	  unreference the taskprocessor when the taskprocessor is no longer
	  needed. A task wraps a callback function pointer and a data
	  pointer and is managed internal to the taskprocessor subsystem.
	  The callback function is responsible for releasing task data.
	  Taskprocessor API * ast_taskprocessor_get(..) - returns a
	  reference to a taskprocessor * ast_taskprocessor_unreference(..)
	  - releases reference to a taskprocessor *
	  ast_taskprocessor_push(..) - push a task into a taskprocessor
	  queue Check doxygen for more details

2008-05-02 14:51 +0000 [r115197-115199]  Mark Michelson <mmichelson@digium.com>

	* res/snmp/agent.c: Make res/snmp/agent.c build

	* /, include/asterisk/sched.h: 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 02:56 +0000 [r115104-115159]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/config.h, include/asterisk/compiler.h: Okay,
	  maybe FreeBSD will like this better.

	* include/asterisk/logger.h, channels/chan_sip.c,
	  include/asterisk/config.h, include/asterisk/sched.h,
	  main/asterisk.c, main/config.c, main/sched.c,
	  apps/app_voicemail.c: Add attributes to various API calls, to
	  help track down bugs (and remove a deprecated function)

	* include/asterisk/res_odbc.h, /: 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 23:09 +0000 [r115078]  Brett Bryant <bbryant@digium.com>

	* channels/chan_zap.c, configure, configure.ac, CHANGES: Add two
	  new console commands "pri show version" and "ss7 show version"
	  that will show the version of each library respectively.

2008-05-01 23:06 +0000 [r115076]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, main/pbx.c, apps/app_read.c,
	  funcs/func_timeout.c, apps/app_readexten.c, apps/app_disa.c,
	  include/asterisk/channel.h, apps/app_queue.c, CHANGES,
	  apps/app_speech_utils.c, main/cli.c, main/channel.c, main/dial.c,
	  main/manager.c, apps/app_dumpchan.c, res/res_agi.c, main/app.c,
	  include/asterisk/pbx.h, apps/app_rpt.c: Modify TIMEOUT() to be
	  accurate down to the millisecond. (closes issue #10540) Reported
	  by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: blitzrage

2008-05-01 19:05 +0000 [r115021]  Russell Bryant <russell@digium.com>

	* doc/smdi.txt, res/res_smdi.c, CHANGES: Merge changes from
	  team/russell/smdi-msg-searching This commit adds some new
	  features to the SMDI_MSG_RETRIEVE() dialplan function.
	  Previously, this function only allowed searching by the
	  forwarding station. I have added some options to allow you to
	  also search for messages in the queue by the message desk
	  terminal ID, as well as the message desk number. This originally
	  came up as a suggestion on the asterisk-dev mailing list.

2008-05-01 19:00 +0000 [r115018]  Tilghman Lesher <tlesher@digium.com>

	* /, main/utils.c: 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 18:28 +0000 [r114977]  Brett Bryant <bbryant@digium.com>

	* funcs/func_speex.c: Add "read" capability to new libspeex
	  functions in func_speex.c. func_speex.c is based on contributions
	  from Switchvox.

2008-05-01 17:28 +0000 [r114931]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt: 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 16:57 +0000 [r114926]  Brett Bryant <bbryant@digium.com>

	* funcs/func_speex.c (added), include/asterisk/audiohook.h,
	  main/audiohook.c, CHANGES: Add two new dialplan functions from
	  libspeex for applying audio gain control and denoising to a
	  channel, AGC() and DENOISE(). Also included, is a change to the
	  audiohook API to add a new function (ast_audiohook_remove) that
	  can remove an audiohook from a channel before it is detached.
	  This code is based on a contribution from Switchvox.

2008-05-01 16:49 +0000 [r114922]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: 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 20:51 +0000 [r114912]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  support for specifying the registration expiry on a per
	  registration basis in the register line. This comes from a
	  Switchvox patch. (issue AST-24)

2008-04-30 19:30 +0000 [r114906]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding new
	  configuration options to app_queue. This adds two new values to
	  announce-position, "limit" and "more," as well as a new option,
	  announce-position-limit. For more information on the use of these
	  options, see CHANGES or configs/queues.conf.sample. (closes issue
	  #10991) Reported by: slavon Patches: app_q.diff uploaded by
	  slavon (license 288) Tested by: slavon, putnopvut

2008-04-30 19:21 +0000 [r114904]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, apps/app_minivm.c, apps/app_morsecode.c,
	  apps/app_macro.c, apps/app_externalivr.c, apps/app_chanspy.c,
	  apps/app_stack.c, apps/app_speech_utils.c, apps/app_voicemail.c,
	  apps/app_while.c: Lock around variables retrieved, and copy the
	  values, if they stay persistent, since another thread could
	  remove them. (closes issue #12541) Reported by: snuffy Patches:
	  bug_12156_apps.diff uploaded by snuffy (license 35) Several
	  additional changes by me

2008-04-30 16:55 +0000 [r114899]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 16:34 +0000 [r114892]  Russell Bryant <russell@digium.com>

	* /, channels/chan_console.c, channels/chan_iax2.c: 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 16:14 +0000 [r114888]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_console.c: 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 14:49 +0000 [r114876-114884]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/iax2.h, channels/chan_iax2.c: 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 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) ........

2008-04-30 05:05 +0000 [r114874]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Document the Incomplete application addition.

2008-04-29 22:54 +0000 [r114866]  Jeff Peeler <jpeeler@digium.com>

	* channels/iax2-provision.c: 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 21:07 +0000 [r114857]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Patching app_chanspy to jibe better with what
	  is documented. This allows for a colon-delimited list of
	  spygroups to be specified when calling the ChanSpy application
	  with the 'g' option. Prior to this, you could only specify a
	  single group when using the 'g' option. I also have upped the
	  maximum number of spygroups to 128 and added a #define so that
	  this can be easily increased or decreased later. (closes issue
	  #12497) Reported by: jsmith Patches:
	  app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license
	  15) Tested by: atis, jvandal

2008-04-29 20:05 +0000 [r114852]  Jason Parker <jparker@digium.com>

	* phoneprov/polycom.xml: Fix formatting

2008-04-29 19:42 +0000 [r114849]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 18:58 +0000 [r114845]  Kevin P. Fleming <kpfleming@digium.com>

	* main/features.c: fix this logic to actually be correct... the fd
	  can't be *both* -1 and an array index to be checked in rfds/efds
	  (bug found by gcc-4.3)

2008-04-29 18:48 +0000 [r114832-114841]  Mark Michelson <mmichelson@digium.com>

	* UPGRADE.txt, apps/app_directory.c: Make app_directory dependent
	  on app_voicemail. This is because the function which says the
	  person's name is handled inside app_voicemail now.

	* apps/app_directory.c, apps/app_voicemail.c: Since there is now a
	  globally available function for saying someone's name, a LOT of
	  functions in app_directory can be removed since the ODBC-specific
	  lookups are accomplished within app_voicemail. This change
	  greatly reduces the amount of lines in app_directory that were
	  solely for the purpose of looking up a name when ODBC_STORAGE is
	  specified for voicemail. This commit also makes the name-saying
	  interruptable via DTMF.

	* apps/app_directory.c: Fix a crash happening in app_directory.
	  This crash would occur if a users.conf existed.

2008-04-29 17:10 +0000 [r114830]  Jason Parker <jparker@digium.com>

	* res/res_config_pgsql.c, /: 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 12:54 +0000 [r114824]  Kevin P. Fleming <kpfleming@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: 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 22:38 +0000 [r114813]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/app.h, main/app.c, CHANGES, apps/app_chanspy.c,
	  apps/app_voicemail.c: Adding a new option 'n' to app_chanspy.
	  This option allows for the name of the spied-on party to be
	  spoken instead of the channel name or number. This was
	  accomplished by adding a new function pointer to point to a
	  function in app_voicemail which retrieves the name file and plays
	  it. This makes for an easy way that applications may play a
	  user's name should it be necessary. app_directory, in particular,
	  can be simplified greatly by this change. This change comes as a
	  suggestion from Switchvox, which already has this feature. AST-23

2008-04-28 17:00 +0000 [r114776]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: 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 16:37 +0000 [r114773]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Add
	  incomplete matching to PBX code and app_dial (closes issue
	  #12351) Reported by: Corydon76 Patches:
	  20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76
	  (license 14) pbx_incomplete_with_timeout.diff uploaded by fabled
	  (license 448) Tested by: Corydon76, fabled

2008-04-28 13:42 +0000 [r114713]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Update autoconf logic with latest API
	  change for libss7.

2008-04-28 04:53 +0000 [r114706-114709]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: 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) ........

	* apps/app_voicemail.c: Fix breakage caused by #12028. (Closes
	  issue #12535)

2008-04-27 22:54 +0000 [r114703]  Russell Bryant <russell@digium.com>

	* channels/chan_skinny.c: s/chan_zap/chan_skinny/

2008-04-27 15:17 +0000 [r114700]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: 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 01:28 +0000 [r114696]  Sean Bright <sean.bright@gmail.com>

	* /, configure, configure.ac: 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 15:08 +0000 [r114683-114692]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Unleak reference

	* /, contrib/scripts/vmail.cgi: 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) ........

	* channels/chan_sip.c: Add 'sip qualify peer <peer>' command (with
	  AMI SIPqualifypeer) (closes issue #12524) Reported by: ctooley
	  Patches: sip_qualify_peer.diff.2 uploaded by ctooley (license
	  136) some modifications for trunk by Corydon76 Tested by:
	  Corydon76

2008-04-25 22:24 +0000 [r114678]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, apps/app_chanspy.c: Adding a new option, 'B' to
	  app_chanspy. This option allows the spy to barge on the call. It
	  is like the existing whisper option, except that it allows the
	  spy to talk to both sides of the conversation on which he is
	  spying. This feature has existed in Switchvox, and this merges
	  the functionality into Asterisk. (AST-32)

2008-04-25 22:04 +0000 [r114674-114676]  Russell Bryant <russell@digium.com>

	* pbx/pbx_lua.c: 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 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 20:20 +0000 [r114665-114667]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, apps/app_waitforring.c, apps/app_minivm.c,
	  apps/app_zapscan.c, apps/app_sms.c, apps/app_externalivr.c,
	  apps/app_followme.c, apps/app_queue.c, apps/app_rpt.c,
	  apps/app_playback.c, apps/app_parkandannounce.c,
	  apps/app_speech_utils.c: Whitespace changes only

	* main/app.c: Oops, this isn't necessarily AGI that is forking
	  anymore

2008-04-25 19:33 +0000 [r114663]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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 18:32 +0000 [r114660]  Jason Parker <jparker@digium.com>

	* apps/app_directed_pickup.c, apps/app_pickupchan.c (removed):
	  Merge app_pickupchan with app_directed_pickup, for AST-27.
	  Initially, this was to be a new feature, with a patch from
	  Switchvox, but after discussions, it was noted that this feature
	  already existed in trunk. The resulting discussions ended in a
	  comment that was along the lines of "the patch provided here is a
	  lot smaller than what is already in trunk, because it doesn't
	  create a new application and duplicate existing code" It was
	  decided that these two applications could be easily merged to
	  reduce code duplication. SO, that's what this does.

2008-04-25 18:18 +0000 [r114656]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: This patch allows for forwarding a message
	  with a "comment" attachment if using IMAP storage for voicemail.
	  The comment will be recorded and attached as a second attachment
	  in addition to the original message. This will be invoked if you
	  choose to prepend a message the way you would with file or ODBC
	  storage (closes issue #12028) Reported by: jaroth Patches:
	  forward_with_comment_v2.patch uploaded by jaroth (license 50)
	  Tested by: jaroth

2008-04-25 18:18 +0000 [r114655]  Russell Bryant <russell@digium.com>

	* main/features.c: Merge code from team/russell/parking_updates Add
	  some additional features to the core park_call_full() function,
	  and expose them as options to the Park() application. The
	  functionality being added is the ability to specify a custom
	  return extension/context/priority, a custom timeout, and a couple
	  of options. The options are to play ringing instead of MOH to the
	  parked caller, and to randomize parking spot selection. (code
	  inspired by the patch in AST-17, code from switchvox)

2008-04-25 16:25 +0000 [r114651]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a memory leak and protect against
	  potential dereferences of a NULL pointer.

2008-04-25 13:56 +0000 [r114644]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_console.c: Speaking of building...

2008-04-24 22:16 +0000 [r114637]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c,
	  channels/chan_zap.c, channels/chan_sip.c, apps/app_disa.c,
	  apps/app_alarmreceiver.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, apps/app_followme.c, apps/app_queue.c,
	  channels/chan_iax2.c, channels/chan_oss.c, main/channel.c,
	  channels/chan_jingle.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_alsa.c, apps/app_externalivr.c,
	  channels/chan_mgcp.c: Pass the hangup cause all the way to the
	  calling app/channel. (closes issue #11328) Reported by: rain
	  Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt
	  uploaded by Corydon76 (license 14) brought up-to-date to trunk by
	  me

2008-04-24 22:11 +0000 [r114635]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Hey look, it builds. (closes issue #12519)
	  Reported by: falves11

2008-04-24 21:35 +0000 [r114625-114633]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 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 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 19:54 +0000 [r114617-114622]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: 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: 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 16:47 +0000 [r114612]  Jason Parker <jparker@digium.com>

	* channels/chan_misdn.c, /: 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 15:56 +0000 [r114609]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 14:59 +0000 [r114606]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 14:55 +0000 [r114598-114604]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Change a verbose message to debug. (closes
	  issue #12514)

	* /, main/http.c: 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 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 18:33 +0000 [r114595]  Jason Parker <jparker@digium.com>

	* /, res/res_musiconhold.c: 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 [r114588-114592]  Russell Bryant <russell@digium.com>

	* main/manager.c, /, include/asterisk/manager.h: 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 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 16:53 +0000 [r114585]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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 14:55 +0000 [r114580]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, /: 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 00:58 +0000 [r114575-114577]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/logger.h, include/asterisk/astobj.h,
	  apps/app_voicemail.c: Round 2 of IMAP_STORAGE app_voicemail.c
	  fixes: This fixes a bug that was thought to be fixed already.
	  app_voicemail, if using IMAP_STORAGE, has a problem because the
	  IMAP header files include syslog.h, which define LOG_WARNING and
	  LOG_DEBUG to be different than what Asterisk uses for those same
	  macros. This was "fixed" in the past by including all the IMAP
	  header files prior to including asterisk.h. This fix worked...
	  unless you were to try to compile with MALLOC_DEBUG. MALLOC_DEBUG
	  prepends the inclusion of astmm.h to every file, which means that
	  no matter what order the includes are in in app_voicemail, the
	  unexpected values for LOG_WARNING and LOG_DEBUG will be in place.
	  The action taken for this fix was to define AST_LOG_* macros in
	  addition to the LOG_* macros already defined. These new macros
	  are used in app_voicemail.c, logger.h, and astobj.h right now,
	  and their use will be encouraged in the future. In consideration
	  of those who have written third-party modules which use the LOG_*
	  macros, these will NOT be removed from the source, however future
	  use of these macros is discouraged.

	* apps/app_voicemail.c: 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 23:58 +0000 [r114572]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: 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 22:17 +0000 [r114559]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 21:57 +0000 [r114553]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (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 20:25 +0000 [r114548]  Russell Bryant <russell@digium.com>

	* main/channel.c: re-add a fix that got lost with a recent change

2008-04-22 18:14 +0000 [r114540]  Jason Parker <jparker@digium.com>

	* main/pbx.c, include/asterisk/pbx.h, apps/app_queue.c: 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 18:04 +0000 [r114533-114538]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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) ........

	* CHANGES, apps/app_jack.c: Add a c() option for the Jack()
	  application and JACK_HOOK() funciton for supplying a custom
	  client name. Using the channel name is still the default. This
	  was done at the request of Jared Smith.

2008-04-22 15:54 +0000 [r114529]  Joshua Colp <jcolp@digium.com>

	* configs/sip_notify.conf.sample, channels/chan_sip.c: Add support
	  for authenticating on a NOTIFY request. This is useful for phones
	  that require it when sending them a special packet to get them to
	  do something (such as reload their configuration). (closes issue
	  #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch
	  uploaded by IgorG (license 20)

2008-04-22 15:46 +0000 [r114527]  Russell Bryant <russell@digium.com>

	* main/manager.c: 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)

2008-04-22 14:38 +0000 [r114520]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c, main/utils.c: Hopefully, this will resolve
	  the issues that russellb had with this log_show_lock(). I
	  gathered the code that filled the string, and put it in a
	  different func which I cryptically call
	  "append_lock_information()". Now, both log_show_lock(), and
	  handle_show_locks() both call this code to do the work. Tested,
	  seems to work fine. Also, log_show_lock was modified to use the
	  ast_str stuff, along with checking for successful ast_str
	  creation, and freeing the ast_str obj when finished. A break was
	  inserted to terminate the search for the lock; we should never
	  see it twice. An example usage in chan_sip.c was created as a
	  comment, for instructional purposes.

2008-04-21 23:42 +0000 [r114487]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_unistim.c, channels/chan_zap.c,
	  channels/chan_sip.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_console.c,
	  channels/chan_iax2.c, configs/features.conf.sample,
	  configs/iax.conf.sample, channels/chan_jingle.c,
	  channels/chan_skinny.c, funcs/func_channel.c, main/features.c,
	  apps/app_dumpchan.c, configs/sip.conf.sample,
	  channels/chan_mgcp.c: (closes issue #6113) Reported by: oej
	  Tested by: jpeeler This patch implements multiple parking lots
	  for parked calls. The default parkinglot is used by default,
	  however setting the channel variable PARKINGLOT in the dialplan
	  will allow use of any other configured parkinglot. See
	  configs/features.conf.sample for more details on setting up
	  another non-default parkinglot. Also, one can (currently) set the
	  default parkinglot to use in the driver configuration file via
	  the parkinglot option. Patch initially written by oej, brought up
	  to date and finalized by mvanbaak, and then stabilized and
	  converted to astobj2 by me.

2008-04-21 22:50 +0000 [r114456]  Doug Bailey <dbailey@digium.com>

	* phoneprov/polycom.xml: Change the time zone offset from a hard
	  code to use res_phoneprov variables

2008-04-21 21:13 +0000 [r114423]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest17, main/ast_expr2.y,
	  doc/tex/channelvariables.tex, doc/tex/ael.tex, CHANGES,
	  pbx/ael/ael-test/ael-ntest24/extensions.ael (added),
	  pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ael-ntest24
	  (added), pbx/ael/ael-test/ref.ael-test19, main/ast_expr2.c,
	  pbx/ael/ael-test/ref.ael-ntest10, main/ast_expr2.h,
	  pbx/ael/ael-test/ref.ael-test1, main/ast_expr2f.c,
	  pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-ntest24
	  (added), pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-vtest13,
	  main/ast_expr2.fl: (closes issue #12467) Reported by: atis Tested
	  by: murf This upgrade adds the ~~ (concatenation) string operator
	  to expr2. While not needed in normal runtime pbx operation, it is
	  needed when raw exprs are being syntax checked. This plays into
	  future syntax- unification plans. By permission of atis, this
	  addition in trunk and the reason of why things are as they are
	  will suffice to close this bug. I also added a short note about
	  the previous addition of "sip show sched" to the CLI in CHANGES,
	  which I discovered I forgot in a previous commit.

2008-04-21 18:44 +0000 [r114389]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Add support for generic name transmission
	  (#12484) on SS7 in chan_zap

2008-04-21 15:34 +0000 [r114327]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_authenticate.c: 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 15:01 +0000 [r114325]  Russell Bryant <russell@digium.com>

	* doc/janitor-projects.txt: Add a simple janitor project

2008-04-21 14:40 +0000 [r114320-114323]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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: 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)

2008-04-20 14:52 +0000 [r114314]  Sean Bright <sean.bright@gmail.com>

	* cdr/cdr_pgsql.c: Minor logging cleanups

2008-04-19 16:58 +0000 [r114303]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: SS7:Added - Generic Name / Access Transport
	  / Redirecting Number handling. #12425

2008-04-19 00:15 +0000 [r114295]  Sean Bright <sean.bright@gmail.com>

	* utils: Ignore refcounter

2008-04-18 21:51 +0000 [r114276-114285]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: 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 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 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 19:35 +0000 [r114261-114271]  Joshua Colp <jcolp@digium.com>

	* channels/chan_unistim.c: 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

	* apps/app_meetme.c, CHANGES: Add MEETME_INFO dialplan function
	  that allows querying various properties of a Meetme conference.
	  (closes issue #11691) Reported by: junky Patches:
	  meetme_info.patch uploaded by jpeeler (license 325)

2008-04-18 18:03 +0000 [r114259]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, /, main/callerid.c: 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 16:11 +0000 [r114254]  Joshua Colp <jcolp@digium.com>

	* res/res_config_ldap.c: If the parsing of the config file fails
	  make sure we unlock ldap_lock. (closes issue #12477) Reported by:
	  IgorG

2008-04-18 16:05 +0000 [r114253]  Doug Bailey <dbailey@digium.com>

	* res/res_http_post.c: Add g__object_unref to clean up gmime
	  message object

2008-04-18 13:38 +0000 [r114246]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: 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 06:53 +0000 [r114243]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_setcallerid.c, /: 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 22:24 +0000 [r114231-114233]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: 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 ........

	* main/utils.c: Make this file compile. The variable str is never
	  set anywhere. Furthermore, it duplicates a lot of code. I will
	  leave it to murf to clean up.

2008-04-17 21:09 +0000 [r114229]  Jeff Peeler <jpeeler@digium.com>

	* CHANGES: added info describing DNS manager

2008-04-17 21:04 +0000 [r114208-114227]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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
	  ........

	* main/frame.c, /, include/asterisk/dsp.h,
	  include/asterisk/frame.h, main/dsp.c: 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 16:25 +0000 [r114205]  Russell Bryant <russell@digium.com>

	* Makefile, /: 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 15:12 +0000 [r114202]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES: fileio.h does not exist; io.h does,
	  though.

2008-04-17 14:45 +0000 [r114201]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: Thanks to snuff for finding these omissions

2008-04-17 13:46 +0000 [r114199]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, res/res_jabber.c: 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 [r114196]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: 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 12:25 +0000 [r114192-114194]  Sean Bright <sean.bright@gmail.com>

	* CHANGES: Update the CHANGES file with yesterday's ChanSpy change.
	  Sorry Kevin, just saw your e-mail.

	* /, apps/app_chanspy.c: 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 23:53 +0000 [r114190]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  doc/chan_sip-perf-testing.txt (added): This is the scariest
	  commit I've done in a long time. This is the astobj2-ification of
	  chan_sip. I've tested a number of scenarios like crazy. It used
	  to have 4x the call setup/teardown performance of trunk, but now
	  it's roughly at parity. I will attempt to find the bottlenecks
	  and get it back to the 4x mark. The changes made were somewhat
	  invasive, but the value to the community of these upgrades
	  outweighs waiting further for more testing. Every change being
	  made to chan_sip was lousing this code up when we tried to merge.
	  Peers, Users, Dialogs, are all now astobj2 objects, indexed via
	  hashtables. Refcounting is used to track objects and free them at
	  the bitter end of their lives. Please file issues on
	  bugs.digium.com, and PLEASE, please, please be patient. One
	  natural advantage to all the hash-table work is that loading
	  large sip.conf files full of thousands of peers now goes much
	  faster. One more please: PLEASE help thrash this code and test
	  it.

2008-04-16 22:57 +0000 [r114188]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/logger.h, apps/app_nbscat.c,
	  include/asterisk/app.h, apps/app_festival.c, apps/app_mp3.c,
	  res/res_agi.c, apps/app_zapras.c, main/logger.c, main/app.c,
	  apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c,
	  main/asterisk.c: Standardized routines for forking processes
	  (keeps all the specialized code in one place).

2008-04-16 20:54 +0000 [r114187]  Steve Murphy <murf@digium.com>

	* main/utils.c, include/asterisk/lock.h: A small enhancement-- I
	  added the routine log_show_lock to utils.c, which if the
	  mentioned lock has been acquired, this routine will log to the
	  console the normal info about that lock you'd see from the CLI
	  when you do a 'core show locks'. It's solely for debug-- if the
	  lock is NOT acquired, there is no output. I use it to show
	  'unexpected' locks, to see where/why a lock is pre-locked. This
	  command is to be called from points of interest, like just before
	  a trylock, and helps to spot fleeting, highly temporal locks that
	  normally are not locked...

2008-04-16 20:47 +0000 [r114185]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: 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
	  ........

2008-04-16 20:28 +0000 [r114172-114183]  Steve Murphy <murf@digium.com>

	* main/event.c: Introducing a small optimization to
	  event_unsubscribe; events now use a Doubly-Linked list for
	  events, gives fast deletions, for the sake of channel driver mwi
	  events. From team/murf/bug11210.

	* include/asterisk/sched.h, CHANGES, main/sched.c: Introducing a
	  small upgrade to the ast_sched_xxx facility, to keep it from
	  eating up lots of cpu cycles. See CHANGES. From the
	  team/murf/bug11210 branch.

	* utils/Makefile, utils/refcounter.c (added),
	  include/asterisk/astobj2.h, CHANGES, main/astobj2.c: Introducing
	  various astobj2 enhancements, chief being a refcount tracing
	  feature, and various documentation updates in astobj2.h, and the
	  addition of standalone utility, refcounter, that will filter the
	  trace output for unbalanced, unfreed objects. This comes from the
	  team/murf/bug11210 branch.

	* tests/test_dlinklists.c (added), include/asterisk/dlinkedlists.h
	  (added), CHANGES: Introducing doubly linked lists to trunk from
	  branch team/murf/bug11210.

2008-04-16 12:23 +0000 [r114165]  Sean Bright <sean.bright@gmail.com>

	* apps/app_chanspy.c: Add the ability to disable channel technology
	  name playback when speaking the current channel name

2008-04-15 20:51 +0000 [r114152]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Oops, buffer wasn't long enough for query

2008-04-15 20:39 +0000 [r114150-114151]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: 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: Adding chanvar to SIPPEER from 1.4 branch

2008-04-15 20:27 +0000 [r114149]  Jason Parker <jparker@digium.com>

	* apps/app_directory.c: If somebody enters a digit during
	  ast_stream_and_wait, the return value is the digit, which we need
	  to use later.

2008-04-15 19:59 +0000 [r114146]  Steve Murphy <murf@digium.com>

	* main/pbx.c: 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 17:54 +0000 [r114143]  Sean Bright <sean.bright@gmail.com>

	* apps/app_chanspy.c: I'm not sure why, but "this" bothers me. Ba
	  dum dum.

2008-04-15 17:21 +0000 [r114131-114141]  Jason Parker <jparker@digium.com>

	* channels/chan_unistim.c: 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 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 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) ........

	* main/channel.c: Convert several DEBUG logs into ast_debug.
	  (closes issue #12444) Reported by: IgorG Patches:
	  channel_c_debug.diff uploaded by IgorG (license 20)

2008-04-14 19:58 +0000 [r114124-114127]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Need a new buffer for each loop

	* res/res_phoneprov.c: Don't unref user twice on failure. Also,
	  when adding sorted list of users, it is best to check the entry
	  already in the list for a "next" entry instead of the newly
	  created entry...

2008-04-14 18:34 +0000 [r114121]  Jason Parker <jparker@digium.com>

	* /, channels/chan_h323.c: 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 [r114118]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: 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 16:32 +0000 [r114115]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astcli: Make tab-completion work for all cases

2008-04-14 16:25 +0000 [r114113]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /, apps/app_queue.c: 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 15:36 +0000 [r114109]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: During hangup it is possible for p->chan
	  or p->owner to be NULL, so just return what the channel is
	  bridged to instead of what they are *really* bridged to. Thanks
	  Matt Nicholson!

2008-04-14 15:01 +0000 [r114107]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: 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 14:53 +0000 [r114101-114104]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 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 02:55 +0000 [r114096-114098]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astcli: Add tab command-line completion (Closes
	  issue #12428)

	* apps/app_meetme.c: Use ast_mkdir instead of mkdir (Closes issue
	  #12430)

2008-04-12 16:21 +0000 [r114092-114093]  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 23:26 +0000 [r114085-114090]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: 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: Make the sample config match the
	  contributed LDAP schema (Closes issue #12421)

	* res/res_config_ldap.c: 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 22:48 +0000 [r114080-114084]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_iax2.c: 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 ........

	* res/res_phoneprov.c: Make sure that ${LINE} is set even if
	  linenumber is not set in users.conf

2008-04-11 22:09 +0000 [r114077]  Doug Bailey <dbailey@digium.com>

	* phoneprov/polycom_line.xml: Change the number of line keys per
	  registration from 2 to 1

2008-04-11 21:04 +0000 [r114067]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Fix the fact that global_variables 1)
	  weren't being updated on reload (thanks for the report, Doug),
	  and 2) weren't actually being appended to the list of profile
	  variables because build_profile was called before the list was
	  populated. Also needed to free the contents returned by
	  load_file().

2008-04-11 15:49 +0000 [r114064]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: 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 14:54 +0000 [r114061]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_ldap.c: 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 22:02 +0000 [r114052]  Mark Michelson <mmichelson@digium.com>

	* 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, utils/conf2ael.c,
	  utils/check_expr.c: 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 20:28 +0000 [r114049]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, CHANGES: A 'b' option has been added which
	  causes chan_local to return the actual channel that is behind it
	  when queried. This is useful for transfer scenarios as the actual
	  channel will be transferred, not the Local channel. If you have
	  been using Local channels as queue members and having issues when
	  the agent did a blind transfer this option may solve the issue.

2008-04-10 19:58 +0000 [r114046]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 19:04 +0000 [r114042]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astcli: 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 [r114036]  Jason Parker <jparker@digium.com>

	* /, main/file.c: 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 15:10 +0000 [r114022-114030]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: 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: 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)

	* channels/chan_unistim.c, channels/chan_skinny.c, main/rtp.c: Fix
	  spelling of existent in a few places. (closes issue #12409)
	  Reported by: candlerb

	* /, channels/chan_sip.c: 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 22:32 +0000 [r113928-113980]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 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 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 19:00 +0000 [r113875]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/cdr.conf.sample, cdr/cdr_csv.c: 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 18:05 +0000 [r113840]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Enable enough RTP bridging to allow P2P to
	  work. (closes issue #11901) Reported by: pj

2008-04-09 17:56 +0000 [r113838]  Jason Parker <jparker@digium.com>

	* contrib/scripts/astcli: Fix a small file handle "leak" pointed
	  out by jjshoe on #asterisk.

2008-04-09 17:48 +0000 [r113836]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: 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 17:41 +0000 [r113834]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Move all messages wrapped in skinnydebug
	  from debug to verbose. (closes issue #12224) Reported by: DEA
	  Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3)

2008-04-09 16:52 +0000 [r113785]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: 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 16:23 +0000 [r113731-113752]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Mark recent additions from #11954 and #12254

	* configs/voicemail.conf.sample, apps/app_voicemail.c: Permit
	  message wrap-around during message retrieval. (closes issue
	  #12254) Reported by: andrew Patches: bug-12253.diff uploaded by
	  snuffy (license 35)

2008-04-09 14:41 +0000 [r113682]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 13:55 +0000 [r113647-113649]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c: Permit callee to continue in the dialplan, after
	  caller has hung up. (closes issue #11954) Reported by: johan
	  Patches: app_dial_rev104031.patch uploaded by johan (license 334)

	* contrib/scripts/astcli: Additional enhancements (closes issue
	  #12390) Reported by: tzafrir Patches: astcli_fixes.diff uploaded
	  by tzafrir (license 46)

2008-04-09 01:36 +0000 [r113597]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_iax2.c: 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 21:33 +0000 [r113559]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astcli (added): 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 [r113403-113505]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: 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 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/features.c, include/asterisk/features.h: Move
	  AST_FEATURE_FLAG_* and FEATURE_RETURN_* to features.h so that
	  they can be used by modules. (closes issue #12384) Reported by:
	  fnordian Patches: features.patch uploaded by fnordian (license
	  110) (patch modified by me, to give FEATURE_RETURN_* an AST_
	  prefix)

	* /, main/asterisk.c: 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 16:54 +0000 [r113349-113400]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/astgenkey.8: 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 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 15:05 +0000 [r113297]  Joshua Colp <jcolp@digium.com>

	* /, main/audiohook.c: 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 22:16 +0000 [r113245]  Tilghman Lesher <tlesher@digium.com>

	* configs/manager.conf.sample: Additional note

2008-04-07 21:49 +0000 [r113243]  Jason Parker <jparker@digium.com>

	* configs/manager.conf.sample: Document 'originate' permission in
	  manager sample config.

2008-04-07 21:35 +0000 [r113241]  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.
	  ........ ................

2008-04-07 20:22 +0000 [r113207]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: This is a "fix" for something that's been
	  bugging the crap out of me for a while. The variable name "flag"
	  to distinguish between whether a message is being forwarded or is
	  new is not a helpful name. The newly added doxygen documentation
	  to app_voicemail is tremendously helpful, but I still just...hate
	  this variable name. I think is_new_message is more indicative of
	  what its purpose is.

2008-04-07 19:06 +0000 [r113172]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: 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 18:57 +0000 [r113170]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: atoi(NULL) is bad

2008-04-07 18:02 +0000 [r113119]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c, configs/skinny.conf.sample: 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 16:12 +0000 [r113066]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: 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 15:18 +0000 [r113013]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: 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 14:54 +0000 [r113009]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c, include/asterisk/slinfactory.h: Put my
	  slinfactory changes back in.

2008-04-05 13:24 +0000 [r112972]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: AsyncAGI should not close the manager session on
	  error. (closes issue #12370) Reported by: srt Patches:
	  asterisk-12370.diff uploaded by srt (license 378)

2008-04-05 07:58 +0000 [r112906-112939]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Clean up some memory leak/ref counting
	  issues

	* phoneprov/000000000000-directory.xml, phoneprov/polycom.xml,
	  res/res_phoneprov.c, phoneprov/polycom_line.xml (added):
	  Multi-line support for phoneprov

2008-04-05 01:33 +0000 [r112874]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: Found a little problem with the sip request
	  handling that could lead to a quick crash of asterisk, and a road
	  to a DOS attack if left unfixed. Attaching to a running asterisk
	  with "telnet hostname 5060", I would input "something", then hit
	  return three times, and asterisk crashes. I traced it to
	  handle_request_do(), which zeroes out the data (an ast_str ptr)
	  if the string is too short. Instead of freeing the struct and
	  nulling the pointer, it now just resets it, because this ast_str
	  is expected by the calling routine to still be there after
	  handle_request_do() returns. This appears to fix the crash. I
	  assume that it was introduced with ast_str's being adopted. It's
	  a subtle and easy-to-miss sort of problem. I also found all the
	  places where the req.data is freed, and made sure the ptr is
	  Nulled out as well; no good leaving bad ptrs laying around-- I
	  didn't need to do this, but it seemed a good thing to do...

2008-04-04 19:28 +0000 [r112785-112821]  Philippe Sultan <philippe.sultan@gmail.com>

	* /, channels/chan_gtalk.c: 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 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 00:57 +0000 [r112714]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/asterisk.c: sleep long enough for the zaptel timer error
	  message to display before exit

2008-04-04 00:53 +0000 [r112712]  Joshua Colp <jcolp@digium.com>

	* /, main/Makefile: 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. ........

2008-04-04 00:32 +0000 [r112653-112708]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /: blocked for trunk....woot

	* main/asterisk.c: satisfy buildbot

	* main/asterisk.c: 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.

2008-04-03 14:35 +0000 [r112600]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, /: 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)
	  ........

2008-04-03 07:49 +0000 [r112520-112564]  Tilghman Lesher <tlesher@digium.com>

	* formats/format_wav.c, main/file.c, include/asterisk/mod_format.h:
	  Use a 32k file buffer on recordings, which increases the
	  efficiency of file recording. (closes issue #11962) Reported by:
	  garlew Patches: recording.patch uploaded by garlew (license 376)
	  bug-11962.diff uploaded by snuffy (license 35)

	* channels/chan_misdn.c: Make MISDN generate channel rename events
	  when the name changes. (closes issue #11142) Reported by:
	  julianjm Patches: chan_misdn_tmpchan_trunk_v1.diff uploaded by
	  julianjm (license 99)

2008-04-02 17:36 +0000 [r112469]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: 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 15:26 +0000 [r112431]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: 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 15:25 +0000 [r112426]  Terry Wilson <twilson@digium.com>

	* build_tools/cflags.xml, include/asterisk/http.h, main/manager.c,
	  res/res_phoneprov.c, main/http.c, res/res_http_post.c (added):
	  Re-add HTTP post support by moving to res_http_post.c

2008-04-02 14:32 +0000 [r112394]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 22:55 +0000 [r112360]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Added dnsmgr status output for sip show
	  registry.

2008-04-01 22:45 +0000 [r112357]  Steve Murphy <murf@digium.com>

	* main/pbx.c: 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 22:25 +0000 [r112351]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a typo that prevented configuration of
	  non-dynamic peers.

2008-04-01 22:07 +0000 [r112321]  Jeff Peeler <jpeeler@digium.com>

	* CHANGES, channels/chan_iax2.c: Existing DNS manager lookups
	  extended to check for SRV records.

2008-04-01 20:02 +0000 [r112289]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (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 18:27 +0000 [r112241-112252]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Minor formatting cleanup. (closes issue
	  #12343) Reported by: travishein Patches:
	  app_voicemail_code_convention.patch uploaded by travishein
	  (license 385)

	* apps/app_voicemail.c: More voicemail doxygen additions/cleanup.
	  (issue #12343) Reported by: travishein Patches:
	  app_voicemail_code_documentation.patch uploaded by travishein
	  (license 385)

2008-04-01 18:23 +0000 [r112234]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_vpb.cc: Fix last commit

2008-04-01 18:06 +0000 [r112210]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: 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 17:53 +0000 [r112207]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c,
	  main/slinfactory.c, CHANGES, channels/chan_iax2.c,
	  include/asterisk/dnsmgr.h, include/asterisk/slinfactory.h: This
	  adds DNS SRV record support to DNS manager. If there is a SRV
	  record for a given domain, the hostname and port listed in the
	  SRV record will be used. If no SRV record exists or a SRV lookup
	  is not attempted, the DNS lookup on the specified domain will be
	  performed as normal. Chan_sip has been modified to take advantage
	  of the new SRV support.

2008-04-01 17:48 +0000 [r112155-112205]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 ........

	* channels/chan_sip.c: Demote a log message down to a warning.
	  (closes issue #12345) Reported by: caio1982 Patches:
	  limit_msg.diff uploaded by caio1982 (license 22)

2008-04-01 17:23 +0000 [r112148]  Mark Michelson <mmichelson@digium.com>

	* /, main/dns.c: 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 16:50 +0000 [r112126]  Joshua Colp <jcolp@digium.com>

	* /, main/slinfactory.c, include/asterisk/slinfactory.h: 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 ........

2008-04-01 16:35 +0000 [r112124]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Now that zaptel trunk has been removed, add
	  the PSTN deprecation notice to chan_zap, as well.

2008-03-31 22:16 +0000 [r112069-112071]  Jason Parker <jparker@digium.com>

	* channels/chan_usbradio.c: I missed a place when this define was
	  changed. (closes issue #12334) Reported by: ovi Patches:
	  12334-asterisk.patch uploaded by dimas (license 88)

	* /, apps/app_voicemail.c: 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 21:01 +0000 [r112033-112035]  Terry Wilson <twilson@digium.com>

	* main/http.c: Yeah, simplify that logic a bit...

	* main/http.c: Handle blank prefix= in http.conf

2008-03-31 17:14 +0000 [r111996-111998]  Russell Bryant <russell@digium.com>

	* Makefile: Ensure configure gets run on a clean checkout. (closes
	  issue #12197) Reported by: juggie Patches: 12197.diff uploaded by
	  juggie (license 24)

	* channels/chan_sip.c: This fixes a high fence violation that
	  MALLOC_DEBUG reported to me.

2008-03-31 14:20 +0000 [r111961]  Joshua Colp <jcolp@digium.com>

	* res/res_config_sqlite.c: Initialize all these here tmp pointers
	  at declaration. They confused some compilers a wee bit. (closes
	  issue #12333) Reported by: ovi

2008-03-28 22:50 +0000 [r111908-111909]  Russell Bryant <russell@digium.com>

	* doc/janitor-projects.txt, include/asterisk/pbx.h: Make some notes
	  about common usage of pbx_builtin_getvar_helper() that is not
	  thread-safe.

	* main/dnsmgr.c: Note a minor race condition that I noticed while
	  reviewing Jeff's changes to this code.

2008-03-28 21:46 +0000 [r111857]  Jason Parker <jparker@digium.com>

	* codecs/gsm/inc/private.h, /: 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 20:03 +0000 [r111777-111811]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This time the fix is proper for issue 12284.
	  I have tested it thoroughly and found that valgrind no longer
	  complains and that calls do complete correctly. The fix is along
	  the same lines as before: Make sure the final null terminator
	  gets copied into the new sip_request's data pointer. Without it,
	  parse_request will read and potentially write past the end of the
	  string, causing potential crashes. (closes issue #12284...for
	  real this time!) reported by falves11

	* channels/chan_sip.c, include/asterisk/strings.h: Temporary revert
	  of 111662. It's causing lots of trouble and appears to not be the
	  proper solution to the problem reported anyway. (related to issue
	  #12884)

2008-03-28 19:08 +0000 [r111721-111774]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Replace magic number size from msgArray
	  array with a define. (same patch as before, I just split this
	  part out) (close issue #12326) Reported by: travishein Patches:
	  app_voicemail_code_documentation.patch uploaded by travishein
	  (license 385)

	* apps/app_voicemail.c: Add a bit of doxygen documentation for
	  app_voicemail. (issue #12326) Reported by: travishein Patches:
	  app_voicemail_code_documentation.patch uploaded by travishein
	  (license 385)

	* /, channels/chan_skinny.c: 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 16:36 +0000 [r111662]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, include/asterisk/strings.h: The copy_request
	  function did not take into account the necessary null terminator
	  for the string to be copied into. This resulted in parse_request
	  reading invalid memory beyond the end of the string, and in some
	  cases led to crashes. Thanks to falves11 for providing the
	  valgrind output which led to the closure of this issue. (closes
	  issue #12284) Reported by: falves11

2008-03-28 16:20 +0000 [r111659]  Jason Parker <jparker@digium.com>

	* /, formats/format_wav_gsm.c: 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 14:37 +0000 [r111606]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/valgrind.txt: 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 00:55 +0000 [r111565]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Forgetting to unregister a manager action is
	  bad, mmmk?

2008-03-28 00:12 +0000 [r111533]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 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 22:10 +0000 [r111500]  Terry Wilson <twilson@digium.com>

	* main/http.c: Fix another little http problem. In making it match
	  coding guidelines, a comparison was dropped

2008-03-27 21:25 +0000 [r111497]  Steve Murphy <murf@digium.com>

	* main/pbx.c: comment cleanup and iron out a really dumb mistake in
	  handling the '.'-wildcard in the new exten pattern matcher.

2008-03-27 19:26 +0000 [r111443]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: 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 13:29 +0000 [r111360-111410]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /, apps/app_playback.c: 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 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 00:27 +0000 [r111246-111295]  Jason Parker <jparker@digium.com>

	* main/frame.c: But we can change the API here.

	* main/frame.c, /: 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 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 21:23 +0000 [r111213]  Terry Wilson <twilson@digium.com>

	* main/http.c: Stupid strcasecmp function :-)

2008-03-26 20:34 +0000 [r111132-111185]  Tilghman Lesher <tlesher@digium.com>

	* channels/misdn_config.c: Oops, missed one

	* include/asterisk/linkedlists.h, main/config.c: Simplify new
	  macro, simplify configfile logic, now that list is sorted

2008-03-26 19:56 +0000 [r111130]  Joshua Colp <jcolp@digium.com>

	* /, contrib/scripts/autosupport: 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 19:52 +0000 [r111127]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: 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 19:39 +0000 [r111123]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: 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 19:29 +0000 [r111083]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Add expiry value to the sip show
	  subscriptions CLI command. (closes issue #12025) Reported by: agx

2008-03-26 19:26 +0000 [r111067]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: 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 19:19 +0000 [r111036]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/linkedlists.h, CHANGES, main/config.c: Add a
	  linkedlist macro that maintains a sorted list

2008-03-26 19:16 +0000 [r111028]  Jason Parker <jparker@digium.com>

	* main/dsp.c: Only try to detect silence when we actually need to,
	  instead of...always. If this is wrong, I'd love to hear why.

2008-03-26 19:08 +0000 [r111025]  Kevin P. Fleming <kpfleming@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh (added), codecs/ilbc:
	  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 19:05 +0000 [r111022]  Jason Parker <jparker@digium.com>

	* channels/chan_usbradio.c, channels/chan_vpb.cc,
	  channels/chan_zap.c, include/asterisk/dsp.h, main/dsp.c: Large
	  cleanup of DSP code Per comments from dimas: 1. The code now
	  generates DTMF_BEGIN frames in addition to DTMF_END ones. 2.
	  "quelching" rewritten - now each detector (MF/DTMF/generic tone)
	  may mark fragment of a frame for suppression (squelching, muting)
	  with a call to mute_fragment. Actual muting happens only once at
	  the very end of ast_dsp_process where all marked fragments are
	  zeroed. This way every detector sees original data in the frame
	  without any piece of a frame being zeroed by a detector which was
	  run before. 3. DTMF detector tries to "mute" one block before and
	  one block after the block where actual tone was detected. Muting
	  of previois block is something new for this patch. Obviously this
	  operation is not always possible - if current frame does not
	  contain data for previous block - it is too late. But at least we
	  make our best. Muting of next block was already done by the old
	  code but it only affects part of the next block which is in the
	  frame being processed. New code keeps this information in state
	  structures so it will mute proper number of samples in the next
	  frame(s) too. 4. Removed ast_dsp_digitdetect and
	  ast_dsp_getdigits APIs because these are not used. 5. DSP API
	  extended a bit - ast_dsp_was_muted() function added which returns
	  true if DSP code was muting any fragment in the last frame.
	  chan_zap uses this function to decide it needs to turn on
	  confmute on the channel. This is to replace AST_FRAME_DTMF
	  'm'/'u' (mute/unmute) functionality. (closes issue #11968)
	  Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas
	  (license 88) v4-11968-zap.patch uploaded by dimas (license 88)
	  Tested by: dimas, qwell

2008-03-26 19:05 +0000 [r111017-111021]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 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 18:41 +0000 [r111012-111013]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Oops, fix this, too

	* main/udptl.c, main/dnsmgr.c, include/asterisk/config.h,
	  channels/iax2-provision.c, main/enum.c, main/rtp.c,
	  main/config.c, main/loader.c, main/cdr.c, main/manager.c,
	  main/features.c, main/logger.c, main/http.c,
	  include/asterisk/udptl.h, main/asterisk.c, main/dsp.c: Add the
	  "config reload <conffile>" command, which allows you to tell
	  Asterisk to reload any file that references a given configuration
	  file.

2008-03-26 17:44 +0000 [r110963]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: 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 17:28 +0000 [r110911-110930]  Donny Kavanagh <donnyk@gmail.com>

	* Makefile: revert something dumb, because i was running svn diff
	  in a subfolder not the root of trunk, before doing my commit and
	  did not see it

	* Makefile, doc/snmp.txt: 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 17:10 +0000 [r110881]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
	  (removed), codecs/ilbc/syntFilter.h (removed),
	  codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
	  (removed), codecs/ilbc/StateConstructW.h (removed),
	  codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
	  (removed), codecs/ilbc/getCBvec.c (removed),
	  codecs/ilbc/LPCdecode.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/iLBC_define.h
	  (removed), codecs/ilbc/FrameClassify.c (removed),
	  codecs/ilbc/enhancer.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/iLBC_decode.c (removed), codecs/ilbc/constants.c
	  (removed), codecs/ilbc/createCB.h (removed), CHANGES,
	  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/StateSearchW.h
	  (removed), codecs/ilbc/iCBConstruct.c (removed): 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 00:02 +0000 [r110831]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: 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 22:51 +0000 [r110780]  Jason Parker <jparker@digium.com>

	* /, cdr/cdr_custom.c: 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 20:02 +0000 [r110726]  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 17:46 +0000 [r110689-110691]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample, configs/extensions.conf.sample:
	  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: 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 15:44 +0000 [r110636-110639]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Oops here too. I need to stop coding for a
	  while...

	* /, channels/chan_sip.c: 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 15:18 +0000 [r110629-110631]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, channels/chan_sip.c, configs/sip.conf.sample,
	  CHANGES: Add a special dialplan variable to chan_sip which will
	  cause an audio file to be played upon completion of an attended
	  transfer. (closes issue #9239) Reported by: sunder

	* Makefile, /, main/app.c, include/asterisk/options.h,
	  main/asterisk.c: 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 ........

2008-03-25 10:54 +0000 [r110625]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Use the "Server" header when responding to
	  SIP requests. (closes issue #12278) Reported by: rjain Patches:
	  chan_sip.c.diff uploaded by rjain (license 226)

2008-03-24 20:14 +0000 [r110619-110621]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove the "Event: registration" header from
	  Asterisk-generated SIP REGISTER requests. rjain points out that
	  RFC 3265 specifies that the Event: header is not a valid header
	  for REGISTER requests and that the "registration" value is not
	  defined at IANA. (closes issue #12279) Reported by: rjain
	  Patches: chan_sip.c.diff uploaded by rjain (license 226)

	* channels/chan_sip.c: 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 [r110615]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: 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 15:28 +0000 [r110610]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: 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 [r110578]  Jason Parker <jparker@digium.com>

	* sounds/Makefile: Update to 1.4.11 core sounds.

2008-03-21 17:58 +0000 [r110542]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/audiohook.h, main/audiohook.c: Merge over
	  ast_audiohook_volume branch. This adds API calls for use by
	  developers to adjust the volume on a channel.

2008-03-21 15:24 +0000 [r110499]  Russell Bryant <russell@digium.com>

	* configs/sip.conf.sample, CHANGES: 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 [r110475]  Jason Parker <jparker@digium.com>

	* /, codecs/gsm/Makefile: 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)
	  ........

2008-03-21 01:44 +0000 [r110444]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES: Add note of the added Directory options, from commit
	  110237 (closes issue #7151)

2008-03-20 23:14 +0000 [r110303-110396]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, /: 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: 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
	  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: 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 18:01 +0000 [r110272]  Mark Michelson <mmichelson@digium.com>

	* main/dial.c: Add missing unlock

2008-03-20 17:45 +0000 [r110268-110270]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, apps/app_minivm.c, include/asterisk/netsock.h,
	  main/netsock.c: Remove astobj.h from some places where it wasn't
	  needed

	* main/channel.c, res/res_musiconhold.c: 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.

2008-03-20 05:06 +0000 [r110211-110237]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_directory.c, sounds/Makefile: Upgrade the sounds
	  version; add several directory enhancements: 1) Number of digits
	  to enter can now be configured 2) The digits can now match on
	  both first AND last name, instead of only one or the other
	  (Closes issue #7151)

	* channels/chan_vpb.cc: Fix recent trunk breakage

2008-03-19 22:58 +0000 [r110164]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: 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 22:25 +0000 [r110132-110161]  Jason Parker <jparker@digium.com>

	* channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c,
	  channels/chan_h323.c, include/asterisk/dsp.h,
	  channels/chan_mgcp.c, main/dsp.c: Rename DSP_FEATURE_DTMF_DETECT,
	  because we are *NOT* only detecting DTMF digits. This was very
	  misleading. Early cleanup for issue #11968

	* channels/chan_usbradio.c, channels/chan_vpb.cc,
	  channels/chan_zap.c, channels/chan_sip.c, include/asterisk/dsp.h,
	  channels/chan_mgcp.c, main/dsp.c: Rename very poorly named
	  function to reflect what it actually does. This was causing quite
	  a bit of confusion for me...

2008-03-19 21:05 +0000 [r110087]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c, CHANGES: This change adds DNS manager
	  support for registrations not referencing a peer entry. It looks
	  like there is support for DNS manager for realtime peers as well,
	  however it is not implemented correctly. The improper usage
	  occurs when ast_dnsmgr_lookup is called with one of the arguments
	  being an address from the stack to be continually updated. The
	  variable from the stack will go out of scope and dnsmgr will
	  continue to try and update the memory there, causing possible
	  stack corruption. This problem will be worked on next as well as
	  adding DNS manager support for peer entries.

2008-03-19 20:34 +0000 [r110084]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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 19:13 +0000 [r110036]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: 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 18:57 +0000 [r110023]  Russell Bryant <russell@digium.com>

	* /: remove svnmerge-blocked property that is not supposed to be
	  here

2008-03-19 18:25 +0000 [r110020]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: 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 17:15 +0000 [r109974]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
	  (added), /: 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 16:54 +0000 [r109970]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, CHANGES: Add the ability to use a pattern match for a
	  hint. (closes issue #7767) Reported by: Corydon76 Patches:
	  20070314__simple_hint_lookup.diff.txt uploaded by Corydon76
	  pbx-trunk-98436.diff uploaded by plack (license 365)

2008-03-19 16:24 +0000 [r109942]  Steve Murphy <murf@digium.com>

	* /, main/config.c: 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 16:21 +0000 [r109912-109926]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_phoneprov.c: ensure that res_phoneprov's HTTP handler
	  tells the dispatcher what method it can handle

	* main/manager.c, main/http.c: actually implement HTTP request
	  dispatching based on both URI and method; reduce duplication of
	  data when generating responses using ast_http_error()

2008-03-19 15:45 +0000 [r109910]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Fix some more breakage that I introduced when
	  changing extension state callbacks to the list macros.

2008-03-19 15:41 +0000 [r109909]  Kevin P. Fleming <kpfleming@digium.com>

	* main/http.c: clean up code to conform to coding guidelines

2008-03-19 15:22 +0000 [r109833-109907]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Remove an unneeded variable. This compiled, but I
	  missed the uninitialized warning because I always compile without
	  optimizations turned on. Sorry!

	* main/pbx.c: Convert handling of extension state callbacks to the
	  list macros.

	* main/pbx.c: Minor coding style changes, including adding handling
	  for memory allocation failure

	* main/http.c: Minor change to use Asterisk macros

	* /, main/utils.c: 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. ........

	* channels/chan_sip.c: Set req->data to NULL after free'ing to
	  ensure that it never gets accidentally double free'd. (reported
	  by dhubbard directly to me)

2008-03-18 23:32 +0000 [r109802]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: Fix a typo which caused a double free in
	  chan_zap. This was discovered by Juggie while attempting to load
	  chan_zap. Apparently this would happen if an error were
	  encountered while trying to process zapata.conf.

2008-03-18 23:22 +0000 [r109775]  Tilghman Lesher <tlesher@digium.com>

	* configs/res_ldap.conf.sample, res/res_config_ldap.c: 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 22:36 +0000 [r109764]  Russell Bryant <russell@digium.com>

	* /, apps/app_chanspy.c: 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 22:32 +0000 [r109762]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/http.h, main/manager.c, res/res_phoneprov.c,
	  main/http.c, include/asterisk/_private.h: start the process of
	  changing HTTP request dispatching to do it based on *both* URI
	  and method, so that POST support can move into a module; move
	  http.c's private function prototypes into _private.h

2008-03-18 20:59 +0000 [r109714]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: 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 20:13 +0000 [r109683]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_ldap.c: Set protocol version, port number
	  correctly. (closes issue #12211, closes issue #12209) Reported
	  by: sylvain

2008-03-18 20:02 +0000 [r109681]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Since a sip request's data field is now a
	  stringfield, we not only have to check if the string is
	  zero-length, but also if the data field is non-null. (closes
	  issue #12250) Reported by: caio1982

2008-03-18 19:53 +0000 [r109680]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/dbsep.cgi: Comment debug

2008-03-18 19:24 +0000 [r109651]  Jason Parker <jparker@digium.com>

	* /, codecs/log2comp.h: 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 18:58 +0000 [r109545-109621]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option
	  'randomperiodicannounce' to queues.conf. Setting this will allow
	  the list of periodic announcments specified to be played in a
	  random order instead of being played sequentially. (closes issue
	  #6681) Reported by: alt_phil Tested by: putnopvut

	* /, channels/chan_agent.c: 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: Add format attribute to printf-style
	  functions in astmm.h

2008-03-18 16:23 +0000 [r109451-109475]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: fix up various warnings found via the
	  addition of format string checking... some of these were really,
	  really bad code

	* configure, include/asterisk/autoconfig.h.in, acinclude.m4: ensure
	  that dependencies on AST_C_DEFINE_CHECK symbols work properly

2008-03-18 15:43 +0000 [r109447]  Terry Wilson <twilson@digium.com>

	* include/asterisk/utils.h, cdr/cdr_sqlite3_custom.c,
	  apps/app_meetme.c, channels/chan_sip.c, apps/app_festival.c,
	  main/translate.c, res/res_phoneprov.c, main/jitterbuf.c,
	  utils/astman.c, main/utils.c, include/jitterbuf.h,
	  apps/app_queue.c, channels/chan_iax2.c, utils/frame.c,
	  main/cli.c, Makefile, funcs/func_enum.c, main/manager.c,
	  channels/chan_misdn.c, include/asterisk/astobj.h, res/res_agi.c,
	  main/features.c, apps/app_minivm.c, res/res_realtime.c,
	  utils/extconf.c, res/res_indications.c,
	  include/asterisk/strings.h, res/res_config_ldap.c,
	  main/asterisk.c, utils/check_expr.c, apps/app_voicemail.c: 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.

2008-03-18 15:13 +0000 [r109396]  Joshua Colp <jcolp@digium.com>

	* main/manager.c, main/logger.c: Make sure values are interpreted
	  as character strings and not format strings. (AST-2008-004)

2008-03-18 15:10 +0000 [r109394]  Jason Parker <jparker@digium.com>

	* /: Block this here. Already committed.

2008-03-18 15:08 +0000 [r109390]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c, main/rtp.c: 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 15:07 +0000 [r109389]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Do not return with a successful
	  authentication if the From header ends up empty. (AST-2008-003)

2008-03-18 14:09 +0000 [r109357]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2,
	  pbx/ael/ael-test/ael-ntest23/t3, /, include/asterisk/extconf.h,
	  pbx/ael/ael-test/ael-ntest23/extensions.ael,
	  pbx/ael/ael-test/ael-ntest23 (added), utils/conf2ael.c,
	  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, utils/ael_main.c,
	  pbx/ael/ael-test/ael-ntest23/t3/i.ael, utils/extconf.c,
	  pbx/ael/ael-test/ael-ntest23/t3/j.ael, res/ael/ael.flex,
	  pbx/ael/ael-test/ael-ntest23/qq.ael: 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 07:23 +0000 [r109316]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  manager peerstatus events when peer can't authenticate. (closes
	  issue #11959) Reported by: mostyn Patches: peerstatus3.patch
	  uploaded by mostyn (license 398)

2008-03-18 00:28 +0000 [r109282]  Sean Bright <sean.bright@gmail.com>

	* configure, configure.ac: Fix a typo

2008-03-17 22:10 +0000 [r109229]  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: 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 22:06 +0000 [r109227]  Mark Michelson <mmichelson@digium.com>

	* /, main/utils.c: 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 17:58 +0000 [r109172]  Michiel van Baak <michiel@vanbaak.info>

	* /: block rev 109171 that is already here

2008-03-17 17:47 +0000 [r109169]  Steve Murphy <murf@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: (closes issue #12238)
	  Reported by: mvanbaak Tested by: murf, mvanbaak Due to a bug that
	  occurred when merge_contexts_and_delete scanned the "old" or
	  existing contexts, and found a context that doesn't exist in the
	  new set, yet owned by a different registrar. The context is
	  created in the new set, with the old registrar, and and all the
	  priorities and extens that have a different registrar are copied
	  into it. But, not the includes, ignorepats, and switches. I added
	  code to do this immediately after the context is created. This
	  still leaves a logical hole in the code. If you define a context
	  in two places, (eg. in extensions.conf and also in
	  extensions.ael), and they both have includes, but different in
	  composition, no new context will be generated, and therefore the
	  'old' includes, switches, and ignorepats will not be copied. I'd
	  have added code to simply add any non-duplicates into the 'new'
	  context that had a different registrar, but there is one big
	  complication: includes, and switches are definitely order
	  dependent. (ignorepats I'm not sure about). And we'll have to
	  develop some sort of policy about how we merge order dependent
	  lists, especially if the intersection of the two sets is empty.
	  (in other words, they do not have any elements in common). Do the
	  new go first, or the old? I've elected to punt this issue until a
	  user complains. Hopefully, this is pretty rare thing.

2008-03-17 17:43 +0000 [r109168]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: 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 17:31 +0000 [r109166]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, configure, configure.ac, acinclude.m4: 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 [r109113]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove something that is never ever used.

2008-03-17 16:37 +0000 [r109111]  Jason Parker <jparker@digium.com>

	* configs/sip_notify.conf.sample: 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)

2008-03-17 16:26 +0000 [r109054-109108]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: 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 ........

	* channels/chan_sip.c: Make sure that the temporary sip_request
	  structure is empty so that copy_request doesn't think it already
	  has an ast_str. (closes issue #12231) Reported by: IgorG

2008-03-17 14:21 +0000 [r109024]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: 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 21:50 +0000 [r108962]  Michiel van Baak <michiel@vanbaak.info>

	* main/dial.c, /: 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) ........

2008-03-16 17:55 +0000 [r108927-108929]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Remove an unnecessary thread attribute
	  instance

	* apps/app_voicemail.c: 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 [r108740-108894]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Remove a double write lock of the contexts lock in
	  ast_wrlock_contexts(). How did this ever work? (closes issue
	  #12219) Reported by: ys Patches: pbx.c.diff uploaded by ys
	  (license 281)

	* include/asterisk/dnsmgr.h: Doxygenify dnsmgr.h

	* Makefile: 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 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 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: 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 16:52 +0000 [r108738]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: 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 14:32 +0000 [r108683]  Jason Parker <jparker@digium.com>

	* /, res/res_musiconhold.c: 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 23:12 +0000 [r108639]  Jeff Peeler <jpeeler@digium.com>

	* doc/externalivr.txt, apps/app_externalivr.c, CHANGES: documenting
	  changes as a result of adding TCP functionality to ExternalIVR

2008-03-13 21:47 +0000 [r108586]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Make this compile

2008-03-13 21:40 +0000 [r108531-108584]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/channel.h,
	  apps/app_chanspy.c: 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) ........

	* /, channels/chan_sip.c: 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 ........

2008-03-13 20:59 +0000 [r108529]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: 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 20:38 +0000 [r108523]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_externalivr.c: set variable to NULL to prevent
	  uninitialized warning

2008-03-13 20:35 +0000 [r108439-108508]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Fix a place where configuration values
	  could cause an overflow of a buffer.

	* /, apps/app_followme.c: 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. ........

	* channels/chan_sip.c: Merge changes from
	  team/jamesgolovich/chan_sip-ast_str This set of changes removes
	  the hard coded maximum packet size of 4kB from chan_sip. It now
	  starts by allocating 1kB, and growing the buffer as needed to
	  accommodate large packets. (closes issue #8556, reported by
	  mikma, patch by jamesgolovich)

2008-03-13 18:59 +0000 [r108404]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_externalivr.c: (closes issue #11827) Reported by:
	  ctooley Patches: eivr_tcp_generic.patch uploaded by jpeeler
	  (license 325) This change adds the ability to communicate over a
	  TCP socket instead of forking a child process.

2008-03-12 22:49 +0000 [r108295-108346]  Russell Bryant <russell@digium.com>

	* main/http.c: Make the default prefix empty, like it was in
	  Asterisk 1.4. (closes issue #12198, reported by bkruse, patched
	  by me)

	* 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: Rename ast_tcptls_server_instance to
	  session_instance, since this pertains to server and client usage.

2008-03-12 22:09 +0000 [r108289-108293]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Let's get this to compile

	* /, channels/chan_sip.c: 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. ........

2008-03-12 21:37 +0000 [r108286]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: add
	  support for named sections in zapata.conf, and fix a few bugs in
	  config file parsing (closes issue #9503) Reported by: tzafrir
	  Patches: fix_cleanups uploaded by tzafrir (license 46)
	  zapata_sections uploaded by tzafrir (license 46)
	  skipchannel_options uploaded by tzafrir (license 46) conf_sample
	  uploaded by tzafrir (license 46) patches updated by me to better
	  conform to coding guidelines and fix some problems

2008-03-12 21:19 +0000 [r108238]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/sched.h: 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 21:06 +0000 [r108226]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c, include/asterisk/slinfactory.h: Doxygenify
	  slinfactory a bit.

2008-03-12 20:27 +0000 [r108191]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: 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 [r108137]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, apps/app_chanspy.c: 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 18:29 +0000 [r108084]  Joshua Colp <jcolp@digium.com>

	* /, include/asterisk/audiohook.h, main/audiohook.c,
	  apps/app_mixmonitor.c: 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 17:06 +0000 [r108032-108034]  Russell Bryant <russell@digium.com>

	* funcs/func_config.c: - Add Tilghman to the copyright info ... he
	  wrote the hard part :) - Remove some magic in unload_module that
	  isn't needed. Module use counts already ensure that the function
	  isn't going to be in use at this point.

	* main/channel.c, /: 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 07:43 +0000 [r107878-107998]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Deadlock fixes (closes issue #12143)
	  Reported by: kactus Patches: 20080312__bug12143__2.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: kactus

	* apps/app_dumpchan.c, apps/app_zapras.c, main/loader.c: Revert
	  several changes from revision 102525, as the changes were not
	  compatible, and, in fact, introduced regressions. (Closes issue
	  #12190)

	* funcs/func_config.c: Cache config files, when possible, for speed

	* contrib/scripts/iax-friends.sql, /,
	  contrib/scripts/sip-friends.sql: 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 [r107827]  Jason Parker <jparker@digium.com>

	* /, doc/voicemail_odbc_postgresql.txt: 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 22:55 +0000 [r107791]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_curl.c, res/res_config_pgsql.c,
	  res/res_config_odbc.c, include/asterisk/config.h,
	  res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c: 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 22:54 +0000 [r107787-107790]  Russell Bryant <russell@digium.com>

	* funcs/func_config.c: remove documentation of an argument that i
	  did not implement

	* funcs/func_config.c (added), CHANGES: Add a trivial new dialplan
	  function, AST_CONFIG(), which allows you to access a variable
	  from an Asterisk configuration file in the dialplan, or anywhere
	  else where dialplan functions can be used. (Inspired by a
	  discussion with Tilghman and Pari)

2008-03-11 21:10 +0000 [r107721-107722]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_odbc.c: Convert prepare_and_execute to direct_execute for
	  speed (closes issue #11935) Reported by: falves11 Patches:
	  20080208__bug11935.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: falves11, Corydon76

	* contrib/scripts/dbsep.cgi (added), configs/dbsep.conf.sample
	  (added): Add contributed script for separation of database access
	  from Asterisk

2008-03-11 20:54 +0000 [r107719]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: This patch adds support for extended help
	  prompts in voicemail. These prompts are in the 1.4.9 sounds
	  release. (closes issue #11705) Reported by: jaroth Patches:
	  helpprompts.patch uploaded by jaroth (license 50)

2008-03-11 20:53 +0000 [r107718]  Jason Parker <jparker@digium.com>

	* channels/chan_jingle.c, /, channels/chan_gtalk.c: 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 20:50 +0000 [r107715]  Kevin P. Fleming <kpfleming@digium.com>

	* /, Makefile.rules, channels/Makefile: 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 20:36 +0000 [r107638-107710]  Joshua Colp <jcolp@digium.com>

	* apps/app_page.c: 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 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 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 16:28 +0000 [r107551]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Whitespace changes only

2008-03-11 15:59 +0000 [r107530]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Remove some redundant logic from
	  wait_for_answer. This also let's us get rid of one of those XXX
	  comments from the code. The redundancy occurs because the
	  'single' flag implies that the 'r' and 'm' flags are not set, so
	  there's no need to explicitly check them again.

2008-03-11 15:39 +0000 [r107466-107525]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_vpb.cc: fix another potential bug found by gcc 4.3

	* /: block fix that is already here

	* codecs/Makefile, /, apps/app_sms.c, apps/app_rpt.c,
	  channels/misdn/isdn_lib.c: 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
	  ........

2008-03-11 15:05 +0000 [r107465]  Joshua Colp <jcolp@digium.com>

	* main/features.c: Clarify comment about masquerading and playback
	  of the parking slot. (closes issue #12180) Reported by: davidw

2008-03-11 14:37 +0000 [r107373-107462]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  apps/app_sms.c: 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 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 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 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 01:09 +0000 [r107292]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: 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 21:48 +0000 [r107231]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: (closes
	  issue #6019) Reported by: ssokol Patches:
	  20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: putnopvut

2008-03-10 20:28 +0000 [r107177]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: 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 20:17 +0000 [r107159-107162]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: 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 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) ........

2008-03-10 20:00 +0000 [r107157]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: 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

2008-03-10 17:13 +0000 [r107100-107103]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: 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 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 15:45 +0000 [r107068]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: 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 14:55 +0000 [r107019]  Steve Murphy <murf@digium.com>

	* main/cdr.c: 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 14:36 +0000 [r107017]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: 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 16:03 +0000 [r106946]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: 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 22:52 +0000 [r106896]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: 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 22:36 +0000 [r106892]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we don't start a call when we have
	  already done so in response to a COT message

2008-03-07 22:15 +0000 [r106843]  Jason Parker <jparker@digium.com>

	* /, main/editline/Makefile.in: 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 ........

2008-03-07 19:33 +0000 [r106789]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: 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 18:57 +0000 [r106757]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pval.h,
	  channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y,
	  apps/app_queue.c, channels/chan_iax2.c, 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, utils/ael_main.c, res/ael/ael.tab.h,
	  utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c,
	  res/ael/ael.flex: (closes issue #6002) Reported by: rizzo Tested
	  by: murf Proposal of the changes to be made, and then an
	  announcement of how they were accomplished:
	  http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
	  and:
	  http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
	  Here is a recap, file by file, of what I have done:
	  pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr
	  to the context list, now will ALSO be passed a hashtab ptr to the
	  same set. Why? because (for the time being), the dialplan is
	  stored in both, to facilitate a quick, low-cost move to
	  hash-tables to speed up dialplan processing. If it was deemed
	  necessary to pass the context LIST, well, it is just as necessary
	  to have the TABLE available. This is because the list/table in
	  question might not be the global one, but temporary ones we would
	  use to stage the dialplan on, and then swap into the global
	  position when things are ready. We now have one external function
	  for apps to use, "ast_context_find_or_create()" instead of the
	  pre-existing "find" and "create", as all existing usages used
	  both in tandem anyway. pbx_config, and pbx_ael, will stage the
	  reloaded dialplan into local lists and tables, and then call
	  merge_contexts_and_delete, which will merge (now) existing
	  contexts and priorities from other registrars into this local set
	  by copying them. Then, merge_contexts_and_delete will lock down
	  the contexts, swap the lists and tables, and unlock (real quick),
	  and then destroy the old dialplan. chan_sip.c chan_iax.c
	  chan_skinny.c All the channel drivers that would add regcontexts
	  now use the ast_context_find_or_create now. chan_sip also
	  includes a small fix to get rid of warnings about removing
	  priorities that never got entered. apps/app_meetme.c
	  apps/app_dial.c apps/app_queue.c All the apps that added a
	  context/exten/priority were also modified to use
	  ast_context_find_or_create instead. include/asterisk/pbx.h
	  ast_context_create() is removed. Find_or_create_ is the new
	  method. ast_context_find_or_create() interface gets the hashtab
	  added. ast_merge_contexts_and_delete() gets the local hashtab arg
	  added. ast_wrlock_contexts_version() is added so you can detect
	  if someone else got a writelock between your readlocking and
	  writelocking. ast_hashtab_compare_contexts was made public for
	  use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like
	  fashion make public. include/asterisk/pval.h ast_compile_ael2()
	  interface changed to include the local hashtab table ptr.
	  main/features.c For the sake of the parking context, we use
	  ast_context_find_or_create(). main/pbx.c I changed all the "tree"
	  names to "table" instead. That's because the original
	  implementation was based on binary trees. (had a free library).
	  Then I moved to hashtabs. Now, the names move forward too.
	  refcount field added to contexts, so you can keep track of how
	  many modules wanted this context to exist. Some log messages that
	  are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added
	  some calls to ast_verb(3,...) for debug messages Lots of little
	  mods to ast_context_remove_extension2, which is now excersized in
	  ways it was not previously; one definite bug fixed.
	  find_or_create was upgraded to handle both local lists/tables as
	  well as the globals. context_merge() was added to do the
	  per-context merging of the old/present contexts/extens/prios into
	  the new/proposed local list/tables
	  ast_merge_contexts_and_delete() was heavily modified.
	  ast_add_extension2() was also upgraded to handle changes. the
	  context_destroy() code was re-engineered to handle the new way of
	  doing things, by exten/prio instead of by context. res/ael/pval.c
	  res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y
	  res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c
	  utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the
	  interface to ast_compile_ael2(), to include the hashtab ptr. This
	  ended up involving several external apps. The main gotcha was I
	  had to include lock.h and hashtab.h in several places. As a side
	  note, I tested this stuff pretty thoroughly, I replicated the
	  problems originally reported by Luigi, and made triply sure that
	  reloads worked, and everything worked thru "stop gracefully". I
	  found a and fixed a few bugs as I was merging into trunk, that
	  did not appear in my tests of bug6002. How's this for verbose
	  commit messages?

2008-03-07 17:17 +0000 [r106684-106707]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/sched.h: 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: fix example usage

2008-03-07 16:26 +0000 [r106553-106654]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: 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 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)
	  ........

	* 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, apps/app_chanspy.c,
	  apps/app_rpt.c, main/asterisk.c, apps/app_speech_utils.c,
	  apps/app_voicemail.c: 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 [r106501-106518]  Russell Bryant <russell@digium.com>

	* doc/smdi.txt: minor text changes

	* doc/smdi.txt: Add updated SMDI documentation that I had only
	  sitting in my email ... oops

	* codecs/codec_g722.c, formats/format_pcm.c, main/file.c,
	  main/rtp.c: 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 22:11 +0000 [r106439]  Jason Parker <jparker@digium.com>

	* main/file.c: 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 22:11 +0000 [r106438]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: 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 19:31 +0000 [r106399]  Donny Kavanagh <donnyk@gmail.com>

	* res/res_agi.c: 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 05:21 +0000 [r106329-106346]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_ldap.c: Missing braces, fix parsing (closes issue
	  #12112) Reported by: