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
|
2009-03-19 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.6.2.0-beta1
2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
2009) | 20 lines Merged revisions 183115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
would erroneously report the device as "in use." A user was
having an issue where if an outgoing SIP call was canceled, the
SIP device would remain in use if we had not received any
response to the initial INVITE we sent out. The SIP device would
remain in use until the autocongestion timer was exhausted. I
tracked down the cause of this to be the section of code I am
removing here. I asked several people what the purpose of this
code was meant to be, but no one could give me any sort of answer
as to why this was here. The person who was having this issue has
been using this patch for several months and it has stopped the
problems they have had. AST-196 ........ ................
2009-03-19 15:45 +0000 [r183068-183111] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
Improve our triggering of a T38 switchover internally when
triggered by a received reinvite. Previously we reached across
the channel bridge to get the other party's SIP dialog structure
in order to trigger an outgoing reinvite. This is extremely
dangerous to do and only works if bridged to another SIP channel.
This patch changes this to use the T38 control frame method of
requesting a switchover. This change also causes the SIP channel
driver to propogate back whether the switchover worked or not
instead of blindly accepting the incoming T38 reinvite. Review:
http://reviewboard.digium.com/r/200/ ........
* main/channel.c, /: Merged revisions 183057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
an issue where a T38 control frame would get dropped. If two
channels were bridged together using a generic bridge the T38
control frame would get passed up instead of being indicated on
the other channel. ........
2009-03-18 21:19 +0000 [r183031] Jeff Peeler <jpeeler@digium.com>
* /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
Mar 2009) | 4 lines Add some code removed by mistake from commit
182722 that works around a file descriptor leak in versions of
PWLib prior to 1.12.0. ........
2009-03-18 14:39 +0000 [r182947] Russell Bryant <russell@digium.com>
* main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
configure, apps/app_mp3.c, res/res_agi.c,
include/asterisk/poll-compat.h, channels/chan_alsa.c,
main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
| 52 lines Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
| 44 lines Fix cases where the internal poll() was not being used
when it needed to be. We have seen a number of problems caused by
poll() not working properly on Mac OSX. If you search around,
you'll find a number of references to using select() instead of
poll() to work around these issues. In Asterisk, we've had poll.c
which implements poll() using select() internally. However, we
were still getting reports of problems. vadim investigated a bit
and realized that at least on his system, even though we were
compiling in poll.o, the system poll() was still being used. So,
the primary purpose of this patch is to ensure that we're using
the internal poll() when we want it to be used. The changes are:
1) Remove logic for when internal poll should be used from the
Makefile. Instead, put it in the configure script. The logic in
the configure script is the same as it was in the Makefile.
Ideally, we would have a functionality test for the problem, but
that's not actually possible, since we would have to be able to
run an application on the _target_ system to test poll()
behavior. 2) Always include poll.o in the build, but it will be
empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
throughout the source tree to ast_poll(). I feel that it is good
practice to give the API call a new name when we are changing its
behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems
where AST_POLL_COMPAT is defined, ast_poll() is redefined to
ast_internal_poll(). 4) Change poll() in main/poll.c to be
ast_internal_poll(). It's worth noting that any code that still
uses poll() directly will work fine (if they worked fine before).
So, for example, out of tree modules that are using poll() will
not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used. (closes issue
#13404) Reported by: agalbraith Tested by: russell, vadim
http://reviewboard.digium.com/r/198/ ........ ................
2009-03-17 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com>
* channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
channels/h323/ast_h323.cxx, configure,
autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
Allow H.323 Plus library to be used in addition to the OpenH323
library Chan_h323 can now be compiled against both the previously
supported versions of OpenH323 as well as the current H.323 Plus
(version 1.20.2). The configure script has been modified to look
in the default install location of h323 to hopefully help avoid
using the environment variables OPENH323DIR and PWLIBDIR. Also,
the CLI command "h323 show version" has been added which
indicates which version of h323 is in use. (closes issue #11261)
Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
uploaded by jthurman (license 614) ........
2009-03-17 16:46 +0000 [r182592] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 182553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
Tweak the handling of the frame list inside of ast_answer(). This
does not change any behavior, but moves the frames from the local
frame list back to the channel read queue using an O(n) algorithm
instead of O(n^2). ........
2009-03-17 15:01 +0000 [r182528-182534] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, /: Merged revisions 182530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
lines correct logic flaw in ast_answer() changes in r182525
........
* main/channel.c, /, main/features.c, include/asterisk/channel.h:
Merged revisions 182525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
lines Improve behavior of ast_answer() to not lose incoming
frames ast_answer(), when supplied a delay before returning to
the caller, use ast_safe_sleep() to implement the delay.
Unfortunately during this time any incoming frames are discarded,
which is problematic for T.38 re-INVITES and other sorts of
channel operations. When a delay is not passed to ast_answer(),
it still delays for up to 500 milliseconds, waiting for media to
arrive. Again, though, it discards any control frames, or
non-voice media frames. This patch rectifies this situation, by
storing all incoming frames during the delay period on a list,
and then requeuing them onto the channel before returning to the
caller. http://reviewboard.digium.com/r/196/ ........
2009-03-17 05:54 +0000 [r182453] Tilghman Lesher <tlesher@digium.com>
* main/db.c, /: Merged revisions 182450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................
r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
| 14 lines Merged revisions 182449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
| 7 lines Fix race in astdb The underlying db1 implementation
does not fully isolate the pages retrieved from astdb, so the
lock protecting accesses needs to be extended until the copy from
the shared memory structure is done. (closes issue #14682)
Reported by: makoto ........ ................
2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
from https://origsvn.digium.com/svn/asterisk/trunk ........
r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
| 8 lines OPENR2 uses an incorrect string value if the extension
delimiter is not present. * Fixed OPENR2 using an incorrect
string value if the extension delimiter is not present in the
Dial() function. This was fixed for SS7 and PRI in trunk
-r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
PRI, and others. * Removed trailing whitespace that appeared with
OPENR2. ........
2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com>
* /: svnmerge init
* / (added): Create a branch for 1.6.2
2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com>
* CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
configure, include/asterisk/autoconfig.h.in, configure.ac,
CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
commit introduces official support for R2 signaling in
chan_dahdi. The modifications to chan_dahdi, and the supporting
library, LibOpenR2, were both written by Moises Silva. Many users
are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
1.6 in Brazil, México and Argentina. An unknown number of users
(but at least 1) are using it in each of the following countries:
Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from
http://www.libopenr2.org/. Information about configuration can be
found in configs/chan_dahdi.conf.sample. The code committed is
the most up to date version, which was being maintained in
svn/asterisk/team/moy/mfcr2/. I would also like to include a
Thank You to the many others that tested this code beyond those
listed in this commit message. These are the names that I could
find in the mantis issue. (closes issue #12509) Reported by: moy
Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
by: moy, korihor, viniciusfontes, Skarmeth, loloski,
asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
ecarruda, rtorresduque, PTorres, ychen Review:
http://reviewboard.digium.com/r/40/
2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16
Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32
byte random padding at the beginning of an encrypted IAX2 frame
turns out to not be all that random at all. This patch calls
ast_random to fill the padding buffer with random data. The
padding is randomized at the beginning of every encrypted call
and for every encrypted retransmit frame. Review:
http://reviewboard.digium.com/r/193/ ........
2009-03-16 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Fix an off-by-one error in the FILE() function,
and extend FILE()'s length parameter to work like variable
substitution. Previously, FILE() returned one less character than
specified, due to the terminating NULL. Both the offset and
length parameters now behave identically to the way variable
substitution offsets and lengths also work. (closes issue #14670)
Reported by: BMC
* channels/chan_local.c, /: Merged revisions 182208 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16
Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak
of a local pvt structure. (closes issue #14656) Reported by:
caspy Patches: 20090313__bug14656__2.diff.txt uploaded by
tilghman (license 14) Tested by: caspy ........
2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com>
* main/channel.c: Fix a memory leak in the ast_answer /
__ast_answer API call. For a channel that is not yet answered
this API call will wait until a voice frame is received on the
channel before returning. It does this by waiting for frames on
the channel and reading them in. The frames read in were not
freed when they should have been.
2009-03-13 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Change faulty comparison used when announcing
average hold minutes and seconds (closes issue #14227) Reported
by: caspy
* main/features.c: Remove ast_ prefix from functions which are not
public.
* /, main/features.c: Merged revisions 181990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
peer when interpreting DTMF. Dynamic features defined in the
applicationmap section of features.conf allow one to specify
whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could
be interpreted to mean that one only needs to set the
DYNAMIC_FEATURES channel variable on the calling channel in order
to allow for the callee to be able to use the features which he
should have permission to use. However, the DYNAMIC_FEATURES
variable would only be read from the channel of the participant
that pressed the DTMF sequence to activate the feature. The
result of this was that the callee was unable to use dynamic
features unless the dialplan writer had taken measures to be sure
that the DYNAMIC_FEATURES variable was set on the callee's
channel. This commit changes the behavior of
ast_feature_interpret to concatenate the values of
DYNAMIC_FEATURES from both parties involved in the bridge. The
features themselves determine who has permission to use them, so
there is no reason to believe that one side of the bridge could
gain the ability to perform an action that they should not have
the ability to perform. Kevin Fleming pointed out on the
asterisk-users list that the typical way that this was worked
around in the past was by setting _DYNAMIC_FEATURES on the
calling channel so that the value would be inherited by the
called channel. While this works, the documentation alone is not
enough to figure out why this is necessary for the callee to be
able to use dynamic features. In this particular case, changing
the code to match the documentation is safe, easy, and will
generally make things easier for people for future installations.
This bug was originally reported on the asterisk-users list by
David Ruggles. (closes issue #14657) Reported by: mmichelson
Patches: 14657.patch uploaded by mmichelson (license 60) ........
2009-03-13 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix an issue with requesting a T38 reinvite
before the call is answered. The code responsible for sending the
T38 reinvite did not check if an INVITE was already being
handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current
INVITE is done being handled. (issue AST-191)
2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: improve a bit of suboptimal code
2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com>
* /: Merged revisions 181898 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
recording the v1.4 change in trunk since it originally came from
here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
(Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
property when generating the version string. Copied the
make_version file from Asterisk trunk. ........
2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Run the macro on the queue member's channel
when he answers, not the caller's channel.
* /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
2009) | 22 lines Properly send a 487 on an INVITE we have not
responded to if we receive a BYE. If we receive an INVITE from an
endpoint and then later receive a BYE from that same endpoint
before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487. There was logic in the code
prior to this commit which seemed to exist solely to handle this
situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the
sip_pvt had no owner channel. This made no sense since we created
the owner channel when we received the INVITE, meaning that the
majority of the time we would never send the 487. The 487 being
sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial
INVITE transaction. With this commit, that logic is now correctly
in place. (closes issue #14149) Reported by: legranjl Patches:
14149.patch uploaded by mmichelson (license 60) Tested by:
legranjl ........
2009-03-12 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com>
* main/translate.c: Adjust translation table column widths based
upon the translation times. Previously, only 5 columns were
displayed, and if a translation time exceeded 99,999 useconds, it
would be displayed as 0, instead of its actual time. (closes
issue #14532) Reported by: pj Patches:
20090311__bug14532.diff.txt uploaded by tilghman (license 14)
Tested by: pj
2009-03-12 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar
2009) | 2 lines Fix incorrect usage of strncasecmp... I really
meant to use strcasecmp. ........
* /, res/res_musiconhold.c: Merged revisions 181659-181660 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
lines Fix another scenario where depending on configuration the
stream would not get read. For custom commands we don't know
whether the audio is coming from a stream or not so we are going
to have to read the data despite no channels. (closes issue
#14416) Reported by: caspy ........ r181660 | file | 2009-03-12
13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
previous commit. ........
* /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar
2009) | 10 lines Fix issue with streaming MOH failing if nobody
is listening. When a music class is setup to actually provide
music on hold from a stream we need to constantly read audio from
it since it will constantly be providing audio. This is now done
despite there being no channels listening to it. (closes issue
#14416) Reported by: caspy ........
* apps/app_dial.c: Fix crash when sleep and retries argument was
not given to RetryDial application. (closes issue #14647)
Reported by: sherpya
2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com>
* build_tools/make_version: Whitespace chages.
* build_tools/make_version: Use the correct branch integrated
property when generating the version string
2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info>
* configs/sip.conf.sample: Provide correct hint to debug SIP
trouble in the default config (closes issue #14646) Reported by:
strk
2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com>
* main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
thread-safe.
2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 181436 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar
2009) | 4 lines Allow prefix to set localstatedir (when used and
different from the default). This is similar to the /etc change
that was made for the non-FreeBSD case. ........
2009-03-11 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com>
* main/channel.c: Make handling of the BRIDGEPVTCALLID variable
thread-safe.
* main/channel.c, /: Merged revisions 181423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
| 9 lines Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without
the channel locked. This patch fixes some places in channel.c
where this was being done, and lead to crashes related to
masquerades. (closes issue #14623) Reported by: guillecabeza
........
2009-03-11 17:34 +0000 [r181371] David Vossel <dvossel@digium.com>
* channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
181340 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
| 11 lines encrypted IAX2 during packet loss causes decryption to
fail on retransmitted frames If an iax channel is encrypted, and
a retransmit frame is sent, that packet's iseqno is updated while
it is encrypted. This causes the entire frame to be corrupted.
When the corrupted frame is sent, the other side decrypts it and
sends a VNAK back because the decrypted frame doesn't make any
sense. When we get the VNAK, we look through the sent queue and
send the same corrupted frame causing a loop. To fix this,
encrypted frames requiring retransmission are decrypted, updated,
then re-encrypted. Since key-rotation may change the key held by
the pvt struct, the keys used for encryption/decryption are held
within the iax_frame to guarantee they remain correct. (closes
issue #14607) Reported by: stevenla Tested by: dvossel Review:
http://reviewboard.digium.com/r/192/ ........
2009-03-11 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
14 lines Fix issue where an attended transfer could not be
completed under a rare scenario. When completing an attended
transfer chan_sip does a check to make sure the extension in the
URI portion of the Refer-To header is a local valid extension. We
don't actually need to check this since we know for sure the
other channel is already up and talking to the extension. Some
devices do not put the extension in the Refer-To header either,
which can cause the extension check to fail. We now no longer do
this check if it is an attended transfer. (closes issue #14628)
Reported by: sverre Patches: 14628.diff uploaded by file (license
11) ........
2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
since it apparently doesn't work too well during startup.
2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
lines Fix a problem with inband DTMF detection on outgoing SIP
calls when dtmfmode=auto. When dtmfmode was set to auto the
inband DTMF detector was not setup on outgoing SIP calls. This
caused inband DTMF detection to fail. The inband DTMF detector is
now setup for both dtmfmode inband and auto. (closes issue
#13713) Reported by: makoto ........
2009-03-11 16:19 +0000 [r181292] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Replace contents of this doc with a
pointer to its new home
2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix segfault when dialing a typo'd queue If
trying to dial a non-existent queue, there would be a segfault
when attempting to access q->weight, even though q was NULL. This
problem was introduced during the queue-reset merge and thus only
affects trunk. (closes issue #14643) Reported by: alecdavis
2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com>
* apps/app_confbridge.c: Don't play the "you are about to be placed
into the conference" and "the leader has left the conference"
sounds if the quiet option is enabled. (reported by Vadim Lebedev
on the asterisk-dev list)
2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com>
* utils/Makefile, include/asterisk/utils.h,
include/asterisk/astmm.h, channels/chan_sip.c,
channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c,
pbx/pbx_config.c: Fix malloc debug macros to work properly with
h323. The main problem here was that cstdlib was undefining free
thereby causing the proper debug macros to not be used.
ast_h323.cxx has been changed to call ast_free instead to avoid
the issue. A few other issues were addressed: - There were a few
instances of functions improperly passing ast_free instead of
ast_free_ptr. - Some clean up was done to avoid the debug macros
intentionally being redefined. (copied below from Kevin's commit,
appreciate the help) - disable astmm.h from doing anything when
STANDALONE is defined, which is used by the tools in the utils/
directory that use parts of Asterisk header files in hackish
ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined. (closes issue #13593) Reported
by: pj
2009-03-11 02:25 +0000 [r181099] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: tabs to spaces
2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add missing comment that quotes RFC 3891
* /, channels/chan_sip.c: Merged revisions 181029,181031 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
2009) | 9 lines Fix incorrect tag checking on transfers when
pedantic=yes is enabled. (closes issue #14611) Reported by:
klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
2009) | 3 lines Remove unused variables. ........
2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com>
* main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
main/heap.c, include/asterisk/strings.h,
include/asterisk/hashtab.h, main/astobj2.c,
include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
APIs, so that memory leaks can be tracked back to their source.
(related to issue #14636)
* main/pbx.c: Spacing changes only
2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com>
* /, configure, configure.ac, autoconf/ast_prog_sed.m4,
autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) |
1 line Make things happier when using autoconf 2.62+ ........
2009-03-10 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Add some notes on getting in
contact with the dev community
* doc/google-soc2009-ideas.txt: Remove difficulty and language
specifiers
* doc/google-soc2009-ideas.txt: Expand upon documentation of
manager event project
2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: list the move of the astvarrundir from /var/run to
/var/run/asterisk (actually its $(localstatedir)/run/asterisk
Makes setups with asterisk as non-root easier to manage because
you can setup permissions on this dir instead of touching a file
and setting permissions on that. Files that come to mind are
asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell
2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: add more projects
* doc/google-soc2009-ideas.txt: add more project ideas
2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com>
* main/manager.c: Reset the thread local string buffer when
handling the UserEvent action. (closes issue #14593) Reported by:
JimDickenson
2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt: Add current mentors list, and first
pass on a project list broken out of "PineMango" I will work on
adding projects that have been sent to be via email tomorrow.
2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/rtp.h, include/asterisk/extconf.h,
main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
include/asterisk/callerid.h, include/asterisk/doxyref.h,
include/asterisk/event.h, include/asterisk/audiohook.h,
include/asterisk/dsp.h, include/asterisk/timing.h,
include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
include/asterisk/utils.h, include/asterisk/devicestate.h,
include/asterisk/taskprocessor.h, include/asterisk/enum.h,
include/asterisk/astobj2.h, include/asterisk/config.h,
include/asterisk/channel.h, include/asterisk/manager.h,
include/asterisk/heap.h, include/asterisk/logger.h,
include/asterisk/http.h, include/asterisk/res_odbc.h,
include/asterisk/app.h, main/tcptls.c,
include/asterisk/linkedlists.h, include/asterisk/sched.h,
include/asterisk/datastore.h, include/asterisk/lock.h,
include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen
documentation for API changes from 1.6.0 to 1.6.1 Copied from my
review board description: This is a continuation of the API
changes documentation started for describing changes between
releases. Most of the API changes were pretty simple needing only
to be brought to attention via the new "Asterisk API Changes"
list. However, if you see anything that needs further explanation
feel free to supplement what is there. The current method of
documenting is to add (in the header file): \version <ver number>
<description of changes> and then to add the function to the
change list in doxyref.h on the AstAPIChanges page. I also made
sure all the functions that were newly added were tagged with
\since 1.6.1. I think this is a good habit to start both for the
historical aspect as well as for the future ability to easily add
a "New Asterisk API" page. Review:
http://reviewboard.digium.com/r/190/
2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com>
* doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
list
2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com>
* contrib/asterisk-ng-doxygen: Make some minor updates to the
doxygen configuration - add bridges directory to be processed -
add some res/ subdirs - alphabetize subdirs - use consistent
indentation
2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
IMAP storage is enabled. ........
2009-03-06 17:26 +0000 [r180534] David Vossel <dvossel@digium.com>
* /, main/enum.c: Merged revisions 180532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
| 9 lines Fix handling of backreferences for ENUM lookups enum.c
did not handle regex backtraces correctly. The '\1' in the regex
is a backreference that requires a pattern match to be inserted.
The way the code used to work is that it would find the
backreference and insert the entire input string minus the '+'.
This is incorrect. The regexec() function takes in a variable
called pmatch which is an array of structs containing the start
and end indexes for each backreference substring. The original
code actually passed the pmatch array pointer into regexec but
never did anything with it. Now when a backtrace is found, the
backtrace number is looked up in the pmatch array and the correct
substring is inserted. (closes issue #14576) Reported by:
chris-mac Review: http://reviewboard.digium.com/r/187/ ........
2009-03-05 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu,
05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when
identical mailbox names were defined in separate contexts. There
was a fix put in a while back so that an X-Asterisk-VM-Context
message header was added to stored IMAP voicemails. This would
allow for us to differentiate if the same mailbox name was used
in multiple contexts. The problem still left was that not all
places where messages were retrieved actually attempted to use
this header for information when retrieving messages. This commit
fixes that so that MWI and message retrieval from VoiceMailMain
work as expected. (closes issue #13853) Reported by: vicks1
Patches: 13853_v2.patch uploaded by mmichelson (license 60)
Tested by: lmadsen ........
* /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
revisions 180380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
2009) | 25 lines Fix broken mailbox parsing when searchcontexts
option is enabled. When using the searchcontexts option in
voicemail.conf, the code made the assumption that all mailbox
names defined were unique across all contexts. However, the code
did nothing to actually enforce this assumption, nor did it do
anything to alert a user that he may have created an ambiguity in
his voicemail.conf file by defining the same mailbox name in
multiple contexts. With this change, we now will issue a nice
long warning if searchcontexts is on and we encounter the same
mailbox name in multiple contexts and ignore any duplicates after
the first box. Whether searchcontexts is enabled or not, if we
come across a duplicate mailbox in the same context, then we will
issue a warning and ignore the duplicated mailbox. I have also
added a small note to voicemail.conf.sample in the explanation
for searchcontexts explaining that you cannot define the same
mailbox in multiple contexts if you have enabled the option.
(closes issue #14599) Reported by: lmadsen Patches: 14599.patch
uploaded by mmichelson (license 60) (with slight modification)
Tested by: lmadsen ........
2009-03-05 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info>
* Makefile: Make sure we terminate the first s| command so we can
actually produce correct files.
2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c, /, include/asterisk/frame.h, main/rtp.c: Merged
revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
2009) | 9 lines Fix problems when RTP packet frame size is
changed During some code analysis, I found that calling
ast_rtp_codec_setpref() on an ast_rtp session does not work as
expected; it does not adjust the smoother that may on the RTP
session, in fact it summarily drops it, even if it has data in
it, even if the current format's framing size has not changed.
This is not good. This patch changes this behavior, so that if
the packetization size for the current format changes, any
existing smoother is safely updated to use the new size, and if
no smoother was present, one is created. A new API call for
smoothers, ast_smoother_reconfigure(), was required to implement
these changes. Review: http://reviewboard.digium.com/r/184/
........
2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com>
* channels/chan_bridge.c (added), main/Makefile,
bridges/bridge_simple.c, bridges/bridge_softmix.c,
include/asterisk/channel.h, bridges/bridge_multiplexed.c,
CHANGES, Makefile, include/asterisk/bridging_technology.h
(added), bridges (added), bridges/bridge_builtin_features.c,
include/asterisk/bridging_features.h (added),
include/asterisk/bridging.h (added), apps/app_confbridge.c
(added), main/bridging.c (added), bridges/Makefile: Merge phase 1
support for the new bridging architecture. This commit brings in
the bridging core, bridging technologies, and the ConfBridge
application. For usage information on the ConfBridge application
please see the output of "core show application ConfBridge" from
the CLI. For API documentation please see the doxygen page
describing the architecture and the documentation for each API
call. Review: http://reviewboard.digium.com/r/93/
2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com>
* contrib/editors/asterisk.vim: Also highlight the preamble and
postamble
* contrib/editors/ael.vim (added), contrib/editors/asterisk.vim
(added), contrib/editors (added), contrib/editors/asteriskvm.vim
(added): Add syntax coloring files for Vim, including a new one
for AEL
2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Resolve object matching issues related to
the removal of the sip_user object. Previously, chan_sip had both
sip_peer and sip_user objects in memory. A patch went in to
remove sip_user to simplify the code, since everything could be
done with just sip_peer. This patch resolves some regressions
found that were introduced by those changes. This code comes from
svn/asterisk/team/group/sip-object-matching/. Here is a list of
the changes that have been made: 1) When doing a match by name
with the find_peer() function, make it much easier to specify
which objects should be matched by having a parameter that
specifies exactly which object types should be considered. Also,
update find_by_name() to handle this parameter. Finally, update
all code to use the new option values. 2) When looking up an
object for an outbound request by name, consider peers only.
(create_addr()) 3) Only match peers on an incoming registration
request. 4) When doing authentication (except for SUBSCRIBE),
look up users by name, instead of all objects by name. 5) When
doing authentication (except for SUBSCRIBE), after looking for a
user by name, look for a peer by IP address, instead of all
objects by IP address. 6) When handling the SIP qualify CLI
command or manager action, look for a peer by name, instead of
any object by name. 7) When handling the SIP unregister CLI
command, look for a peer by name, instead of any object by name.
9) In sip_do_debug_peer(), search for a peer by name, instead of
any object by name. 9) When handling the SIPPEER() dialplan
function, search for a peer by name, instead of any object by
name. 10) In the following session timer related functions,
st_get_se(), st_get_refresher(), and st_get_mode(), when looking
for an object for a given sip_pvt using pvt->peername, look for a
peer by name, instead of any object by name. 11) Fix build_peer()
to properly handle the case where separate type=peer and
type=user entries were specified in sip.conf. (closes issue
#14505) Reported by: lmadsen Review:
http://reviewboard.digium.com/r/172/
2009-03-04 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com>
* main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c,
main/alaw.c: Spacing changes only
2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com>
* /, main/callerid.c: Merged revisions 180194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
lines Look for the number in a callerid string starting from the
end. This way a value using <> can exist in the name portion.
(issue #AST-194) ........
2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic"
pickups to work when we wish to ignore the context When the
subscription context for a call pickup subscription differs from
the context of the call pickup target, there's not an easy way to
divine what context should be used for the pickup. The way to
work around this is to use PICKUPMARK as the context for the
pickup. This has been documented in the sip.conf.sample file
(ABE-1708) closes issue #14567 submitted by: alecdavis
2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Remove duplicate 'k' and 'K' Dial options.
(closes issue #14601) Reported by: alecdavis Patches:
app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
2009-03-03 23:35 +0000 [r180079] Steve Murphy <murf@digium.com>
* utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by
mistake. Already done to 1.6.x
2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com>
* main/channel.c, include/asterisk/app.h, apps/app_read.c,
main/app.c: app_read does not break from prompt loop with user
terminated empty string In app.c, ast_app_getdata is called to
stream the prompts and receive DTMF input. If ast_app_getdata()
receives an empty string caused by the user inputing the end of
string character, in this case '#', it should break from the
prompt loop and return to app_read, but instead it cycles through
all the prompts. I've added a return value for this special case
in ast_readstring() which uses an enum I've delcared in apps.h.
This enum is now used as a return value for ast_app_getdata().
(closes issue #14279) Reported by: Marquis Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested
by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/
2009-03-03 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com>
* /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
180006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
2009) | 17 lines Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank" value
for a sound file in queues.conf and have no sound played back.
The problem with this is that it would result in some ugly CLI
warnings from file.c. This commit introduces a check when playing
a file in app_queue to see if the name of the file is zero-length
and return early if that is the case. Also, the ability to
specify the blank sound files in queues.conf is now mentioned
more clearly in queues.conf.sample (closes issue #14227) Reported
by: caspy ........
2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com>
* utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h,
main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl,
main/ast_expr2.c: Merged revisions 179807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
work to do to port these changes to trunk; the check_expr stuff
hasn't been updated here for quite some time, it appears. I added
some more tests to the check_expr2 suite. I had to play around
with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
ast_expr2.y so as not to conflict structure with aelparse.
........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
2009) | 19 lines These changes allow AEL to better check ${}
constructs within $[...], that are concatenated with text. I
modified and added rules in ast_expr2.fl to better handle the
concatenations. I added some default routines to ast_expr2.y so
the standalone would compile. It also looks like I haven't run
this thru bison since 2.1, so it's good to get this updated. The
Makefile has comments added now for check_expr2 and check_expr to
explain what they are for, and how to run them. The testexpr2s
stuff has been removed, in favor of check_expr2. expr2.testinput
has been updated to include the two expressions that inspired
these changes (from mcnobody on #asterisk this morning) The
regression has been run and all looks well. ........
2009-03-03 22:01 +0000 [r179972] David Vossel <dvossel@digium.com>
* apps/app_meetme.c: app_meetme not setting filename and fileformat
correctly for realtime When app_meetme finds a realtime
conference, it doesn't get the filename and fileformat correctly
when 'r' is set. Now app_meetme first checks to see if fileformat
and filename are declared in the db, if they're not it checks the
.conf file, if its not declared there either it then uses
defaults. (closes issue #14545) Reported by: dalbaech Patches:
app_meetme-realtime5.patch uploaded by dvossel (license 671)
Realtime_Conference_Record_workaround.txt uploaded by dalbaech
(license 705) Tested by: dvossel, dalbaech Review:
http://reviewboard.digium.com/r/180/
2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com>
* res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation
for timing modules used in Asterisk This document specifies the
timing modules available in Asterisk beginning with Asterisk
1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used
for in Asterisk. There is also a section which can be used to
help customize your setup or to troubleshoot timing issues you
may have. I also added messages to the DAHDI timing test used in
res_timing_dahdi.c that points to this new documentation if
people experience problems. Big thanks to all who contributed
comments on this. (closes issue #14490) Reported by: mmichelson
Patches: timing.txt uploaded by mmichelson (license 60) Review:
http://reviewboard.digium.com/r/164/
2009-03-03 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com>
* apps/app_directed_pickup.c: fix a leaked channel lock (and future
deadlock) when we try to pick up our own channel
2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 179840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
lines Do not assume that the bridge_cdr is still attached to the
channel when the 'h' exten is finished executing. It is possible
for a masquerade operation to occur when the 'h' exten is
operating. This operation moves the CDR records around causing
the bridge_cdr to no longer exist on the channel where it is
expected to. We can not safely modify it afterwards because of
this, so don't even try. (closes issue #14564) Reported by: meric
........
2009-03-03 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com>
* pbx/pbx_spool.c: Convert pbx_spool to use string fields instead
of statically-sized buffers. In tests run after making this
conversion, I noticed an approximate 85% reduction in memory
usage for call file processing. Review:
http://reviewboard.digium.com/r/168/
2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 179741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
| 6 lines Ensure chan->fdno always gets reset to -1 after
handling a channel fd event. Since setting fdno to -1 had to be
moved, a couple of other code paths that do process an fd event
return early and do not pass through the code path where it was
moved to. So, set it to -1 in a few other places, too. ........
2009-03-03 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Please prefix default values with DEFAULT
2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 179671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
lines Move where fdno is set to the default value to *after* the
read callback of the channel driver is called. We have to do this
as the underlying channel driver may need the fdno value to
determine what to read. ........
2009-03-03 13:54 +0000 [r179609] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 179608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
| 9 lines Make it easier to detect an improper call to
ast_read(). When you call ast_waitfor() on a channel, the index
into the channel fds array that holds the file descriptor that
poll() determines has input available is stored in fdno. This
patch clears out this value after a call to ast_read() and also
reports errors if ast_read() is called without an fdno set. From
a discussion on the asterisk-dev list. ........
2009-03-03 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 179536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
| 15 lines Fix bridging regression from commit 176701 This fixes
a bad regression where the bridge would exit after an attended
transfer was made. The problem was due to nexteventts getting set
after the masquerade which caused the bridge to return
AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
tim_ringenbach ........
2009-03-02 23:36 +0000 [r179533] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 179532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
| 40 lines Move ast_waitfor() down to avoid the results of the
API call becoming stale. This call to ast_waitfor() was being
done way too soon in this section of code. Specifically, there
was code in between the call to waitfor and the code that uses
the result that puts the channel in autoservice. By putting the
channel in autoservice, the previous results of ast_waitfor()
become meaningless, as the autoservice thread will do it's own
ast_waitfor() and ast_read() on the channel. So, when we came
back out of autoservice and eventually hit the block of code that
calls ast_read() on the channel, there may not actually be any
input on the channel available. Even though the previous call to
ast_waitfor() in app_meetme said there was input, the autoservice
thread has since serviced the channel for some period of time.
This bug manifested itself while dvossel was doing some testing
of MeetMe in Asterisk trunk. He was using the timerfd timing
module. When the code hit ast_read() erroneously, it determined
that it must have been called because of input on the timer fd,
as chan->fdno was set to AST_TIMING_FD, since that was the cause
of the last legitimate call to ast_read() done by autoservice. In
this test, an IAX2 channel was calling into the MeetMe
conference. It was _much_ more likely to be seen with an IAX2
channel because of the way audio is handled. Every audio frame
that comes in results in a call to ast_queue_frame(), which then
uses ast_timer_enable_continuous() to notify the channel thread
that a frame is waiting to be handled. So, the chances of
ast_waitfor() indicating that a channel needs servicing due to a
timer event on an IAX2 event is very high. Finally, it is
interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When
ast_read() is called and erroneously thinks that there is a timer
event to handle, it calls the ast_timer_ack() function. The
pthread and dahdi timing modules handle the ack() function being
called when there is no event by simply ignoring it. In the case
of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused
Asterisk to lock up very quickly. Thanks to dvossel and
mmichelson for the fun debugging session. :-) ........
2009-03-02 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com>
* /, main/app.c: Merged revisions 179468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
| 10 lines When ending a recording with silence detection,
remember to reduce the duration. The end of the recording is
correspondingly trimmed, but the duration was not trimmed by the
number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration. (closes issue #14406)
Reported by: sasargen Patches: 20090226__bug14406.diff.txt
uploaded by tilghman (license 14) Tested by: sasargen ........
2009-03-02 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com>
* res/res_timing_timerfd.c: Fix a reference leak in
timerfd_set_rate(). (found during a debugging session with
dvossel and mmichelson.)
* main/channel.c, /: Merged revisions 179461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
| 8 lines Ensure that only one thread is calling ast_settimeout()
on a channel at a time. For example, with an IAX2 channel, you
can have both the channel thread and the chan_iax2 processing
threads calling this function, and doing so twice at the same
time is a bad thing. (Found in a debugging session with dvossel
and mmichelson) ........
2009-03-02 20:16 +0000 [r179396] Jason Parker <jparker@digium.com>
* /, main/editline/configure, main/editline/np/unvis.c,
main/editline/sys.h, main/editline/configure.in: Merged revisions
179395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
1 line Remove several silly warnings in editline. One about a
broken preprocessor directive, and another about strlcpy/strlcat.
(closes issue #14264) Reported by: dimas ........
2009-03-02 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe
if db is not loaded)
2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Do not try to remove a registration
scheduled item if the scheduler context has already been
destroyed. (closes issue #14580) Reported by: alecdavis
* main/audiohook.c: Fix issue where changing the volume of both
directions of audio did not work. (closes issue #14574) Reported
by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK
(license 545)
2009-03-01 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com>
* apps/app_speech_utils.c: Swap reversed timevals. This was pointed
out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
* channels/chan_sip.c: Properly free memory and remove scheduler
entries when a transmission failure occurs. Previously, only the
"data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_ERROR was returned by __sip_xmit. When
retrans_pkt was called, this inevitably resulted in the reading
and writing of freed memory. XMIT_ERROR is a condition meaning
that we don't want to attempt resending the packet at all. The
proper action to take is to remove the scheduler entry we just
created, free the packet's data as well as the packet itself, and
unlink it from the list of packets on the sip_pvt structure.
(closes issue #14455) Reported by: Nick_Lewis Patches:
14455.patch uploaded by mmichelson (license 60) Tested by:
Nick_Lewis
2009-02-27 21:47 +0000 [r179164] Russell Bryant <russell@digium.com>
* res/res_ais.c, doc/distributed_devstate.txt,
configs/ais.conf.sample: Mark res_ais as experimental, as the
binary event format is subject to change.
2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c: If config file is blank, don't load
module. (Closes issue #14563)
2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com>
* UPGRADE.txt: Add a note about the ordering of entries in sip.conf
in 1.6.1.
2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Add reload support to chan_skinny.
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the
way we configure devices and lines. (closes issue #10297)
Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by
wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA
(license 3) With mods by me based on feedback from wedhorn and
Russell and seanbright Tested by: DEA, mvanbaak, pj Review:
http://reviewboard.digium.com/r/130/
2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com>
* doc/tex/channelvariables.tex: Update documentation for DIALEDTIME
and ANSWEREDTIME variables. (closes issue #14566) Reported by:
klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
klaus3000 (license 65)
2009-02-27 15:51 +0000 [r179021] Russell Bryant <russell@digium.com>
* sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound
packages. In passing, also fix downloading SLIN16 extra sound
packages. (closes issue #14565) Reported by: jtodd
2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com>
* /, main/features.c, configs/features.conf.sample: Merged
revisions 178956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
case, it's just a matter of reducing the default timeouts from
2000 to 1000 msec, as the max def feature digit timeout is no
longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
-0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
feature digit timeout to 1000 ms from the previous default of
500. As per bug 14515, a dev discussion arrived at a "mediated
concensus" of a default feature digit timeout of 1.0 sec. Some
voted for 1300; ctooley thought 1500 for distracted phone users
in phone booths; kpfleming put his foot down at 1.0 sec. Users
who found the previous default max delay of 250 msec perfect, are
welcome to override the new default. Notice that I said that 250
msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for
14515, we found that 500 msec was actually enforcing a max of
250. The bug fix would restore 500 msec, but we felt even that
was a bit tight for most users... 2000 msec was pushed earlier by
mmichelson, so that reduces to 1000 msec after the bug fix.
Enjoy! ........
2009-02-26 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com>
* main/features.c, CHANGES, configs/features.conf.sample: Sound
confirmation of call pickup success. (closes issue #13826)
Reported by: azielke Patches: pickupsound2-trunk.patch uploaded
by azielke (license 548) __20081124_bug_13826_updated.patch
uploaded by lmadsen (license 10) Tested by: lmadsen
2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last
fix A return statement was missing which caused unexpected cli
output. issue #14479
2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com>
* apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent
compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to
break my dev-mode build. Not a problem in 1.6.x.
* /, main/features.c: Merged revisions 178804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
28 lines This patch prevents the feature detection timeout from
being cut in half. Because the ast_channel_bridge() call will
return 0 and pass a frame pointer for both DTMF_BEGIN and
DTMF_END, the feature_timer field in hte config struct is getting
decremented twice, which effectively cuts the digittimeout in
half. I added conditions to the if statement to only let DTMF_END
frames to flow thru, which solved the problem. Also, when the
frame pointer is null, let control flow thru-- this usually
happens on timeouts. I added a comment to the code to explain
what's going on and why. Many thanks to sodom for reporting this
problem. Personnally, it always seemed like something was wrong
with the featuredigittimeout, but I never could quite decide
what... and was too busy to investigate. This bug forced the
issue, and now we know. Sodom had other issues in 14515, but I
couldn't reproduce them. If he still has problems, and wants to
get them solved, he is welcome to reopen 14515. (closes issue
#14515) Reported by: sodom Patches: 14515.patch uploaded by murf
(license 17) Tested by: murf, sodom ........
2009-02-26 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com>
* main/file.c: Fix an issue where the timer for file playback would
not be stopped if DAHDI was not installed. (closes issue #14541)
Reported by: grant
2009-02-26 15:50 +0000 [r178767] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: IAX2 prune realtime fix Iax2 prune realtime
had issues. If "iax2 prune realtime all" was called, it would
appear like the command was successful, but in reality nothing
happened. This is because the reload that was supposed to take
place checks the config files, sees no changes, and does nothing.
If there had been a change in the the config file, the realtime
users would have been marked for deletion and everything would
have been fine. Now prune_users() and prune_peers() are called
instead of reload_config() to prune all users/peers that are
realtime. These functions remove all users/peers with the
rtfriend and delme flags set. iax2_prune_realtime() also lacked
the code to properly delete a single friend. For example. if iax2
prune realtime <friend> was called, only the peer instance would
be removed. The user would still remain. (closes issue #14479)
Reported by: mousepad99 Review:
http://reviewboard.digium.com/r/176/
2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com>
* main/indications.c: Ensure there is a valid tone part before
trying to play tones. (closes issue #14558) Reported by:
alecdavis
2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net>
* configs/res_snmp.conf.sample: Clarifications on the different
models and reference to further docs.
2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com>
* README: another minor commit to test post-commit script changes
(now testing post-revprop-change as well, third try)
* README: minor commit to test post-commit script changes
2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Picky, picky buildbots
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/stdtime/localtime.c: Use notification when timezone files
change and re-scan then. (closes issue #14300) Reported by:
jamessan Patches: 20090127__bug14300.diff.txt uploaded by
tilghman (license 14) 20090224__bug14300.diff uploaded by
jamessan (license 246) Tested by: jamessan Review:
http://reviewboard.digium.com/r/136/
* res/res_odbc.c: Oops, wrong direction of command
2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 178508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
| 2 lines Update the copyright year for the main page of the
doxygen documentation. ........
2009-02-24 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com>
* /, configs/extensions.conf.sample: Merged revisions 178445 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
| 5 lines Add section about the #exec command in configuration
files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
with additional notes by tilghman (license 14) ........
* main/asterisk.c: Apparently, a void cast doesn't override
warn_unused_result.
* main/asterisk.c: The 3 possible errors with pipe(2) are all
impossible in this situation.
2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 178373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
| 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
to 0 properly. (issue #14460) Reported by: moliveras Tested by:
russell ........
2009-02-24 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com>
* utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the
process, instead of depending upon the astcanary process being
inherited by init.
* utils/astcanary.c: Cause astcanary to exit if Asterisk exits
abnormally and doesn't kill astcanary. Also, add some
documentation supporting the use of astcanary. (closes issue
#14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff
uploaded by KNK (license 545)
2009-02-24 17:42 +0000 [r178300] David Vossel <dvossel@digium.com>
* doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows
manager command to see if IAX link is trunked and encrypted.
Displays what kind of encryption is enabled as well. Manager
command "iaxpeers" now shows if a link is trunked and encrypted.
Instead of encryption saying simply "yes" or "no", it now
displays what type of encryption is enabled and if keyrotation is
on or not. (closes issue #14427) Reported by: snuffy Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by
mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review:
http://reviewboard.digium.com/r/173/
2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 178205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
lines Skip check for extension when subscribing for MWI. Since
the remote side is not actually subscribing to a specific
extension when subscribing for MWI just skip the check to see if
the extension exists. They can't use it to specify the mailbox
either since we require configuration of that in sip.conf (closes
issue #14531) Reported by: festr ........
2009-02-23 23:11 +0000 [r178142] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 178141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
| 14 lines Fix infinite DTMF when a BEGIN is received without an
END. This commit is related to rev 175124 of 1.4 where a previous
attempt was made to fix this problem. The problem with the
previous patch was that the inserted code needed to go _before_
setting the lastrxts to the current timestamp. Because those were
the same, the dtmfcount variable was never decremented, and so
the END was never sent. In passing, I removed the dtmfsamples
variable which was completed unused. I also removed a redundant
setting of the lastrxts variable. (closes issue #14460) Reported
by: moliveras ........
2009-02-23 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Permit emailsubject and emailbody to be set per mailbox. (closes
issue #14372) Reported by: fhackenberger Patches:
voicemail_individual_subject_and_body_1.6.1 uploaded by
fhackenberger (license 592) with additional fixes by Corydon76
(license 14)
2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: update the new manager commands in
chan_skinny to match chan_sip's headers. requested by oej.
2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Changes the way keyrotation is enabled by
default Key rotation was enabled by default by setting the global
encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this
is that if encryption is not enabled, and the encryption method
is set to anything except 0, the peer appears to have encryption
enabled when issuing a "iax2 show peers". Rather than have the
key rotation bit always set by default, it is now only set when
an encryption method is enabled. (closes issue #14523) Reported
by: mvanbaak
2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: list the addition of the SKINNY manager actions in the
CHANGES file.
2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com>
* tests/test_sched.c, main/sched.c: Fix a regression in scheduler
entry ordering, and add a regression test for it. (closes issue
#14522) Reported by: pj Tested by: russell
2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info>
* doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of
manager commands to chan_skinny Added: SKINNYdevices
SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521)
Reported by: mvanbaak Review:
http://reviewboard.digium.com/r/170/
2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: On update, test against the existence of
sipregs.
2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: add extra check for sysinfo/sysctl (closes issue
#14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff
uploaded by snuffy (license 35)
2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com>
* main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace,
minor coding guideline fixes, and start beefing up the hashtab
documentation a bit.
2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com>
* include/asterisk/indications.h: Fix build issues on Solaris and
OpenBSD. (closes issue #14512) Reported by: snuffy
2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info>
* Makefile, contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.archlinux.asterisk,
contrib/scripts/safe_asterisk: set
ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When
running asterisk as non-root and without this patch the pidfile
wants to go into /var/run/asterisk.pid. This directory is not
writable for the non-root user and changing permissions is not an
option. Putting it in /var/run/asterisk/asterisk.pid makes it
possible to set permissions on the /var/run/asterisk dir so
everything works as it should be. Patched committed is based on
pabelanger's patch. (closes issue #13153) Reported by: pabelanger
Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by
mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/
* channels/chan_sip.c: make chan_sip.c compile on OpenBSD again.
2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 177786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
| 9 lines Don't print the CR-NL combination when we aren't
outputting to the manager. An embedded CR-NL in a CLI command
screws up several AMI parsers that don't expect to see that
combination in the middle of output. (Closes issue #14305)
Reported by: martins Patch by: tilghman ........
* /, include/asterisk/threadstorage.h: Merged revisions 177701 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
| 3 lines This exception does not appear to still be true for
Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel. ........
2009-02-20 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4
Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a
pages_transferred integer to indicate the number of pages
transferred (so far) during the fax session. The
spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and
pages_rx. This revision uses the new integers for
spandsp-0.0.6pre4 while maintaining backwards compatibility for
previous spandsp releases.
2009-02-20 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, main/app.c, apps/app_system.c: Allow
semicolons to be escaped, when passing arguments to the System
command. (closes issue #14231) Reported by: jcovert Patches:
20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
corrected_20090113__bug14231__2.diff.txt uploaded by jcovert
(license 551) Tested by: jcovert
* apps/app_voicemail.c: Oops, merge broke trunk
2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Set sip_request ast_str data to NULL so
ast_str_copy allocates space properly in copy_request (issue
#14478) Reported by: erik_dedecker
2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com>
* /, main/Makefile, main/ast_expr2f.c: Merged revisions 177540 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was
already pretty 8-bit clean; but I'm still removing the --full
from the flex command so everything is uniform. ........ r177540
| murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2
scanner. The real culprit was the --full argument to flex in the
Makefile! This causes a 7-bit scanner to be generated. I reviewed
the rules and found one rule where I needed to specifically
include 8-bit chars for a token. I tested against the text
supplied by ibercom, and all looks very well. This has been there
a surprisingly long time! (closes issue #14498) Reported by:
ibercom Patches: 14498.patch uploaded by murf (license 17) Tested
by: murf ........
2009-02-19 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19
Feb 2009) | 7 lines Fix up potential crashes, by reducing the
sharing between interactive and non-interactive threads. (closes
issue #14253) Reported by: Skavin Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin ........
* doc/database_transactions.txt (added): Document how to use
database transactions
2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/channel.h: Fix another merge error from 176708
2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com>
* /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb
2009) | 3 lines If we are able to create a speech structure unset
the ERROR variable in case it was previously set. (issue
#LUMENVOX-13) ........
2009-02-19 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Fix mismerge from revision 176708 pointed out by
Kaloyan Kovachev on the asterisk-dev mailing list. Thanks!
2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES,
res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction
support
2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com>
* CHANGES: Update CHANGES file to include MWI subscription support
that was added some time ago.
2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com>
* main/strings.c: Handle negative length and eliminate a condition
that is always true.
2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com>
* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
34 lines This patch fixes a regression of sorts that was
introduced in rev 24425. It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only
handle about 4893 of the 6000 extens in the single context. This
was due to the regression I mentioned. To get rid of shift/reduce
conflicts, Luigi set up right-recursive lists for globals,
context elements, switch lists, and statements. Right recursive
lists got rid of the warnings, but instead, they use up a
tremendous amount of stack space when the lists are long. I saw
this a few years back, and resolved not to fix it until someone
complained. That day has arrived! After the changes were made, I
ran the regression test suite, and there were no problems. I took
the test case the user provided, and added 100,000 extensions to
the single context, that already had 6,000 extens in it. (I'll
see your 6, and raise you 100!) It takes a few minutes to read it
all in, check it and generate code for it, but no problems. So, I
think I can say that fundamentally, there are no longer any
limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints. ........
2009-02-18 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: fix two very minor bugs: if anyone ever uses
SLINEAR16 as a format in RTP, ensure that the samples are
byte-swapped to network order if needed. also, when a smoother is
operating on a format that has a sample rate other than 8000
samples per second, use the proper sample rate for computing
delivery timestamps.
2009-02-18 22:51 +0000 [r177226] David Vossel <dvossel@digium.com>
* main/features.c: Locking issue in action_bridge and bridge_exec
action_bridge() and bridge_exec() both search for the channels to
bridge to, and then immediately drop the lock. Instead, they
should hold the lock until the masquerade is complete. This will
guarantee the channel remains and prevent any other weirdness
from occurring. In action_bridge() some more weirdness comes into
play. Both channels are needlessly locked at the same time and
perform the exact same logic. It makes sense from a coding
organizational standpoint, but could cause a theoretical deadlock
so I split the code up. There is an issue associated with this,
but since its a rather complicated thing to reproduce I'm not
certain this alone will close it. issue# 14296 Review:
http://reviewboard.digium.com/r/167/
2009-02-18 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com>
* channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx,
channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added),
configure, channels/h323/compat_h323.h, configure.ac,
channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
channels/h323/ast_h323.h, channels/h323/chan_h323.h,
channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib
as well as the older PWLib Several changes in PTLib have occurred
requiring build time detection. Changes accounted for include the
library name change, config option change, install location
change, and a boolean type change which is handled by
ast_ptlib.h. Also, the sed check has been modified to properly
work with autoconf >= 2.62. (closes issue #14224) Reported by:
bergolth Patches: asterisk-autoconf-sed.patch uploaded by
bergolth (license 661) asterisk-pwlib-v3.patch uploaded by
bergolth (license 661) Tested by: jpeeler
2009-02-18 19:12 +0000 [r177101] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev
62297. Enabling this option by default proved to be a bad idea,
as the talker detection is not very reliable. So, make it
optional again, and off by default. (issue #13801) Reported by:
justdave
2009-02-18 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/config.h: Merged revisions 177096 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009)
| 2 lines Document the return value of the update method (as
requested on -dev list) ........
2009-02-18 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com>
* main/utils.c: Fixed error where a check for an zero length,
terminated string was needed.
2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix ordering of output for a ChannelUpdate
manager event. (closes issue #14497) Reported by: vinsik Patches:
chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
2009-02-18 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com>
* main/utils.c: Need to take into account the \0 terminator of the
old string to determine the amount available.
2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com>
* main/pbx.c: This patch fixes merge_contexts_and_delete so it does
not deadlock when hints are present. Reason: when I re-engineered
the merge_and_delete func to reduce its lock time, I failed to
notice that the functions it calls still also do locking as
before. This leads to deadlocks on dialplan reloads, when there
are actually living, subscribed hints registered in the system.
While the reporter come across this problem while using AEL, I
might note that these deadlocks should also happen if
extensions.conf were used. Here I added these routines to pbx.c:
ast_add_extension_nolock add_pri_lockopt
ast_add_extension2_lockopt find_context add_hint_nolock All of
the above routines are static and restricted to be used only
within pbx.c, and more specifically within the
merge_contexts_and_delete routine. They are pretty much the same
as their counterparts except they don't lock contexts or hints.
Most of them now do the real work of their name-alike, with
optional locking via extra arguments, and are called by their
name-alike. The goal was to have the original functions so they
would behave exactly as before. Both PJ and I tested these fixes,
and the deadlocking problem is no longer encountered. (closes
issue #14357) Reported by: pj Patches: 14357.diff uploaded by
murf (license 17) Tested by: pj, murf
2009-02-18 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com>
* include/asterisk/heap.h: Add example code for a heap traversal.
* main/pbx.c: Fix a number of incorrect uses of strncpy(). The big
problem here is that the 3rd argument provided in these uses of
strncpy() did not reserve a byte for the null terminator, leaving
the potential for writing one byte past the end of the buffer.
Aside from this, there were coding guidelines violations with
regards to spacing, as well as hard coded lengths being used
instead of sizeof().
2009-02-18 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c: T38 faxdetect should jump to the 'fax'
extension for incoming calls only The previous implementation of
T38 faxdetect resulted in both sides of the call jumping to a fax
extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current
context. This revision will jump to a 'fax' extension on incoming
calls only.
2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c: suppress smoothers for Siren codecs as well as Speex
and G.723.1
2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com>
* apps/app_milliwatt.c: Remove a dependency that no longer exists.
2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
with G723. This commit brings in the changes that were living out
on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder
branch. codec_dahdi.c now always uses signed linear as the simple
codec so that a soft g729 codec will not end up being preferred
to the hardware codec. There are also changes to allow
codec_dahdi.c to feed packets to the hardware in the native
sample size of the codec. This solves problems with choppy audio
when using G723.
2009-02-17 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /, main/features.c, include/asterisk/channel.h:
Merged revisions 176701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009)
| 17 lines Modify bridging to properly evaluate DTMF after first
warning is played The main problem is currently if the Dial flag
L is used with a warning sound, DTMF is not evaluated after the
first warning sound. To fix this, a flag has been added in
ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played
back (due to a feature evaluation or waiting for digits).
ast_channel_bridge was modified to store the nexteventts in the
ast_bridge_config structure as that information was lost every
time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations. (closes issue #14315) Reported by:
tim_ringenbach Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/ ........
2009-02-17 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com>
* tests/test_sched.c: Use constants from inttypes.h to clear up
32-bit compilation errors
2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c: create a UDPTL structure in
create_addr_from_peer() if it does not already exist for T38 This
is required to create a UDPTL structure in
create_addr_from_peer() to handle the scenario where
't38pt_udptl=yes' is not defined in the [general] section of
sip.conf but is defined the peer's context. I tested this patch
by enabling t38pt_udptl in the [general] section on one system
and only enabling t38pt_udptl in a peer's context on the system
sending a fax. Without the patch, the sending system will fail to
initiate T38 negotiation with the warning message, "No way to add
SDP without an UDPTL structure". When this patch is applied the
sending side will successfully initiate T38 negotiation.
2009-02-17 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com>
* include/asterisk/frame.h: Clear up documentation of
AST_FRIENDLY_OFFSET in frame.h
2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 176661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009)
| 9 lines Backport change to 1.4: Prior to masquerade, move the
group definitions to the channel performing the masq, so that the
group count lingers past the bridge. (closes issue #14275)
Reported by: kowalma Patches: 20090216__bug14275.diff.txt
uploaded by Corydon76 (license 14) Tested by: kowalma ........
2009-02-17 21:22 +0000 [r176666] Russell Bryant <russell@digium.com>
* main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
res/res_timing_timerfd.c, include/asterisk/timing.h,
main/timing.c: Update the timing API to have better support for
multiple timing interfaces. 1) Add module use count handling so
that timing modules can be unloaded. 2) Implement unload_module()
functions for the timing interface modules. 3) Allow multiple
timing modules to be loaded, and use the one with the highest
priority value. 4) Report which timing module is being use in the
"timing test" CLI command. (closes issue #14489) Reported by:
russell Review: http://reviewboard.digium.com/r/162/
2009-02-17 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: Prior to masquerade, move the group
definitions to the channel performing the masq, so that the group
count lingers past the bridge. (closes issue #14275) Reported by:
kowalma Patches: 20090216__bug14275.diff.txt uploaded by
Corydon76 (license 14) Tested by: kowalma
2009-02-17 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com>
* tests/test_sched.c (added), main/sched.c: Significantly improve
scheduler performance under high load. This patch changes the
scheduler to use a max-heap to store pending scheduler entries
instead of a fully sorted doubly linked list. When the number of
entries in the scheduler gets large, this will perform much
better. For much more detailed information on this change, see
the review request. Review: http://reviewboard.digium.com/r/160/
* tests/test_heap.c (added): Add a test module for the heap
implementation. Review: http://reviewboard.digium.com/r/160/
* main/Makefile, main/heap.c (added), include/asterisk/heap.h
(added): Add an implementation of the heap data structure. A heap
is a convenient data structure for implementing a priority queue.
Code from svn/asterisk/team/russell/heap/. Review:
http://reviewboard.digium.com/r/160/
2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net>
* include/asterisk/config.h: Typo
2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com>
* channels/chan_unistim.c, main/pbx.c, apps/app_read.c,
configs/indications.conf.sample, apps/app_playtones.c (added),
include/asterisk/indications.h, apps/app_readexten.c,
apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h,
include/asterisk/_private.h, main/indications.c, main/loader.c,
main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
funcs/func_channel.c, res/snmp/agent.c, main/app.c,
res/res_indications.c (removed), main/asterisk.c: Merge a large
set of updates to the Asterisk indications API. This patch
includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object
management in this API was significantly flawed, and a number of
trivial situations could cause crashes. The changes included are:
1) Remove the module res_indications. This included the critical
functionality that actually loaded the indications configuration.
I have seen many people have Asterisk problems because they
accidentally did not have an indications.conf present and loaded.
Now, this code is in the core, and Asterisk will fail to start
without indications configuration. There was one part of
res_indications, the dialplan applications, which did belong in a
module, and have been moved to a new module, app_playtones. 2)
Object management has been significantly changed. Tone zones are
now managed using astobj2, and it is no longer possible to crash
Asterisk by issuing a reload that destroys tone zones while they
are in use. 3) The API documentation has been filled out. 4) The
API has been updated to follow our naming conventions. 5) Various
bits of code throughout the tree have been updated to account for
the API update. 6) Configuration parsing has been mostly
re-written. 7) "Code cleanup" The code is from
svn/asterisk/team/russell/indications/. Review:
http://reviewboard.digium.com/r/149/
2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to
track down a refcount leak. (closes issue #14485) Reported by:
davevg
2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com>
* main/pbx.c, apps/app_queue.c: Fix a race condition that caused
device states to become incorrect for hints. The problem here is
that the hint processing code was subscribed to the wrong event
type. So, it started processing state for a hint too soon, before
the device state cache had been updated. Also, fix a similar bug
in app_queue, as it was also subscribed to the wrong event type.
(closes issue #14461) Reported by: alecdavis
2009-02-17 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net>
* configs/extconfig.conf.sample: Typo
* main/config.c: If there are no realtime engines, there's no
reason to check for realtime families
2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: In this version, we can combine the queries,
because we support dropping nonexistent columns.
* /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
| 10 lines After a 'sip reload', qualifies for realtime peers
weren't immediately restarted, instead waiting until the next
registration. We're now caching the qualify across a
reload/restart and starting the qualify immediately upon loading
the peer. (closes issue #14196) Reported by: pdf Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf ........
* main/strings.c: Might want to update the buffer pointer after a
realloc (or we crash) (closes issue #14485) Reported by: davevg
2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com>
* sounds/sounds.xml: add support for Siren7 and Siren14 flavors of
prompts and music on hold
2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16
Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not
being relayed correctly during bridging This should have been
committed with rev176247, but I missed it. srcupdate frames no
longer break out of the native bridge, but are not being sent to
the other call leg either. This fixs that. issue #13749 ........
2009-02-16 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com>
* channels/chan_skinny.c: Use the correct list macros for deleting
an item from the middle of a list. (issue #13777) Reported by: pj
Patches: 20090203__bug13777.diff.txt uploaded by Corydon76
(license 14) Tested by: pj
2009-02-16 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com>
* /, main/utils.c, include/asterisk/stringfields.h: Merged
revisions 176216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb
2009) | 3 lines fix a flaw in the ast_string_field_build() family
of API calls; these functions made no attempt to reuse the space
already allocated to a field, so every time the field was written
it would allocate new space, leading to what appeared to be a
memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46
-0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the
last stringfields commit... don't mark additional space as
allocated if the string was built using already-allocated space
........
2009-02-16 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com>
* /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon,
16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it
to be nonblocking atomically Apparently on FreeBSD, attempting to
set the O_NONBLOCKING flag separately from opening the file was
causing an "inappropriate ioctl for device" error. While I cannot
fathom why this would be happening, I certainly am not opposed to
making the code a bit more compact/efficient if it also fixes a
bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
uploaded by ys (license 281) Tested by: ys ........ r176252 |
mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
lines Remove unused variable and make dev-mode compilation happy
........
2009-02-16 21:30 +0000 [r176248] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Merged revisions 175597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
Fixed iax2 key rotation backwards compatibility Turns key
rotation back on by default. Added bit into encryption IE to
indicate whether or not key rotation is supported or not. If it
is not supported then it is not enabled, which insures backwards
compatibility. This eliminates the need for the keyrotate option
in iax.conf, so it has been removed. ........
2009-02-16 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com>
* main/logger.c: Assist proper thread synchronization when stopping
the logger thread. I was finding that on my dev box, occasionally
attempting to "stop now" in trunk would cause Asterisk to hang. I
traced this to the fact that the logger thread was waiting on a
condition which had already been signalled. The logger thread
also need to be sure to check the value of the
close_logger_thread variable. The close_logger_thread variable is
only checked when the list of logmessages is empty. This allows
for the logger thread to print and free any pending messages
before exiting.
2009-02-16 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Can't set debug level 2 (intense
debugging) unless the syntax matches
2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com>
* channels/chan_features.c (removed): Remove chan_features. Review:
http://reviewboard.digium.com/r/161/
2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
lines Don't have the Via header stored as a stringfield as it can
change often during the lifetime of a dialog. This issue crept up
with subscriptions on the AA50. When an outgoing NOTIFY is sent a
new branch value is created and the Via header is changed to
reflect it. Since this was a stringfield a new spot in the pool
was used for the value while the old was left untouched/unused.
If the current pool was full a new pool was created. This would
cause memory usage to increase steadily. (issue #AA50-2332)
........
2009-02-16 02:54 +0000 [r175983] Russell Bryant <russell@digium.com>
* main/channel.c: Make the causes array static, and remove the type
name as it is not needed.
2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_unistim.c, /, channels/chan_sip.c,
include/asterisk/manager.h, doc/unistim.txt: Merged revisions
175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
| 3 lines fix mis-spelling of the word registered. Reported by
De_Mon on #asterisk-dev. ........
2009-02-15 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com>
* include/asterisk/sched.h, main/sched.c: Make ast_sched_report()
and ast_sched_dump() thread safe.
* channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix
a number of problems with ast_sched_report(). 1) It had numerous
coding guidelines violations with regards to formatting. 2) It
allocated memory using ast_calloc() that was never freed. 3) It
didn't check for failure from the allocation. 4) It used
sprintf() and strcat() to build the result, doing zero checking
to prevent writing past the end of the provided buffer. The
function also lacks API documentation, but that has not been
addressed in this commit.
2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net>
* formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb
2009) | 2 lines format_ilbc does not depend on codec libraries
and can therefore always be made. My mistake. Ursäkta! ........
* formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb
2009) | 2 lines Disable format_ilbc.so by default, like
codec_ilbc.so ........
* /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2
lines Make sure that the debug line is not printed on debug level
0 ........
2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com>
* doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset
branch to Asterisk From a user point-of-view, this adds new CLI
commands and Manager Actions to better facilitate the reloading
of queues and the resetting of their statistics. The new CLI
commands are the "queue reload" and "queue reset stats" commands.
The new manager actions are the QueueReload and QueueReset
commands. Review: http://reviewboard.digium.com/r/115
* doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for
chanspy starting or stopping (closes issue #14469) Reported by:
caio1982 Patches: chanspy_events2.diff uploaded by caio1982
(license 22)
2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com>
* res/res_jabber.c: fix a few more XML documentation problems
* main/pbx.c: add missing </para>
2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c:
Fixed iax2 key rotation backwards compatibility Turns key
rotation back on by default. Added bit into encryption IE to
indicate whether or not key rotation is supported or not. If it
is not supported then it is not enabled, which insures backwards
compatibility. This eliminates the need for the keyrotate option
in iax.conf, so it has been removed. Review:
http://reviewboard.digium.com/r/159/
2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri,
13 Feb 2009) | 16 lines Fix a potential crash situation when
using IMAP voicemail If calling into VoiceMailMain when using
IMAP storage, it was possible to crash Asterisk by hanging up the
phone when prompted for a voicemail mailbox. This patch fixes the
issue. While it may appear that this patch is superficial, it
allows code execution to continue to the failure case just below
the IMAP_STORAGE code block where this patch has been applied
(closes issue #14473) Reported by: dwpaul Patches:
voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
689) ........
2009-02-13 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com>
* apps/app_record.c: Add an option to keep the recorded file upon
hangup. (closes issue #14341) Reported by: fnordian
2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: document G.722.1/.1C support
* main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h,
channels/chan_h323.c, include/asterisk/frame.h,
formats/format_siren14.c (added), main/rtp.c,
formats/format_siren7.c (added): Add basic (passthrough,
playback, record) support for ITU G.722.1 and G.722.1C (also
known as Siren7 and Siren14) This patch adds passthrough, file
recording and file playback support for the codecs listed above,
with negotiation over SIP/SDP supported. Due to Asterisk's
current limitation of treating a codec/bitrate combination as a
unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are
supported. Along the way, some related work was done: 1) The
rtpPayloadType structure definition, used as a return result for
an API call in rtp.h, was moved from rtp.c to rtp.h so that the
API call was actually usable. The only previous used of the API
all was chan_h323.c, which had a duplicate of the structure
definition instead of doing it the right way. 2) The hardcoded
SDP sample rates for various codecs in chan_sip.c were removed,
in favor of storing these sample rates in rtp.c along with the
codec definitions there. A new API call was added to allow
retrieval of the sample rate for a given codec. 3) Some basic
'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip
*must* decline any media streams offered for these codecs that
are not at the bitrates that we support (otherwise Bad Things
(TM) would result). Review: http://reviewboard.digium.com/r/158/
2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* CHANGES: add 'faxbuffers' configuration option information to
CHANGES
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
dynamic fax buffer configuration option to chan_dahdi.conf When
the 'faxdetect' configuration option is used, one may also want
to use the 'faxbuffers' configuration option in chan_dahdi.conf.
This option will dynamically use the configured 'faxbuffers'
buffer policy on a channel for the life of the call following the
detection of fax tones. The faxbuffers buffer policy will be
reverted during call teardown. An example use of 'faxbuffers' is
below. This example would switch to using 6 buffers with a full
buffer policy. faxbuffers=>6,full
2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove useless string copy, and make sscanf
safe again
2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
force encryption option to iax.conf This patch adds
forceencryption=yes as an iax.conf option. When force encryption
is enabled, no unencrypted connections are allowed. This insures
all connections are encrypted. This is a new feature, so CHANGES
and iax.conf.sample are updated as well. (closes issue #13285)
Reported by: sgofferj Tested by: russell Review:
http://reviewboard.digium.com/r/150/
2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c, /: Merged revisions 175311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
| 9 lines Fix crashes when receiving certain T.38 packets. Also,
increase the maximum size of T.38 packets and warn users when
they try to set the limits above those maximums. (closes issue
#13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
uploaded by Corydon76 (license 14) Tested by: schern ........
2009-02-12 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 175294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
| 9 lines Fix ParkedCall event information for From field in the
case of a blind transfer If the parker information can not be
obtained from the peer, try and see if the BLINDTRANSFER channel
variable has been set. Previously, a blind transfer to the
ParkAndAnnounce app would return nothing for the From. Closes
AST-189 ........
2009-02-12 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Avoid using ast_strdupa() in a loop.
* build_tools/cflags.xml: Don't enable something by default that
has a dependency on something _not_ enabled by default.
menuselect was not happy with this.
2009-02-12 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: correct warning message to not refer
specifically to DAHDI
2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 175187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
| 6 lines Fix crash in event of failed attempt to transfer to
parking The peer may not necessarily exist, such as in the case
of a transfer to ParkAndAnnounce. In this case don't try to play
a sound to it. ........
2009-02-12 17:07 +0000 [r175127] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Setting key rotation to be off by default
Key rotation breaks compatibility between (trunk/1.6.1) and
(1.2/1.4/1.6.0). As a follow up to this, I am investigating
possible ways to allow key rotation to be on by default and not
affect the other branches, but for now it must be turned off.
2009-02-12 16:57 +0000 [r175125] Russell Bryant <russell@digium.com>
* /, main/rtp.c: Merged revisions 175124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
| 27 lines Don't send DTMF for infinite time if we do not receive
an END event. I thought that this was going to end up being a
pretty gnarly fix, but it turns out that there was actually
already a configuration option in rtp.conf, dtmftimeout, that was
intended to handle this situation. However, in between Asterisk
1.2 and Asterisk 1.4, the code that processed the option got
lost. So, this commit brings it back to life. The default timeout
is 3 seconds. However, it is worth noting that having this be
configurable at all is not really the recommended behavior in RFC
2833. From Section 3.5 of RFC 2833: Limiting the time period of
extending the tone is necessary to avoid that a tone "gets
stuck". Regardless of the algorithm used, the tone SHOULD NOT be
extended by more than three packet interarrival times. A slight
extension of tone durations and shortening of pauses is generally
harmless. Three seconds will pretty much _always_ be far more
than three packet interarrival times. However, that behavior is
not required, so I'm going to leave it with our legacy behavior
for now. Code from svn/asterisk/team/russell/issue_14460 (closes
issue #14460) Reported by: moliveras ........
2009-02-12 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: Make lock information
for ao2_trylock be more useful and gnarly Core show locks
information involving an ao2_trylock did not show the function
that called ao2_trylock, but would instead show ao2_trylock as
the source of the lock. This is not useful when trying to debug
locking issues. One bizarre note is that this logic is already in
1.4 but somehow did not get merged to trunk or the 1.6.X
branches.
2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Issue a warning message if our candidate's
IP is the loopback address. (closes issue #13985) Reported by:
jcovert Tested by: phsultan
* /, channels/chan_gtalk.c: Merged revisions 175029 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12
Feb 2009) | 12 lines Set the initiator attribute to lowercase in
our replies when receiving calls. This attribute contains a JID
that identifies the initiator of the GoogleTalk voice session.
The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters. (closes issue
#13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
by jcovert (license 551) Tested by: jcovert ........
2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a bit of odd logic for announcing position.
Sync with 1.6.0's logic
* apps/app_queue.c: Fix odd "thank you" sound playing behavior in
app_queue.c If someone has configured the queue to play an
position or holdtime announcement, then it is odd and potentially
unexpected to hear a "Thank you for your patience" sound when no
position or holdtime was actually announced. This fixes the
announcement so that the "thanks" sound is only played in the
case that a position or holdtime was actually announced. There is
a way that the "thank you" sound can be played without a position
or holdtime, and that is to set announce-frequency to a value but
keep announce-position and announce-holdtime both turned off.
(closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
uploaded by putnopvut (license 60) Tested by: caspy
* apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c,
apps/app_waitforsilence.c, include/asterisk/channel.h: Fix 'd'
option for app_dial and add new option to Answer application The
'd' option would not work for channel types which use RTP to
transport DTMF digits. The only way to allow for this to work was
to answer the channel if we saw that this option was enabled. I
realized that this may cause issues with CDRs, specifically with
giving false dispositions and answer times. I therefore modified
ast_answer to take another parameter which would tell if the CDR
should be marked answered. I also extended this to the Answer
application so that the channel may be answered but not CDRified
if desired. I also modified app_dictate and app_waitforsilence to
only answer the channel if it is not already up, to help not
allow for faulty CDR answer times. All of these changes are going
into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the
changes except for the change to the Answer application will go
in since we do not introduce new features into stable branches
(closes issue #14164) Reported by: DennisD Patches: 14164.patch
uploaded by putnopvut (license 60) Tested by: putnopvut Review:
http://reviewboard.digium.com/r/145
2009-02-11 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com>
* main/channel.c: Tell the device state core a change happened when
a channel is freed but not a specific state. We need to do this
because while we know that the freeing of the channel may cause
something to become not in use we do not know this for sure.
There may be another channel that is still up which would cause
it to be in use. (closes issue #13238) Reported by: kowalma
Patches: 20090121__bug13238.diff.txt uploaded by Corydon76
(license 14) Tested by: alecdavis
2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Fix potential for stack overflows in
app_chanspy.c When using the 'g' or 'e' options, the stack
allocations that were used could cause a stack overflow if a
spyer stayed on the line long enough without actually
successfully spying on anyone. The problem has been corrected by
using static buffers and copying the contents of the appropriate
strings into them instead of using functions like alloca or
ast_strdupa
* main/manager.c: Fix an fd leak that would occur in HTTP AMI
sessions The explanation behind this fix is a bit complicated,
and I've already typed it up in the code as a huge comment inside
of manager.c, so I'll give the abridged version here. We needed a
way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to
not have to change every single manager action was to rename the
current mansession structure and wrap it inside a new mansession
structure which actually contains action- specific data. (closes
issue #14364) Reported by: awk Patches: 14364_better.patch
uploaded by putnopvut (license 60) Tested by: putnopvut Review:
http://reviewboard.digium.com/r/148/
2009-02-10 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only decrease inringing count if above zero.
(issue #13238) Reported by: kowalma
2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com>
* main/slinfactory.c, include/asterisk/slinfactory.h: improve
slinfactory API to remove implicit sample rate and require
explicit sample rate selection by creator of the slinfactory
2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com>
* /, main/jitterbuf.c: Merged revisions 174583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
2009) | 18 lines Improve behavior of jitterbuffer when
maxjitterbuffer is set. This change improves the way the
jitterbuffer handles maxjitterbuffer and dramatically reduces the
number of frames dropped when maxjitterbuffer is exceeded. In the
previous jitterbuffer, when maxjitterbuffer was exceeded, all new
frames were dropped until the jitterbuffer is empty. This change
modifies the code to only drop frames until maxjitterbuffer is no
longer exceeded. Also, previously when maxjitterbuffer was
exceeded, dropped frames were not tracked causing stats for
dropped frames to be incorrect, this change also addresses that
problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
by mnicholson (license 96) Tested by: mnicholson Review:
http://reviewboard.digium.com/r/144/ ........
2009-02-10 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Set the type for the peer structure to be a
peer as the default. (closes issue #14447) Reported by: triccyx
* channels/chan_sip.c: Make the logic for inuse and inringing
manipluation match that of 1.4. The old broken logic would reset
the values back to 0 during certain scenarios causing the wrong
state to be reported. (closes issue #14399) Reported by: caspy
(issue #13238) Reported by: kowalma
2009-02-10 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c, apps/app_voicemail.c: Fix0ring build
* apps/app_stack.c: Remove the usage of the KeepAlive app, as it no
longer exists.
2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com>
* apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c. (closes issue #14435) Reported by: D_McNaul
* apps/app_rpt.c: More intptr_t work.
* /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
lines This patch solves some compiler complaints in both 32 and
64-bit environments. ........
2009-02-09 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix something I messed up in the merge I
just did
2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com>
* apps/app_externalivr.c: Fixes issue with hangups not being sent
and external process never terminating. The ignore_hangup,
run_dead, and noanswer flags were never initilized to zero
causing hangups to never be issued. If the external script
expects to be notified of a hangup and never receives one, it
runs indefinitely. (closes issue #14251) Reported by: chris-mac
Tested by: dvossel
2009-02-09 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
2009) | 12 lines Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname if a port has
been specified, so that's what we're going to do. See section
4.2. (closes issue #14419) Reported by: klaus3000 Patches:
patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
(license 65) ........
2009-02-09 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb
2009) | 4 lines Don't overwrite our pointer to the music class
when music on hold stops. We will use this if it starts again to
see if we can resume the music where it left off. (closes issue
#14407) Reported by: mostyn ........
2009-02-07 16:16 +0000 [r174149] Russell Bryant <russell@digium.com>
* /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
| 2 lines Fix a race condition that could cause a crash. ........
2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
| 5 lines check ast_strlen_zero() before calling ast_strdupa() in
sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to
determine the changes, then modified the suggested changes to
create a proper fix. The summary above is a complete description
of the changes. (closes issue #13547) Reported by: tecnoxarxa
Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa ........
2009-02-06 20:12 +0000 [r174046] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
immediate yes/no option to iax.conf This is very similar to the
DAHDI immediate=yes option. When the phone is picked up, instead
of giving a dialtone it connects directly to the "s" extension.
Changes where implemented in chan_iax2.c to directly connect to
the "s" extension in the appropriate context when this option is
enabled. Examples explaining its use are added to
iax2.conf.sample. CHANGES has been updated as well. (closes issue
#14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk
uploaded by jcovert (license 551) iax.conf.sample.patch uploaded
by jcovert (license 551) Tested by: jcovert, dvossel Review:
http://reviewboard.digium.com/r/143/
2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo
channels. It is futile. This solves an issue where duplicated
pseudo channels would cause a crash because the first one would
unsubscribe and the next one would also try to unsubscribe the
same subscription. (closes issue #14322) Reported by: amessina
* /, channels/chan_sip.c: Merged revisions 173967-173968 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
lines Some clients do not put the call-id for replaces at the
beginning, so support it being anywhere in the string. (closes
issue #14350) Reported by: fhackenberger ........ r173968 | file
| 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
debug message I put in by accident. ........
2009-02-06 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
2009) | 7 lines Limit the addition of the Contact header in SIP
responses according to various SIP RFCs. (closes issue #13602)
Reported by: hjourdain Tested by: mnicholson ........
2009-02-06 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com>
* main/audiohook.c, apps/app_chanspy.c: Always detach and destroy
the whisper and barge audiohooks. Additionally also allow an
audiohook to be detached if it has not been attached. (closes
issue #14414) Reported by: bluecrow76
2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com>
* include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add
a common implementation of a scheduler context with a dedicated
thread. This commit expands the Asterisk scheduler API to include
a common implementation of a scheduler context being processed by
a dedicated thread. chan_iax2 has been updated to use this new
code. Also, as a result, this resolves some race conditions
related to the previous chan_iax2 scheduler handling. Related to
rev 171452 which resolved the same issues in 1.4. Code from
team/russell/sched_thread2 Review:
http://reviewboard.digium.com/r/129/
* main/manager.c: Resolve a memory leak that would occur on an
invalid channel given to Action: Status
2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com>
* configs/extensions.conf.sample: Update extensions.conf.sample to
be correct. In trunk, the only necessary change pointed out was
that the call to ChanIsAvail uses an option that has been
removed. For the 1.6.1 branch, however, it appears that the
sample file is badly in need of updating since there are |'s used
all over the place there. My tentative plan is just to copy
trunk's sample config file to those branches since the info there
is most up-to-date and should be correct for use in 1.6.1 Thanks
to macli in #asterisk-dev for bringing this up
* apps/app_voicemail.c: Properly set "seen" and "unseen" flags when
moving messages from the new to the old folder when using IMAP
for voicemail storage (closes issue #13905) Reported by: jaroth
Patches: foldermove_v2.patch uploaded by jaroth (license 50)
2009-02-05 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05
Feb 2009) | 12 lines Add new configuration option to make shared
IMAP mailboxes function as expected. The new option is
"imapvmshareid" which is an ID to tag multiple mailboxes using
the same IMAP storage location to function as one mailbox. This
allows all messages to be retrieved for any user in the group.
The patch alters the 'X-Asterisk-VM-Extension' header that is
responsible for matching voicemails for a given user. (closes
issue #13673) Reported by: howardwilkinson ........
2009-02-05 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
2009) | 12 lines Fix situations where queue members could be
autopaused unexpectedly Specifically, this patch prevents us from
autopausing members when we receive a busy or congestion frame
from them. (closes issue #14376) Reported by: fiddur Patches:
14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
........
2009-02-05 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com>
* res/res_config_sqlite.c: Change the first field, or we don't get
the necessary field separation.
2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com>
* /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu,
05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor
........
* /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu,
05 Feb 2009) | 25 lines Fix a problem where a channel pointer
becomes invalid due to masquerading or hanging up. app_mixmonitor
runs its own thread to monitor the channel's activity and write
the mixed audio to a file. Since this thread runs independently
of the channel, it is possible that the mixmonitor thread's
channel pointer will point to freed memory when the channel
either is masqueraded or hangs up (technically, both cases are
hangups, but we need to handle the cases slightly differently).
The solution for this is to employ a datastore, which has the
nice benefit of allowing us to hook into channel masquerades and
hangups and update our pointer as necessary. If this looks
familiar, this same technique is employed in app_chanspy.
app_chanspy is a bit more involved since it does a lot more
operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't
have, though. Since there is a thread race between the channel's
thread and the mixmonitor thread on a hangup, we em- ploy a
condition-and-boolean combination to ensure that the channel
thread finishes with our structure before the mixmonitor thread
attempts to free it. No crashes! (closes issue #14374) Reported
by: aragon Patches: 14374.patch uploaded by putnopvut (license
60) Tested by: aragon, putnopvut ........
* apps/app_queue.c: Fix some areas where the incorrect interface
was passed to ast_device_state I swear it feels like I already
did this once... (closes issue #14359) Reported by: francesco_r
2009-02-04 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com>
* res/res_jabber.c: Add XML documentation for the applications and
functions in res_jabber (closes issue #14405) Reported by: snuffy
Patches: xml_jabber.diff uploaded by snuffy (license 35)
2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com>
* channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with
IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a
call that was being transfered would never be handed off
correctly to the call ends because of how call numbers were
stored in a hash table. The hash table, "iax_peercallno_pvt",
storing all the current call numbers did not take into account
the complications associated with transferring a call, so a
separate hash table was required. This second hash table
"iax_transfercallno_pvt" handles calls being transfered, once the
call transfer is complete the call is removed from the transfer
hash table and added to the peer hash table resuming normal
operations. Addition functions were created to handle storing,
removing, and comparing items in the iax_transfercallno_pvt
table. The changes reverted in 116884 caused backwards
compatibility issues involving iax2 transfer with 1.6.0, 1.4, and
1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel
2009-02-04 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c, include/asterisk/features.h: Merged revisions
173211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
| 17 lines Parking attempts made to one end of a bridge no longer
will hang up due to a parking failure. Parking attempts made
using either one-touch, or doing either a blind or assisted
transfer to the parking extension now keep up the bridge instead
of hanging up the attempted parked party. Normal causes for the
parking attempt to fail includes the specific specified extension
(via PARKINGEXTEN) not being available or if all the parking
spaces are currently in use. To avoid having to reverse a
masquerade park_space_reserve was made to provide foresight if a
parking attempt will succeed and if so reserve the parking space.
(closes issue #13494) Reported by: mdu113 Reviewed by Russell:
http://reviewboard.digium.com/r/133/ ........
2009-02-04 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com>
* main/tcptls.c: When using a socket as a FILE *, the stdio
functions will sometimes try to do an fseek() on the stream,
which is an invalid operation for a socket. Turning off buffering
explicitly lets the stdio functions know they cannot do this,
thus avoiding a potential error. (closes issue #14400) Reported
by: fnordian Patches: tcptls.patch uploaded by fnordian (license
110)
2009-02-04 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com>
* /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
2009) | 3 lines Revert my previous change because it was stupid
........
* /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
matter, but it's needed. ........
* main/file.c: Fix a problem where file playback would cause fds to
remain open forever The problem came from the fact that a frame
read from a format interpreter was not freed. Adding a call to
ast_frfree fixed this. The explanation for why this caused the
problem is a bit complex, but here goes: There was a problem in
all versions of Asterisk where the embedded frame of a filestream
structure was referenced after the filestream was freed. This was
fixed by adding reference counting to the filestream structure.
The refcount would increase every time that a filestream's frame
pointer was pointing to an actual frame of data. When the frame
was freed, the refcount would decrease. Once the refcount reached
0, the filestream was freed, and as part of the operation, the
open files were closed as well. Thus it becomes more clear why a
missing ast_frfree would cause a reference leak and cause the
files to not be closed. You may ask then if there was a frame
leak before this patch. The answer to that is actually no! The
filestream code was "smart" enough to know that since the frame
we received came from a format interpreter, the frame had no
malloced data and thus didn't need to be freed. Now, however,
there is cleanup that needs to be done when we finish with the
frame, so we do need to call ast_frfree on the frame to be sure
that the refcount for the filestream is decremented
appropriately. (closes issue #14384) Reported by: fiddur Patches:
14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
putnopvut
2009-02-04 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, pbx/pbx_config.c: Ensure that commas placed in the
middle of extension character classes do not interfere with
correct parsing of the extension. Also, if an unterminated
character class DOES make its way into the pbx core (through some
other method), ensure that it does not crash Asterisk. (closes
issue #14362) Reported by: Nick_Lewis Patches:
20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
2009-02-03 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Broke up the large conditional blocks so
it is easy to see if a function is compiled.
2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/xml.c, include/asterisk/compiler.h, apps/app_stack.c,
include/asterisk/optional_api.h: 1. Make OS X compile cleanly
with app_stack. 2. Use curl to download sound files, as curl is
installed natively on OS X, whereas wget and fetch are not.
(closes issue #14332) Reported by: oej Tested by: Corydon76
* /, configs/extensions.conf.sample: Merged revisions 173070 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
| 5 lines Add warning to standard config, that globals may be
overridden by other dialplan configuration files. (closes issue
#14388) Reported by: macli ........
2009-02-02 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 173066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
| 2 lines Fix a feature inheritance bug I added after code review
........
2009-02-02 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com>
* main/manager.c, CHANGES: Reverting commit number 173028 as there
are some potential issues
* main/manager.c, CHANGES: Add a CLI command to log out a manager
user (closes issue #13877) Reported by: eliel Patches:
cli_manager_logout.patch.txt uploaded by eliel (license 64)
Tested by: eliel, putnopvut
2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 172962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009)
| 11 lines channels/chan_dahdi.c * Added doxygen comments to the
major dahdi structures. * Fixed PRI using an incorrect string
value if the extension delimiter is not present in the Dial()
function. * Fixed some uninitialized string variables on FXS
ports. configs/chan_dahdi.conf.sample * Updated some
documentation. These changes are already in trunk -r172400
........
2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/features.c, CHANGES,
include/asterisk/features.h: This reverts the changes I made for
11583; will reviewboard this before committing again... reopened
11583 until all Russell's issues are resolved.
2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample: Update the res_ldap.conf file with
a better working example. (closes issue #13861) Reported by:
scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by
blitzrage (license 10) Tested by: jcovert
2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/features.c, CHANGES,
include/asterisk/features.h: This change allows the disconnect
feature (as in "one-touch" in features.c) to be used within the
dial app, before a call is bridged. Many thanks to sobomax for
submitting this patch. Quoting from bug 11582: "So the goal of
the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function
is not well suited for this purpose as it uses much call bridge
specific data and doesn't separate a detection of feature from a
feature handler call. So a new function ast_feature_detect() has
been extracted off the ast_feature_interpret() function but
keeping the original logic intact except some insignificant
changes to locking. "Having created the ast_feature_detect()
function the possibility to use feature detection in almost any
place of the asterisk code. So a call to this function has been
added to wait_for_answer() function of app_dial.so module. This
code doesn't call the feature handler however and uses old call
leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature
currently is the only one supported during call setup as other
features as call parking and call transfer don't make much sense
during call setup. However if need in some of the features would
arise it is much easier to implement as the infrastructure
changes are already in place with this patch." I have cleaned up
the patch somewhat, and verified that the existing functionality
is not harmed, and that the new functionality works. Terry has
committed his stuff, and there were no conflicts (see 14274).
(closes issue #11583) Reported by: sobomax Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license
359) patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax
(license 359) 11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax
(license 359) 11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix a spelling mistake.
2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add a todo. I do need to really check what's
going on with this kill-the-user business ;-) Why do we suddenly
have two flags to set peer type?
* channels/chan_sip.c: Small formatting change
* channels/chan_sip.c: Add some well-needed improvements to the
wishlist in the code, so that we can close some bug reports.
2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: The CID lookup feature wasn't actually
working properly with dialog-info+xml supporting devices. The
devices (snoms, specifically) need to receive a SIP URI instead
of just an extension. This adds that functionality.
2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Blank argument crashes Asterisk (closes
issue #14377) Reported by: amorsen
* funcs/func_strings.c: Don't increment the loop, now that
incrementing is taken care of by the decoder function. (closes
issue #14363) Reported by: andrew53 Patches:
func_strings_filter.patch uploaded by andrew53 (license 519)
2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com>
* include/asterisk/channel.h: Fix redefinition of flag in channel.h
2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com>
* configs/features.conf.sample: Remove incorrect line from sample
config
* apps/app_dial.c, main/global_datastores.c, main/features.c,
include/asterisk/global_datastores.h, CHANGES,
configs/features.conf.sample: Merged revisions 172517 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
| 37 lines Fix feature inheritance with builtin features When
using builtin features like parking and transfers, the
AST_FEATURE_* flags would not be set correctly for all instances
when either performing a builtin attended transfer, or parking a
call and getting the timeout callback. Also, there was no way on
a per-call basis to specify what features someone should have on
picking up a parked call (since that doesn't involve the Dial()
command). There was a global option for setting whether or not
all users who pickup a parked call should have
AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
variable which can be set either in the dialplan or with setvar
in channels that support it. This variable can be set to any
combination of 't', 'k', 'w', and 'h' (case insensitive matching
of the equivalent dial options), to set what features should be
activated on this channel. The patch moves the setting of the
features datastores into the bridging code instead of app_dial to
help facilitate this. 2) adds global options parkedcallparking,
parkedcallhangup, and parkedcallrecording to be similar to the
parkedcalltransfers option for globally setting features. 3) has
builtin_atxfer call builtin_parkcall if being transfered to the
parking extension since tracking everything through multiple
masquerades, etc. is difficult and error-prone 4) attempts to fix
all cases of return calls from parking and completed builtin
transfers not having the correct permissions (closes issue
#14274) Reported by: aragon Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy
(license 396) Tested by: aragon, otherwiseguy Review
http://reviewboard.digium.com/r/138/ ........
2009-01-30 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com>
* funcs/func_aes.c: Parameter position reversed in documentation
* /, autoconf/ast_func_fork.m4, configure, main/app.c,
apps/app_rpt.c, main/asterisk.c: Merged revisions 172438 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
startup. Otherwise, if Asterisk runs as a non-root user and the
administrator does a 'restart now', Asterisk loses the ability to
set QOS on packets. (closes issue #14004) Reported by: nemo
Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76 ........
2009-01-29 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com>
* main/cli.c: Remove tabs from comment
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
channels/chan_dahdi.c * Added doxygen comments to the major dahdi
structures. * Fixed PRI and SS7 using an incorrect string value
if the extension delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as
long as the stripmsd option. * Fixed PRI not handling unknown
TON/NPI prefix letters correctly. * Fixed some uninitialized
string variables on FXS ports. configs/chan_dahdi.conf.sample *
Updated some documentation.
* include/asterisk/say.h: Fixed some doxygen comments
2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net>
* channels/chan_local.c: Revert two lines that was extra, but only
on fridays.
* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered
elsewhere" through app_queue with members in chan_local. Also,
implement a private cause code (as suggested by Tilghman). This
works with chan_sip, but doesn't propagate through chan_local.
2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample: Better document mode=multirow,
based upon a conversation with Jared.
2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script
is missing a couple of fields. closes issue #14339) Reported by:
fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur
(license 678)
2009-01-29 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample, CHANGES: Update documentation
* include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make
sure we set setvar= variables on outbound calls too, not only
inbound calls. - Also, change a function in app.c to return a
userful value instead of always returning 0. Patch by fnordian,
changed by Corydon76 and myself. This does not close the bug
report, as fnordian had an additional change we're still
discussing. (related to issue #14059) Reported by: fnordian
Patches: chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
* channels/chan_sip.c: Make sure register= line supports both port
and expiry at the same time. (closes issue #14185) Reported by:
Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by
Nick (license 657) Tested by: Nick_Lewis
* /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
lines Make sure that we always add the hangupcause headers. In
some cases, the owner was disconnected before we checked for the
cause. This patch implements a temporary storage in the pvt and
use that instead. The code is based on ideas from code from
Adomjan in issue #13385 (Add support for Reason: header) Thanks
to Klaus Darillion for testing! (closes issue #14294) related to
issue #13385 Reported by: klaus3000 and adomjan Patches:
bug14294b.diff uploaded by oej (license 306) Based on
20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
(license 487) Tested by: oej, klaus3000 ........
2009-01-28 22:52 +0000 [r172132] Steve Murphy <murf@digium.com>
* channels/chan_misdn.c: A further correction: cast the sizeof to
an int.
2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c: Fix how we skip fields (to avoid fields
which don't exist) when doing an UPDATE. (closes issue #14205)
Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt
uploaded by Corydon76 (license 14) Tested by: blitzrage
2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com>
* channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2
didn't like the \%ld and issued a warning, breaking my dev-mode
build. This fixes it.
* apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
main/features.c, include/asterisk/channel.h: Merged revisions
172030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
46 lines This patch fixes h-exten running misbehavior in
manager-redirected situations. What it does: 1. A new Flag value
is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop level
instead. 2. In the manager Redirect code, I set this flag on the
channel if the channel has a non-null pbx pointer. I did the same
for the second (chan2) channel, which gets run if name2 is set...
and the first succeeds. 3. I restored the ending of the cdr for
the pbx loop h-exten running code. Don't know why it was removed
in the first place. 4. The first attempt at the fix for this bug
was to place code directly in the async_goto routine, which was
called from a large number of places, and could affect a large
number of cases, so I tested that fix against a fair number of
transfer scenarios, both with and without the patch. In the
process, I saw that putting the fix in async_goto seemed not to
affect any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to its
current scope, and jmls tested both. In the process, tho, I saw
that blind xfers in one situation, when the one-touch blind-xfer
feature is used by the peer, we got strange h-exten behavior. So,
I inserted code to swap CDRs and to set the HANGUP_DONT field, to
get uniform behavior. 5. I added code to the bridge to obey the
HANGUP_DONT flag, skipping both publishing the bridge CDR, and
running the h-exten; they will be done at the pbx-loop (higher)
level instead. 6. I removed all the debug logs from the patch
before committing. 7. I moved the AUTOLOOP set/reset in the
h-exten code in res_features so it's only done if the h-exten is
going to be run. A very minor performance improvement, but
technically correct. (closes issue #14241) Reported by: jmls
Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
uploaded by murf (license 17) Tested by: murf, jmls ........
2009-01-28 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
Jan 2009) | 2 lines Clarify log message (suggested by manxpower
on #asterisk-dev) ........
2009-01-28 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net>
* CHANGES: Yep. Documentation is important.
* apps/app_queue.c: Add final part of previously committed work for
answered elsewhere in queue - the missing piece that started with
app_dial() earlier on. This is to avoid having the list and
counter of missed calls being touched by queue calls. Add the C
option to queue() and nothing will be logged on phones that
support the Reason: header on SIP cancel, like the SNOM phones.
* configs/sip.conf.sample: Add some more notes about device
matching.
* /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan
2009) | 2 lines Add a better explanation of the difference
between the device namespace and the dialplan for newbies.
........
2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com>
* funcs/func_aes.c: Fix some signedness problems in func_aes.c
2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Don't complain about lack of D-channels on
PTMP connections
2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com>
* funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and
AES_DECRYPT dialplan functions. (closes issue #14301) Reported
by: amorsen review: http://reviewboard.digium.com/r/128/
2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Merged revisions 171689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
2009) | 39 lines Fix devicestate problems for "always-on" agent
channels A revision to chan_agent attempted to "inherit" the
device state of the underlying channel in order to report the
device state of an agent channel more accurately. The problem
with the logic here is that it makes no sense to use this for
always-on agents. If the agent is logged in, then to the
underlying channel, the agent will always appear to be "in use,"
no matter if the agent is on a call or not. The reason is that to
the underlying channel, the channel is currently in use on a call
to the AgentLogin application. The most common cause that I found
for this issue to occur was for a SIP channel to be the
underlying channel type for an Agent channel. If the SIP phone
re-registers, then the registration will cause the device state
core to query the device state of the SIP channel. Since the SIP
channel is in use, the Agent channel would also inherit this
status. Once the agent channel was set to "in use" there was no
way that the device state could change on that channel unless the
agent logged out. The solution for this problem is a bit
different in 1.4 than it is in the other branches. In 1.4, there
will be a one-line fix to make sure that only callback agents
will inherit device state from their underlying channel type. For
the other branches of Asterisk, since callback support has been
removed, there is also no need for device state inheritance in
chan_agent, so I will simply be removing it from the code. In
addition, the 1.4 source is getting a new comment to help the
next person who edits chan_agent.c. I'm adding a comment that a
agent_pvt's loginchan field may be used to determine if the agent
is a callback agent or not. (closes issue #14173) Reported by:
nathan Patches: 14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez ........
* /, main/slinfactory.c: Merged revisions 171621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
2009) | 18 lines Prevent a crash from occurring when a jitter
buffer interpolated frame is removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to
determine the amount of data contained within the frame. In
certain cases, such as jitter buffer interpolated frames, the
frame would have a non-zero value for "samples" but have NULL
"data" This caused a problem when a memcpy call in
ast_slinfactory_read would attempt to access invalid memory. The
solution in use here is to never feed frames into the slinfactory
if they have NULL "data" (closes issue #13116) Reported by:
aragon Patches: 13116.diff uploaded by putnopvut (license 60)
........
* apps/app_queue.c: Fix queue crashes that would occur after the
calling channel was masqueraded. The data passed to the
end_bridge_callback was assumed to be data which was still
stack'd. The problem was that with some call features, attended
transfers in particular, a new bridge thread is started once the
feature completes, meaning that when the end_bridge_callback is
called, the end_bridge_callback_data was invalid. To fix this
problem, there are two measures taken 1. Instead of pointing to
stacked data, we now used heap-allocated data for passing to the
end_bridge_callback in app_queue 2. Since bridges can end
multiple times on a single logical call, we wait until the final
bridge is broken to actually set any queue variables. This is
accomplished through reference-counting and the use of an
end_bridge_callback_data_fixup function in app_queue.c (closes
issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
by putnopvut (license 60) Tested by: ccesario
2009-01-27 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there
are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis
Tested by: dbailey
2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Solving the same issue, but a bit
different in trunk... Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
lines Use the same branch tag in CANCEL as in INVITE Originally
putnopvut implemented some changes in revision 142079 that
according to the bug report seemed to have worked then, but
somehow fails now. I guess code, as humans, get old and forget
stuff. Anyway, this bug caused CANCEL not to work with picky
systems. Thanks Fredrik for pointing out where the bug in the SIP
messaging was. (closes issue #14346) Reported by: oej Patches:
bug14346.diff uploaded by oej (license 306) Tested by: oej
........
* channels/chan_sip.c: Moving generic setting to friends
* channels/chan_sip.c: Continue to move variables into the sip_cfg
structure to make them easier to handle in the future as a group
of settings for a group of devices. At some point, I want one
sip_cfg per domain handled, so we can have "group" settings.
* channels/chan_sip.c: Just moving around variable declarations so
that we have all globals in the same place. Default setting is
set before we activate the channel or at reloads, not where we
declare the variable.
* /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
lines Don't retransmit 401 on REGISTER requests when
alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000 ........
* main/channel.c: Add extensions and context on manager event when
new channel is created.
2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
| 6 lines Correctly track the hookstate (closes issue #13686)
Reported by: itiliti Patches: 20081013__bug13686.diff.txt
uploaded by Corydon76 (license 14) ........
2009-01-25 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: dont segfault when a MWI event occurs on
a line without a registered device
* configs/skinny.conf.sample: Make the sample skinny.conf work
(closes issue #14325) Reported by: DEA Patches:
skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com>
* /, apps/app_page.c: Merged revisions 170979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
2009) | 9 lines Resolve a logic error that was causing Page() to
crash when more than one channel was specified. (closes issue
#14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
uploaded by seanbright (license 71) Tested by: kc0bvu ........
2009-01-25 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe.
When the second part of this macro is written as 0[a] instead of
a[0], it will force a failure if the macro is used on a C++
object that overloads the [] operator.
* res/res_agi.c: Add a todo to finish the XML docs in this module
2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com>
* /, configs/res_odbc.conf.sample: Merged revisions 170836 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009)
| 2 lines Remove superfluous implementation note (closes issue
#14319) ........
2009-01-23 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com>
* doc/tex/Makefile: Fix asterisk.pdf generation if branch name has
an underscore in it.
2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com>
* doc/tex/Makefile: Don't blow up if a branch name has an
underscore in it
2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com>
* /, configs/res_odbc.conf.sample: Merged revisions 170719 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
2009) | 8 lines Add notes to the idlecheck explanation in
res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
klaus3000 (license 65) ........
* /, contrib/i18n.testsuite.conf: Merged revisions 170671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
deprecated syntax * Convert Wait,1 to Wait(1) * Convert
SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
priorities beyond the first Also added test for Chinese numbers,
too. (closes issue #14320) Reported by: dant Patches:
i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
670) ........
2009-01-23 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 170648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
lines When a channel is answered make sure any indications
currently playing stop. Usually the phone would do this but if
the channel was already answered then they are being generated by
Asterisk and we darn well need to stop them. (closes issue
#14249) Reported by: RadicAlish ........
2009-01-23 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
Jan 2009) | 2 lines Additions to AST-2009-001 ........
2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 170568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
lines When a call is forwarded stop any active indications. The
new channel will provide an indication, if need be, itself.
(closes issue #14310) Reported by: RadicAlish ........
* /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
lines Use the on hold flag to see if the call is on hold or not.
It is possible that our address for them will still be valid even
though they are on hold. (closes issue #14295) Reported by:
klaus3000 ........
2009-01-23 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_h323.c: let's use SENTINEL where needed
2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Reset the ast_str used for escape
substitution. We need to do this since it is a thread local
variable that may contain the value of a previous substitution.
(closes issue #14312) Reported by: pj
2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: We should not do restart messages if we're
in PTMP mode
2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Dont clear the display of skinny phones
when not needed. (closes issue #13182) Reported by: pj Patches:
2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak
(license 7) Tested by: mvanbaak, pj
2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: MWI messages included in CID spill was not
being properly handled and prevented the call from being
processed (issue #14313) Reported by: seandarcy Tested by:
dbailey
2009-01-23 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 170392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
2009) | 28 lines Fix broken call pickup There was a subtle change
in ast_do_masquerade which resulted in failed attempts to pickup
calls. The problem was that the value of the AST_FLAG_OUTGOING
flag was copied from the clone to the original channel. In the
case of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting to
execute the pickup. Because this flag was not set, when ast_read
came across an answer frame, it ignored it. The result of this
was that the calling channel was never properly answered. This
fix changes the behavior in ast_do_masquerade to set the flags on
the original channel to the union of the flags on the clone
channel. This way, if the AST_FLAG_OUTGOING flag is set on either
of the two channels involved in the masquerade, the resulting
channel will have the flag set as well. (closes issue #14206)
Reported by: francesco_r Patches: 14206.patch uploaded by
putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
........
2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Make sure we don't set the channel to be
inalarm for a D-channel drop on PTMP connections
2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com>
* main/abstract_jb.c: Create logfile safely. (closes issue #14160)
Reported by: tzafrir Patches: 20090104__bug14160.diff.txt
uploaded by Corydon76 (license 14)
2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 170239 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
lines Don't crash if RTCP is not enabled on an RTP structure but
statistics are output. (closes issue #14234) Reported by: jcovert
Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
2009-01-22 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com>
* /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
| 6 lines Allow global variables after substitution to be as long
as other variables. (closes issue #14263) Reported by: markd
Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
(license 14) ........
2009-01-22 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
lines If we are unable to request a DAHDI pseudo channel and we
are using the user introduction without review option make sure
it gets unset so other code does not blindly assume a DAHDI
pseudo channel exists. (closes issue #14282) Reported by:
cheesegrits ........
2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: change VMWI to
use new DAHDI_VMWI ioctl call. Change configure script to detect
the new ioctl call data structure. (issue #14104) Reported by:
alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded
by dbailey (license ) Tested by: alecdavis, dbailey
2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com>
* main/pbx.c, /: Merged revisions 170050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
lines Do a string comparison instead of pointer comparison since
some people specify the context they are actually in as an
argument to get around some funkiness. (closes issue #14011)
Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
(license 665) ........
* apps/app_parkandannounce.c: Clear the autoloop flag when parsing
and setting the context/extension/priority to go back to. When
the channel executes a PBX again we want it to start out at the
point we explicitly say and at that point it will not yet be
doing autoloop. (closes issue #14304) Reported by: jcovert
2009-01-22 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: * Adjust some conditionals to balance
curly braces. * Other minor changes.
2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 169943 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
| 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
wanted to ask is whether autoconf detected a static initializer
value. This fixes rwlocks on all such platforms (mainly, Mac OS
X). (closes issue #13767) Reported by: jcovert Patches:
20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
Tested by: jcovert, Corydon76 ........
2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Whitespace changes only
2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com>
* main/pbx.c, /: Merged revisions 169867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
lines Read lock the contexts to maintain the locking order when
we are notified that the state of a device has changed. (closes
issue #13839) Reported by: mcallist ........
2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Test commit for test issue #14303
* main/say.c: Fix a crash when saying certain numbers in Chinese
This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0. This
also removes some places where a "zero" is spoken when it should
not be. (closes issue #14291) Reported by: dant Patches:
say.c-14291.diff uploaded by dant (license 670) Tested by: dant
2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info>
* doc/tex/extensions.tex: remove duplicated sentence.
2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Further fix some oddities in sip show users
and sip show peers logic ccesario on IRC pointed out that his sip
peers were not displayed properly when he would issue the command
"sip show peers." The problem was that the onlymatchonip field
was used to determine if the endpoint was a "peer" or "user." The
tricky part is that a "friend" is supposed to be treated as both
a "user" and a "peer" but the logic would not allow "friends" to
show up as "peers" since onlymatchonip was set to FALSE for
friends. I have modified the sip_peer structure to more
explicitly keep track of what type endpoint it is so that the
various manager and CLI commands will display the expected
information Reported by ccesario via IRC Tested by ccesario
2009-01-21 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 169722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
| 8 lines Extra NULLs in the output cause some terminal types to
abort in the middle of a color code, causing terminal weirdness.
(closes issue #14130) Reported by: coolmig Patches:
20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, coolmig ........
2009-01-21 17:21 +0000 [r169673] Steve Murphy <murf@digium.com>
* utils/refcounter.c: This patch corrects a segfault reported in
14289, due to a null ptr being refd. Yes, seanbright is right in
the bug comments, that is the fix. Sorry for this oversight; I
guess my personal usage didn't have this happen! murf (closes
issue #14289) Reported by: jamesgolovich
2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant <russell@digium.com>
* /: Remove properties that erroneously got merged into trunk
* main/tcptls.c: Fix a regression in TCP support. This patch fixes
a problem that caused chan_sip to think that every open TCP
session was to a remote address of 0.0.0.0:0. (closes issue
#14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt
uploaded by jamesgolovich (license 176)
2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix device state parsing issues for channel
names with multiple slashes The fix being applied is a bit
different for trunk and the 1.6.X branches. For trunk, we only
wish to strip off the characters beyond the second slash if the
channel is a Local channel (i.e. we are removing the /n from the
device name). Other channel technologies with multiple slashes
(e.g. DAHDI) need the information after the second slash in order
to get the proper device state information. In addition to this
fix, the 1.6.X branches are receiving a much more important fix
as well. The problem in 1.6.X is that the member's device name
was being directly changed instead of having a copy changed. This
meant that we would strip off the second slash and trailing
characters and then leave the member's device name like that
permanently thereafter. (closes issue #14014) Reported by:
kebl0155 Patches: 14014_number2.patch uploaded by putnopvut
(license 60) Tested by: kebl0155
* apps/app_queue.c: Use the default timeout for a queue instead of
-1 (closes issue #14272) Reported by: timking
* /, channels/chan_sip.c: Convert the character pointers in a
sip_request to be pointer offsets When an ast_str expands to hold
more data, any pointers that were pointing to the data prior to
the expansion will be pointing at invalid memory. This change
makes such pointers used in chan_sip.c instead be offsets from
the beginning of the string so that the same math may be applied
no matter where in memory the string resides. To help ease this
transition, a macro called REQ_OFFSET_TO_STR has been added to
chan_sip.c so that given a sip_request and an offset, the string
at that offset is returned. (closes issue #14220) Reported by:
riksta Tested by: putnopvut Review
http://reviewboard.digium.com/r/126/
2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com>
* main/features.c: Make a proper builtin attended transfer to
parking work This is an ugly hack from 1.4 that allows the
timeout callback from a parked call to use the right channel name
for the callback when the park is done with a builtin attended
transfer (that isn't completed early). This hasn't ever worked in
trunk and no one has complained yet, so eh.
* /, main/features.c: Merged revisions 169485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
| 6 lines Don't play audio to the channel if we've masqueraded
(closes issue #14066) Reported by: bluefox Tested by:
otherwiseguy, bluefox ........
2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/res_odbc.h, funcs/func_odbc.c,
include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is
*not* part of the ast_str API, it's part of the ast_odbc API and
just happens to use an ast_str as the buffer; move all of it to
res_odbc.c and res_odbc.h, renaming appropriately along the way
fix some minor coding style issues in strings.h and add some
attribute_pure annotations to functions in the ast_str API
2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: fix assignment in swapmode plug. Spotted and fix
provided by ys (closes issue #14129) Reported by: ys Tested by:
ys
* channels/chan_skinny.c: Redo the event-based MWI in chan_skinny.
Dan saw regular segfaults with the old implementation and rewrote
it to make it really eventbased. I altered it to be trunk
compatible and wedhorn gave some feedback and ideas how to make
it even better. (closes issue #13821) Reported by: DEA Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by:
mvanbaak, DEA "no probs by me" from wedhorn
2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /, apps/app_userevent.c: Merged revisions 169364
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
| 4 lines Truncate userevents at the end of a line, when the
command exceeds the buffer. (closes issue #14278) Reported by:
fnordian ........
2009-01-19 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: Make asterisk compile on non-amd64 versions of
OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD
and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when
HW_PHYSMEM64 is not available. (closes issue #14129) Reported by:
ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak
(license 7) Tested by: mvanbaak, jtodd
2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Get rid of magic number and replace with
DAHDI_VMWI_NUMBER_MASK when determining the number of messages
pending for MWI call
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
enhanced MWI generation to take advantage of new dahdi line
reversal MWI ability. (closes issue #14104) Reported by:
alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey
(license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis
(license 585) Tested by: alecdavis, dbailey
2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c, /: Merged revisions 169210 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon,
19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a
potential NULL pointer dereference Move the check for if both
channels on a local_pvt have generators to below where p->chan is
checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL. (closes issue #14189) Reported by:
sascha Patches: 14189.patch uploaded by putnopvut (license 60)
Tested by: sascha ........
2009-01-17 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
discriminator for when ring pulse alert signal is used to preface
MWI spills This prevents the situation when MWI messages are
added to caller ID spills causing the channel to be hung up
2009-01-17 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com>
* pbx/pbx_dundi.c: Change intializer types. Found while working on
asterisk-cpp. I have a new favorite error message from g++:
pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
initializers not supported I like it when compilers are
apologetic.
2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com>
* main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix
qualify for TCP peer (closes issue #14192) Reported by:
pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
jamesgolovich (license 176) Tested by: jamesgolovich
* channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when
outboundproxy used (closes issue #14233) Reported by: chris-mac
Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich
(license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy
2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
2009) | 18 lines Account for possible NULL pointer when we
receive a 408 in response to a REGISTER It may be that by the
time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the
corresponding sip_pvt has gone away. This situation was handled
properly for a 200 OK response, but the 408 case assumed that the
sip_registry struct was non-NULL, thus potentially causing a
crash This commit fixes this assumption and prints out a message
to the console if we should receive a late 408 response to a
REGISTER (closes issue #14211) Reported by: aborghi Patches:
14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
........
2009-01-16 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 168716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
| 12 lines Convert call to park_call_full to
masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
return value, we need to use masqueraded parking, otherwise we
will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash. (closes issue
#14215) Reported by: waverly360 Tested by: otherwiseguy (closes
issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
2009-01-16 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com>
* res/res_timing_timerfd.c: Fix a logic error that occur when using
the timerfd interface This sequence of events posed a problem
timerfd_timer_open timerfd_timer_enable_continuous
timerfd_timer_set_rate timerfd_timer_disable_continuous The
reason was that the timing module was written under the
assumption that timerfd_timer_set_rate would not be called
between enabling and disabling continuous mode. What happened in
this situation was that timerfd_timer_enable_continuous saved off
our previously set timer (in this situation a 0 timer, meaning it
never runs out). Then timerfd_timer_disable_continuous would
restore this 0 timer, even though it logically should set the
timer to be whatever was set in timerfd_timer_set_rate. Now the
behavior in timerfd_timer_set_rate is to overwrite the saved
timer that may or may not have been set in
timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called,
this will not harm the operation. Thanks to Terry Wilson for
discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c:
Merged revisions 168828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
| 6 lines Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475) Reported by: chappell Patches:
vm_multilang.patch uploaded by chappell (license 8) ........
2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com>
* CHANGES: Fix a spelling mistake.
* channels/chan_misdn.c: build in dev mode
2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com>
* res/ael/pval.c, /: Merged revisions 168745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
14 lines This patch fixes a problem where a goto (or jump, in
this case) fails a consistency check because it can't find a
matching extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern into
a regex and uses the regex code to determine the match. I tested
using the AEL code the user supplied, and now, the consistency
check passes. (closes issue #14141) Reported by: dimas ........
* main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch
allows null args in ast_expr2 func calls, and fixes commas being
converted to pipes, which was 1.4 type stuff. If the user says
count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it
won't complain about the empty arg (c,,...) and fabled's patch
won't let it swap the commas for pipes. Ran it thru my dialplan
and no complaints. (closes issue #14169) Reported by: fabled
Patches: function-argument-separator-fix.diff uploaded by fabled
(license 448)
2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c,
cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c,
apps/app_voicemail.c: remove the PBX_ODBC logic from the
configure script, and add GENERIC_ODCB logic that includes
copying the relevant LIB and INCLUDE data from either UnixODBC or
iODBC, based on which was found; if both were found, prefer
UnixODBC this stops modules from being linked against both sets
of libraries on systems that have both installed
2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add missing brace
* channels/chan_sip.c: Fix the compactheaders option in sip.conf
* channels/chan_sip.c: Remove an unneeded condition for line
addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the
sip_request structure had a statically allocated buffer to hold
the text of the request. There was a check in the add_line
function to not attempt to write the line into the buffer if we
did not have room for it. In trunk and Asterisk versions starting
with 1.6.1, an expandable ast_str structure is used to hold the
text. Since it may grow to fit an arbitrarily sized string, this
check in add_line is no longer valid. I found this oddity while
attempting to fix issue #14220; however, I do not believe that
this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be
printed had this condition been satisfied.
2009-01-15 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net>
* /, configs/extconfig.conf.sample: Merged revisions 168721 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
lines Meetme actually has realtime but wasn't documented ........
2009-01-15 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/strings.h: Resolve issue with negative vs
non-negative length parameters. (closes issue #14245) Reported
by: dveiga
2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Make sure that we have the same terminology
in sip.conf.sample and the source code warning. Thanks Nick Lewis
for pointing this out in the bug tracker.
* configs/sip.conf.sample: Clarify some misunderstandings and make
it even more clear that you can refer to a peer in the register=
line.
2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com>
* apps/app_meetme.c: Add a missing unlock and properly handle the
'maxusers' setting on MeetMe conferences. We were using the 'user
number' field to compare against the maximum allowed users, which
works assuming users with lower user numbers didn't leave the
conference. (closes issue #14117) Reported by: sergedevorop
Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
(license 71) Tested by: sergedevorop
2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net>
* CREDITS, CHANGES: Related to issue #14246 Update changes for
SIPRemoveHeader()
* channels/chan_sip.c: Add capability to remove added SIP headers
*before* INVITE is generated. (closes issue #14246) Reported by:
klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt
uploaded by klaus3000 (license 65)
* apps/app_queue.c: Add support for setting the Reason header when
cancelling a call in the queue because someone else answered.
Previously, only dial() was supported. EDV-102
2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
2009) | 16 lines Fix some crashes from bad datastore handling in
app_queue.c * The queue_transfer_fixup function was searching for
and removing the datastore from the incorrect channel, so this
was fixed. * Most datastore operations regarding the
queue_transfer datastore were being done without the channel
locked, so proper channel locking was added, too. (closes issue
#14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
putnopvut (license 60) Tested by: ZX81, festr ........
2009-01-14 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com>
* main/cli.c: Don't crash when typing 'core set verbose' or 'core
set debug' by themselves. (closes issue #14219) Reported by:
jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded
by jamesgolovich (license 176)
2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn/isdn_lib.c: Merged revisions 168622 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
| 4 lines * Fixed create_process() allocation of process ID
values. The allocated process IDs could overflow their respective
NT and TE fields. Affects outgoing calls. ........
2009-01-14 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: This fixes a problem where MWI FSK spills
were being injected onto off hook fxs lines. (closes issue
#14143) Reported by: alecdavis Patches:
chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested
by: alecdavis
2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com>
* /, contrib/scripts/autosupport: Merged revisions 168614 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
2009) | 9 lines Update autosupport script to supply info for both
Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
and trunk instead of zttest. (closes issue #14132) Reported by:
dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
by dsedivec (license 638) ........
2009-01-14 20:51 +0000 [r168613] Steve Murphy <murf@digium.com>
* /, apps/app_page.c: Merged revisions 168608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
line app_page was failing to compile in dev-mode on my gcc-4.2.4
system. This change gets rid of the warning. ........
2009-01-14 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Restore the "sip show users" and "sip show
user" CLI commands (closes issue #14180) Reported by: amorsen
Patches: sip_show_users_161v3.diff uploaded by putnopvut (license
60) Tested by: blitzrage, amorsen
2009-01-14 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info>
* main/asterisk.c: Fix compilation on FreeBSD and OSX This started
as work to fix the 'core show sysinfo' CLI command but while
working on it oej pointed out that read_credentials did not
compile neither. So while being there, fix that as well. Thanks
for all the testing oej! (closes issue #14129) Reported by: ys
Tested by: oej, mvanbaak
2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c, /: Merged revisions 168603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
| 7 lines Don't read into a buffer without first checking if a
value is beyond the end. (closes issue #13600) Reported by: atis
Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
(license 14) Tested by: atis ........
* channels/chan_misdn.c: Mostly spacing changes; no functionality
change at all.
2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com>
* /, apps/app_page.c: Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
| 20 lines Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded
to 128. Proper bounds checking was not in place to enforce this
limit, so if more than 128 extensions were passed to the Page()
app, Asterisk would crash. This patch instead dynamically
allocates memory for the ast_dial structures and removes the
(non-functional) arbitrary limit. This issue would have special
importance to anyone who is dynamically creating the argument
passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external
interface. The patch posted by a_villacis was slightly modified
for some coding guidelines and other cleanups. Thanks,
a_villacis! (closes issue #14217) Reported by: a_villacis
Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
uploaded by a (license 660) Tested by: otherwiseguy ........
2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com>
* channels/chan_misdn.c: Janitor patch for chan_misdn (make channel
variable access safe) (closes issue #12887) Reported by: pputman
Patches: chan_misdn_threadsafe.patch uploaded by pputman (license
81)
2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com>
* res/res_http_post.c: Fully overwrite a same-named file when
uploading (closes issue #14190) Reported by: timking
* Makefile, include/asterisk/options.h, main/asterisk.c: Add option
to hide console connect messages (closes issue #14222) Reported
by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded
by jamesgolovich (license 176) Tested by: otherwiseguy
2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Clarify a message that app_queue prints and
change to a debug-level message The "No one is answering..."
verbose message contained 3 numbers that were not explained in
any way to whoever was viewing the message. It is more helpful
now since the message explains what the numbers mean. Also, the
message has been downgraded to "DEBUG" level. (closes issue
#14172) Reported by: caio1982 Patches: queue_answering_debug.diff
uploaded by caio1982 (license 22)
2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
| 7 lines Don't pass a value with a side effect to a macro
(closes issue #14176) Reported by: paraeco Patches:
chan_sip.c.diff uploaded by paraeco (license 658) ........
2009-01-13 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow
specifying a port number in the user portion of a register =>
line in sip.conf With this commit, a register => line in sip.conf
may contain a port number in the "user" section of the line.
Please see CHANGES and sip.conf.sample for more details regarding
this. (closes issue #14198) Reported by: Nick_Lewis Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com>
* channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /,
include/asterisk/indications.h, apps/app_readexten.c,
apps/app_disa.c, include/asterisk/channel.h, main/indications.c,
main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
funcs/func_channel.c, main/app.c, res/snmp/agent.c,
res/res_indications.c: Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
| 2 lines Revert unnecessary indications API change from rev
122314 ........
2009-01-13 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
| 6 lines If either conditional is NULL, don't try copying it.
(closes issue #14226) Reported by: caspy Patches:
20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
........
2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* main/taskprocessor.c: correct a CLI description
2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31
Dec 2008) | 5 lines Repeat attempts to write when we receive
-EAGAIN from the driver, as detailed in the ALSA sample code (see
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
Reported by: Jerry Geis (via the -users list) Fixed by: me
(license 14) ........
2009-01-12 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com>
* main/srv.c: bump the verbosity of a message in srv.c up by one.
It used to be at this level prior to a large patch merge which
converted ast_verbose calls to ast_verb (closes issue #14221)
Reported by: jcovert Patches: srv.c.patch uploaded by jcovert
(license 551)
2009-01-12 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/app.c: Some platforms (notably, the BSDs) have a more
efficient implementation called closefrom(3).
2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com>
* /, res/res_agi.c: Merged revisions 168516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
| 5 lines (closes issue #13881) Reported by: hoowa Update the app
CDR field for AGI commands that are not executing an application
via "exec". ........
* /, channels/chan_agent.c: Merged revisions 168507 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12
Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG
Tested by: denisgalvao This gits rid of the notion of an
owning_app allowing the request and hangup to be initiated by
different threads. Originating from an active agent channel
requires this. The implementation primarily changes __login_exec
to wait on a condition variable rather than a lock. Review:
http://reviewboard.digium.com/r/35/ ........
2009-01-12 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net>
* apps/app_minivm.c: Better to use the proper app name
2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Merged revisions 168482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan
2009) | 5 lines I am reverting the fix made in revision 168128
(and its upward merges) after being contacted by Olle Johansson
and being shown how this fix is incorrect. Thanks to Olle for
clearing this up for me. ........
2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com>
* /, configs/indications.conf.sample: Merged revisions 168480 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
| 2 lines s/ringdance/ringcadence/ for Bulgaria ........
2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net>
* main/asterisk.c: Don't include swap.h unless we have swapctl
2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for
reconstructing a field value.
2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com>
* /, sounds/Makefile: Merged revisions 168267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
2009) | 1 line update to use new sound file packages that include
license files ........
2009-01-09 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Spacing change
2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/sip_nat_settings (added), CHANGES: Add a script
to find out the correct settings for Asterisk behind NAT (closes
issue #13065) Reported by: tzafrir Patches: sip_nat_settings
uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by
mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and
moi
2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09
Jan 2009) | 2 lines Make this compile for mvanbaak ........
2009-01-09 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
2009) | 13 lines Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
checking the topmost Via to determine where to send the response.
Adding check_via calls to those request handlers solves this.
(closes issue #13071) Reported by: baron Patches: check_via.patch
uploaded by baron (license 531) Tested by: baron ........
2009-01-09 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09
Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 *
Miscellaneous doxygen comments added. ........
2009-01-09 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not
exist (closes issue #14203) Reported by: jamesgolovich Patches:
asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
(license 176)
2009-01-09 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c, include/asterisk/strings.h: When using ast_str
with a non-ast_str-enabled API, we need to update the buffer or
otherwise, we cannot use ast_str_strlen().
2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com>
* main/logger.c: Added a comment to logger.c about where to put
includes
* main/logger.c: Use ast_safe_system() in logger.c instead of
system() (closes issue #14194) Reported by: pabelanger
2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com>
* apps/app_originate.c: Set ORIGINATE_STATUS instead of
OUTGOING_STATUS to match the documentation
* apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN
and MACRO_CONTEXT will be set
2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com>
* /, res/res_agi.c: Merged revisions 167840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
| 6 lines Don't truncate database results at 255 chars. (closes
issue #14069) Reported by: evandro Patches:
20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
........
2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Revert chan_sip changes which were
accidentally committed in revision 167792
2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com>
* apps/app_minivm.c: Fix variables to comply with documentation
changes
* apps/app_minivm.c: Textual changes, consistency in status
variable naming, and other minor bugs. (closes issue #13943)
Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded
by Marquis (license 32)
2009-01-08 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average
talk time for a queue This patch adds the functionality to
app_queue of calculating the average amount of time that channels
are bridged for a queue. The algorithm used to calculate the
average is the same exponential average currently used to
calculate the average holdtime. See the CHANGES file to see the
methods you may use to view this information. (closes issue
#13960) Reported by: coolmig Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, CHANGES: Convert dialplan application
DAHDISendCallreroutingFacility to use commas. (closes issue
#13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded
by eliel (license 64)
2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
2009) | 1 line remove an unnecessary argument to queue_request()
........
* channels/chan_sip.c: Merged revisions 167620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
2009) | 5 lines When a SIP request or response arrives for a
dialog with an associated Asterisk channel, and the lock on that
channel cannot be obtained because it is held by another thread,
instead of dropping the request/response, queue it for later
processing when the channel lock becomes available.
http://reviewboard.digium.com/r/123/ ........
2009-01-08 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql: Oops... fix the fieldname I
changed yesterday to be right.
2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com>
* /, main/file.c: Merged revisions 167566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
| 2 lines Fix the last couple of places where free() was
improperly used directly. ........
* /, main/file.c: Merged revisions 167554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
| 2 lines Don't fclose() the file early, the filestream
destructor will handle it. ........
* /, main/file.c: Merged revisions 167545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
| 2 lines Only try to close the file if one was actually opened
........
* /, main/file.c: Merged revisions 167541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
| 4 lines Don't use free() directly. This caused a crash since
ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
#asterisk-dev ........
2009-01-07 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com>
* apps/app_followme.c: Answer the channel if it has not already
been answered and we've already found a valid profile for
followme. (closes issue #14140) Reported by: dimas Patches:
14140.patch uploaded by dimas
2009-01-07 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com>
* configs/queues.conf.sample: Update queues.conf.sample
documentation. Update the queues.conf.sample documentation to
mention that you need to preload chan_local.so as well if you
plan on using Local channels for queue members, and you're
preloading pbx_config.so. (closes issue #14179) Reported by:
CrashHD Tested by: CrashHD
2009-01-07 17:35 +0000 [r167442] Russell Bryant <russell@digium.com>
* /, main/indications.c: Merged revisions 167432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
| 4 lines Treat an empty string the same way as a NULL country
argument. In passing, simplify the handling of returning a
default tone zone. ........
2009-01-07 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected
during mwi spill. Correct logic error in handling events when
sending mwi spill (closes issue #14143) Reported by: alecdavis
Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by
dbailey
2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/sip-friends.sql: Update the sip-friends.sql file
to use the non-deprecated 'defaultname' instead of 'username' and
remove an extra comma that would cause the script to fail as-is
2009-01-06 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com>
* /, main/db.c: Merged revisions 167299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
2009) | 8 lines Use the correct variable when creating the format
string (closes issue #14177) Reported by: nic_bellamy Patches:
asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
(license 299) ........
2009-01-06 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
(Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
Jan 2009) | 2 lines Security fix AST-2009-001. ........
................
2009-01-05 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
2009) | 41 lines A couple of changes to T.38 SDP attribute
handling There are some boolean attributes for T.38 such as
T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we should treat
these as a "true" value. The current code, however, was requiring
a 1 or 0 as the value of the attribute in order to parse it. This
is due to the fact that there are some T.38 endpoints and
gateways that also transmit this information incorrectly. This
patch follows the "be liberal in what you accept and strict in
what you send" philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending information as
it is supposed to be sent. It was also discovered that a
particular type of T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
it used T38MaxDatagram and T38FaxMaxRate respectively. We now
will properly accept these attributes as well. Note that there
are a lot of patches cited in the below commit message template.
This is because the person who submitted these patches is an
awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
issue #13976) Reported by: linulin Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
(license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
(license 648) Tested by: arcivanov ........
2009-01-05 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com>
* UPGRADE-1.6.txt: More clearly explain that quote marks are no
longer necessary. (closes issue #13718) Reported by: davidw
Patches: 20081020__bug13718.diff.txt uploaded by Corydon76
(license 14) Tested by: blitzrage
2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com>
* main/asterisk.c: When parsing environment variable
ASTERISK_PROMPT, make sure to proceed to the next character when
a non format specifier is used (no %). Otherwise, the while loop
looking for the null byte will never exit.
2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com>
* doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README:
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a
couple places and fix a few typos I found in the
CODING_GUIDELINES.
2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com>
* main/xmldoc.c: Don't forget to free typename
2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Change some incorrect syntax for pri set
debug and correct an off-by-one error in ss7 set debug command
2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com>
* main/ast_expr2.h, main/ast_expr2.c: That was weird...
* channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c:
Merged revisions 166953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
| 5 lines Also inherit the musiconhold class. (Closes #14153)
Reported by: Jerry Geis, via the users list. Patch by: me
(license 14) ........
2008-12-30 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c, doc/sip-retransmit.txt,
doc/tex/phoneprov.tex, res/res_http_post.c,
phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some
svn:keywords
2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with
the removal of AST_PBX_KEEPALIVE When placing a call to a queue
which ran a gosub on the member's channel, Asterisk would crash
every time, stemming from the fact that the member's channel was
being hung up unexpectedly when the Gosub completed. The
necessary change was pretty much copied and pasted from
app_dial's similar changes made last week. I also took the
opportunity to change a LOG_DEBUG message in app_dial to use
ast_debug. I am guessing this was due to a direct merge from 1.4
that was not corrected to use trunk's preferred syntax.
2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com>
* funcs/func_audiohookinherit.c: Fix a typo in the XML
documentation of the AUDIOHOOK_INHERIT dialplan function.
2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com>
* /, channels/misdn_config.c: Merged revisions 166772 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28
Dec 2008) | 4 lines Use strncat() instead of an sprintf() in
which source and target buffers overlap
http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
........
2008-12-24 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I
believe 26.2.2 is what was meant: Note that in the SIPS URI
scheme, transport is independent of TLS, and thus
"sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Allow semicolons and extended characters in
user-specified SIP headers. (closes issue #14110) Reported by:
gork Patches: 20081222__bug14110__2.diff.txt uploaded by
Corydon76 (license 14) Tested by: gork, putnopvut
2008-12-23 18:13 +0000 [r166665] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/pbx.c, /, main/features.c,
apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c,
include/asterisk/features.h: Merged revisions 166093 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4 In
order to merge this 1.4 patch into trunk, I had to resolve some
conflicts and wait for Russell to make some changes to res_agi. I
re-ran all the tests; 39 calls in all, and made fairly careful
notes and comparisons: I don't want this to blow up some aspect
of asterisk; I completely removed the KEEPALIVE from the pbx.h
decls. The first 3 scenarios involving feature park; feature xfer
to 700; hookflash park to Park() app call all behave the same,
don't appear to leave hung channels, and no crashes. ........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
131 lines This merges the masqpark branch into 1.4 These changes
eliminate the need for (and use of) the KEEPALIVE return code in
res_features.c; There are other places that use this result code
for similar purposes at a higher level, these appear to be left
alone in 1.4, but attacked in trunk. The reason these changes are
being made in 1.4, is that parking ends a channel's life, in some
situations, and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing the
channel pointer, which could lead to memory corruption and
crashes. Calling the masq_park function eliminates this danger in
higher levels. A series of previous commits have replaced some
parking calls with masq_park, but this patch puts them ALL to
rest, (except one, purposely left alone because a masquerade is
done anyway), and gets rid of the code that tests the KEEPALIVE
result, and the NOHANGUP_PEER result codes. While bug 13820
inspired this work, this patch does not solve all the problems
mentioned there. I have tested this patch (again) to make sure I
have not introduced regressions. Crashes that occurred when a
parked party hung up while the parking party was listening to the
numbers of the parking stall being assigned, is eliminated. These
are the cases where parking code may be activated: 1. Feature one
touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
hookflash xfer to 700) 4. Run Park via manager. The interesting
testing cases for parking are: I. A calls B, A parks B a. B hangs
up while A is getting the numbers announced. b. B hangs up after
A gets the announcement, but before the parking time expires c. B
waits, time expires, A is redialed, A answers, B and A are
connected, after which, B hangs up. d. C picks up B while still
in parking lot. II. A calls B, B parks A a. A hangs up while B is
getting the numbers announced. b. A hangs up after B gets the
announcement, but before the parking time expires c. A waits,
time expires, B is redialed, B answers, A and B are connected,
after which, A hangs up. d. C picks up A while still in parking
lot. Testing this throroughly involves acting all the
permutations of I and II, in situations 1,2,3, and 4. Since I
added a few more changes (ALL references to KEEPALIVE in the
bridge code eliimated (I missed one earlier), I retested most of
the above cases, and no crashes. H-extension weirdness. Current
h-extension execution is not completely correct for several of
the cases. For the case where A calls B, and A parks B, the 'h'
exten is run on A's channel as soon as the park is accomplished.
This is expected behavior. But when A calls B, and B parks A,
this will be current behavior: After B parks A, B is hung up by
the system, and the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
and B is DAHDI/2, the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
those relating to Channel A. And, in the case where A is
reconnected to B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten will be run at all.
In the case where C picks up A from the parking lot, when either
A or C hang up, the h-exten will be run for the C channel. CDR's
are a separate issue, and not addressed here. As to WHY this
strange behavior occurs, the answer lies in the procedure
followed to accomplish handing over the channel to the parking
manager thread. This procedure is called masquerading. In the
process, a duplicate copy of the channel is created, and most of
the active data is given to the new copy. The original channel
gets its name changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original thread (preserving its
role as a call originator, if it had this role to begin with),
while the new channel is without this info and becomes a call
target (a "peer"). In this case, the parking lot manager thread
is handed the new (masqueraded) channel. It will not run an
h-exten on the channel if it hangs up while in the parking lot.
The h exten will be run on the original channel instead, in the
original thread, after the bridge completes. See bug 13820 for
our intentions as to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/ ........
2008-12-23 16:04 +0000 [r166625] Russell Bryant <russell@digium.com>
* CHANGES: Fix spelling error.
2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 166568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
2008) | 12 lines Fix a crash resulting from a datastore with
inheritance but no duplicate callback The fix for this is to
simply set the newly created datastore's data pointer to NULL if
it is inherited but has no duplicate callback. (closes issue
#14113) Reported by: francesco_r Patches: 14113.patch uploaded by
putnopvut (license 60) Tested by: francesco_r ........
2008-12-23 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 166509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
| 4 lines Use the integer form of condition for integer
comparisons. (closes issue #14127) Reported by: andrew ........
2008-12-22 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com>
* res/res_agi.c: Always use the value of the AGISIGHUP when running
an AGI. Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel. (closes issue #13711) Reported by:
fmueller Patches: 13711.patch uploaded by putnopvut (license 60)
2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Cosmetic change - don't mix struct
initializer styles.
2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon,
22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks
and autoservice It has been discovered that if a channel is
locked prior to a call to ast_autoservice_stop, then it is likely
that a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure that
the thread running autoservice is not currently trying to
manipulate the channel we are about to pull out of autoservice.
The autoservice thread, however, cannot advance beyond where it
currently is, though, because it is trying to acquire the lock of
the channel for which autoservice is attempting to be stopped.
The gist of all this is that a channel MUST NOT be locked when
attempting to stop autoservice on the channel. In this particular
case, the channel was locked by a call to ast_read. A call to
ast_exists_extension led to autoservice being started and stopped
due to the existence of dialplan switches. It may be that there
are future commits which handle the same symptoms but in a
different location, but based on my looks through the code, it is
very rare to see a construct such as this one. (closes issue
#14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
by putnopvut (license 60) Tested by: rtrauntvein Review:
http://reviewboard.digium.com/r/107/ ........
2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Fix a bad typo.
* main/astobj2.c: Remove some error messages. This is the default
handler that is valid to use.
* /, main/utils.c: Merged revisions 166297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
| 2 lines Fix up timeout handling in ast_carefulwrite(). ........
* include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce
ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file
stream which will handle timeouts and partial writes. It is
currently used in AMI to address the issue that has been
reported. However, there are probably a few other places where
this could be used. (closes issue #13546) Reported by: srt Tested
by: russell http://reviewboard.digium.com/r/104/
* res/res_musiconhold.c: Re-work ref count handling of MoH classes
using astobj2 to resolve crashes. (closes issue #13566) Reported
by: igorcarneiro Tested by: russell Review:
http://reviewboard.digium.com/r/106/
2008-12-22 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com>
* main/dnsmgr.c: Record the previous port in the temporary address
structure so that the comparison does not treat the host as
having changed even if it did not. This would have been
uninitialized before and would have led to a baddddd port.
(closes issue #13628) Reported by: pananix Patches:
bug13628.patch uploaded by jpeeler (license 325) Tested by: file,
blitzrage
2008-12-22 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com>
* funcs/func_timeout.c, main/file.c: Fix a file playback crash and
explicitly initialize values in func_timeout.c A crash was
brought up on the bugtracker. The first run through valgrind was
full of legitimate complaints of uninitialized values in
func_timeout when setting a response timeout. These were fixed
but the crash persisted. A second run through showed the real
problem. The reference counting used for filestreams was
incorrect because there were some missing increments when a frame
was read from a format module. (closes issue #14118) Reported by:
blitzrage Patches: 14118v2.patch uploaded by putnopvut (license
60) Tested by: blitzrage
2008-12-22 14:16 +0000 [r166258] Russell Bryant <russell@digium.com>
* res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This
patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The
only usage was for the AGI command, "asyncagi break". This patch
removes this feature. Normally, a feature would not be removed
like this. However, this code is broken and usage of it will
result in a memory leak. Usage of this feature will make the AGI
code return a result of AST_PBX_KEEPALIVE. The PBX handler
assumes that another thread has assumed ownership of the channel.
The channel thread will exit without destroying the channel.
Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here: 1) The only
way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old
ast_channel structure will be lost. 2) Until the channel redirect
happens, there is no code servicing the channel. That means
nothing is reading audio or handling events coming from the
channel. This is very bad. The recommended way to get this same
"break" functionality is to issue the redirect while the channel
is still being handled by the AGI code. That way, there will be
no memory leak, and there will be no period of time that the
channel is not being serviced.
2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h: Make a note about formatting the
review URL in commit messages
2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com>
* main/audiohook.c: Get rid of an extra space. I don't know how
this crept back in when I had already fixed it earlier
* funcs/func_audiohookinherit.c: Remove the verbatim tag from the
author line I could have sworn I already did that before,
though...
* main/channel.c, funcs/func_audiohookinherit.c (added),
include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a
new dialplan function AUDIOHOOK_INHERIT This function is being
added as a method to allow for an audiohook to move to a new
channel during a channel masquerade. The most obvious use for
such a facility is for MixMonitor when a transfer is performed.
Prior to the addition of this functionality, if a channel running
MixMonitor was transferred by another party, then the recording
would stop once the transfer had completed. By using
AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the
call even after the transfer has completed. It has also been
determined that since this is seen by most as a bug fix and is
not an invasive change, this functionality will also be
backported to 1.4 and merged into the 1.6.0 branches, even though
they are feature-frozen. (closes issue #13538) Reported by: mbit
Patches: 13538.patch uploaded by putnopvut (license 60) Tested
by: putnopvut Review: http://reviewboard.digium.com/r/102/
2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Add configuration
support for half_full DAHDI buffer policy
2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_record.c: Fix the XML documentation for Record().
<value> tags inside <variable> elements must have CDATA and no
another XML node.
2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com>
* /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
| 9 lines Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user
on a BSD system (closes issue #14111) Reported by: ys Patches:
app_chanspy.c.diff uploaded by ys (license 281) ........
* include/asterisk/doxyref.h: Disable some automatic links
generated by doxygen.
* include/asterisk/doxyref.h: Introduce commit message formatting
guidelines. This documents the recommended outline to use for
commit message. It also covers information on special tags that
can be used in commit messages, as well as the template
functionality that is available on bugs.digium.com. Review:
http://reviewboard.digium.com/r/96/
* /, main/utils.c: Merged revisions 165796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
| 11 lines Make ast_carefulwrite() be more careful. This patch
handles some additional cases that could result in partial writes
to the file description. This was done to address complaints
about partial writes on AMI. (issue #13546) (more changes needed
to address potential problems in 1.6) Reported by: srt Tested by:
russell Review: http://reviewboard.digium.com/r/99/ ........
2008-12-18 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: (closes issue #13993) Reported by: mika Add
ActionID response to ping if sent with request.
2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18
Dec 2008) | 8 lines Add mutexes around accesses to the IMAP
library interface. This prevents certain crashes, especially when
shared mailboxes are used. (closes issue #13653) Reported by:
howardwilkinson Patches:
asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
howardwilkinson (license 590) Tested by: jpeeler ........
2008-12-18 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c,
channels/chan_oss.c: Numerous documentation updates. (closes
issue #13970) Reported by: pkempgen Patches:
__20081217_cli_usage_fixes.patch.txt uploaded by blitzrage
(license 10)
2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com>
* res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was
being set NULL just prior to being dereferenced in an ao2_link
call. I have moved the setting of the variable to NULL until
after the ao2_link call.
2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com>
* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the
need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This
is part of an effort to completely remove AST_PBX_KEEPALIVE and
other similar return codes from the source. While this usage was
perfectly safe, there are others that are problematic. Since we
know ahead of time that we do not want to PBX to destroy the
channel, the PBX API has been changed so that information can be
provided as an argument, instead, thus removing the need for the
KEEPALIVE return value. Further changes to get rid of KEEPALIVE
and related code is being done by murf. There is a patch up for
that on review 29. Review: http://reviewboard.digium.com/r/98/
* /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18
Dec 2008) | 7 lines Set the process group ID on the MOH process
so that all children will get killed (closes issue #14099)
Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2
uploaded by caspy (license 645) ........
2008-12-18 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Fix 2 resource leaks and fix another
pipe-to-comma conversion
2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 165591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
lines Only care about a compatible codec for early bridging if we
are actually bridging to another channel. If we are not we
actually want to bring the audio back to us. (closes issue
#13545) Reported by: davidw ........
2008-12-18 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Fix reference counts of the class and add an
assertion to the end.
2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com>
* main/strings.c, include/asterisk/strings.h: Remove duplicate code
from the ast_str API. We now use __AST_STR_* to access 'struct
ast_str' members, but this must only be used inside the API
implementation. (closes issue #14098) Reported by: eliel Patches:
ast_str.patch uploaded by eliel (license 64)
2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com>
* apps/app_originate.c: Add a \todo note for app_originate. Jared
Smith suggested that we add a way to be able to set variables and
functions on the outbound channel. I think that it's a great
idea, so I have added it as a todo so that it gets done at some
point.
* apps/app_originate.c (added), CHANGES: Add a new application,
Originate. (closes issue #14075) Reported by: rcasas Patches:
app_originate.c uploaded by rcasas (license 641), heavily
modified by me Tested by: russell Review:
http://reviewboard.digium.com/r/95/
2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com>
* apps/app_record.c: Add RECORD_STATUS variable, as requested on
the -users list. Patch by me (license 14)
2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson <mmichelson@digium.com>
* res/res_odbc.c: Fix a refcount leak in res_odbc
* apps/app_meetme.c, res/res_realtime.c: Fix the build
2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher <tlesher@digium.com>
* apps/app_macro.c: Oops, broke trunk
* /, apps/app_macro.c: Merged revisions 165317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
| 4 lines Reverse the fix from issue #6176 and add proper
handling for that issue. (Closes issue #13962, closes issue
#13363) Fixed by myself (license 14) ........
2008-12-17 21:17 +0000 [r165318] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c,
apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
2008) | 7 lines Fix some memory leaks found while looking at how
realtime configs are handled. Also cleaned up some coding
guidelines violations in app_realtime.c, mostly related to
spacing ........
2008-12-17 20:50 +0000 [r165254] Steve Murphy <murf@digium.com>
* utils/extconf.c: This patch is here committed to satisfy the
buildbot, who has a problem with the const.
2008-12-17 19:55 +0000 [r165219] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Polycom phones close the connection after
reading a little bit of the firmware files, we should stop
sending in that case. Also, make that case print out a debug
statement instead of a scary WARNING.
2008-12-17 19:52 +0000 [r165216] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Call proxy_update so that the IP address
gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue
#14055) Reported by: chris-mac
2008-12-17 18:49 +0000 [r165180] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will
treat any SDP data it receives as new data and update the media
stream accordingly. By default, Asterisk will only modify the
media stream if the SDP session version received is different
from the current SDP session version. This option is required to
interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with
Microsoft OCS which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/ (closes issue #13958)
Reported by: toc Tested by: toc
2008-12-17 17:56 +0000 [r165145] Russell Bryant <russell@digium.com>
* doc/appdocsxml.dtd: argsep is used as an attribute for an
argument, as well
2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: And actually assign the function to a
pointer...
* apps/app_voicemail.c: Use the create_vm_state_from_user function
in a place where it was not being used before. Also, I've moved
the urgent folder check in messagecount() up a bit so that the
flow is a bit better. This was something I noticed while taking a
look at issue #13973, although I don't think this is the
underlying cause of the issue.
2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming <kpfleming@digium.com>
* utils: ignore this copied file
2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy <murf@digium.com>
* utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c,
utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix"
for a "horribly broken" situation. As stuff shifts around in the
asterisk code, the miscellaneous inclusions from the standalone
stuff gets broken. There's no easy fix for this situation. I made
sure that everything in utils builds without problem ***AND***
that aelparse runs the regressions correctly with the following
make menuselect options both on and off: DONT_OPTIMIZE
DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE
DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE
I think from now on, I'm going to #undef all these features in
the various utils native files; I guess I could do the same for
the copied-in files, surrounded by STANDALONE ifdef. A standalone
isn't going to care about threads, mutexes, etc.
* pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions
2008-12-16 23:06 +0000 [r164978] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
2008) | 7 lines After looking through SIP registration code most
of the day, this is one of the few things I could find that was
just plain wrong. Even though it probably isn't possible for it
to happen, it seems weird to have code that checks if a pointer
is NULL and then immediately dereferences that pointer if it was
NULL. ........
2008-12-16 22:57 +0000 [r164976] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, doc/api-1.6.2-changes.txt (added),
funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c,
CHANGES, configs/extensions.conf.sample: Add timezone to the
possible fields in a timespec. (closes issue #14028) Reported by:
mostyn Patches: timezone-v2.patch uploaded by mostyn (license
398) (with additional code guideline fixes and a memory leak fix
by me - license 14)
2008-12-16 22:45 +0000 [r164942] Jeff Peeler <jpeeler@digium.com>
* apps/app_record.c: (closes issue #13669) Reported by: pj Delete
file recording if recording terminated from a hangup.
2008-12-16 22:31 +0000 [r164941] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Make a note of the feature request in bug
#11157 as per the reporter and oej, and suspend the bug since no
one seems to be keen on implementing it any time soon.
2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant <russell@digium.com>
* /, main/utils.c: Merged revisions 164881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
| 9 lines Fix an issue where DEBUG_THREADS may erroneously report
that a thread is exiting while holding a lock. If the last lock
attempt was a trylock, and it failed, it will still be in the
list of locks so that it can be reported. (closes issue #13219)
Reported by: pj ........
* /, apps/app_macro.c: Merged revisions 164876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
| 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
been returned. This is a bug I noticed while looking at the code
for app_macro. This return code means that another thread has
assumed ownership of the channel and it can no longer be touched.
(I hate this return code with a passion, by the way.) ........
* main/asterisk.c: Fix build issues on Linux after sysinfo related
changes
2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify
trumps poke per lmadsen.
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
configuration options for finer control over how Asterisk handles
having to poke all peers at seemingly the same time. (closes
issue #13217) Reported by: cervajs
2008-12-16 20:41 +0000 [r164807] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 164806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
| 9 lines Add "restart gracefully" to the AMI blacklist of CLI
commands. "module unload" was already identified as a command
that can not be used from the AMI. "restart gracefully"
effectively unloads all modules, and will run in to the same
problems. (closes issue #13894) Reported by: kernelsensei
........
2008-12-16 20:08 +0000 [r164802] Michiel van Baak <michiel@vanbaak.info>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/asterisk.c: introduce 'core show sysinfo' for systems that
dont have the Linux-ish sysinfo stuff but do have sysctl. (closes
issue #13433) Reported by: mvanbaak Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license
7) with two free calls replaced with ast_free based on feedback
on reviewboard Review: http://reviewboard.digium.com/r/91/
2008-12-16 20:04 +0000 [r164801] Steve Murphy <murf@digium.com>
* main/pbx.c: (closes issue #14076) Reported by: toc Tested by:
murf OK, Well this issue has had its share of flip-flopping. I
found the following: 1. the code in question, in ext_cmp1 in
pbx.c, would not allow two extensions that vary only by any
dashes contained within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored. So, skipping
them when sorting patterns seemed a bit silly. Thus, you might
declare ext 891 in a context, but if you try dialing 8-9-1, it
will NOT match 891. So, I proposed to remove the code from
ext_cmp1 to skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context, forcing users to
generate otherwise uselessly obfuscated dialplan code to get the
same effect. Then, I tried out 1.4, and found that: 1. you can
declare 891 and 8-9-1 in the same context! 2. You can't define
891, and have 8-9-1 match it! Nor can you define 8-9-1, and have
891 match it! So, it appears that my proposal simply restores the
pbx to behaving as it did in 1.4.
2008-12-16 19:54 +0000 [r164798] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/safe_asterisk: Set up umask as a possible
configuration option. (closes issue #13753) Reported by: irroot
2008-12-16 17:14 +0000 [r164737] Russell Bryant <russell@digium.com>
* /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged
revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
| 14 lines Fix memory leak and invalid reporting issues with
DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
being used within the context of the thread local data
destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all.
This led to a memory leak. Another issue was an invalid argument
being provided to the the object_add API call. (closes issue
#13678) Reported by: ys Tested by: Russell ........
2008-12-16 16:50 +0000 [r164733] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: Be more detailed about why the include did not
get included. (closes issue #14071) Reported by: kshumard
Patches: pbx_config.patch.improvederroroutput.txt uploaded by
kshumard (license 92)
2008-12-16 16:00 +0000 [r164675] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
| 11 lines Fix a memory leak related to the use of the "setvar"
configuration option. The problem was that these variables were
being appended to the list of vars on the sip_pvt every time a
re-registration or re-subscription came in. Since it's just a
waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying
the vars. (closes issue #14037) Reported by: marvinek Tested by:
russell ........
2008-12-16 15:44 +0000 [r164659] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When using externhost make sure the port
gets set to the bindaddr port if one was not specified in the
externhost value itself. (closes issue #13634) Reported by:
performer
2008-12-16 15:31 +0000 [r164648] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 164634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
lines I added a sentence to clarify why - and ' ' are ignored in
patterns as per bug 14076. Leif says he'll put some stuff about
it in the extensions.conf sample, etc. ........
2008-12-16 15:00 +0000 [r164602-164623] Russell Bryant <russell@digium.com>
* apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also,
remove a variable that was not needed. (closes issue #14081)
Reported by: pkempgen
* /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16
Dec 2008) | 5 lines Don't try to change working directory if a
directory was not configured. (closes issue #14089) Reported by:
caspy ........
* channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes
issue #14090) Reported by: alecdavis Patches:
chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
585)
2008-12-16 01:52 +0000 [r164565] Sean Bright <sean.bright@gmail.com>
* doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also
change a bit of minor formatting.
2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Open a timer before loading configuration
so that the trunking configuration option will take effect.
(closes issue #14082) Reported by: seandarcy
* channels/chan_iax2.c: Fix log message to refer to the generic
timing interface, not DAHDI specifically (inspired by issue
#14082)
* main/frame.c: Make sure we handle a uint32_t payload in
ast_frdup() (closes issue #14080) Reported by: fnordian Patches:
frame.patch uploaded by fnordian (license 110)
2008-12-15 21:17 +0000 [r164485] Tilghman Lesher <tlesher@digium.com>
* configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow
disabling pattern match searches within the Realtime dialplan
switch. (closes issue #13698) Reported by: fhackenberger Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson <mmichelson@digium.com>
* apps/app_page.c: Add an 'i' option to app_page. This option works
the same as the 'i' options for app_dial and app_queue, in that
they will ignore any attempts by phones to forward the call.
(closes issue #13977) Reported by: putnopvut Patches:
page_ignore_forwards.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, acunningham
* /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon,
15 Dec 2008) | 3 lines Add the deadlock note to
ast_spawn_extension as well ........
* /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
revisions 164416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
2008) | 4 lines Add notes to autoservice and pbx doxygen
regarding a potential deadlock scenario so that it is avoided in
the future ........
2008-12-15 19:48 +0000 [r164417] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str
opacity in chan_sip for now, since something wasn't quite right
in the merge.
2008-12-15 19:42 +0000 [r164415] Steve Murphy <murf@digium.com>
* include/asterisk/strings.h: I was getting this warning during a
compile on a 64-bit machine running ubuntu server 8.10, and
gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings
being treated as errors In file included from
/home/murf/asterisk/trunk/include/asterisk/utils.h:671, from
chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function
‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error:
comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2
which this fix silences
2008-12-15 18:12 +0000 [r164351] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
lines Do not try to unlock a non-existant channel if the transfer
fails. (closes issue #13800) Reported by: dwagner Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
608) ........
2008-12-15 18:09 +0000 [r164349] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: When querying for the structure of the CDR
table, remove the schema, if it exists. (Closes issue #14058)
2008-12-15 17:24 +0000 [r164312] Joshua Colp <jcolp@digium.com>
* main/file.c: Use ast_seekstream to return the file stream back to
the beginning instead of directly seeking to zero. This is
because some audio formats have headers at the front that need to
be skipped, which will be done by the format module. (closes
issue #14079) Reported by: elguero
2008-12-15 17:21 +0000 [r164272-164309] Russell Bryant <russell@digium.com>
* channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a
couple more build issues related to ast_str_opaque
* pbx/pbx_dundi.c: When a reload is issued, always process the
configuration for dundi.conf. The reason is that a reload can be
used to refresh DNS lookups for defined peers. Even if the config
file hasn't changed, we want to process it for that purpose.
(closes issue #13776) Reported by: kombjuder
2008-12-15 16:16 +0000 [r164268-164270] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a compile warning and a logic error that
could have been bad for non-realtime queues
* apps/app_queue.c: Fix up a few issues with regards to queues *
Fix reference counting used in the __queues_show function * Add
code to be sure that the "queue show" command does not print
information for a realtime queue which has been deleted from the
backend * Add a missing unref to the realtime queue loading
function for the case where a queue is in the module's container
but has been deleted from the realtime backend (closes issue
#14033) Reported by: cristiandimache Patches: 14033.patch
uploaded by putnopvut (license 60) Tested by: cristiandimache
2008-12-15 15:41 +0000 [r164208-164257] Joshua Colp <jcolp@digium.com>
* configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
configure.ac: Make app_fax compatible with newer versions of
spandsp. This remains backwards compatible with earlier versions
though so do not fret. (closes issue #14073) Reported by:
seandarcy
* main/utils.c: Update to work with new ast_str changes.
2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant <russell@digium.com>
* main/channel.c, /, main/features.c: Merged revisions 164201 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
| 31 lines Handle a case where a call can be bridged to a channel
that is still ringing. The issue that was reported was about a
case where a RINGING channel got redirected to an extension to
pick up a call from parking. Once the parked call got taken out
of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing
indication. This patch fixes this case so that the caller
receives ringback once it comes out of parking until the other
side answers. The fixes are: - Make sure we remember that a
channel was an outgoing channel when doing a masquerade. This
prevents an erroneous ast_answer() call on the channel, which
causes a bogus 200 OK to be sent in the case of SIP. - Add some
additional comments to explain related parts of code. - Update
the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that
are events or commands should be ignored. - When a bridge first
starts, check to see if the peer channel needs to be given
ringing indication because the calling side is still ringing. -
Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747) Reported by: davidw Tested by: russell
Review: http://reviewboard.digium.com/r/90/ ........
* apps/app_jack.c: Fix build WRT ast_str_opaque
2008-12-14 18:16 +0000 [r164168] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/strings.h: Don't pass a negative to an unsigned
type and expect things to work correctly.
2008-12-14 15:26 +0000 [r164054-164137] Sean Bright <sean.bright@gmail.com>
* doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art.
* res/snmp/agent.c: Use ast_str_strlen() instead of recalculating
the string length.
2008-12-13 13:26 +0000 [r164028] Michiel van Baak <michiel@vanbaak.info>
* res/snmp/agent.c: nuke another use of the ast_str internals.
2008-12-13 08:36 +0000 [r163991] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c, apps/app_meetme.c,
funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c,
main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h,
cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c,
res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c,
channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c,
configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c,
main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c,
res/res_config_ldap.c, include/asterisk/threadstorage.h,
cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c,
res/res_config_pgsql.c, main/strings.c (added), main/pbx.c,
channels/chan_sip.c, main/Makefile, main/translate.c,
include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c,
funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c,
main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c,
utils/hashtest2.c, include/asterisk/strings.h,
include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c,
apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque
branch (discontinue usage of ast_str internals)
2008-12-13 03:03 +0000 [r163951-163952] Sean Bright <sean.bright@gmail.com>
* doc/tex/asterisk.tex: This shouldn't have gotten commited. We
might want to generate this into a separate file instead of the
version controlled one.
* doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead
of ASCII art ones.
2008-12-13 00:59 +0000 [r163912] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c: Only detach and destroy the whisper
audiohooks if they are actually in use.
2008-12-12 23:48 +0000 [r163873] Terry Wilson <twilson@digium.com>
* apps/app_queue.c: When using realtime queues, app_queue wasn't
updating the strategy if it was changed in the realtime backend.
This patch resolves the issue for almost all situations. It is
currently not supported to switch to the linear strategy via
realtime since the ao2_container for members will have been set
to have multiple buckets and therefore the members would be
unordered. (closes issue #14034) Reported by: cristiandimache
Tested by: otherwiseguy, cristiandimache
2008-12-12 23:06 +0000 [r163828] Russell Bryant <russell@digium.com>
* res/res_clioriginate.c: Add a note to indicate why this only
supports one channel for now.
2008-12-12 22:04 +0000 [r163762] Tilghman Lesher <tlesher@digium.com>
* main/editline/read.c, /, main/asterisk.c: Merged revisions 163761
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
| 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
but also add a pointer inside editline to look back to
asterisk.c, so others don't spend as much time as I did looking
(in the wrong place) for the appropriate function. Reported by:
ZX81, via the #asterisk-users channel Fixed by: me (license 14)
........
2008-12-12 20:12 +0000 [r163716] Russell Bryant <russell@digium.com>
* CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel
redirect", which is similar in operation to AMI Redirect. Review:
http://reviewboard.digium.com/r/89/
2008-12-12 19:16 +0000 [r163675] Steve Murphy <murf@digium.com>
* channels/chan_dahdi.c: demote always-appearing debug message (for
certain boards) to ast_debug lev 3 msg instead
2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant <russell@digium.com>
* main/tcptls.c, channels/chan_sip.c: Rename a number of
tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a
relic from when the struct was a server_instance, not a
session_instance. It was renamed since it represents both a
server or client connection.
* channels/chan_sip.c: Fix a small race condition in
sip_tcp_locate(). We must increase the reference count on the
tcptls_session _before_ unlocking the thread list.
* channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with
qualify. The problem was a reference count error on the
tcptls_session structure. (closes issue #13989) Reported by:
Nugget
2008-12-12 18:17 +0000 [r163629] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When a device registers we need to unlink
them (if linked) from the peers_by_ip container and link them
back in since their IP address has changed. This would have
manifested itself if you configured a new device (as type=peer),
registered, and then tried to place a call from the device. Since
the peer was not linked into the peers_by_ip container it would
have never been found. (closes issue #13811) Reported by: pj
2008-12-12 17:22 +0000 [r163582-163612] Michiel van Baak <michiel@vanbaak.info>
* res/res_monitor.c: Document default Monitor file location.
(closes issue #14065) Reported by: kshumard Patches:
res_monitor.documentation.patch.txt uploaded by kshumard (license
92)
* channels/chan_skinny.c: Fix codec capability setup in chan_skinny
Behaviour now is that general codec config flows to default_line
and default_device. [devices] stuff amends default_device and
similar for [lines]. These are copied to individual device and
line as they are created. Added confcapability and confprefs for
the configured stuff which doesn't change as device and so on are
connected. prefs are based on line prefs if they exist, else the
device prefs are used (prefs identifies codec order). (closes
issue #13806) Reported by: pj Patches: codecs.diff uploaded by
wedhorn (license 30) Tested by: pj and me
2008-12-12 16:55 +0000 [r163579] Joshua Colp <jcolp@digium.com>
* main/channel.c, channels/chan_sip.c: Since chan_sip is callback
devicestate driven do not pass in actual states, pass in unknown
so we get asked. Additionally do not pass in an actual device
state value in ast_setstate since the channel may be callback
driven. (closes issue #13525) Reported by: pj
2008-12-12 15:10 +0000 [r163516] Doug Bailey <dbailey@digium.com>
* configs/phoneprov.conf.sample: Add internationalization to sample
configuration file
2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant <russell@digium.com>
* /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)
| 5 lines Specify uint32_t for variables storing a CRC32 so that
it is actually 32 bits on 64-bit machines, as well. (inspired by
issue #13879) ........
* main/channel.c, main/autoservice.c, /,
include/asterisk/channel.h: Merged revisions 163448 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12
Dec 2008) | 26 lines Resolve issues that could cause DTMF to be
processed out of order. These changes come from
team/russell/issue_12658 1) Change autoservice to put digits on
the head of the channel's frame readq instead of the tail. If
there were frames on the readq that autoservice had not yet read,
the previous code would have resulted in out of order processing.
This required a new API call to queue a frame to the head of the
queue instead of the tail. 2) Change up the processing of DTMF in
ast_read(). Some of the problems were the result of having two
sources of pending DTMF frames. There was the dtmfq and the more
generic readq. Both were used for pending DTMF in various
scenarios. Simplifying things to only use the frame readq avoids
some of the problems. 3) Fix a bug where a DTMF END frame could
get passed through when it shouldn't have. If code set
END_DTMF_ONLY in the middle of digit emulation, and a digit
arrived before emulation was complete, digits would get processed
out of order. (closes issue #12658) Reported by: dimas Tested by:
russell, file Review: http://reviewboard.digium.com/r/85/
........
2008-12-11 23:38 +0000 [r163384] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 163383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008)
| 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on
certain shells, the terminal is messed up. By intercepting those
events with a signal handler in the remote console, we can avoid
those issues. (closes issue #13464) Reported by: tzafrir Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage ........
2008-12-11 22:49 +0000 [r163317] Matthew Nicholson <mnicholson@digium.com>
* /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec
2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes
issue #13819) Reported by: adomjan Patches:
pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
dundi_clearecache3.diff uploaded by mnicholson (license 96)
Tested by: adomjan ........
2008-12-11 21:48 +0000 [r163241-163254] Russell Bryant <russell@digium.com>
* /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions
163253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008)
| 8 lines Fix some observed slowdowns in dialplan processing. The
change is to remove autoservice usage from dialplan functions
that do not need it because they do not perform operations that
potentially block. (closes issue #13940) Reported by: tbelder
........
* res/res_timing_pthread.c: Fix a problem where continuous mode
will get inadvertently get turned off if set_rate() is used while
continuous mode was already turned on. (closes issue #13738)
Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix
(license 547)
2008-12-11 20:57 +0000 [r163198-163213] Mark Michelson <mmichelson@digium.com>
* configs/voicemail.conf.sample, apps/app_voicemail.c: Add an
option to voicemail.conf to allow urgent messages to be forwarded
as not urgent. (closes issue #14063) Reported by: jaroth Patches:
urgfwd_v2.patch uploaded by jaroth (license 50)
* main/features.c: Add an appropriate goto if ast_call fails
2008-12-11 20:07 +0000 [r163171] Russell Bryant <russell@digium.com>
* main/channel.c: Fix the "failed" extension for outgoing calls.
The conversion to use ast_check_hangup() everywhere instead of
checking the softhangup flag directly introduced this problem.
The issue is that ast_check_hangup() checked for tech_pvt to be
NULL. Unfortunately, this will be NULL is some valid
circumstances, such as with a dummy channel. The fix is simple.
Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets
called. This happens inside of ast_hangup(), which is the same
function responsible for freeing the channel. Any code calling
ast_check_hangup() better not be calling it after that point, and
if so, we have a bigger problem at hand. (closes issue #14035)
Reported by: erogoza
2008-12-11 20:02 +0000 [r163168] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Sometimes even Linux needs -lm to link
libtonezone, such as when libtonezone is compiled statically.
(closes issue #13887) Reported by: tzafrir
2008-12-11 19:40 +0000 [r163166] Mark Michelson <mmichelson@digium.com>
* main/features.c: Reduce indentation level of
ast_feature_request_and_dial
2008-12-11 17:06 +0000 [r163094] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 163092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008)
| 11 lines Fix an issue that made it so you could only have a
single caller executing a custom feature at a time. This was
especially problematic when custom features ran for any
appreciable amount of time. The fix turned out to be quite
simple. The dynamic features are now stored in a read/write list
instead of a list using a mutex. (closes issue #13478) Reported
by: neutrino88 Fix suggested by file ........
2008-12-11 16:52 +0000 [r163089] Tilghman Lesher <tlesher@digium.com>
* /, res/res_agi.c: Merged revisions 163088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008)
| 6 lines Don't wait forever, if there's a specified recording
timeout. (closes issue #13885) Reported by: bamby Patches:
res_agi.c.patch uploaded by bamby (license 430) ........
2008-12-11 16:47 +0000 [r163081-163085] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 163084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec
2008) | 4 lines Revert this cast to long. Using time_t here
causes build failures on a FreeBSD 32-bit build. ........
* /, apps/app_queue.c: Merged revisions 163080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec
2008) | 14 lines Fix a potential crash due to unsafe datastore
handling. This patch also contains a conversion from using long
to time_t for representing times for a queue, as well as some
whitespace fixes. (closes issue #14060) Reported by: nivek
Patches: datastore_fixup.patch.corrected uploaded by nivek
(license 636) with slight modification from me Tested by: nivek
........
2008-12-11 15:40 +0000 [r163037] Sean Bright <sean.bright@gmail.com>
* doc/tex/qos.tex: Fix some of the grammar issues in
doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard
Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92)
(Slight modifications by seanbright)
2008-12-11 15:05 +0000 [r162997] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When a device registers to use it is
entirely possible that they may be in use, so tell the core that
we don't know the devstate and have it ask us for it. (closes
issue #13525) Reported by: pj
2008-12-10 23:01 +0000 [r162930] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Previously missing line, now the substitution works
correctly
2008-12-10 22:53 +0000 [r162927] Jeff Peeler <jpeeler@digium.com>
* /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10
Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check.
Pointed out by mmichelson, thanks! ........
2008-12-10 22:48 +0000 [r162923] Joshua Colp <jcolp@digium.com>
* res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to
changes done to turn it into a single memory allocation we can't
just use the existing CLI alias structure. We have to destroy all
existing ones and then create new ones. (closes issue #14054)
Reported by: pj
2008-12-10 22:48 +0000 [r162922] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Checking global variables here actually overwrote the
previous substitution by channel variables, and in any case, was
redundant; pbx_substitute_variables_helper ALREADY does
substitution for global variables. (closes issue #13327) Reported
by: pj
2008-12-10 22:11 +0000 [r162891] Jeff Peeler <jpeeler@digium.com>
* /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10
Dec 2008) | 5 lines (closes issue #13229) Reported by:
clegall_proformatique Ensure that moh_generate does not return
prematurely before local_ast_moh_stop is called. Also, the sleep
in mp3_spawn now only occurs for http locations since it seems to
have been added originally only for failing media streams.
........
2008-12-10 19:02 +0000 [r162739-162805] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6
lines Fix subscription based MWI up a bit. We only want to put
sip: at the beginning of the URI if it is not already there and
revert code to ignore destination check if subscribing for MWI.
(closes issue #12560) Reported by: vsauer Patches: patch001.diff
uploaded by ramonpeek (license 266) ........
* /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6
lines When a SIP peer unregisters set the expiry time back to 0
so that the 200 OK contains an expires of 0. (closes issue
#13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded
by hjourdain (license 583) ........
2008-12-10 17:09 +0000 [r162687] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk.h, main/asterisk.c, main/cli.c: add tab
completion for 'core set debug X filename.c' (closes issue
#13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt
uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel
2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson <mmichelson@digium.com>
* doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec
2008) | 8 lines Add missing documentation to misdn.txt (closes
issue #14052) Reported by: festr Patches: misdn.txt.patch
uploaded by festr (license 443) ........
* /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec
2008) | 11 lines Revert fix for issue 13570. It has caused more
problems than it helped to fix. (closes issue #13783) Reported
by: navkumar (closes issue #14025) Reported by: ffs ........
2008-12-10 16:11 +0000 [r162619-162660] Joshua Colp <jcolp@digium.com>
* res/res_http_post.c: FreeBSD also needs libgen.h (closes issue
#14051) Reported by: ys Patches: res_http_post.c.diff uploaded by
ys (license 281)
* /, main/rtp.c: Merged revisions 162653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6
lines Increment the sequence number on the end packets for
RFC2833. After reading the RFC some more and doing some testing I
agree with this change. (closes issue #12983) Reported by: vt
Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
520) ........
* channels/chan_sip.c: When transmitting a register set the socket
port to the local one for the transport being used, not the port
for the remote server. (closes issue #13633) Reported by:
performer
2008-12-10 11:34 +0000 [r162583] Michiel van Baak <michiel@vanbaak.info>
* res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD
uses an old version of gcc which throws an error if you use a
macro that's not #defined
2008-12-10 01:09 +0000 [r162542] Joshua Colp <jcolp@digium.com>
* doc/janitor-projects.txt, channels/iax2-parser.c,
apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and
remove it as a janitor project. (closes issue #14032) Reported
by: bkruse Patches: 14032.patch uploaded by bkruse (license 132)
2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/stringfields.h: it does help if the compiler
attribute syntax is correct
2008-12-09 23:10 +0000 [r162466] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09
Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........
2008-12-09 22:38 +0000 [r162414-162418] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
main/asterisk.c: Add some additional Asterisk project developer
documentation. After the nightly update of the documentation on
asterisk.org, I'll post an update to asterisk-dev with a pointer
to the changes. This covers some release branch and commit policy
information. None of this should be a surprise, since it's just
documenting what we have already been doing.
* include/asterisk/utils.h, /, main/utils.c, main/asterisk.c:
Merged revisions 162413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008)
| 8 lines Remove the test_for_thread_safety() function
completely. The test is not valid. Besides, if we actually
suspected that recursive mutexes were not working, we would get a
ton of LOG_ERROR messages when DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list) ........
2008-12-09 21:57 +0000 [r162355] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09
Dec 2008) | 4 lines We appear to have documented tz= in the
[general] section of voicemail.conf, without actually having
implemented it. Oops. (Reported by Olivier on the -users list)
........
2008-12-09 21:16 +0000 [r162342] Joshua Colp <jcolp@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 162341 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4
lines Add 'down' as a valid state for directed call pickup. This
creeps up when we receive session progress when dialing a device
and not ringing. (closes issue #14005) Reported by: ddl ........
2008-12-09 20:59 +0000 [r162291] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008)
| 9 lines Fix an issue where callers on an incoming call on an
SLA trunk would not hear ringback. We need to make sure that we
don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will
treat it as inband progress, even though all they are getting is
silence. (closes issue #12471) Reported by: mthomasslo ........
2008-12-09 20:46 +0000 [r162275] Joshua Colp <jcolp@digium.com>
* /, apps/app_festival.c: Merged revisions 162273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4
lines Fix double declaration of 'x' on the PPC platform. (closes
issue #14038) Reported by: ffloimair ........
2008-12-09 20:40 +0000 [r162271] Steve Murphy <murf@digium.com>
* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1
line In discussion with seanbright on #asterisk-dev, I have added
a default rule, and an option to suppress the default rule from
being generated in the flex output, for the sake of those OS's
where they didn't tweak flex's ECHO macro, and the compiler
doesn't like it. The regressions are OK with this. ........
2008-12-09 20:30 +0000 [r162266] Mark Michelson <mmichelson@digium.com>
* main/pbx.c, /: Merged revisions 162265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec
2008) | 6 lines If we fail to start a thread for the pbx to run
in, we need to be sure to decrease the number of active calls on
the system. This fix may relate to ABE-1713, but it is not
certain yet. ........
2008-12-09 19:48 +0000 [r162197-162205] Joshua Colp <jcolp@digium.com>
* /, main/rtp.c: Merged revisions 162204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7
lines Make sure that the timestamp for DTMF is not the same as
the previous voice frame and do not send audio when transmitting
DTMF as this confuses some equipment. (closes issue #13209)
Reported by: ip-rob Patches: 13209.diff uploaded by file (license
11) Tested by: ip-rob, bujones ........
* /, main/rtp.c: Merged revisions 162188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4
lines Take video into account when early bridging RTP. (closes
issue #13535) Reported by: davidw ........
2008-12-09 18:35 +0000 [r162079-162140] Steve Murphy <murf@digium.com>
* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1
line Previous fix used ast_malloc and ast_copy_string and messed
up the standalone stuff. Fixed. ........
* res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c,
res/ael/ael.flex: Merged revisions 162013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) |
45 lines (closes issue #14019) Reported by: ckjohnsonme Patches:
14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme,
murf This crash was the result of a few small errors that would
combine in 64-bit land to result in a crash. 32-bit land might
have seen these combine to mysteriously drop the args to an
application call, in certain circumstances. Also, in trying to
find this bug, I spotted a situation in the flex input, where, in
passing back a 'word' to the parser, it would allocate a buffer
larger than necessary. I changed the usage in such situations, so
that strdup was not used, but rather, an ast_malloc, followed by
ast_copy_string. I removed a field from the pval struct, in u2,
that was never getting used, and set in one spot in the code. I
believe it was an artifact of a previous fix to make switch cases
work invisibly with extens. And, for goto's I removed a '!' from
before a strcmp, that has been there since the initial merging of
AEL2, that might prevent the proper target of a goto from being
found. This was pretty harmless on its own, as it would just
louse up a consistency check for users. Many thanks to
ckjohnsonme for providing a simplified and complete set of
information about the bug, that helped considerably in finding
and fixing the problem. Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state, so I can run
the regression suite! ........
2008-12-09 16:47 +0000 [r161951-162016] Russell Bryant <russell@digium.com>
* /, apps/app_disa.c: Merged revisions 162014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008)
| 5 lines Allow DISA to handle extensions that start with #.
(closes issue #13330) Reported by: jcovert ........
* /, main/app.c: Merged revisions 161948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008)
| 15 lines Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old
channel to the new one. However, it did nothing with the group
settings that were already on the new channel. This patch removes
all group settings that already existed on the new channel. I
have a more complicated version of this patch which addresses
only the most blatant problem with this, which is that a channel
can end up with multiple group settings in the same category.
However, I could not think of a use case for keeping any of the
group settings from the old channel, so I went this route for
now. (closes AST-152) ........
2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons <eliels@gmail.com>
* funcs/func_odbc.c: Avoid allocating memory for a thread that
don't need it. Also, this memory was not being freed until the
main thread ends. (That is never). (closes issue #14040) Reported
by: eliel Patches: func_odbc.c.patch uploaded by eliel (license
64)
2008-12-08 23:04 +0000 [r161911] Brandon Kruse <bkruse@digium.com>
* main/pbx.c: Note that the recently changed waittime parameter is
in milliseconds.
2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp <jcolp@digium.com>
* formats/format_pcm.c: Add alw as a valid file extension for alaw
and ulw as a valid file extension for ulaw. (closes issue #14001)
Reported by: henrikw Patches: alw.diff uploaded by henrikw
(license 627)
* contrib/scripts/autosupport.8, contrib/scripts/autosupport:
Update autosupport script with a few changes.
2008-12-08 18:49 +0000 [r161790] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Allocate enough space initially for the message.
(closes issue #14027) Reported by: junky Patches: M14027.diff
uploaded by junky (license 177)
2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Fix a regression introduced when the PBX timeouts
were converted to milliseconds. collect_digits now gets
milliseconds fed to it, not seconds. (closes issue #14012)
Reported by: dveiga Patches: 14012.patch uploaded by bkruse
(license 132)
* /, channels/chan_sip.c: Merged revisions 161725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6
lines Make the usereqphone option work again. (closes issue
#13474) Reported by: mmaguire Patches: 20080912_bug13474.diff
uploaded by mmaguire (license 571) ........
2008-12-08 17:23 +0000 [r161721] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Fix a crash that can occur on a transfer in
chan_sip when attempting to collect rtp stats. (closes issue
#13956) Reported by: chris-mac Tested by: chris-mac
2008-12-08 16:02 +0000 [r161679] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, CHANGES: Add the ability to play a courtesy
tone to the transfer target in a native SIP attended transfer by
setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: - Fix a leak while printing an argument
description. - Avoid printing the name of an argument in the
[Arguments] tag if there is no description for that argument.
* apps/app_voicemail.c: Add voicemail related applications and
functions XML documentation: applications: - VoiceMail() -
VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: -
MAILBOX_EXISTS()
* apps/app_sms.c: Introduce SMS() application XML documentation.
2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_speech_utils.c: Move Speech* applications and functions
documentation to XML.
2008-12-05 23:24 +0000 [r161493] Mark Michelson <mmichelson@digium.com>
* apps/app_stack.c: If the autoloop flag is set on a channel, then
we need to add 1 to the priority when checking if the extension
exists. Otherwise, gosubs will fail. This was discovered when
investigating an asterisk-users mailing list post made by Gary
Hawkins.
2008-12-05 21:08 +0000 [r161349-161427] Sean Bright <sean.bright@gmail.com>
* /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
161426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r161426 | seanbright | 2008-12-05 16:02:20 -0500
(Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
int). (closes issue #14006) Reported by: alphaque Patches:
astobj2.h-patch uploaded by alphaque (license 259) (Slightly
modified by seanbright) ........ ................
* apps/app_voicemail.c: Use ast_free() instead of free(), pointed
out by eliel on IRC.
* apps/app_voicemail.c: When using IMAP_STORAGE, it's important to
convert bare newlines (\n) in emailbody and pagerbody to CR-LF so
that the IMAP server doesn't spit out an error. This was
informally reported on #asterisk-dev a few weeks ago. Reviewed by
Mark M. on IRC.
2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant <russell@digium.com>
* main/pbx.c, /: Merged revisions 161287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008)
| 2 lines Fix a NULL format string warning found by buildbot.
........
* apps/app_minivm.c: Resolve a compiler warning from buildbot about
a NULL format string.
2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons <eliels@gmail.com>
* main/udptl.c, main/frame.c, res/res_musiconhold.c,
channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c,
main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c,
channels/chan_skinny.c, res/res_agi.c, main/features.c,
apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c,
res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c,
apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c,
channels/chan_sip.c, main/translate.c, channels/chan_agent.c,
res/res_convert.c, res/res_crypto.c, apps/app_queue.c,
channels/chan_oss.c, apps/app_playback.c,
channels/chan_usbradio.c, main/file.c, main/astmm.c,
pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c,
apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c,
apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible.
(closes issue #13990) Reported by: eliel Patches: array_len.diff
uploaded by eliel (license 64)
2008-12-05 05:41 +0000 [r161181] Tilghman Lesher <tlesher@digium.com>
* main/config.c: The first file should have a blank config filename
in the structure, so that when a save occurs to a different
filename, everything goes to the alternate filename, instead of
appending to the original. This is important for the AMI command
UpdateConfig. (closes issue #13301) Reported by: trevo Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license
14) Tested by: Corydon76, blitzrage
2008-12-05 02:47 +0000 [r161147] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: Check the return value of fread/fwrite so
the compiler doesn't complain. Only a problem when IMAP_STORAGE
is enabled.
2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If
'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it
exists) after T38 is negotiated. Terry Wilson created the
original patch for this functionality, which I slightly modified
and added the faxdetect=yes|no configuration option. This patch
is only for T38 fax detection and does not do anything for G711
over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/ This
functionality is for issue AST-140.
2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons <eliels@gmail.com>
* main/cli.c: Fix minor coding guidelines introduced with CLI
permissions.
2008-12-04 18:32 +0000 [r161014] Jeff Peeler <jpeeler@digium.com>
* /, main/rtp.c: Merged revisions 161013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008)
| 9 lines (closes issue #13835) Reported by: matt_b Tested by:
jpeeler This mirrors a check that was present in ast_rtp_read to
also be in ast_rtp_raw_write to not schedule sending the receiver
report if the remote RTCP endpoint address isn't present in the
RTCP structure. Closes AST-142. ........
2008-12-04 16:45 +0000 [r160945] Mark Michelson <mmichelson@digium.com>
* /, main/callerid.c: Merged revisions 160943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec
2008) | 15 lines Fix a callerid parsing issue. If someone
formatted callerid like the following: "name <number>" (including
the quotation marks), then the parts would be parsed as name:
"name number: number This is because the closing quotation mark
was not discovered since the number and everything after was
parsed out of the string earlier. Now, there is a check to see if
the closing quote occurs after the number, so that we can know if
we should strip off the opening quote on the name. Closes AST-158
........
2008-12-04 16:37 +0000 [r160938] Michiel van Baak <michiel@vanbaak.info>
* build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug
flag so skinny debug will show information about packets. We dont
want to scare users with this, so we added a devmode compile flag
(closes issue #13952) Reported by: wedhorn Patches:
packetdebug3.diff uploaded by wedhorn (license 30) Tested by:
mvanbaak, wedhorn
2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c: Added XML documentation for the following AGI
commands: - get option - get variable - hangup - noop
2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett <rmudgett@digium.com>
* funcs/func_callerid.c: Jcolp pointed out that num will also match
number
* funcs/func_callerid.c: * Found a couple more places where
num/number needed to be done so 1.4 upgraders will not have
problems. * Added curly braces and minor tweaks.
2008-12-03 21:58 +0000 [r160791] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 160770 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03
Dec 2008) | 2 lines Some compilers warn on null format strings;
some don't (caught by buildbot) ........
2008-12-03 21:09 +0000 [r160760] Steve Murphy <murf@digium.com>
* /, funcs/func_callerid.c: Merged revisions 160703 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec
2008) | 11 lines (closes issue #13597) Reported by: john8675309
Patches: patch.13597 uploaded by murf (license 17) Tested by:
murf, john8675309 This patch causes the setcid func to update the
CDR clid after setting the channel field. I also notice that in
trunk, the num/number of 1.4 is left out; I decided to include
the option to use either in trunk, so as not to have 1.4
upgraders not to have problems. ........
2008-12-03 20:35 +0000 [r160699-160700] Jason Parker <jparker@digium.com>
* main/manager.c: Another place this is missing
* main/manager.c: Fix typo when ListCategories returns none.
(closes issue #13994) Reported by: mika Patches:
ListCategoriesActionPatch.diff uploaded by mika (license 624)
2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons <eliels@gmail.com>
* channels/iax2-provision.c: - iax2-provision was not freeing
iax_templates structure when unloading the chan_iax2.so module. -
Move the code to start using the LIST macros. Review:
http://reviewboard.digium.com/r/72 (closes issue #13232) Reported
by: eliel Patches: iax2-provision.patch.txt uploaded by eliel
(license 64) (with minor changes pointed by Mark Michelson on
review board) Tested by: eliel
2008-12-03 18:37 +0000 [r160626] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: Add some
safety measures when using gosub, especially when using the
options for app_dial and app_queue to run a gosub when the call
is answered. * Check for the existence of the gosub target in
gosub_exec. If it is nonexistent, then this will cause errors
when we attempt to actually run the gosub, including a definite
memory leak and potential crashes. Return an error in this
situation * Check the return value of pbx_exec in app_dial and
app_queue before attempting to actually run the gosub routine. If
there was an error, we should not attempt to run the gosub. *
Change a '|' to a ',' in app_queue. * Add some extra curly braces
where they had been missing previously. (closes issue #13548)
Reported by: fiddur
2008-12-03 17:48 +0000 [r160562] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_minivm.c: - Add <variable /> tags when naming a channel
variable. - Add <filename /> tags when naming a filename. -
Simplify the xml formatting putting some enters.
2008-12-03 17:38 +0000 [r160559] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 160558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008)
| 7 lines If an entry is added to the directory during a scan
when another entry expires, then that new entry will not be
processed promptly, but must wait for either a future entry to
start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch
expire, then the new entries must wait until another new entry is
added to be processed. This was a rather weird race condition,
really. Fixes AST-147. ........
2008-12-03 17:07 +0000 [r160555] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: When investigating issue #13548, I found that
gosub handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being done on
the incorrect channel. With this set of changes, a gosub will
correctly run on the answering queue member's channel. There are
still crash issues which occur if there are dialplan syntax
errors, so I cannot yet close the referenced issue.
2008-12-03 17:01 +0000 [r160481-160552] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 160551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008)
| 4 lines Don't start scanning the directory until all modules
are loaded, because some required modules (channels, apps,
functions) may not yet be in memory yet. Fixes AST-149. ........
* /, channels/chan_sip.c: Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
| 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I
guess that having only ip-phones in mind is not a good approach.
Since it is possible to have a sip proxy connected to asterisk we
could receive a 407 (unauthorized) or 483 (too many hops) as
response and dialog ending would not be a good behavior." So
modified. ........
2008-12-03 11:01 +0000 [r160447] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are
using XML documentation (or the xml documentation wont be
loaded). - Use <variable></variable> to refer to a dialplan
variable.
2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher <tlesher@digium.com>
* CHANGES: Info on LOCAL_PEEK function.
* apps/app_stack.c: Add LOCAL_PEEK function, as requested by
lmadsen.
2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: remove duplicate comment that I
accidentally merged
* channels/chan_dahdi.c: (closes issue #13786) Reported by: tzafrir
Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260
which fixes not being able to make outgoing calls on some FXO
adapters:
http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
2008-12-02 17:56 +0000 [r160208-160308] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 160297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
| 10 lines When the text does not match exactly (e.g. RTP/SAVP),
then the %n conversion fails, and the resulting integer is
garbage. Thus, we must initialize the integer and check it
afterwards for success. (closes issue #14000) Reported by: folke
Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke
(license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by
folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
uploaded by folke (license 626) ........
* main/pbx.c, main/frame.c, /, channels/chan_features.c,
include/asterisk/stringfields.h, apps/app_voicemail.c,
main/cli.c: Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
| 3 lines Ensure that Asterisk builds with --enable-dev-mode,
even on the latest gcc and glibc. ........
2008-12-01 23:37 +0000 [r160170-160172] Sean Bright <sean.bright@gmail.com>
* main/manager.c, /: Merged revisions 159976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
| 3 lines Get rid of the useless format string and argument in
the Bogus/ manager channelname. Noted by kpfleming and name
Bogus/manager suggested by eliel ........
* channels/chan_phone.c: Silence a build warning.
(chan_phone.c:810: warning: value computed is not used)
* utils/smsq.c: Pay attention to the return value of system(), even
if we basically ignore it.
2008-12-01 21:23 +0000 [r160097] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Use AST_EXT_LIB_SETUP before using
AST_EXT_LIB_CHECK or bad things happen.
2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons <eliels@gmail.com>
* configs/cli_permissions.conf.sample (added), configure,
include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/cli.h, include/asterisk/_private.h, CHANGES,
main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on
cli_permissions.conf configuration file, we are able to permit or
deny cli commands based on some patterns and the local user and
group running rasterisk. (Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue
#11123) Reported by: eliel Tested by: eliel, IgorG, Laureano,
otherwiseguy, mvanbaak
2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 160003 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01
Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to
iax2_setoption(), as well, since they both have the potential to
send control frames in the middle of call setup. We have to wait
until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will
consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
........
* /, .cleancount: Merged revisions 159900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
| 2 lines Force a "make clean" to avoid a bizarre build issue ...
........
2008-12-01 14:09 +0000 [r159898] Michiel van Baak <michiel@vanbaak.info>
* main/manager.c, /: Merged revisions 159897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
| 4 lines make manager compile on OpenBSD. The last (10th)
argument to ast_channel_alloc here should be a pointer and NULL
is not really a pointer. ........
2008-11-29 18:33 +0000 [r159853] Tilghman Lesher <tlesher@digium.com>
* apps/app_readexten.c: Allow the '#' sign to exist within an
extension (inspired by issue #13330)
2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c,
utils/frame.c, include/asterisk/astmm.h, configure,
include/asterisk/compat.h, main/features.c,
include/asterisk/module.h, main/logger.c,
include/asterisk/dlinkedlists.h, main/dns.c,
include/asterisk/utils.h, include/asterisk/devicestate.h,
channels/chan_sip.c, include/asterisk/dundi.h,
include/asterisk/enum.h, configure.ac, channels/chan_agent.c,
include/asterisk/config.h, utils/astman.c,
include/asterisk/cli.h, include/asterisk/channel.h,
include/jitterbuf.h, include/asterisk/manager.h,
utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c,
include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h,
main/srv.c, channels/chan_misdn.c,
include/asterisk/linkedlists.h, main/event.c,
include/asterisk/lock.h, include/asterisk/strings.h,
utils/extconf.c, makeopts.in, include/asterisk/stringfields.h,
main/xmldoc.c, utils/check_expr.c: incorporates r159808 from
branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
2008) | 7 lines update dev-mode compiler flags to match the ones
used by default on Ubuntu Intrepid, so all developers will see
the same warnings and errors since this branch already had some
printf format attributes, enable checking for them and tag
functions that didn't have them format attributes in a consistent
way
------------------------------------------------------------------------
in addition: move some format attributes from main/utils.c to the
header files they belong in, and fix up references to the
relevant functions based on new compiler warnings
* Makefile, funcs/func_sprintf.c (added), main/Makefile,
channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt,
res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile:
we can now build with -Wformat=2, which found a couple of real
bugs because SPRINTF() use non-literal format strings (which
cannot be checked), move it into its own module so the rest of
func_strings can benefit from format string checking
2008-11-28 14:20 +0000 [r159734] Michiel van Baak <michiel@vanbaak.info>
* res/Makefile: Make res_config_ldap compile with the official
OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define
from their source and since we are using a couple of deprecated
function calls we should define it with a CFLAG. Tested by me on
OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps
compiling. It shouldn't break, we only define the LDAP_DEPRECATED
with this which is what all 2.2.X and older versions of OpenLDAP
did in their own tree.
2008-11-27 20:29 +0000 [r159701] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Removed duplicate code
2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant <russell@digium.com>
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
* main/pbx.c: Make a formatting change to test a new post-commit
hook for reviewboard. http://reviewboard.digium.com/r/65/
2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h, configure,
include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen,
autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c,
apps/app_stack.c, include/asterisk/optional_api.h (added):
improve handling of API calls provided by loaded modules through
use of some GCC features; this makes app_stack's usage of AGI
APIs even cleaner, and will allow it to work 'as expected' either
with or without res_agi being loaded reviewed at
http://reviewboard.digium.com/r/62
* main/manager.c, CHANGES: add support for event suppression for
AMI-over-HTTP
2008-11-26 19:57 +0000 [r159554] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Add some necessary hangup commands in the case
that forwarding a call fails 1) Hang up the original destination
if the local channel cannot be requested. 2) Hang up the local
channel (in addition to the original destination) if ast_call
fails when calling the newly created local channel. This prevents
channels from sticking around forever in the case of a botched
call forward (e.g. to an extension which does not exist). (closes
issue #13764) Reported by: davidw Patches: 13764_v2.patch
uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
2008-11-26 19:08 +0000 [r159534] Kevin P. Fleming <kpfleming@digium.com>
* agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
Makefile.rules: Merged revisions 159476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
2008) | 7 lines simplify (and slightly bug-fix) the recent
developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
removes dependency files for .i files that are created in
COMPILE_DOUBLE mode ........
2008-11-26 18:33 +0000 [r159475] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c: If the config file does not exist, then the first
use crashes Asterisk. (closes issue #13848) Reported by:
klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64)
Tested by: blitzrage
2008-11-26 14:58 +0000 [r159437] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Don't allow for configuration options to
overwrite options set via channel variables on a reload. (closes
issue #13921) Reported by: davidw Patches: 13921.patch uploaded
by putnopvut (license 60) Tested by: davidw
2008-11-26 03:18 +0000 [r159402] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Always parse arguments in park_call_exec so that
app_args is valid. This prevents a crash when executing Park from
the dialplan with no arguments.
2008-11-25 23:03 +0000 [r159360] Steve Murphy <murf@digium.com>
* main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159316 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
15 lines (closes issue #12694) Reported by: yraber Patches:
12694.2nd.diff uploaded by murf (license 17) Tested by: murf,
laurav Thanks to file (Joshua Colp) for his IAX fix. the change
to cdr.c allows no-answer to percolate up into CDR's, and feels
like the right place to locate this fix; if BUSY is done here,
no-answer should be, too. ........
2008-11-25 22:45 +0000 [r159276-159317] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option,
waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up. (closes issue #12382)
Reported by: one47 Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license
23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed
(license 463) Tested by: fleed
* /, channels/chan_iax2.c: Merged revisions 159269 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25
Nov 2008) | 7 lines Don't try to send a response on a NULL pvt.
(closes issue #13919) Reported by: barthpbx Patches:
chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
barthpbx ........
2008-11-25 21:49 +0000 [r159250] Mark Michelson <mmichelson@digium.com>
* apps/app_followme.c: Make the options for the general and
profiles more consistent for the "pls_hold_prompt" option. This
does not affect any released version of Asterisk, so there is no
need to update the CHANGES file for this. (closes issue #13893)
Reported by: eliel
2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 159246 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
(Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008)
| 7 lines Regression fix for last security fix. Set the iseqno
correctly. (closes issue #13918) Reported by: ffloimair Patches:
20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
Tested by: ffloimair ........ ................
* pbx/pbx_realtime.c: Don't actually do anything with a negative
priority, because we ignore it in the result, anyway.
* main/pbx.c: Don't limit the length of the hint at the final step
(from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80
chars max). This will allow more channels to be used in a single
hint.
2008-11-25 16:18 +0000 [r159093] Terry Wilson <twilson@digium.com>
* apps/app_festival.c: Add missing variable declaration for PPC
code
2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher <tlesher@digium.com>
* apps/app_readexten.c: Copyright clarification; also, have
variable set to "t" or "i" on timeout or invalid extension,
respectively. (closes issue #13944) Reported by: chappell
* channels/chan_usbradio.c, /, configure,
include/asterisk/autoconfig.h.in, configure.ac,
channels/xpmr/xpmr.c, apps/app_rpt.c: Merged revisions 159025 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008)
| 3 lines System call ioperm is non-portable, so check for its
existence in autoconf. (Closes issue #13863) ........
2008-11-25 03:49 +0000 [r158992] Terry Wilson <twilson@digium.com>
* channels/chan_usbradio.c: Make chan_usbradio compile under dev
mode
2008-11-25 01:01 +0000 [r158959] Sean Bright <sean.bright@gmail.com>
* funcs/func_dialgroup.c, channels/chan_sip.c,
include/asterisk/astobj2.h, res/res_phoneprov.c,
main/taskprocessor.c, channels/chan_console.c,
channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c,
main/config.c, main/manager.c, res/res_timing_pthread.c,
main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c,
res/res_clialiases.c: This is basically a complete rollback of
r155401, as it was determined that it would be best to maintain
API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to
ao2_callback() except that it allows you to pass arbitrary data
to the callback. Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson <mnicholson@digium.com>
* main/file.c: Fix compiling in dev mode.
* UPGRADE.txt, apps/app_queue.c: Make the Join event from app_queue
use CallerIDNum insead of CallerID for indicating the callerid
number just like the rest of asterisk. (closes issue #13883)
Reported by: davidw
* main/manager.c, res/res_agi.c, include/asterisk/manager.h: Added
EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes
issue #13873) Reported by: fnordian Patches: ami_agievent.patch
uploaded by fnordian (license 110)
2008-11-24 21:52 +0000 [r158857] Tilghman Lesher <tlesher@digium.com>
* main/dsp.c: Add a bit of documentation (thanks, I-MOD) on what
the silence threshold constant actually does and what values are
valid for it.
2008-11-24 21:27 +0000 [r158851] Matthew Nicholson <mnicholson@digium.com>
* main/file.c: Make ast_streamfile() check the result of
ast_openstream() before doing anything with it. (closes issue
#13955) Reported by: chris-mac
2008-11-24 18:11 +0000 [r158808] Terry Wilson <twilson@digium.com>
* apps/app_minivm.c: This patch adds a new application for sending
MWI to phones via Asterisk's event subsystem. Also, the minivm
documentation is all converted to use xmldocs. (closes issue
#13946) Reported by: Marquis Patches:
minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32)
Tested by: otherwiseguy, Marquis
2008-11-23 03:36 +0000 [r158754-158756] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c, configs/sip.conf.sample: If you enabled
'notifycid' one of the limitations is that the calling channel is
only found if it dialed the extension that was subscribed to. You
can now specify 'ignore-context' for the 'notifycid' option in
sip.conf which will, as it's value implies, ignore the current
context of the caller when doing the lookup.
* channels/chan_sip.c: No need to use a separate structure for this
since we can just pass our sip_pvt pointer in directly.
2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_realtime.c: last commit worked on OpenBSD but still
generated warning on Ubuntu. Initialise a variable so
--enable-dev-mode does not complain
* channels/chan_skinny.c: dont send reorder tone after a device is
hungup if a dialout is abandoned or failed. Without this reorder
tone will play after hangup and both wedhorn's and my wife have
threatened to use an axe on our asterisk box (closes issue
#13948) Reported by: wedhorn Patches: switch.diff uploaded by
wedhorn (license 30)
* channels/chan_skinny.c: Add Media Source Update to skinny's
control2str (issue #13948)
* channels/chan_skinny.c: fix a very occasional core dump in
chan_skinny found by wedhorn. (issue #13948)
* funcs/func_realtime.c: make this compile under devmode
2008-11-21 23:40 +0000 [r158606] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 158603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) |
11 lines In reference to the fix made for 13871, I was merging
the fix into 1.6.0 and realized I missed the code in the h-exten
block, and didn't catch it because my test case had the h-exten
commented out. So, this corrects the code I missed, as a
preventative against another crash report. Tested with the
h-exten defined, all is well. ........
2008-11-21 23:33 +0000 [r158602-158605] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Allow space within an extension, when the space is
within a character class. (requested by lmadsen on -dev, patch by
me)
* main/pbx.c, /: Merged revisions 158600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
| 5 lines The passed extension may not be the same in the list as
the current entry, because we strip spaces when copying the
extension into the structure. Therefore, use the copied item to
place the item into the list. (found by lmadsen on -dev, fixed by
me) ........
2008-11-21 22:12 +0000 [r158540] Russell Bryant <russell@digium.com>
* /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
158539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
| 2 lines When compiling with DEBUG_THREADS, report the real
file/func/line for ao2_lock/ao2_unlock ........
2008-11-21 21:47 +0000 [r158484] Steve Murphy <murf@digium.com>
* /, main/features.c: Merged revisions 158483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
11 lines (closes issue #13871) Reported by: mdu113 This one is
totally my fault. The code doesn't even create a bridge CDR if
the channel CDR has POST_DISABLED. I didn't check for that at the
end of the bridge. Fixed with a few small insertions. Tested.
Looks good. No cdr generated, no crash, no unnecc. data objects
created either. ........
2008-11-21 21:06 +0000 [r158482] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Fix for #13963. Make physical channel
mapping unconfigured default
2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming <kpfleming@digium.com>
* UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt,
CHANGES: as suggested by jtodd, document the purposes of the
CHANGES and UPGRADE files
2008-11-21 19:40 +0000 [r158414] Jason Parker <jparker@digium.com>
* main/manager.c: Make sure we add the Event header for
CoreShowChannels. (closes issue #13334) Reported by: srt Patches:
13334_missing_event_header_in_core_show_channel.diff uploaded by
srt (license 378)
2008-11-21 17:08 +0000 [r158374] Terry Wilson <twilson@digium.com>
* cdr/cdr_csv.c: Reloading the config and having no changes still
initialized some settings to 0. Initialize settings after doing
all of the cfg checks. (closes issue #13942) Reported by: davidw
Patches: cdr_diff.txt uploaded by otherwiseguy (license 396)
Tested by: davidw
2008-11-21 15:53 +0000 [r158315] Doug Bailey <dbailey@digium.com>
* channels/chan_sip.c: Add fix to prevent crash during reload if
there is an outstanding MWI registration message pending.
2008-11-21 01:22 +0000 [r158230-158266] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use a more expressive constant for a 64-bit
scanned int
* channels/chan_sip.c: Use some magic constants to get the right
size for this sscanf statement. Thanks Richard!
* channels/chan_sip.c: Fix the build for 32-bit systems. %lu is
only 32-bits on 32-bit systems, so we need to use %llu instead.
Of course %llu is 128-bits on 64-bit systems, so we have to cast
to unsigned long long. No harm, but it's sure annoying.
* channels/chan_sip.c: Change the remote user agent session version
variable from an int to a uint64_t. This prevents potential
comparison problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could not
properly place a call on hold since the version in the SDP of the
re-INVITE to place the call on hold greatly exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends using
an NTP timestamp for the version (which is a 64-bit number).
(closes issue #13531) Reported by: sgofferj Patches: 13531.patch
uploaded by putnopvut (license 60) Tested by: sgofferj
2008-11-20 19:41 +0000 [r158188] Sean Bright <sean.bright@gmail.com>
* res/ael/pval.c: Fix one case where the application argument was
not converted from a pipe to a comma. This was causing problems
with switch statements with empty expressions. (closes issue
#13901) Reported by: smurfix Patches: 20081118_bug13901.diff
uploaded by seanbright (license 71) Tested by: seanbright
Reviewed by: murf
2008-11-20 18:20 +0000 [r158082-158133] Mark Michelson <mmichelson@digium.com>
* include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c,
main/file.c, include/asterisk/frame.h: Merged revisions 158072
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20
Nov 2008) | 2 lines Begin on a crusade to end trailing
whitespace! ........
* /, channels/chan_sip.c: Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
2008) | 16 lines We don't handle 4XX responses to BYE well.
According to section 15 of RFC 3261, we should terminate a dialog
if we receive a 481 or 408 in response to our BYE. Since I am
aware of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog. (closes issue
#12994) Reported by: pabelanger Patches: 12994.patch uploaded by
putnopvut (license 60) Closes AST-129 ........
2008-11-20 17:53 +0000 [r158078] Ryan Brindley <rbrindley@digium.com>
* main/config.c: more formatting corrections :: one line for loops
and if statements still need {}
2008-11-20 17:48 +0000 [r158072] Terry Wilson <twilson@digium.com>
* cdr/cdr_sqlite3_custom.c, cdr/cdr_sqlite.c, cdr/Makefile,
cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
cdr/cdr_csv.c: Begin on a crusade to end trailing whitespace!
2008-11-20 17:46 +0000 [r158070] Ryan Brindley <rbrindley@digium.com>
* main/config.c: formatting changes :: one line for loops and if
statements should have {}
2008-11-20 17:39 +0000 [r158066] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158053
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
2008) | 12 lines Make sure to set the hangup cause on the calling
channel in the case that ast_call() fails. For incoming SIP
channels, this was causing us to send a 603 instead of a 486 when
the call-limit was reached on the destination channel. (closes
issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded
by putnopvut (license 60) Tested by: blitzrage ........
2008-11-20 17:37 +0000 [r158062] Jeff Peeler <jpeeler@digium.com>
* main/file.c: (closes issue #12929) Reported by: snyfer This
handles the case for a zero length file to attempt to be
streamed. Instead of failing from not playing any data, go ahead
and return success as ast_streamfile should consider playing
nothing a success when there is nothing to play.
2008-11-20 17:37 +0000 [r158061] Jason Parker <jparker@digium.com>
* README: Whitespace fix
2008-11-20 00:08 +0000 [r157974] Kevin P. Fleming <kpfleming@digium.com>
* main/stdtime, /, main/db1-ast/hash, codecs/gsm/Makefile,
Makefile.moddir_rules, main/db1-ast/db, channels/misdn,
main/db1-ast/mpool, res/ais, res/Makefile, pbx/Makefile,
Makefile.rules, res/snmp, main/stdtime/Makefile, codecs/gsm/src,
main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno,
res/ael, pbx/ael, channels, main/db1-ast/Makefile: Merged
revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
2008) | 7 lines the gcc optimizer frequently finds broken code
(use of uninitalized variables, unreachable code, etc.), which is
good. however, developers usually compile with the optimizer
turned off, because if they need to debug the resulting code,
optimized code makes that process very difficult. this means that
we get code changes committed that weren't adequately checked
over for these sorts of problems. with this build system change,
if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is
turned on, when a source file is compiled it will actually be
preprocessed (into a .i or .ii file), then compiled once with
optimization (with the result sent to /dev/null) and again
without optimization (but only if the first compile succeeded, of
course). while making these changes, i did some cleanup work in
Makefile.rules to move commonly-used combinations of flag
variables into their own variables, to make the file easier to
read and maintain ........
2008-11-20 00:06 +0000 [r157973] Terry Wilson <twilson@digium.com>
* res/res_timing_timerfd.c: Fix compiling
2008-11-19 23:30 +0000 [r157906-157940] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add a space to the output
* apps/app_queue.c: Add a RES_NOT_DYNAMIC case for the CLI command
'queue remove member'
* CHANGES: Commit CHANGES change I promised when submitting
res_timing_timerfd
2008-11-19 22:01 +0000 [r157893] Tilghman Lesher <tlesher@digium.com>
* CHANGES: Add info about REALTIME_FIELD and REALTIME_HASH
2008-11-19 21:55 +0000 [r157874] Mark Michelson <mmichelson@digium.com>
* res/res_timing_timerfd.c: Cast this value since a uint64_t is not
the same as an unsigned long long on a 64-bit machine. Reported
by kpfleming on IRC
2008-11-19 21:54 +0000 [r157870] Tilghman Lesher <tlesher@digium.com>
* funcs/func_realtime.c: Two new functions, REALTIME_FIELD, and
REALTIME_HASH, which should make querying realtime from the
dialplan a little more consistent and easy to use. The original
REALTIME function is preserved, for those who are already
accustomed to that interface. (closes issue #13651) Reported by:
Corydon76 Patches: 20081119__bug13651__2.diff.txt uploaded by
Corydon76 (license 14) Tested by: blitzrage, Corydon76
2008-11-19 19:37 +0000 [r157820] Mark Michelson <mmichelson@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, res/res_timing_pthread.c,
configure.ac, res/res_timing_dahdi.c, res/res_timing_timerfd.c
(added), makeopts.in: Merge the changes from the
res_timing_timerfd branch. This provides a new timing interface.
In order to use it, you must be running a Linux with a kernel
version of 2.6.25 or newer and glibc 2.8 or newer. This timing
interface is a good alternative if a timing source is necessary
(e.g. for IAX trunking) but DAHDI is otherwise unnecessary for
the system. For now, this commit contains the actual work done in
the res_timing_timerfd branch. There are no notices in the README
or CHANGES files yet, but they will be added in my next commit.
The timing API of Asterisk also needs to have a bit of work done
with regards to choosing which timing interface to use. This
commit makes the choice a build-time decision, by only allowing
one of the timer interfaces to be chosen in menuselect. It would
be preferable if the choice could be made at run-time, however.
The preferred timing interface could be loaded and tested, and if
it does not work, choice number two may be used instead. That
sort of thing. That is beyond the scope of work in this branch
though.
2008-11-19 19:25 +0000 [r157818] Terry Wilson <twilson@digium.com>
* channels/chan_vpb.cc, cdr/cdr_sqlite3_custom.c,
channels/iax2-provision.c, cdr/cdr_adaptive_odbc.c,
cdr/cdr_pgsql.c, cdr/cdr_radius.c, cdr/cdr_tds.c,
channels/misdn_config.c, cdr/cdr_csv.c, channels/chan_usbradio.c,
channels/chan_skinny.c, main/logger.c, res/ais/evt.c,
pbx/pbx_dundi.c, cdr/cdr_odbc.c, cdr/cdr_custom.c,
cdr/cdr_manager.c, main/xmldoc.c, res/res_clialiases.c: Fix
checking for CONFIG_STATUS_FILEINVALID so that modules don't
crash upon trying to parse an invalid config
2008-11-19 18:28 +0000 [r157784] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Add check for t38_terminal_init in
spandsp (not found in 0.0.6, so it should fail reasonably)
(closes issue #13473) Reported by: genie Patches:
20080916__bug13473.diff.txt uploaded by Corydon76 (license 14)
2008-11-19 13:45 +0000 [r157706-157743] Kevin P. Fleming <kpfleming@digium.com>
* res/res_agi.c: correct small bug introduced during API conversion
* UPGRADE.txt, UPGRADE-1.6.txt: move relevant entries into
UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches
* include/asterisk/agi.h, res/res_agi.c, UPGRADE.txt,
UPGRADE-1.6.txt (added), apps/app_stack.c: make some corrections
to the ast_agi_register_multiple(), ast_agi_unregister_multiple()
and ast_agi_fdprintf() API calls to be consistent with API
guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the
new UPGRADE.txt contain information about upgrading between
Asterisk 1.6 releases
2008-11-19 05:37 +0000 [r157675] Terry Wilson <twilson@digium.com>
* configs/cdr_adaptive_odbc.conf.sample: Comment out config line
that is in a commented out context
2008-11-19 01:02 +0000 [r157639] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/logger.h, main/logger.c, main/utils.c,
include/asterisk/strings.h: Starting with a change to ensure that
ast_verbose() preserves ABI compatibility in 1.6.1 (as compared
to 1.6.0 and versions of 1.4), this change also deprecates the
use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already
require C99 for other purposes, so this isn't really a big change
anyway. This change also simplifies some of the core ast_str_*
functions.
2008-11-19 00:59 +0000 [r157632] Mark Michelson <mmichelson@digium.com>
* main/astmm.c: If malloc returns NULL, we need to return NULL
immediately or else Asterisk will crash when attempting to
dereference the NULL pointer (closes issue #13858) Reported by:
eliel Patches: astmm.c.patch uploaded by eliel (license 64)
2008-11-19 00:27 +0000 [r157600] Sean Bright <sean.bright@gmail.com>
* Makefile, build_tools/make_version, configure, configure.ac,
build_tools/make_buildopts_h, makeopts.in: Fix a few build
problems on Solaris (and check for an md5 utility in configure
instead of the icky loop I was doing before). (closes issue
#13842) Reported by: snuffy Patches: bug13842_20081106.diff
uploaded by snuffy (license 35) 13842.diff uploaded by seanbright
(license 71) Tested by: snuffy
2008-11-18 23:59 +0000 [r157496-157592] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c: This change prevents a crash from
occurring if res_musiconhold.so is unloaded and then Asterisk is
stopped. The problem was that we are not unregistering the
ast_moh_destroy function at exit. (closes issue #13761) Reported
by: eliel Patches: res_musiconhold.c.patch uploaded by eliel
(license 64)
* Makefile: Add some missing $(DESTDIR)s to the bininstall target
of the Makefile. (closes issue #13875) Reported by: pabelanger
Patches: Makefile.155928 uploaded by pabelanger (license 224)
* apps/app_voicemail.c: Fix the logic for when delete=yes when IMAP
storage is in use so that the message is deleted from both local
and IMAP storage. (closes issue #13642) Reported by: jaroth
Patches: deleteyes.patch uploaded by jaroth (license 50)
* channels/chan_sip.c: Merged revisions 157503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
2008) | 13 lines Add some missing invite state changes necessary
in the sip_write function. Not setting the invite state correctly
on the call was resulting in the Record application leaving empty
files. I also have updated the doxygen comment next to the
declaration of the INV_EARLY_MEDIA constant to reflect that we
also use this state when we *send* a 18X response to an INVITE.
(closes issue #13878) Reported by: nahuelgreco Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
(license 162) Tested by: putnopvut ........
* channels/chan_sip.c: Based on Russell's advice on the
asterisk-dev list, I have changed from using a global lock in
update_call_counter to using the locks within the sip_pvt and
sip_peer structures instead.
2008-11-18 21:15 +0000 [r157460-157463] Jason Parker <jparker@digium.com>
* Makefile: Remove echo line that is unnecessary (Thanks
seanbright).
* contrib/init.d/rc.archlinux.asterisk: Make this executable
* Makefile, contrib/init.d/rc.archlinux.asterisk (added): Add init
script for ArchLinux (closes issue #13667) Reported by: sherif
Patches: archlinux_rc_makefile.patch uploaded by sherif (license
591) archlinux_rc_makefile-2.patch uploaded by mvanbaak (license
7)
2008-11-18 20:23 +0000 [r157427] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: * Add a lock to be used in the
update_call_counter function. * Revert logic to mirror 1.4's in
the sense that it will not allow the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP
peer to appear to be busy forever. (closes issue #13668) Reported
by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by
wolfelectronic (license 586)
2008-11-18 19:16 +0000 [r157366] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 157365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008)
| 6 lines (closes issue #13899) Reported by: akkornel This fix is
the result of a bug fix in ast_app_separate_args r124395. If an
argument does not exist it should always be set to a null string
rather than a null pointer. ........
2008-11-18 18:31 +0000 [r157306] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, channels/chan_local.c, /, main/features.c,
include/asterisk/channel.h, apps/app_followme.c: Merged revisions
157305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
2008) | 12 lines Fix a crash in the end_bridge_callback of
app_dial and app_followme which would occur at the end of an
attended transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due to a
masquerade. This commit adds a "fixup" callback to the
bridge_config structure to allow for end_bridge_callback_data to
be changed in the case that a new channel pointer is needed for
the end_bridge_callback. ........
2008-11-18 18:07 +0000 [r157302] Steve Murphy <murf@digium.com>
* main/config.c: (closes issue #13420) Reported by: alex70 Patches:
13420.13539.patch uploaded by murf (license 17) Tested by: murf,
awk This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only generated when
category had no previous comments. The other problem: Insertions
could get the line-numbering out of whack and generate negative
line numbers, causing chunks of line numbers to be emitted, on
the scale of the number of lines up to that point in the file. In
such cases, abort the looping, and all is well.
2008-11-17 22:25 +0000 [r157253] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c: Can't use items duplicated off the stack frame
in an element returned from a function: in these cases, we have
to use the heap, or garbage will result. (closes issue #13898)
Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt
uploaded by Corydon76 (license 14) Tested by: alecdavis
2008-11-15 19:51 +0000 [r157105-157167] Kevin P. Fleming <kpfleming@digium.com>
* Makefile.rules: ensure that if a .i file (preprocessed source) is
present, the .o file is made from it, not from the .c file (this
only works because GNU makes respects the order the rules are
defined)
* Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
revisions 157162-157163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
2008) | 1 line dist-clean should remove dependency information
files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
+0100 (Sat, 15 Nov 2008) | 1 line when an individual directory
dist-clean is run, run clean in that directory first, and when
running top-level dist-clean, do not run subdirectory clean
operations twice ........
* /, contrib/asterisk-ng-doxygen: Merged revisions 157104 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov
2008) | 13 lines major update to doxygen configuration file: 1)
update to doxygen 1.5.x style file, as used in trunk 2) tell
doxygen where are header files are, so include-file processing
can be done 3) make all macros that are used to define
variables/functions be expanded, so that doxygen will properly
document the resulting variable/function 4) make all macros that
are used to provide the contents of a variable (structure) be
expanded, so that doxygen will be able to document the resulting
fields 5) suppress compiler attributes (__attribute__(xxx)) from
being seen by doxygen, so it will properly match up function
definition and usage (for an example of th effect of this, look
at the doxygen docs for ast_log() from before and afte this
commit) ........
2008-11-15 15:37 +0000 [r157073] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: Avoid a not needed cast, making code more
readable.
2008-11-15 04:25 +0000 [r157039-157041] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, main/features.c, main/taskprocessor.c: Fix a
few more places where the case insensitive hash should be used
since the comparison is case insensitive.
* channels/chan_console.c: Use the new case insensitive hash
function for console interfaces. The comparison function is case
insensitive.
2008-11-14 22:36 +0000 [r157006] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
Allow setting static values in CDRs
2008-11-14 21:19 +0000 [r156962] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Revision 155513 of chan_sip.c in trunk
inadvertently removed a very important line to set the "len"
field for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk could do
no meaningful processing of anything SIP-related
2008-11-14 17:35 +0000 [r156916-156918] Terry Wilson <twilson@digium.com>
* res/res_phoneprov.c: Cleanup whitespace issues
* res/res_phoneprov.c: Use Mark's new ast_str_case_hash function
instead of jumping through hoops to do insensitive case lookups
2008-11-14 17:02 +0000 [r156911] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Ping is missing the standard double-newline after
the event. (closes issue #13903) Reported by: kebl0155
2008-11-14 16:53 +0000 [r156883] Mark Michelson <mmichelson@digium.com>
* UPGRADE.txt, include/asterisk/strings.h, apps/app_queue.c: Fix
some refcounting in app_queue.c and change the hashing used by
app_queue.c to be case-insensitive. This is accomplished by
adding a new case-insensitive hashing function. This was
necessary to prevent bad refcount errors (and potential crashes)
which would occur due to the fact that queues were initially read
from the config file in a case-sensitive manner. Then, when a
user issued a CLI command or manager action, we allowed for
case-insensitive input and used that input to directly try to
find the queue in the hash table. The result was either that we
could not find a queue that was input or worse, we would end up
hashing to a completely bogus value based on the input. This
commit resolves the problem presented in issue #13703. However,
that issue was reported against 1.6.0. Since this fix introduces
a behavior change, I am electing to not place this same fix in to
the 1.6.0 or 1.6.1 branches, and instead will opt for a change
which does not change behavior.
2008-11-14 16:34 +0000 [r156874] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Remove some useless locking and make sure
we hangup channels on a link when we get a GRS.
2008-11-14 15:20 +0000 [r156817] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 156816 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri,
14 Nov 2008) | 10 lines If the prompt to reenter a voicemail
password timed out, it resulted in the password not being saved,
even if the input matched what you gave when first prompted to
enter a new password. This is because the return value of
ast_readstring was checked, but not checked properly. This bug
was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it! ........
2008-11-14 00:43 +0000 [r156690-156756] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_while.c: Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
| 6 lines ast_waitfordigit() requires that the channel be up, for
no good logical reason. This prevents While/EndWhile from working
within the "h" extension. Reported by: jgalarneau (for ABE C.2)
Fixed by: me ........
* main/manager.c, /: Merged revisions 156688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
| 7 lines Provide more space for all the data which can appear in
an originating channel name. (closes issue #13398) Reported by:
bamby Patches: manager.c.diff uploaded by bamby (license 430)
........
2008-11-13 19:17 +0000 [r156649] Jeff Peeler <jpeeler@digium.com>
* main/pbx.c: (closes issue #13891) Reported by: smurfix This
reverts a change I made in 116297. At the time it seemed the
change was required to solve an issue with attempting a transfer
but then letting it timeout without dialing any digits. However,
I didn't realize that having an empty extension was possible. I'm
removing the immediate return that was added in
pbx_find_extension if the extension is null.
2008-11-13 19:10 +0000 [r156647] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Command offsets were not changed correctly
when the command syntax for 'pri set debug' was changed from 'pri
debug'.
2008-11-13 17:07 +0000 [r156612] Mark Michelson <mmichelson@digium.com>
* configure, autoconf/ast_c_compile_check.m4: Kevin sent a note
indicating that this change is not necessary, so I am reverting
it
2008-11-13 15:46 +0000 [r156535-156575] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_meetme.c, doc/appdocsxml.dtd, main/xmldoc.c: Introduce
XML documentation for: - MeetMe() - MeetMeCount() -
MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk()
- Add an attribute to optionlist 'hasparams' with the same
functionality as the one we have in <parameter> and <argument>
(the DTD was updated) - Fix a leak when getting an attribute
while parsing an <optionlist>.
* main/xmldoc.c: Fix a typo introduced when changing
xmldoc_has_arguments() to xmldoc_has_inside() we need to pass the
name of the node that we are looking for.
* include/asterisk/xml.h, include/asterisk/xmldoc.h, main/xmldoc.c:
Remove trailing whitespaces using ':%s/\s\+$//' pointed by
seanbright on #asterisk-dev
2008-11-12 23:13 +0000 [r156443] Sean Bright <sean.bright@gmail.com>
* /: Use the reviewboard:url SVN property so post-review will work
without modification.
2008-11-12 21:34 +0000 [r156388] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, /: Merged revisions 156386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
| 5 lines When using call limits under 1 second, infinite call
lengths are allowed, instead. (closes issue #13851) Reported by:
ruddy ........
2008-11-12 20:27 +0000 [r156355] Eliel C. Sardanons <eliels@gmail.com>
* res/res_clialiases.c: - Make alias->real_cmd point to the
allocated space outside alias->alias. - Register the aliased cli
command (or we will not alias anything). - Use ARRAY_LEN() when
possible.
2008-11-12 19:47 +0000 [r156299] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 156297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
18 lines It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with jsmith on
#asterisk-dev; he was teaching a class and disconcerted that this
published rule was not being followed, with patterns _NXX,
_[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
have been. This change, tested on these 3 patterns now picks the
proper one. However, this change may surprise users who set up
dialplans based on previous behavior, which has been there for
what, 2 and half years or so now. ........
2008-11-12 19:38 +0000 [r156298] Russell Bryant <russell@digium.com>
* res/res_clialiases.c: Fix a bug caused by using sizeof(pointer)
instead of sizeof(the struct)
2008-11-12 19:28 +0000 [r156295] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 156294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
| 6 lines If the SLA thread is not started, then reload causes a
memory leak. (closes issue #13889) Reported by: eliel Patches:
app_meetme.c.patch uploaded by eliel (license 64) ........
2008-11-12 19:11 +0000 [r156290] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 156289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
| 3 lines For whatever reason, gcc only warned me about the
possible use of an uninitialized variable when compiling 1.6.1.
........
2008-11-12 18:55 +0000 [r156243] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 156229 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12
Nov 2008) | 11 lines Revert revision 132506, since it
occasionally caused IAX2 HANGUP packets not to be sent, and
instead, schedule a task to destroy the iax2 pvt structure 10
seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed. (closes issue
#13645) Reported by: dzajro Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
........
2008-11-12 18:32 +0000 [r156228] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 156178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
| 8 lines (closes issue #13173) Reported by: pep This change adds
an announce_thread responsible for playing announcements to an
existing conference. This allows all announcing to be immediately
stopped if necessary but more importantly allows other threads
that need to play something to not block. There are multiple
benefits to this, but the actual bug is for solving the scenario
for a channel to be unusable after hang up for the entire
duration of the parting announcement. The parting announcement
can be extremely long depending on what the user recorded upon
joining the conference. Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/ ........
2008-11-12 17:41 +0000 [r156169] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, /: Merged revisions 156167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
2008) | 7 lines When doing some tests, I was having a crash at
the end of every call if an attended transfer occurred during the
call. I traced the cause to the CDR on one of the channels being
NULL. murf suggested a check in the end bridge callback to be
sure the CDR is non-NULL before proceeding, so that's what I'm
adding. ........
2008-11-12 17:38 +0000 [r156166] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 156164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
| 7 lines Move the sanity check that makes sure "always fork" is
not set along with the console option to be after the code that
reads options from asterisk.conf. This resolves a situation where
Asterisk can start taking up 100% when misconfigured. (Thanks to
Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.) ........
2008-11-12 17:28 +0000 [r156162] Eliel C. Sardanons <eliels@gmail.com>
* main/xmldoc.c: - The paramname is a pointer allocated with
strdup() or malloc(), so, we need to free it with ast_free().
2008-11-12 15:33 +0000 [r156127] Mark Michelson <mmichelson@digium.com>
* configure, autoconf/ast_c_compile_check.m4: Add a couple of
AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing
calls were discovered when working on timerfd support in a
separate branch.
2008-11-12 13:43 +0000 [r156125] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c: Add XML documentation for AGI commands: - database
deltree - database get - exec - get data - get full variable
2008-11-12 06:46 +0000 [r156120] Michiel van Baak <michiel@vanbaak.info>
* main/udptl.c, main/pbx.c, channels/chan_sip.c,
configs/cli_aliases.conf.sample (added), include/asterisk/cli.h,
CHANGES, res/res_jabber.c, main/rtp.c, main/cli.c, main/cdr.c,
channels/chan_skinny.c, res/res_agi.c, pbx/pbx_ael.c,
pbx/pbx_dundi.c, funcs/func_devstate.c, main/asterisk.c,
channels/chan_mgcp.c, res/res_clialiases.c (added): This commit
does two things: - Add CLI aliases module to asterisk. - Remove
all deprecated CLI commands from the code Initial work done by
file. Junk-Y and lmadsen did a lot of work and testing to get the
list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module, see
cli_aliases.conf for more info about that. ok russellb@ via
reviewboard (closes issue #13735) Reported by: mvanbaak
2008-11-12 02:20 +0000 [r156051-156087] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c, doc/appdocsxml.dtd: - Add 'database del',
'database put' and 'set music' AGI commands XML documentation. -
Add to the DTD the possibility to put a parameter inside an
<enum>.
* include/asterisk/agi.h, res/res_agi.c, doc/appdocsxml.dtd,
main/xmldoc.c: Implement AGI XML documentation parsing functions.
A new <agi> element is used to describe the XML documentation. We
have the usual synopsis,syntax,description and seealso for AGI
commands. The CLI 'agi show commands' command was changed to show
all the documentation se ctions.
2008-11-11 23:32 +0000 [r156017-156018] Pari Nannapaneni <paripurnachand@digium.com>
* main/manager.c: changing comment style to conform coding
guidelines
* main/manager.c: Patch by Ryan Brindley -- Make sure that manager
refuses any duplicate 'new category' requests in updateconfig
2008-11-11 17:57 +0000 [r155967] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/strings.h: use some fancy compiler magic (thanks
to Matthew Woehlke on the gcc-help mailing list) to restore
type-safety to S_OR by going back to a macro, but preserve the
side-effect-safe usage of the macro arguments
2008-11-11 16:46 +0000 [r155934] Doug Bailey <dbailey@digium.com>
* res/res_phoneprov.c, phoneprov/polycom_line.xml: Add LINEKEYS
variable to allow for a user to set the number of keys assigned
to a line on a polycom phone
2008-11-11 16:07 +0000 [r155929] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove commentary from the issues list for
SIP TCP/TLS
2008-11-10 21:14 +0000 [r155863] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_agent.c: Merged revisions 155861 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon,
10 Nov 2008) | 14 lines Channel drivers assume that when their
indicate callback is invoked, that the channel on which the
callback was called is locked. This patch corrects an instance in
chan_agent where a channel's indicate callback is called directly
without first locking the channel. This was leading to some
observed locking issues in chan_local, but considering that all
channel drivers operate under the same expectations, the generic
fix in chan_agent is the right way to go. AST-126 ........
2008-11-10 21:12 +0000 [r155763-155862] Tilghman Lesher <tlesher@digium.com>
* res/res_realtime.c: Make documentation of update method match
documentation and update update2 method to match. Reported by:
atis, via -dev mailing list. Fixed by: me
* /, doc/valgrind.txt: Merged revisions 155803 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008)
| 1 line I got tired of saying this in every single bugnote
referring to this file. ........
* main/editline/readline.c: Fix memory leak when MALLOC_DEBUG is
enabled. (closes issue #13864) Reported by: eliel Patches:
readline.c.patch uploaded by eliel (license 64)
2008-11-10 13:53 +0000 [r155711] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, main/Makefile, include/asterisk/xmldoc.h (added),
include/asterisk/term.h, include/asterisk/_private.h,
main/asterisk.c, main/xmldoc.c (added): Move all the XML
documentation API from pbx.c to xmldoc.c. Export the XML
documentation API: ast_xmldoc_build_synopsis()
ast_xmldoc_build_syntax() ast_xmldoc_build_description()
ast_xmldoc_build_seealso() ast_xmldoc_build_arguments()
ast_xmldoc_printable() ast_xmldoc_load_documentation()
2008-11-09 16:30 +0000 [r155554-155671] Sean Bright <sean.bright@gmail.com>
* configs/chan_dahdi.conf.sample: Fix this as well. Pointed out by
tzafrir.
* configs/chan_dahdi.conf.sample: Fix some spelling errors, and
convert tabs to spaces.
* main/channel.c, channels/chan_sip.c, apps/app_directed_pickup.c,
main/features.c, include/asterisk/channel.h: In order to move
away from nested function use, some changes to the recently
introduced ast_channel_search_locked need to be made.
Specifically, the caller needs to be able to pass arbitrary data
which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
* apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
apps/app_followme.c, apps/app_queue.c: Merged revisions 155553
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
2008) | 6 lines Use static functions here instead of nested ones.
This requires a small change to the ast_bridge_config struct as
well. To understand the reason for this change, see the following
post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
........
2008-11-08 21:46 +0000 [r155513-155516] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, include/asterisk/strings.h: - Check for
failure when putting the packet in the ast_str - fix a spelling
error in a header file
* channels/chan_sip.c: Remove some code that is basically a no-op.
Code above this already ensures that the buffer is terminated.
2008-11-07 23:41 +0000 [r155467] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Set the invite state to INV_CANCELLED in a
place that makes more sense. Where it was set before, it was
impossible to actually delay sending a CANCEL if we had not yet
received a provisional response to an INVITE. (closes issue
#13626) Reported by: atis Patches: 13626.patch uploaded by
putnopvut (license 60) Tested by: atis
2008-11-07 22:39 +0000 [r155401] Sean Bright <sean.bright@gmail.com>
* main/manager.c, channels/chan_sip.c, funcs/func_dialgroup.c,
res/res_timing_pthread.c, include/asterisk/astobj2.h,
main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c,
channels/chan_iax2.c, main/astobj2.c, main/config.c: Add ability
to pass arbitrary data to the ao2_callback_fn (called from
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to
either of these mandates that the passed 'arg' is a hashable
object, making searching for an ao2 object based on outside
criteria difficult. Reviewed by Russell and Mark M. via
ReviewBoard: http://reviewboard.digium.com/r/36/
2008-11-07 22:28 +0000 [r155395-155399] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 155398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
| 7 lines Clarify error message. (closes issue #13809) Reported
by: denke Patches: 20081104__bug13809.diff.txt uploaded by
Corydon76 (license 14) Tested by: denke ........
* funcs/func_odbc.c: Two bugs relating to colnames found by
Marquis42 on #asterisk-dev
2008-11-07 21:14 +0000 [r155360] Mark Michelson <mmichelson@digium.com>
* configs/voicemail.conf.sample: Remove one more instance of the
sample configuration lying about what's possible. The tz cannot
be set in a context like this. It can only be set in the general
section or per-mailbox. Thanks to sasargen on #asterisk-dev for
pointing this out
2008-11-07 20:13 +0000 [r155324] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Send call release with unallocated cause
instead of normal call clearing, when invalid extension is
called. (closes issue #13408) Reported by: adomjan Patches:
chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487)
2008-11-07 16:18 +0000 [r155284] Sean Bright <sean.bright@gmail.com>
* include/asterisk/indications.h, res/res_indications.c,
main/indications.c: Convert open-coded linked list in indications
to the AST_LIST_* macros. This cleans the code up some and should
make it more maintainable as time goes on. Reviewed by Russell,
Kevin, Mark M., and Tilghman via ReviewBoard:
http://reviewboard.digium.com/r/34/
2008-11-07 15:52 +0000 [r155282] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: stringfields conversion for struct sip_peer,
as requested :-)
2008-11-07 15:42 +0000 [r155241-155264] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove a bogus ast_free() that Kevin
noticed. This was probably just left over from pre-astobj2ified
chan_sip.
* include/asterisk/astobj2.h: Clarify which part of OBJ_MULTIPLE is
not implemented, and under what case it is perfectly fine to use.
(Inspired by a question I received about my last commit.)
* main/pbx.c, channels/chan_sip.c: Fix some code in chan_sip that
was intended to unlink multiple objects from a container. The
OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
2008-11-07 03:09 +0000 [r155206] Kevin P. Fleming <kpfleming@digium.com>
* pbx/pbx_config.c: correct logic error noticed by rmudgett
(thanks!)
2008-11-07 03:02 +0000 [r155175-155204] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c: If 'asterisk.conf' is not found, instead of giving
up, load documentation for the 'en_US' language (fix my last
commit).
* main/pbx.c: Fix an asterisk crash if no asterisk.conf
configuration file is present.
2008-11-06 22:49 +0000 [r155066-155121] Kevin P. Fleming <kpfleming@digium.com>
* res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: don't
blindly assume that Darwin and Cygwin need GLOB_ABORTED defined;
only define it if it is not already defined
* pbx/pbx_config.c: coding style/guidelines cleanup, plus use new
side-effect safe S_OR
* include/asterisk/strings.h: make S_OR and S_COR safe to use even
if the parameters are function calls or have side effects. it
still bothers me that these are called S_OR and not something
like ast_string_or, but that's water over the bridge
* channels/chan_dahdi.c: put ifdef protection around the rest of
the libpri function calls that were added at the same time as
progress_with_cause move parsing of the qsig channel mapping
configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and
into a properly ifdef'd block
2008-11-06 19:46 +0000 [r155012] Mark Michelson <mmichelson@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 155011 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
2008) | 8 lines The documentation listed the ability to set
'maxmsg' per context. The truth is that you can only set this in
the general section or per mailbox. Thus I am updating the sample
config file to be more accurate. Thanks to sasargen on IRC for
bringing up this issue. ........
2008-11-06 18:19 +0000 [r154967] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c: Simplify the output of [See Also]. Functions are
printed without parenthesis like: FUNTION Applications are
printed with parenthesis like: AppName() Cli commands are printed
like: 'core show application' The other type of references are
printed as they are inside the <ref> tag.
2008-11-05 22:22 +0000 [r154923-154926] Sean Bright <sean.bright@gmail.com>
* apps/app_directed_pickup.c: Fix some whitespace.
* apps/app_directed_pickup.c, main/features.c: Update a couple
places to use the new ast_channel_search_locked API call.
2008-11-05 22:19 +0000 [r154922] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Don't read history on -rx commands. (Closes
issue #13571) Reported by: tzafrir Patch
'0001-no-need-for-history-on-asterisk-rx.patch' uploaded by
tzafrir.
2008-11-05 22:01 +0000 [r154919] Sean Bright <sean.bright@gmail.com>
* include/asterisk.h: Fix a problem found while building res_snmp.
2008-11-05 21:58 +0000 [r154915] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, funcs/func_strings.c, main/app.c,
CHANGES: Add LISTFILTER dialplan function, along with supporting
documentation. See documentation for more information on how to
use it.
2008-11-05 20:45 +0000 [r154875] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Make compilation
of chan_dahdi so that it does not require the new
pri_progress_with_cause function to have libpri support work.
2008-11-05 20:33 +0000 [r154839] Michiel van Baak <michiel@vanbaak.info>
* res/res_http_post.c: make this compile on OpenBSD again.
2008-11-05 20:17 +0000 [r154796-154837] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_agent.c: Add AgentLogin(), AgentMonitorOutgoing()
applications and AGENT() function XML documentation.
* apps/app_test.c: Add TestClient() and TestServer() applications
XML documentation.
* apps/app_mixmonitor.c: Add more [see also] references based on
TFOT.
* apps/app_macro.c: Add Macro(), MacroExit(), MacroExclusive() and
MacroIf() applications XML documentation. (closes issue #13699)
Reported by: snuffy Patches: bug13699_20081016.diff uploaded by
snuffy (license 35)
2008-11-05 16:11 +0000 [r154687] Steve Murphy <murf@digium.com>
* main/channel.c, /: Merged revisions 154685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1
line This fix was prompted by communication from user, who was
seeing thousands of error logs... looks like EAGAIN. Made such
uninteresting. ........
2008-11-05 14:37 +0000 [r154467-154647] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, apps/app_privacy.c, apps/app_sayunixtime.c,
main/features.c, apps/app_morsecode.c, apps/app_alarmreceiver.c,
apps/app_amd.c: Add more SeeAlso references based on TFOT.
* doc/appdocsxml.dtd: We now can have a reference to a filename
inside a <see-also> tag.
* apps/app_parkandannounce.c: - Add ParkAndAnnounce() application
XML documentation.
* main/pbx.c, apps/app_page.c, apps/app_authenticate.c,
apps/app_dumpchan.c, apps/app_disa.c, apps/app_image.c,
apps/app_chanspy.c, apps/app_stack.c, apps/app_adsiprog.c: - Add
more <see-also> based on TFOT. - Add the 'filename' type to the
see-also ref. To be able to reference a filename.
* apps/app_readfile.c, funcs/func_db.c, apps/app_sendtext.c,
funcs/func_blacklist.c, apps/app_url.c, apps/app_queue.c,
apps/app_senddtmf.c, apps/app_db.c: - Add some see-also
references based on TFOT.
* apps/app_read.c: - Add Read() application XML documentation.
* apps/app_followme.c: - Add FollowMe() application XML
documentation.
* apps/app_forkcdr.c, res/res_indications.c: - Add PlayTones() and
StopPlayTones() applications XML documentation. - Fix a dot that
was outside of the <para> in the ForkCDR() XML documentation.
2008-11-04 23:23 +0000 [r154429] Sean Bright <sean.bright@gmail.com>
* main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
Introduce a new API call ast_channel_search_locked, which
iterates through the channel list calling a caller-defined
callback. The callback returns non-zero if a match is found. This
should speed up some of the code that I committed earlier today
in chan_sip (which is also updated by this commit). Reviewed by
russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
2008-11-04 23:03 +0000 [r154366-154428] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Switch to using a thread condition to
signal that a child thread is ready for work, rather than a busy
wait. (closes issue #13011) Reported by: jpgrayson Patches:
chan_iax2_find_idle.patch uploaded by jpgrayson (license 492)
* /, channels/chan_iax2.c: Merged revisions 154365 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04
Nov 2008) | 9 lines On busy systems, it's possible for the values
checked within a single line of code to change, unless the
structure is locked to ensure a consistent state. (closes issue
#13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
uploaded by Corydon76 (license 14) Tested by: kowalma ........
2008-11-04 20:12 +0000 [r154329] Eliel C. Sardanons <eliels@gmail.com>
* Makefile: We need to pass the DTD to xmlstarlet to validate
against it the XML. (I thought it was being read within the
DOCTYPE definition inside the XML).
2008-11-04 19:07 +0000 [r154268] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 154266 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04
Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the
wrong state when it receives the indication AST_CONTROL_RINGING.
........
2008-11-04 18:59 +0000 [r154260-154264] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_skinny.c, channels/chan_h323.c: Recorded merge
of revisions 154263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
| 3 lines Make the monitor thread non-detached, so it can be
joined (suggested by Russell on -dev list). ........
* include/asterisk/devicestate.h, main/manager.c, apps/app_page.c,
include/asterisk/config.h, main/features.c, main/devicestate.c,
apps/app_queue.c, main/config.c, apps/app_voicemail.c: Slightly
optimize ast_devstate_str and rename global functions
devstate2str and config_text_file_save to have an ast_ prefix
2008-11-04 18:06 +0000 [r154225] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_forkcdr.c: Add XML documentation for the ForkCDR()
application.
2008-11-04 17:23 +0000 [r154186-154191] Sean Bright <sean.bright@gmail.com>
* main/pbx.c: GLOB_BRACE is already added to MY_GLOB_FLAGS if it is
supported on the platform. This should resolve some build errors
on Solaris. (issue #13704) Reported by: dougm
* channels/chan_sip.c, configs/sip.conf.sample: Allow devices that
accept dialog-info+xml (like snoms) to get the Caller ID of the
calling party when subscribed to the state of an extension that
is ringing. This has some limitations which are documented in
sip.conf.sample. (closes issue #13827) Reported by: seanbright
Patches: issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
* main/Makefile: Fix build errors.
2008-11-04 15:07 +0000 [r154151] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_vpb.cc, res/res_crypto.c, configure.ac,
cdr/cdr_adaptive_odbc.c, channels/chan_oss.c,
channels/chan_usbradio.c, res/res_config_odbc.c,
apps/app_osplookup.c, funcs/func_odbc.c, configure,
build_tools/menuselect-deps.in, channels/chan_alsa.c,
makeopts.in, cdr/cdr_odbc.c, res/res_odbc.c,
apps/app_voicemail.c: improve configure script to remember the
previous value of each dependency in build_tools/menuselect-deps,
so that (once it has been written) menuselect can use this
information to warn the user when a previously met dependency is
no longer met along the way, change tags used in configure
script, menuselect-deps and code for various dependencies to be
consistently named
2008-11-04 14:38 +0000 [r154149] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_dahdi.c: Add XML documentation for: Applications -
DAHDISendKeypadFacility() - DAHDISendCallreroutingFacility()
2008-11-03 22:28 +0000 [r154023-154072] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 154066 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03
Nov 2008) | 5 lines Attempting to expunge a mailbox when the
mailstream is NULL will crash Asterisk. (Closes issue #13829)
Reported by: jaroth Patch by: me (modified jaroth's patch)
........
* /, main/rtp.c: Merged revisions 154060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
| 3 lines Remove the potential for a division by zero error.
(Closes issue #13810) ........
* funcs/func_odbc.c: Should have passed the string pointer, not the
ast_str structure. (closes issue #13830) Reported by: Marquis
2008-11-03 18:02 +0000 [r153983] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Updating docs
2008-11-03 17:11 +0000 [r153947] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_stack.c: Add LOCAL() function XML documentation.
2008-11-03 15:25 +0000 [r153904-153905] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Spaces to replace tabs...
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding a
separation of remote authentication and our authentication.
remotesecret => our password for a remote service secret => our
authentication when someone calls us Secret => still has both
functions if remotesecret is not used.
2008-11-03 13:33 +0000 [r153803-153852] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_iax2.c: Add XML documentation for: Functions -
IAXPEER() - IAXVAR()
* channels/chan_sip.c: Add XML documentation for: Applications -
SIPDtmfMode() - SIPAddHeader() Functions - SIP_HEADER() -
SIPPEER() - SIPCHANINFO() - CHECKSIPDOMAIN()
2008-11-03 12:26 +0000 [r153787] Kevin P. Fleming <kpfleming@digium.com>
* configure, autoconf/ast_ext_lib.m4: when --without-<foo> is
passed to the configure script, explicitly inform menuselect that
the package was disabled by the user
2008-11-03 01:01 +0000 [r153747] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_waitforring.c, apps/app_waitforsilence.c, apps/app_db.c,
apps/app_ivrdemo.c: Add XML documentation for: - WaitForSilence()
- WaitForNoise() - WaitForRing() - IVRDemo() - DBDel() -
DBDeltree() (issue #13699) Reported by: snuffy Patches:
bug13699_20081016.diff uploaded by snuffy (license 35) (With
minor changes)
2008-11-02 23:34 +0000 [r153709] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h, configure,
include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
configure.ac, include/asterisk/compiler.h, apps/app_stack.c:
instead of trying to forcibly load res_agi when app_stack is
loaded (even if the administrator didn't want it loaded), use GCC
weak symbols to determine whether it was loaded already or not;
if it was loaded, then use it.
2008-11-02 20:06 +0000 [r153652] Russell Bryant <russell@digium.com>
* /, include/asterisk/features.h: Merged revisions 153651 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008)
| 2 lines features.h depends on linkedlists.h, so include it
........
2008-11-02 19:39 +0000 [r153616-153650] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: fix one more warning missed because i did
not have new enough libpri installed
* res/res_musiconhold.c: fix small bug introduced while cleaning up
compiler warnings
* /: mark this revision as merged manually
* utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c,
main/utils.c, formats/format_wav_gsm.c, res/res_http_post.c,
res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c,
res/res_config_sqlite.c, utils/frame.c, utils/stereorize.c,
main/channel.c, channels/chan_dahdi.c, main/manager.c,
res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c,
res/res_agi.c, main/http.c, main/logger.c, formats/format_gsm.c,
apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c,
apps/app_festival.c, formats/format_wav.c, res/ael/ael.y,
main/db1-ast/hash/hash_page.c, agi/eagi-test.c, res/res_crypto.c,
utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c,
utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c,
agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c,
main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex,
pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c,
utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c:
bring over all the fixes for the warnings found by gcc 4.3.x from
the 1.4 branch, and add the ones needed for all the new code here
too
2008-11-02 06:24 +0000 [r153582] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_iax2.c: Add IAX2Provision() application XML
documentation.
2008-11-02 05:56 +0000 [r153577-153580] Russell Bryant <russell@digium.com>
* Makefile: validate-docs is a PHONY target
* Makefile, configure, configure.ac, makeopts.in: Add a handy
makefile target so that you can validate the documentation
against the DTD by running "make validate-docs"
* Makefile: Modify the Makefile logic for extracting documentation.
- Build the documentation when you run "make", as opposed to
"make install" - Only rebuild the documentation when source code
has been changed
2008-11-02 05:10 +0000 [r153541-153543] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_flash.c: Add Flash() application XML documentation.
* apps/app_talkdetect.c: Fix a typo in the name of the application.
2008-11-02 04:14 +0000 [r153472-153507] Sean Bright <sean.bright@gmail.com>
* channels/Makefile: There is a troublesome assert() in the
alsa/control.h header that causes GCC 4.3.2 to complain that the
passed argument will always evaluate to true. So to get things to
compile, disable assert when building chan_usbradio.so.
* apps/app_record.c: Another little one.
2008-11-02 02:55 +0000 [r153362-153470] Russell Bryant <russell@digium.com>
* apps/app_page.c: fix a typo (thanks sean)
* apps/app_dial.c, funcs/func_speex.c, apps/app_page.c,
apps/app_record.c, funcs/func_env.c, apps/app_dahdiras.c,
funcs/func_math.c, funcs/func_strings.c, apps/app_userevent.c,
apps/app_exec.c, apps/app_chanspy.c, apps/app_playback.c: Fix
various spelling and grammatical issues in documentation
* apps/app_voicemail.c: - Use a for loop instead of a while loop -
Get rid of an unnecessary variable
* apps/app_directed_pickup.c: Instead of doing a couple of strlen()
calls each iteration of the loop, only do it once at the
beginning of the function
* channels/chan_sip.c: Don't ignore the result of find_peer() when
looking for a peer by IP in check_peer_ok().
* funcs/func_speex.c, apps/app_dahdibarge.c, funcs/func_rand.c,
apps/app_readfile.c, funcs/func_module.c, funcs/func_dialgroup.c,
include/asterisk/autoconfig.h.in, funcs/func_env.c,
apps/app_dahdiscan.c, apps/app_record.c, funcs/func_strings.c,
apps/app_sayunixtime.c, include/asterisk/extconf.h,
apps/app_alarmreceiver.c, apps/app_image.c,
apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
main/config.c, main/term.c, include/asterisk/compat.h, configure,
funcs/func_shell.c, apps/app_skel.c, apps/app_dumpchan.c,
include/asterisk/module.h, main/features.c, apps/app_amd.c,
apps/app_url.c, apps/app_milliwatt.c, apps/app_dial.c,
main/pbx.c, include/asterisk/xml.h (added), apps/app_page.c,
funcs/func_timeout.c, main/Makefile, apps/app_privacy.c,
apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c,
funcs/func_math.c, apps/app_dahdiras.c, configure.ac,
apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c,
apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c,
doc/tex/asterisk-conf.tex, Makefile, apps/app_sendtext.c,
funcs/func_channel.c, funcs/func_cdr.c, apps/app_zapateller.c,
build_tools/get_documentation (added), funcs/func_iconv.c,
apps/app_mixmonitor.c, apps/app_chanspy.c, main/asterisk.c,
apps/app_cdr.c, funcs/func_base64.c, funcs/func_md5.c,
apps/app_dictate.c, apps/app_authenticate.c,
apps/app_readexten.c, apps/app_userevent.c, funcs/func_vmcount.c,
main/xml.c (added), funcs/func_sha1.c, funcs/func_logic.c,
funcs/func_uri.c, apps/app_controlplayback.c, funcs/func_enum.c,
apps/app_setcallerid.c, funcs/func_groupcount.c,
funcs/func_config.c, funcs/func_volume.c, funcs/func_odbc.c,
apps/app_mp3.c, apps/app_directory.c, apps/app_jack.c,
apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c,
funcs/func_dialplan.c, funcs/func_db.c, funcs/func_version.c,
apps/app_festival.c, funcs/func_lock.c, apps/app_waituntil.c,
doc, include/asterisk/term.h, include/asterisk/_private.h,
apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
funcs/func_global.c, funcs/func_extstate.c,
funcs/func_realtime.c, apps/app_channelredirect.c,
funcs/func_blacklist.c, apps/app_directed_pickup.c,
include/asterisk/pbx.h, include/asterisk/strings.h, makeopts.in,
apps/app_senddtmf.c, funcs/func_devstate.c,
funcs/func_callerid.c, doc/appdocsxml.dtd (added),
apps/app_verbose.c, apps/app_stack.c: Merge changes from
team/group/appdocsxml This commit introduces the first phase of
an effort to manage documentation of the interfaces in Asterisk
in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions
to the new format are also included. For more information, see
the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
* channels/chan_sip.c: Ensure that the sip_pvt properly has its
refcount incremented when the scheduler holds a reference to it
for session timer processing.
2008-11-01 01:55 +0000 [r153296] Sean Bright <sean.bright@gmail.com>
* configs/sip.conf.sample: The default in chan_sip for
notifyringing is yes, so update the sample conf to reflect that.
2008-10-31 20:05 +0000 [r153223] Mark Michelson <mmichelson@digium.com>
* main/dial.c, apps/app_page.c, include/asterisk/dial.h, CHANGES: *
Fixed timeout logic in the dialing API as setting timeouts had no
effect * Updated dialing API documentation to indicate that
timeouts are specified in milliseconds * Added a new timeout
argument to the Page application. If time expires, any endpoints
which have not answered will be hung up.
2008-10-31 18:55 +0000 [r153181] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, main/features.c, include/asterisk/channel.h,
apps/app_followme.c, apps/app_queue.c: Recent CDR fixes moved
execution of the 'h' exten into the bridging code, so variables
that were set after ast_bridge_call was called would not show up
in the 'h' exten. Added a callback function to handle setting
variables, etc. from w/in the bridging code. Calls back into a
nested function within the function calling ast_bridge_call
(closes issue #13793) Reported by: greenfieldtech
2008-10-31 17:18 +0000 [r153122-153124] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES:
Failover for func_odbc, allowing an INSERT query to be performed
when the UPDATE query initially affects 0 rows. (closes issue
#13083) Reported by: Corydon76 Patches:
20081031__bug13083.diff.txt uploaded by Corydon76 (license 14)
* /, channels/chan_sip.c: Merged revisions 153114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008)
| 3 lines Turn off qualify on uncached realtime peers. (Closes
issue #13383) ........
2008-10-31 09:31 +0000 [r153057] Russell Bryant <russell@digium.com>
* main/channel.c: Use the ast_str API call to reset the string
instead of manually editing its internals (closes issue #13816)
Reported by: eliel Patches: channel.c.patch uploaded by eliel
(license 64)
2008-10-30 20:59 +0000 [r152993] Sean Bright <sean.bright@gmail.com>
* /, bootstrap.sh: Merged revisions 152992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct
2008) | 2 lines The -I argument to aclocal needs a space before
the include directory name. ........
2008-10-30 20:46 +0000 [r152990] Russell Bryant <russell@digium.com>
* include/asterisk/timing.h: Add a todo for a new timing API
implementation that would work for Linux systems as of kernel
2.6.25 and glibc 2.8
2008-10-30 20:35 +0000 [r152923-152969] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_h323.c: Merged revisions 152958 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30
Oct 2008) | 3 lines Cannot join detached threads. See
http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400) ........
* channels/chan_local.c, /: Merged revisions 152922 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152922 | tilghman | 2008-10-30 14:43:38 -0500 (Thu, 30
Oct 2008) | 6 lines Unlock before returning, when extension
doesn't exist. (closes issue #13807) Reported by: eliel Patches:
chan_local.c.patch uploaded by eliel (license 64) ........
2008-10-30 19:40 +0000 [r152887-152920] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix the sip_peer reference count with
respect to scheduler entries for scheduling peer pokes, and
scheduling peer poke expirations.
* channels/chan_sip.c: Fix the sip_peer reference count with
respect to scheduler entries for registration expirations.
* include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF().
The reference count of the object _must_ be increased before
creating the scheduler entry. Otherwise, you create a race
condition where the reference count may hit zero and the object
can disappear out from under you. This could also would have
incorrectly decreased the reference count in the case that the
scheduler add failed.
2008-10-30 19:23 +0000 [r152879] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: I just noticed this construct and thought it
was silly to have a bunch of case statements with duplicated code
in each case. Instead, just use the built-in fallthrough
capability of case statements and reduce the code to a single
instance
2008-10-30 19:21 +0000 [r152875-152877] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Modify the documentation of the sip_registry
struct - Remove a comment that says that the monitor thread is
the only one that ever touches these objects. This is no longer
the case with TCP. Also, I would eventually like to get the
scheduler in its own thread, so this is just a poor assumption to
make. - Note that reference counting of these objects with
respect to scheduler entries is not complete. There are some
leaked references when deleting scheduler entries.
* funcs/func_db.c: - spaces to tabs - add some braces - remove
unnecessary cast
2008-10-30 16:54 +0000 [r152809-152812] Kevin P. Fleming <kpfleming@digium.com>
* main/cdr.c, /: Merged revisions 152811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct
2008) | 3 lines instead of comparing the string pointer to 0,
let's compare the value that was actually parsed out of the
string (found by sparse) ........
* include/asterisk/buildinfo.h (added): try to get this committed
before the buildbot complains about a broken tree
* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
main/dial.c, main/dnsmgr.c, main/buildinfo.c,
codecs/lpc10/chanwr.c, utils/astcanary.c,
channels/misdn/isdn_lib.c, main/asterisk.c, apps/app_adsiprog.c:
fix a few small things found by using sparse
2008-10-30 16:38 +0000 [r152807] Mark Michelson <mmichelson@digium.com>
* main/features.c, CHANGES, configs/features.conf.sample: After
seeing another problem in #asterisk stemming from the low default
value of featuredigittimeout, I decided it was high time to
change it. I have changed the default to 2000 ms based on a
suggestion from Leif Madsen.
2008-10-30 04:26 +0000 [r152689-152765] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Set up an example stdexten that
preserves the original context and extension in the CDR. (Related
to issue #13799) Reported by: davidw
* CHANGES, apps/app_directory.c: Pay attention to the
searchcontexts entry in voicemail.conf (related to AST-125)
* main/pbx.c: Track down and fix annoying lock errors
2008-10-29 20:53 +0000 [r152646] Mark Michelson <mmichelson@digium.com>
* apps/app_directory.c: If there was no named defined in a
voicemail.conf mailbox entry, then app_directory would crash when
attempting to read that entry from the file. We now check for the
NULL or empty string properly so that there will be no crash.
(closes issue #13804) Reported by: bluecrow76
2008-10-29 05:47 +0000 [r152605] Steve Murphy <murf@digium.com>
* apps/app_dial.c, /, apps/app_queue.c,
configs/features.conf.sample: Merged revisions 152538 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) |
14 lines A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying to get the
one-touch parking stuff working, because it didn't occur to me
that I had to also have the corresponding options in the dial
command! Duh! (In all this time, I never set this up before!) So,
to keep some poor fool from suffering the same fate, I made the
features.conf.sample file mention the corresponding opts in
dial/queue; and the docs for dial/app specifically mention the
corresponding decls in the feature.conf file. I hope this doesn't
spoil some vast, eternal plan... ........
2008-10-29 05:34 +0000 [r152569] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 152539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008)
| 7 lines Fix an incorrect usage of sizeof() (closes issue
#13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch
uploaded by andrew53 (license 519) ........
2008-10-29 05:01 +0000 [r152536] Steve Murphy <murf@digium.com>
* apps/app_dial.c, /, main/features.c, include/asterisk/pbx.h,
apps/app_queue.c, include/asterisk/features.h: Merged revisions
152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) |
46 lines The magic trick to avoid this crash is not to try to
find the channel by name in the list, which is slow and resource
consuming, but rather to pay attention to the result codes from
the ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when
a channel is parked. Why? because CDR's aren't generated via
parking, so nothing is needed, but if a transfer occurred, there
are critical things I need. If you get AST_PBX_KEEPALIVE, then
don't touch the channel pointer. If you get
AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
don't touch the peer pointer. Updated the several places where
the results from a bridge were not being properly obeyed, and
fixed some code I had introduced so that the results of the
bridge were not overridden (in trunk). All the places that
previously tested for AST_PBX_NO_HANGUP_PEER now have to check
for both AST_PBX_NO_HANGUP_PEER and
AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
while A is getting the parking slot announcement, immediately
after being put on hold. 2. A calls B; B answers; A parks B; B
hangs up after A has been hung up, but before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting
the parking slot announcement, immediately after being put on
hold. 4. A calls B; B answers; B parks A; A hangs up after B has
been hung up, but before the park times out. No crash. I also ran
the scenarios above against valgrind, and accesses looked good.
........
2008-10-28 22:33 +0000 [r152467] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 152463 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28
Oct 2008) | 3 lines Quoting in the wrong direction (Fixes
AST-107) ........
2008-10-28 22:26 +0000 [r152448] Doug Bailey <dbailey@digium.com>
* configs/phoneprov.conf.sample: Add more polycom firmware files to
static mapping
2008-10-28 21:38 +0000 [r152369-152442] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: Only re-add the io port if it was closed,
otherwise reload causes a memory leak. (closes issue #13785)
Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel
(license 64)
* apps/app_dial.c, /: Merged revisions 152368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008)
| 8 lines Reset all DIAL variables back to blank, in case Dial is
called multiple times per call (which could otherwise lead to
inconsistent status reports). (closes issue #13216) Reported by:
ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76
(license 14) Tested by: ruddy ........
2008-10-27 23:31 +0000 [r152287] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 152286 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27
Oct 2008) | 2 lines Buffer policy setting for half is not needed.
........
2008-10-27 21:34 +0000 [r152134-152216] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 152215 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27
Oct 2008) | 6 lines Inherit ALL elements of CallerID across a
local channel. (closes issue #13368) Reported by: Peter Schlaile
Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
(license 14) ........
* apps/app_stack.c: Set ARGC in subroutines with the number of
arguments passed.
* apps/app_stack.c: Oops, only delete the ARG variables once upon
release. The following section would have removed them again
(removing variables from 2 stack frames, instead of just one).
2008-10-27 16:03 +0000 [r152132] Jason Parker <jparker@digium.com>
* apps/app_transfer.c: Remove options argument parsing/syntax (it
isn't used any longer) (closes issue #13789) Reported by: IgorG
Patches: app_transfer.c.diff uploaded by IgorG (license 20)
2008-10-26 20:25 +0000 [r152060] Sean Bright <sean.bright@gmail.com>
* /, funcs/func_strings.c: Merged revisions 152059 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun,
26 Oct 2008) | 7 lines Since passing \0 as the second argument to
strchr is valid (and will match the trailing \0 of a string) we
need to check that first, otherwise we end up with incorrect
results. Fix suggested by reporter. (closes issue #13787)
Reported by: meitinger ........
2008-10-26 10:23 +0000 [r151980-152020] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Trying to fix the user/peer matching
correctly. This will need some testing before getting merged into
1.6.1
* channels/chan_sip.c: Moving more variables to the sip_cfg
structure, as I have some future ideas for the usage of that
structure.
* channels/chan_sip.c: Doxygen changes and some formatting.
2008-10-25 11:02 +0000 [r151906] Russell Bryant <russell@digium.com>
* /, main/asterisk.c: Merged revisions 151905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008)
| 8 lines Move AMI initialization to occur after loading modules.
This prevents a deadlock when someone tries to initiate a module
reload from the AMI just as Asterisk is starting. (closes issue
#13778) Reported by: hotsblanc Fix suggested by hotsblanc
........
2008-10-23 21:27 +0000 [r151830] Terry Wilson <twilson@digium.com>
* funcs/func_odbc.c: allow to compile under --enable-dev-mode (gcc
didn't actually complain when I was using ccache...)
2008-10-23 15:54 +0000 [r151762] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/vmdb.sql: Clarify documentation, following merge
of realtime_update2 branch
2008-10-23 15:38 +0000 [r151739-151761] Olle Johansson <oej@edvina.net>
* CHANGES: Thanks russellb for reminding an old man....
* channels/chan_sip.c, doc/tex/channelvariables.tex: Adding a small
new feature. Setting _SIPFROMDOMAIN in a channel will set the
domain we use for the URI in the outbound call leg.
2008-10-23 15:28 +0000 [r151732] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Simplify some nested functions, as suggested
by Russell on -dev
2008-10-23 15:09 +0000 [r151722] Doug Bailey <dbailey@digium.com>
* res/res_http_post.c: Add patch to handle how IE7 issues POST
requests using Window path spec including backslash delimiters
2008-10-22 22:11 +0000 [r151682] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, CHANGES: Added debugging CLI functions
2008-10-22 20:45 +0000 [r151642] BJ Weschke <bweschke@btwtech.com>
* channels/chan_sip.c: revert the changes in issue #13705 - it's
being re-opened as while the results fixed the complaint in the
issue, it introduced other more undesirable issues than what was
already reported
2008-10-22 20:05 +0000 [r151601] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/live_ast (added): Add a contributed script for
running Asterisk without installing it, first. (closes issue
#11680) Reported by: tzafrir Patches: live_ast_6 uploaded by
tzafrir (license 46)
2008-10-22 20:05 +0000 [r151600] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Change some logical ands to bitwise ands
and add messages alerting that a channel is being ignored if the
PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue
#13759) Reported by: smurfix Patches: dahdi.patch uploaded by
smurfix (license 547)
2008-10-22 17:45 +0000 [r151554-151555] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Print out the right var in the log message
* channels/chan_sip.c: Fix this check to use the proper variable
(the result from get_in_brackets)
2008-10-22 15:08 +0000 [r151420-151512] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: The logic of a strncasecmp call was
reversed. (closes issue #13706) Reported by: andrew53 Patches:
sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519)
* channels/chan_sip.c: Make the sip_standard_port function more
granular by allowing separate type and port arguments. This is
necessary because when building our From and Contact headers, we
need to be absolutely sure that we are placing our source port
there and not the peer's source port. (closes issue #12761)
Reported by: asbestoshead Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license
455)
* channels/chan_sip.c: Get this compiling in dev-mode
* channels/chan_sip.c: If a peer uses any transport other than UDP,
then MWI will fail for that peer since sip_alloc will allocate a
sip_pvt with a default transport of UDP. This change resets the
socket type immediately after allocating the sip_pvt in
sip_send_mwi_from_peer, so that the proceeding call to
create_addr_from_peer does not fail right away. The socket data
from the peer is properly copied to the sip_pvt in
create_addr_from_peer. (closes issue #13710) Reported by:
andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53
(license 519)
* channels/chan_sip.c: When attempting to resolve hostnames, we
need to be sure to remove any parameters from the string so that
name resolution succeeds. (closes issue #13727) Reported by:
fnordian Patches: resolvewithouturiparameter.patch uploaded by
fnordian (license 110)
2008-10-21 15:20 +0000 [r151371] Tilghman Lesher <tlesher@digium.com>
* apps/app_mixmonitor.c: Default file modes should always be full
read and write, to allow the system administrator to make the
decision of what permissions will actually be given, through the
use of the process umask. (Closes issue# 13751)
2008-10-21 11:02 +0000 [r151327] BJ Weschke <bweschke@btwtech.com>
* channels/chan_sip.c: Fix configuration parsing so type=friend
still identifies "friend" as a peer even though it is now a
legacy configuration verb. (closes issue #13705) reported by:
blitzrage patched by: bweschke
2008-10-20 05:07 +0000 [r151246] BJ Weschke <bweschke@btwtech.com>
* pbx/pbx_config.c, main/config.c: Do NOT attempt to do anything
with the ast_config struct when it's been returned as INVALID by
the config file interpreter. (closes issue #13741)
2008-10-20 05:00 +0000 [r151242-151243] Kevin P. Fleming <kpfleming@digium.com>
* autoconf/ast_check_pwlib.m4, /, autoconf/ast_check_openh323.m4,
configure.ac: Merged revisions 151241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct
2008) | 2 lines rename this macro to properly reflect what it
does ........
* autoconf/ast_prog_egrep.m4, autoconf/ast_c_define_check.m4,
autoconf/ast_ext_tool_check.m4 (added),
autoconf/ast_check_mandatory.m4 (added), /,
autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
autoconf/ast_prog_sed.m4, acinclude.m4 (removed),
autoconf/ast_check_pwlib.m4, autoconf (added),
autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
autoconf/ast_gcc_attribute.m4, bootstrap.sh,
autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4: Merged
revisions 151240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct
2008) | 3 lines break up acinclude.m4 into individual files,
which will make it easier to maintain, easier to add new macros
(less patching) and will ease maintenance of these macros across
Asterisk branches ........
2008-10-19 20:30 +0000 [r151188-151190] BJ Weschke <bweschke@btwtech.com>
* /: Block 151167 from coming forward into the /trunk this is a 1.4
fix only.
* /: Block 151100 from coming forward into the /trunk this is a 1.4
fix only.
2008-10-19 19:11 +0000 [r151101] Kevin P. Fleming <kpfleming@digium.com>
* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the
TCP/TLS socket API: 1) rename 'struct server_args' to 'struct
ast_tcptls_session_args', to follow coding guidelines 2) make
ast_make_file_from_fd() static and rename it to something that
indicates what it really is for (again coding guidelines) 3)
rename address variables inside 'struct ast_tcptls_session_args'
to be more descriptive (dare i say it... coding guidelines) 4)
change ast_tcptls_client_start() to use the new 'remote_address'
field of the session args for the destination of the connection,
and use the 'local_address' field to bind() the socket to the
proper source address, if one is supplied 5) in chan_sip, ensure
that we pass in the PP address we are bound to when creating
outbound (client) connections, so that our connections will
appear from the correct address
2008-10-19 13:10 +0000 [r151060] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: dont segfault when placing a call to a
line that has no registered device.
2008-10-19 07:20 +0000 [r151019] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding changes from train and flight back
home from SIPit23 in Lannion, France. - Additional comments on
TCP/TLS implementation - Some additions for new drafts/rfcs (no
new functionality really, mostly documentation) - Other random
small fixes
2008-10-18 10:27 +0000 [r150930-150971] Michiel van Baak <michiel@vanbaak.info>
* Makefile: Make sure we support nested functions and generation of
trampolines under OpenBSD. (closes issue #13724) Reported by:
mvanbaak
* contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.suse.asterisk: dont use deprecated commands in
the init scripts. (closes issue #13720) Reported by:
decryptus_proformatique Patches:
contrib_initd_module_reload.patch uploaded by decryptus (license
555) With mods by me to fix stop commands as well
2008-10-18 03:35 +0000 [r150773-150887] BJ Weschke <bweschke@btwtech.com>
* apps/app_authenticate.c, CHANGES: Give app_authenticate the
ability to select a prompt other than the default. (closes issue
#13734) reported and patched by: jvandal
* main/manager.c, /: Using the GetVar handler in AMI is potentially
dangerous (insta-crash [tm]) when you use a dialplan function
that requires a channel and then you don't provide one or provide
an invalid one in the Channel: parameter. We'll handle this
situation exactly the same way it was handled in pbx.c back on
r61766. We'll create a bogus channel for the function call and
destroy it when we're done. If we have trouble allocating the
bogus channel then we're not going to try executing the function
call at all and run the risk of crashing. (closes issue #13715)
reported by: makoto patch by: bweschke
* doc/manager_1_1.txt, CHANGES, apps/app_queue.c: The QueueEntry
event now has the uniqueid of the channel included. (closes issue
#13731) reported and patched by: caio1982
2008-10-17 21:48 +0000 [r150731] Matthew Fredrickson <creslin@digium.com>
* configure, configure.ac: Update configure check to check for new
function in libpri (pri_progress_with_cause)
2008-10-17 21:35 +0000 [r150729] Jason Parker <jparker@digium.com>
* codecs/codec_adpcm.c, codecs/ex_g722.h (added),
codecs/codec_gsm.c, codecs/ex_adpcm.h (added), codecs/ex_alaw.h
(added), codecs/ex_g726.h (added), codecs/ex_gsm.h (added),
codecs/slin_ulaw_ex.h (removed), codecs/slin_lpc10_ex.h
(removed), codecs/codec_resample.c, codecs/slin_g722_ex.h
(removed), codecs/g722_slin_ex.h (removed), codecs/ex_ulaw.h
(added), codecs/adpcm_slin_ex.h (removed), codecs/ex_ilbc.h
(added), codecs/slin_adpcm_ex.h (removed), codecs/g726_slin_ex.h
(removed), codecs/slin_g726_ex.h (removed), codecs/codec_lpc10.c,
codecs/gsm_slin_ex.h (removed), codecs/slin_gsm_ex.h (removed),
codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/ex_lpc10.h
(added), codecs/codec_alaw.c, codecs/codec_speex.c,
codecs/codec_g726.c, include/asterisk/slin.h (added),
codecs/ex_speex.h (added), codecs/slin_resample_ex.h (removed),
codecs/ulaw_slin_ex.h (removed), codecs/slin_ilbc_ex.h (removed),
codecs/ilbc_slin_ex.h (removed), codecs/lpc10_slin_ex.h
(removed), codecs/codec_ulaw.c, codecs/codec_ilbc.c,
codecs/speex_slin_ex.h (removed), codecs/slin_speex_ex.h
(removed): Merge codec_consistency branch. This should make
sample usage much happier.
2008-10-17 17:31 +0000 [r150664] Michiel van Baak <michiel@vanbaak.info>
* main/cli.c: Fix CLI command 'channel request hangup' Prodded on
IRC by Russell and fixed by eliel (closes issue #13730) Reported
by: eliel Patches: main_cli.patch uploaded by eliel (license 64)
2008-10-17 17:25 +0000 [r150640] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Merge in
patch for #13454. Includes CallRereouting dialplan application,
option for discard of remote hold messages, and using the
alternate logical channel mapping in Q.SIG instead of the default
physical channel mapping.
2008-10-17 17:09 +0000 [r150580-150635] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Make helper call a little safer (suggested
by Russell on IRC)
* include/asterisk/sched.h, channels/chan_iax2.c: Fix the FRACK!
warnings in chan_iax2 when POKE/LAGRQ packets are not answered.
2008-10-17 08:42 +0000 [r150469-150510] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding some additional thoughts on
configuration changes to TCP/TLS
* Makefile: Make sure we support nested functions with GCC 4.01
OS/X. This might not be OS/X only, but I'll leave it to kpfleming
to add this to the configure script for testing.
2008-10-17 06:00 +0000 [r150426] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c, UPGRADE.txt, configs/skinny.conf.sample,
CHANGES: Break up skinny.conf into seperate sections for devices
and lines. (closes issue #13412) Reported by: wedhorn Patches:
config-restruct-v4.diff uploaded by wedhorn (license 30)
2008-10-17 04:28 +0000 [r150384] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Fix option handling code. (closes issue
#11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt
uploaded by DEA (license 3) with additional fixes by me
2008-10-17 00:18 +0000 [r150311] Mark Michelson <mmichelson@digium.com>
* doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Add an
IAXregistry manager command. See doc/manager_1_1.txt for more
details of this command. (closes issue #13326) Reported by: ib2
Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license
35)
2008-10-17 00:14 +0000 [r150309] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: Initialize character arrays as they are not
guaranteed to be set.
2008-10-17 00:13 +0000 [r150207-150307] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: After a long discussion on #asterisk-bugs,
it seems kind of odd that a channel would be named after the
originating port. For endpoints that always include ":5060" as
part of the From: header, it will mean that you have a ton of
channels with names like "SIP/5060-3ea38a8b." I am boldly moving
forward with this change in trunk, but I'm not touching other
branches with this one since this definitely would qualify as a
behavior change. If there is a problem with this commit, and I
haven't seen the obvious reason why you'd want to name the
channel after the port from which the call originated, then
please feel free to revert this
* main/manager.c, /: Merged revisions 150304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct
2008) | 6 lines Reverting changes from commits 150298 and 150301
since I was mistakenly under the assumption that dialplan
functions *always* required that a channel be present. I need to
go home earlier, I think :) ........
* main/manager.c: Merged revisions 150298,150301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct
2008) | 10 lines Don't try to call a dialplan function's read
callback from the manager's GetVar handler if an invalid channel
has been specified. Several dialplan functions, including CHANNEL
and SIP_HEADER, do not check for NULL-ness of the channel being
passed in. (closes issue #13715) Reported by: makoto ........
r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct
2008) | 3 lines And don't forget to return on the error condition
........
* apps/app_sms.c: Answer the channel prior to checking for the 'a'
option in app_sms. (closes issue #13675) Reported by: alecdavis
Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis
(license 585)
* apps/app_skel.c: Updating app_skel.c to follow coding guidelines
with regards to braces used on if statements. (closes issue
#13696) Reported by: alecdavis Patches:
app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license
585)
* channels/chan_iax2.c: Remove an odd redundant comparison
* configure, configure.ac: Change configure script to search for
openais in both /usr/lib and /usr/lib64 since some distros place
64-bit libraries only in the /usr/lib64 directory. (closes issue
#13721) Reported by: jcollie Patches:
0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie
(license 412)
* channels/chan_sip.c: INVITES with proxy auth were sent with a
different branch than what was in the invite_branch of a sip_pvt,
meaning that if a CANCEL were sent later, the branch in the
CANCEL would not match the branch in the latest INVITE sent out,
leading to some endpoints responding to the CANCEL with a 481.
(closes issue #13714) Reported by: fnordian Patches:
invite_branch.patch uploaded by fnordian (license 110)
2008-10-16 16:04 +0000 [r150125] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 150124 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16
Oct 2008) | 1 line Fix memory leak found by customer ........
2008-10-16 15:48 +0000 [r150118-150121] Terry Wilson <twilson@digium.com>
* configs/modules.conf.sample: This is nolonger needed
* res/res_phoneprov.c: func_strings isn't a dependency of this
module anymore
2008-10-16 15:02 +0000 [r150052] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: ensure that type=peer entries are only
matched on IP/port, not on name (after oej audits all the calls
to find_peer() to make sure that forcenamematch is set correctly
in each case)
2008-10-16 15:00 +0000 [r150008-150051] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Doxygen addition
* channels/chan_sip.c: Add some notes on problems with the TCP/TLS
implementation
2008-10-16 13:28 +0000 [r149917-149981] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: return this logic to where it used to be,
*after* the dialog->needdestroy flag has been determined to be
set; otherwise, we generate these debug messages every time we
inspect every active dialog
* channels/chan_sip.c: some additional debugging tools added at
SIPit23: - move all setting of 'needdestroy' on dialog structures
into the history - report all tags involved when a pedantic check
fails on a REFER
* res/res_phoneprov.c: inter-module dependencies should be included
in the source code, not just in sample config files
* res/res_phoneprov.c: correct file name in message
* configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
support relative paths in musiconhold.conf, which makes moh work
by default when Asterisk was configured using --prefix and 'make
samples' is run
2008-10-15 21:36 +0000 [r149848] BJ Weschke <bweschke@btwtech.com>
* /: Blocking 149840 from coming forward.
2008-10-15 20:55 +0000 [r149802] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Make the sip_proxy struct reference counted.
This is necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has been freed.
(closes issue #13700) Reported by: fnordian Patches: 13700.patch
uploaded by putnopvut (license 60) Tested by: fnordian
2008-10-15 20:14 +0000 [r149756] BJ Weschke <bweschke@btwtech.com>
* configs/agents.conf.sample, /: Merged revisions 149683 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008)
| 4 lines An update to the documentation/example of
agents.conf.sample with the correct parameter for this feature as
defined in chan_agent.c (closes issue #13709) ........
2008-10-15 19:07 +0000 [r149588-149687] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Permit data fields to contain more than 255
characters. (closes issue #13631) Reported by: seanbright
Patches: 20081015__bug13631.diff.txt uploaded by Corydon76
(license 14) Tested by: blitzrage
* funcs/func_odbc.c: Only set buf to blank before the goto.
* codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks
memory, because it matches a library malloc() with an ast_free
(which, of course, doesn't match up with known allocated memory,
so the free fails). (closes issue #13702) Reported by: eliel
Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64)
* apps/app_echo.c: Minor spacing change (closes issue #13697)
Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt
uploaded by alecdavis (license 585)
2008-10-15 13:52 +0000 [r149542] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding a note about a missing part of
"kill-the-user" - I got lost in the Ao2 world... We're going to
try to get time to fix this and kpfleming believes that there's
code in ao2 so that we can solve it...
2008-10-15 11:26 +0000 [r149384-149487] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 149452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct
2008) | 3 lines fix some problems when parsing SIP messages that
have the maximum number of headers or body lines that we support
........
* configure, configure.ac: reverting this change... had not read
the commit list yet, didn't realize the code had been upgraded
* configure, configure.ac: do complete version check for SpanDSP,
since the app_fax code is not compatible with 0.0.6 yet
* apps/app_stack.c: building this module depends on res_agi being
built as well
2008-10-15 07:45 +0000 [r149342] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Fixing sytax errors ;-)
2008-10-14 23:57 +0000 [r149201-149279] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, CHANGES: When specifying an invalid timeout to
Dial, take it to mean that no timeout is desired. (closes issue
#13625) Reported by: atis
* /, channels/chan_sip.c: Merged revisions 149266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct
2008) | 4 lines Change this warning to an error message.
Suggestion comes from Sean Bright. Thanks Sean! ........
* /, channels/chan_sip.c: Merged revisions 149207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct
2008) | 9 lines Call register_peer_exten even in the case that
the peer's IP/port does not change. (closes issue #13309)
Reported by: dimas Patches: v2-13309.patch uploaded by dimas
(license 88) ........
* /, include/asterisk/audiohook.h, main/audiohook.c: Merged
revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct
2008) | 12 lines Add a tolerance period for sync-triggered
audiohooks so that if packetization of audio is close (but not
equal) we don't end up flushing the audiohooks over small
inconsistencies in synchronization. Related to issue #13005, and
solves the issue for most people who were experiencing the
problem. However, a small number of people are still experiencing
the problem on long calls, so I am not closing the issue yet
........
* /, apps/app_queue.c: Merged revisions 149200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct
2008) | 12 lines Update the queue with the correct number of
calls and whether the call was completed within the service level
when a transfer takes place. This way, we do not "break" the
leastrecent and fewestcalls strategies by not logging a call
until after the transferred call has ended. (closes issue #13395)
Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
by Marquis (license 32) ........
2008-10-14 22:38 +0000 [r149199] Tilghman Lesher <tlesher@digium.com>
* main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c,
include/asterisk/chanvars.h, include/asterisk/config.h,
include/asterisk/strings.h, res/res_indications.c,
include/asterisk/hashtab.h, main/chanvars.c, main/config.c: Add
additional memory debugging to several core APIs, and fix several
memory leaks found with these changes. (Closes issue #13505,
closes issue #13543) Reported by: mav3rick, triccyx Patches:
20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
Tested by: mav3rick, triccyx
2008-10-14 21:08 +0000 [r149131] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 149130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct
2008) | 7 lines Don't allow reserved characters to be used in
register lines in sip.conf. (closes issue #13570) Reported by:
putnopvut ........
2008-10-14 20:16 +0000 [r149062] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_waitforsilence.c: Merged revisions 149061 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008)
| 6 lines Check correct values in the return of ast_waitfor();
also, get rid of a possible memory leak. (closes issue #13658)
Reported by: explidous Patch by: me ........
2008-10-14 19:35 +0000 [r149040] Leif Madsen <lmadsen@digium.com>
* doc/manager_1_1.txt: Add missing documentation for
SipShowRegistry action and RegistryEntry event. (closes issue
#13342) Reported and patch by: Laureano
2008-10-14 19:03 +0000 [r148917-148988] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 148987 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14
Oct 2008) | 2 lines Some compilers warn, some don't. Fixing.
........
* apps/app_sms.c: App is ignoring 'p' parameter -- initial pause.
(closes issue #13617) Reported by: alecdavis Patches:
app_sms.13oct.diff.txt uploaded by alecdavis (license 585)
* /, apps/app_voicemail.c: Merged revisions 148916 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14
Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean,
even when UTF-8 characters are used in headers like 'Subject' and
'To'. Closes AST-107. ........
2008-10-14 17:38 +0000 [r148913] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c, /: Merged revisions 148912 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue,
14 Oct 2008) | 9 lines Deadlock prevention in chan_local. (closes
issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
by putnopvut (license 60) Tested by: tacvbo ........
2008-10-14 15:15 +0000 [r148868] Tilghman Lesher <tlesher@digium.com>
* apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher
(closes issue #13688) Reported by: irroot Patches:
app_fax-span6.patch uploaded by irroot (license 52) with minor
modifications by me
2008-10-14 15:00 +0000 [r148867] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix reference count issue that Russell
brought up in SIP MWI NOTIFY support. Bump the reference count up
before we add it to the scheduler, duh.
2008-10-14 14:18 +0000 [r148825] Doug Bailey <dbailey@digium.com>
* phoneprov/polycom.xml: Allow MWI registration for all configured
lines.
2008-10-14 11:31 +0000 [r148695-148754] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: fix some references to the owner of a
private structure that may not be present
* Makefile, /: Merged revisions 148736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct
2008) | 3 lines on Ubuntu (at least), recent versions of ld in
binutils delete all debugging symbols when -x is supplied; since
the reasons why -x is being passed are lost in the mists of time,
remove it so debugging will work properly ........
* channels/chan_sip.c: this structure should be static
* channels/chan_sip.c: ensure that *all* fields in the req
structure are cleared out before reusing it; has_to_tag was not
cleared, which caused the second incoming call over a TCP socket
to fail if pedantic checking was enabled
2008-10-14 09:16 +0000 [r148679] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding some clarifications
2008-10-14 08:06 +0000 [r148612] Kevin P. Fleming <kpfleming@digium.com>
* /, main/translate.c: Merged revisions 148611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct
2008) | 3 lines it would be nice if this message printing code
had actually been tested before it was committed... ........
2008-10-14 00:08 +0000 [r148570] 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_realtime.c, include/asterisk/strings.h,
res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c,
apps/app_voicemail.c: Merge realtime_update2 branch, which adds a
new realtime API call named 'update2', which permits updates
which match across multiple columns, instead of requiring all
tables to have a single unique identifier. All of the other API
calls with the exception of 'update' already had the ability to
match on multiple fields, so it was a missing and very desireable
feature that an API call implementing an update should have this,
too. This does not change any outward performance of Asterisk,
but it should make life easier for application developers who use
the RealTime framework.
2008-10-13 17:14 +0000 [r148519] Steve Murphy <murf@digium.com>
* main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the
trie info when they do 'dialplan show ...' (even with debug set
to non-zero); so I set up a 'dialplan debug [context]' cli
command instead, to explicitly show just the trie info. I even
added an extension_exists() call to make sure the trie info is
built. I moved the explanatory header to above the extension loop
to ensure it only prints once. And it will do this now, whether
debug is set or not. I removed the trie printing from the
'dialplan show' command entirely.
2008-10-13 15:56 +0000 [r148471-148474] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Doxygen formatting. (tss tss) - Fixing
language
* main/tcptls.c, channels/chan_sip.c: Highlightning even more bugs
in the current tcp/tls implementation.
* channels/chan_sip.c: Sending a 403 after a 200 is considered very
bad. (found at SIPit)
2008-10-12 09:19 +0000 [r148425] Michiel van Baak <michiel@vanbaak.info>
* res/res_agi.c: fix the 'agi show commands' CLI function. (closes
issue #13666) Reported by: eliel Patches: res_agi.c.patch
uploaded by eliel (license 64)
2008-10-10 21:21 +0000 [r148373-148376] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: The logic used when checking a peer got
changed subtly in the "kill the user" commit and caused calls
relying on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that commit.
(closes issue #13644) Reported by: pj Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by:
pj
* channels/chan_sip.c: Make sure that the inUse and inRinging
fields for a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well. (closes
issue #13668) Reported by: mjc
2008-10-10 18:59 +0000 [r148268-148329] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_config.c: Reset continuation items at the beginning of
each context (suggested by kpfleming).
* CHANGES, pbx/pbx_config.c: Add keyword "same", which allows you
to create multiple steps in a dialplan, without needing to
respecify an extension pattern multiple times. (closes issue
#13632) Reported by: blitzrage Patches:
20081006__bug13632.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
* /, apps/app_voicemail.c: Merged revisions 148257 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10
Oct 2008) | 7 lines User not notified of temporary greeting, if
ODBC storage is in use. (closes issue #13659) Reported by:
moliveras Patches: 20081009__bug13659.diff.txt uploaded by
Corydon76 (license 14) Tested by: moliveras ........
2008-10-10 00:42 +0000 [r148200] Sean Bright <sean.bright@gmail.com>
* include/asterisk.h, main/tdd.c, main/cryptostub.c,
res/res_config_sqlite.c, apps/app_voicemail.c: Don't include
logger.h in asterisk.h by default as it is causing problems
building app_voicemail. Instead, include it where it is needed.
This turned out to be a relatively minor issue because other
headers include logger.h as well. Need to test -addons before
merging this back to 1.6.0. (closes issue #13605) Reported by:
tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright
(license 71) Tested by: mmichelson
2008-10-09 23:54 +0000 [r148144-148160] Mark Michelson <mmichelson@digium.com>
* main/manager.c: The priority was unnecessary for the manager
atxfer, so it has been removed. Furthermore, now we actually use
the Context argument passed to set the transfer context and don't
error out if no context is specified. This addresses the actual
problems outlined in issue 12158. Regarding the other points
brought up, regarding the inability to not transfer to extensions
which cannot be represented by DTMF, it is not enough of a
constraint that it is worth attempting to rework the feature.
(closes issue #12158) Reported by: davidw
* apps/app_voicemail.c: Read the callerid in the correct order and
make sure to read the Urgent flag value from the IMAP headers.
(closes issue #13652) Reported by: jaroth Patches:
imapheaders.patch uploaded by jaroth (license 50)
2008-10-09 23:25 +0000 [r148120] Tilghman Lesher <tlesher@digium.com>
* configs/res_ldap.conf.sample: Fix example schema (closes issue
#12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded
by flyn (license 503)
2008-10-09 23:15 +0000 [r148112] Mark Michelson <mmichelson@digium.com>
* /, main/features.c: Merged revisions 146026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) |
18 lines (closes issue #13579) Reported by: dwagner (closes issue
#13584) Reported by: dwagner Tested by: murf, putnopvut The
thought occurred to me that the res= from the extension spawn was
ending up being returned from the bridge. "Thou shalt not poison
the return value". Made the change and it appears to allow blind
xfers to work as normal. If I'm wrong, reopen the bugs. But it
looks good to me! Many thanks to putnopvut for helping me
reproduce this! ........
2008-10-09 21:47 +0000 [r148000-148071] Tilghman Lesher <tlesher@digium.com>
* formats/format_wav.c, apps/app_minivm.c, channels/chan_agent.c,
main/file.c, res/res_monitor.c, apps/app_voicemail.c: Reverting
format addition for now
* apps/app_minivm.c, channels/chan_agent.c, main/file.c,
res/res_monitor.c, apps/app_voicemail.c: Fudges for wav16, just
like wav49
* formats/format_wav.c: Add native 16kHz format for wav file
format. (Closes issue #13657)
* sounds/sounds.xml, sounds/Makefile: Publish MOH files in sln16
format
* /, apps/app_voicemail.c: Merged revisions 147997 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09
Oct 2008) | 4 lines When blank, callerid name and number should
display "unknown caller" in voicemail emails. (Closes issue
#13643) ........
2008-10-09 19:27 +0000 [r147952] Jeff Peeler <jpeeler@digium.com>
* main/features.c: (closes issue #13139) Reported by: krisk84
Tested by: krisk84 This change prevents a call that is placed in
the parkinglot to be picked up before the PBX is finished. If
another extension dials the parking extension before the PBX
thread has completed at minimum warnings will occur about the PBX
not properly being terminated. At worst, a crash could occur.
2008-10-09 17:48 +0000 [r147899] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk/endian.h: only include this for OpenBSD. At
least FreeBSD is borked when including it (closes issue #13649)
Reported by: ys
2008-10-09 17:46 +0000 [r147896] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Remove "second form" of
extensions, as it no longer applies. Also, cleanup the grammar,
formatting, and introduce several clarifications to the text.
(Closes issue #13654)
2008-10-09 17:04 +0000 [r147854] Terry Wilson <twilson@digium.com>
* phoneprov/000000000000.cfg, res/res_phoneprov.c,
configs/phoneprov.conf.sample: Make phoneprov case-insensitive to
remove the func_strings dependency of the default config
2008-10-09 17:01 +0000 [r147853] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_h323.c: fix some CLI commands we borked during
devcon2008 Thanks rmudget for letting me know and providing hints
on how to fix it best.
2008-10-09 14:17 +0000 [r147807] Steve Murphy <murf@digium.com>
* main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES,
include/asterisk/autoconfig.h.in, channels/vcodecs.c,
main/translate.c, configure.ac, channels/console_video.c,
channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c,
main/rtp.c, main/config.c, main/cli.c, channels/chan_usbradio.c,
configure, channels/console_gui.c, utils/extconf.c: (closes issue
#13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by
nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by
nickpeirson (license 579) Tested by: nickpeirson, murf 1.
replaced all refs to bzero and bcopy to memset and memmove
instead. 2. added a note to the CODING-GUIDELINES 3. add two
macros to asterisk.h to prevent bzero, bcopy from creeping back
into the source 4. removed bzero from configure, configure.ac,
autoconfig.h.in
2008-10-09 01:43 +0000 [r147760-147761] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample: *whistle*
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
support for subscribing to a voice mailbox on a remote SIP server
and making the new/old message count available to local devices.
(issue #AST-77)
2008-10-08 22:32 +0000 [r147714] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c: Some small tweaks regarding realtime
conference announcements. (closes issue #13522) Reported by: DEA
Patches: meetme-rt-fixes.txt uploaded by DEA (license 3)
2008-10-08 22:26 +0000 [r147689] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 147681 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08
Oct 2008) | 3 lines when parsing a text configuration option,
ensure that the buffer on the stack is actually large enough to
hold the legal values of that option, and also ensure that
sscanf() knows to stop parsing if it would overrun the buffer
(without these changes, specifying "buffers=...,immediate" would
overflow the buffer on the stack, and could not have worked as
expected) ........
2008-10-08 20:07 +0000 [r147635] Sean Bright <sean.bright@gmail.com>
* configs/voicemail.conf.sample: Add some examples of IMAP
accounts.
2008-10-08 19:08 +0000 [r147592] Tilghman Lesher <tlesher@digium.com>
* apps/app_sms.c: Correct a typo in the help; also, ensure that the
date and time are correctly set, if not specified in the message.
(Closes issue #13594, closes issue #13595) Reported by: alecdavis
Patches: 20081001__bug13595.diff.txt uploaded by Corydon76
(license 14) Tested by: alecdavis
2008-10-08 14:53 +0000 [r147518] Joshua Colp <jcolp@digium.com>
* /, apps/app_speech_utils.c: Merged revisions 147517 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct
2008) | 2 lines If we receive DTMF make sure that the state of
the speech structure goes back to being not ready. (issue
#LUMENVOX-8) ........
2008-10-08 12:28 +0000 [r147476] Bradley Latus <brad.latus@gmail.com>
* configs/iax.conf.sample: Adjust commented default trunkmtu value
to match documentation above it
2008-10-08 12:15 +0000 [r147388-147457] Sean Bright <sean.bright@gmail.com>
* funcs/func_curl.c, apps/app_meetme.c, cdr/cdr_adaptive_odbc.c,
res/res_odbc.c: Keep up with shadow warnings. One day I'll
actually enable this in the Makefile.
* utils/Makefile: When echoing our copies, strip off ASTTOPDIR from
the front of the source file.
* apps/app_dial.c, channels/chan_dahdi.c, channels/chan_iax2.c:
Move the DAHDI-to-DAHDI operator mode check from app_dial into
chan_dahdi so we don't have to hardcode anything. (closes issue
#13636) Reported by: seanbright Patches: 13636.diff uploaded by
seanbright (license 71) Reviewed by: russellb, putnopvut
2008-10-07 20:15 +0000 [r147266-147347] Michiel van Baak <michiel@vanbaak.info>
* configure, configure.ac: Make format_vorbis_ogg compile on
OpenBSD (closes issue #13639) Reported by: mvanbaak Patches:
2008100700_oggsupportOBSD.diff.txt uploaded by mvanbaak (license
7) 2008100700_oggsupportOBSD-configurescript.diff.txt uploaded by
mvanbaak (license 7) Tested by: mvanbaak
* channels/Makefile: make this work on OpenBSD
* configure, configure.ac: Make sure the configs on OpenBSD are in
/etc/asterisk by default (closes issue #13641) Reported by: jtodd
* contrib/scripts/safe_asterisk_restart,
contrib/scripts/safe_asterisk: use pkill instead of killall to be
more portable
2008-10-07 18:00 +0000 [r147265] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: This was flawed. The issue that I was
trying to address was addressed by adding the imapsecret alias
for imappassword. Will rethink this one and give it another shot
on a rainy day TBD.
2008-10-07 17:49 +0000 [r147264] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: fix wording as pointed out by Corydon
2008-10-07 17:44 +0000 [r147262] Tilghman Lesher <tlesher@digium.com>
* UPGRADE.txt, include/asterisk/options.h, main/asterisk.c,
main/term.c: Allow people to select the old console behavior of
white text on a black background, by using the startup flag '-B'.
2008-10-07 16:52 +0000 [r147191-147194] Sean Bright <sean.bright@gmail.com>
* /, apps/app_voicemail.c: Merged revisions 147193 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue,
07 Oct 2008) | 2 lines Make 'imapsecret' an alias to
'imappassword' in voicemail.conf. ........
* apps/app_voicemail.c: Or not.
* apps/app_voicemail.c: There was a boo-boo in TFOT that is causing
some confusion on the mailing lists so include 'imapsecret' as an
alias to 'imappassword' (and print a little notice nudging users
toward the right option name).
2008-10-07 16:04 +0000 [r147146] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Explicitly setting these fields to NULL was done
because I wasn't sure if they would be NULL otherwise. Since they
will be set automatically, removing.
2008-10-07 14:59 +0000 [r147050-147099] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: If we encounter something in mailbox
options that we don't grok, then spit out a warning instead of
just silently ignoring it.
* apps/app_dial.c: Make sure to compare the correct number of
characters when special-casing our DAHDI operator mode stuff.
Technically, it would work fine, as 'DAH' is currently unique
amongst our channel technologies, but as Jared points out:
<@jsmith> Sure... as long as the technology starts whith DAH....
but it could be DAHDOO!
2008-10-07 02:02 +0000 [r147011] Richard Mudgett <rmudgett@digium.com>
* funcs/func_callerid.c: Independent change from branch issue8824
that is not part of COLP. (-r142574 rmudgett)
2008-10-07 00:02 +0000 [r146970] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: A blind transfer to the parking thread would
cause a segfault because copy_request accesses dst->data w/o
being able to tell whether it is proerly initialized
2008-10-06 23:21 +0000 [r146928] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/threadstorage.h: Update documentation;
AST_THREADSTORAGE() in trunk only takes a single argument.
2008-10-06 23:14 +0000 [r146925] Michiel van Baak <michiel@vanbaak.info>
* res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
funcs/func_odbc.c, include/asterisk/autoconfig.h.in,
configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c,
makeopts.in, res/res_odbc.c, apps/app_voicemail.c: All ODBC parts
can now use either unixodbc or iodbc. This allows for the ODBC
parts to work on OpenBSD as well. 99.99% of the work is done by
seanbright (bow, bow) and I actually did nothing but test and
yell at him that it still didn't work :) Thanks for helping out !
2008-10-06 23:08 +0000 [r146875-146923] Jeff Peeler <jpeeler@digium.com>
* main/features.c, res/res_agi.c, include/asterisk/features.h:
Similar to r143204, masquerade the channel in the case of Park
being called from AGI.
* include/asterisk/endian.h: Mvanbaak said this was needed to
compile on OpenBSD, so put it in the OpenBSD section.
* main/features.c: This commit squashes together three commits
because the wrong approach was originally used. (One of the
commits was only one line.) 1) r143204: The main change here was
to masquerade the channel if the channel that was to be parked
was running a PBX on it. The PBX thread can then maintain full
control of the channel (the zombie) as it expects to while
allowing the parking thread full control of the real (parked)
channel. 2) r143270: Changed park_call_full to hold the
parkinglot lock a little longer, which protects the parkeduser
struct from being freed out from underneath. Made sure that the
parking extension is added to the parking context while holding
the lock thereby ensuring that there are no spurious warnings
from removal attempts when a hangup occurs while the parking lot
is being announced. 3) r143475: (the one liner) compare peer and
chan instead of looking at the parked user (pu), which could have
possibly already have been freed by the parking thread
* main/features.c: fix some comment placement
* main/features.c: Explicitly set args in park_call_exec NULL so in
the case of no options being passed in, there is no garbage
attempted to be used. Also, do not set args to unknown value
again if there are no options passed in.
2008-10-06 21:18 +0000 [r146807] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk/endian.h: make aescrypt.c compile on OpenBSD
again
2008-10-06 21:09 +0000 [r146802] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, /,
channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c,
funcs/func_callerid.c, apps/app_speech_utils.c: Merged revisions
146799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008)
| 8 lines Dialplan functions should not actually return 0, unless
they have modified the workspace. To signal an error (and no
change to the workspace), -1 should be returned instead. (closes
issue #13340) Reported by: kryptolus Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
2008-10-06 17:32 +0000 [r146738] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: Pretty-print a couple configure options
2008-10-06 16:52 +0000 [r146713] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 146711 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06
Oct 2008) | 9 lines Check whether an extension exists in the
_call method, rather than the _alloc method, because we need to
evaluate the callerid (since that data affects whether an
extension exists). (closes issue #13343) Reported by: efutch
Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
(license 14) Tested by: efutch ........
2008-10-06 16:03 +0000 [r146644] Kevin P. Fleming <kpfleming@digium.com>
* /: Merged revisions 146643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146643 | kpfleming | 2008-10-06 10:57:49 -0500 (Mon, 06 Oct
2008) | 8 lines ensure that the private structure for pseudo
channels is created without 'leaking' configuration data from
other configured channels (closes issue #13555) Reported by:
jeffg Patches: issue_13555.patch uploaded by kpfleming (license
421) Tested by: jeffg ........
2008-10-06 15:29 +0000 [r146640] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: This
commit introduces a change to how the "joinempty" and
"leavewhenempty" options are configured in queues.conf. Instead
of using vague terms like "yes," "no," "loose," and "strict," we
now accept a comma-separated list of values to determine when to
consider a member available. Extended details can be found in the
queues.conf.sample file. Note also that the above four referenced
values are still accepted for backwards-compatibility, but are
mapped internally to the new method of representing the option.
AST-105
2008-10-06 00:36 +0000 [r146555-146597] Sean Bright <sean.bright@gmail.com>
* utils/Makefile: Make NOISY_BUILD work for the calls to cp in
utils/Makefile
* utils/Makefile: Quote arguments to cp so we can handle spaces in
our paths.
2008-10-05 22:11 +0000 [r146514] Russell Bryant <russell@digium.com>
* utils/muted.c: Make this build on my mac.
2008-10-05 21:21 +0000 [r146449] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Recorded merge of revisions 146448 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) |
1 line Fix silly formatting. ........
2008-10-05 01:59 +0000 [r146312-146407] Sean Bright <sean.bright@gmail.com>
* build_tools/make_buildopts_h: This is far from optimal, but I
just found a FreeBSD system without md5 installed on it. So look
around for all of the different binaries that we could possibly
use. I'd wager this gets completely replaced by someone else in
less than 24 hours... :)
* main/asterisk.c: Fix a bug with the last item in CLI history
getting duplicated when read from the .asterisk_history file (and
subsequently being duplicated when written). We weren't checking
the result of fgets() which meant that we read the same line
twice before feof() actually returned non- zero. Also, stop
writing out an extra blank line between each item in the history
file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.
* configs/sip_notify.conf.sample: Add ability to remotely reboot
snom phones. Also cleaned up and reorganized
sip_notify.conf.sample a bit as well. Tested snom reboot on snom
360 and verified snom-check-cfg worked as well. (closes issue
#13601) Reported by: mjc Tested by: seanbright
2008-10-03 22:40 +0000 [r146242] Jeff Peeler <jpeeler@digium.com>
* main/features.c: remove superfluous reference counting operations
in manage_parkinglot since ao2_interator_next increments the ref
count automatically
2008-10-03 22:10 +0000 [r146198] Sean Bright <sean.bright@gmail.com>
* main/cli.c: Resolve a subtle bug where we would never
successfully be able to get the first item in the CLI entry list.
This was preventing '!' from showing up in either 'help' or in
tab completion. (closes issue #13578) Reported by: mvanbaak
2008-10-03 18:30 +0000 [r146081] Tilghman Lesher <tlesher@digium.com>
* CHANGES: document meetme schedule changes (related to issue
#11040)
2008-10-03 17:36 +0000 [r146053] Michiel van Baak <michiel@vanbaak.info>
* CHANGES: put a note in CHANGES about the cli_cleanup done during
AstriDevCon
2008-10-03 17:35 +0000 [r146052] Terry Wilson <twilson@digium.com>
* main/dial.c: The dialing API should inherit datastores as well as
variables
2008-10-02 19:30 +0000 [r145959-145962] Russell Bryant <russell@digium.com>
* CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0
* CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0
2008-10-02 18:02 +0000 [r145915] Michiel van Baak <michiel@vanbaak.info>
* apps/app_meetme.c: fix the 'meetme list', 'meetme list concise',
'meetme list $confno' and 'meetme list $confno concise' CLI
commands (closes issue #13586) Reported by: john8675309 Help and
feedback from eliel, thanks!
2008-10-02 17:16 +0000 [r145846] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Permit
the syntax and synopsis fields to be set (for func_odbc).
2008-10-02 16:42 +0000 [r145842] Michiel van Baak <michiel@vanbaak.info>
* apps/app_meetme.c: make this compile under devmode again
2008-10-02 15:28 +0000 [r145771] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: This is much cleaner, methinks.
2008-10-02 15:17 +0000 [r145752] Tilghman Lesher <tlesher@digium.com>
* /, res/res_odbc.c: 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 23:48 +0000 [r145692] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: 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 23:02 +0000 [r145649] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, funcs/func_strings.c,
include/asterisk/localtime.h, main/stdtime/localtime.c: Add
schedule extensions to app_meetme. In addition, the reporter
found a problem within strptime(3), which we are correcting here
with ast_strptime(). (closes issue #11040) Reported by: DEA
Patches: 20080910__bug11040.diff.txt uploaded by Corydon76
(license 14) Tested by: DEA
2008-10-01 22:23 +0000 [r145553-145606] Mark Michelson <mmichelson@digium.com>
* main/features.c: 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: if (!(x) == 0) is the same as if (x).
* main/features.c: 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 17:26 +0000 [r145487] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/realtime_pgsql.sql, /: 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 15:44 +0000 [r145428] Tilghman Lesher <tlesher@digium.com>
* apps/app_sms.c: 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
2008-10-01 14:44 +0000 [r145381] Mark Michelson <mmichelson@digium.com>
* Makefile: Too many times have I mistyped the word 'install' as
'isntall' Now this typo shall no longer be a problem since 'make
isntall' just builds the 'install' target.
2008-10-01 12:29 +0000 [r145329] Russell Bryant <russell@digium.com>
* CHANGES: tabs to spaces
2008-09-30 22:21 +0000 [r145249] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: (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.
2008-09-30 21:32 +0000 [r145226] Russell Bryant <russell@digium.com>
* channels/chan_sip.c, CHANGES: Add support for call pickup on Snom
phones. Asterisk now includes a magic call-id in the dialog-info
event package used with extension state subscriptions on Snom
phones. Then, when the phone sends an INVITE with Replaces for
the special callid, Asterisk will perform a pickup on the
extension that was subscribed to. The original code on this issue
was submitted by xylome. However, contributions have been made by
(at least) mgernoth and pkempgen. The final patch was written by
seanbright, and includes the necessary logic to allow this work
in a technology independent way. (closes issue #5014) Reported
by: xylome Patches: issue5014-trunk.diff uploaded by seanbright
(license 71)
2008-09-30 21:00 +0000 [r145200] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.h, doc/tex/misdn.tex,
channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Miscellaneous
formatting changes to make v1.4 and trunk more merge compatible
in the mISDN area. channels/chan_misdn.c * Eliminated redundant
code in cb_events() EVENT_SETUP
2008-09-28 23:32 +0000 [r145121] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_unistim.c, res/res_config_pgsql.c,
apps/app_meetme.c, res/ais/clm.c, res/res_limit.c,
main/taskprocessor.c, channels/chan_console.c, apps/app_queue.c,
channels/chan_oss.c, main/astobj2.c, main/cli.c,
channels/chan_dahdi.c, main/manager.c, channels/chan_misdn.c,
channels/chan_features.c, res/res_agi.c, channels/chan_h323.c,
res/ais/evt.c, res/res_config_ldap.c, apps/app_mixmonitor.c,
res/res_clioriginate.c: Merge the cli_cleanup branch. This work
is done by lmadsen, junky and mvanbaak during AstriDevCon. This
is the second audit the CLI got, and this time lmadsen made sure
he had _ALL_ modules loaded that have CLI commands in them.
2008-09-28 21:39 +0000 [r145076] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c: Change several improper "sizeof" to
"strlen", as sizeof in that context would incorrectly use the
size of a pointer, rather than the length of a string. (Closes
issue #13574)
2008-09-28 17:08 +0000 [r145027] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: rename chandup() and clarify its usage
2008-09-27 16:17 +0000 [r144949-144951] Kevin P. Fleming <kpfleming@digium.com>
* utils/Makefile: remove incorrect comment
* agi/Makefile, utils/Makefile, include/asterisk/astmm.h: fix bugs
caused by r144949 when MALLOC_DEBUG is defined
* include/asterisk.h, /, main/Makefile, main/ast_expr2.y,
Makefile.moddir_rules, utils/astman.c, main/ast_expr2.c,
Makefile, utils/Makefile, main/ast_expr2f.c, pbx/pbx_ael.c,
main/astmm.c, utils/ael_main.c, main/stdtime/localtime.c,
utils/extconf.c, main/ast_expr2.fl: 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 ........
2008-09-27 00:49 +0000 [r144879] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_dahdi.c, apps/app_queue.c: fix a couple of CLI
commands that did not have a help description.
2008-09-26 23:12 +0000 [r144829] Joshua Colp <jcolp@digium.com>
* configs/rtp.conf.sample: Update documentation to include default
setting. This is for you jtodd!
2008-09-26 18:02 +0000 [r144482-144681] Steve Murphy <murf@digium.com>
* pbx/pbx_lua.c: (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 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): Matt suggests we
remove utils/build-extensions-conf.lua, as per bug 12961, it is
no longer necessary.
* main/pbx.c, funcs/func_cut.c, channels/chan_oss.c,
apps/app_playback.c: (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: (closes issue #13559) Reported by: mnicholson
Patches: pbx_lua8.diff uploaded by mnicholson (license 96)
* pbx/pbx_lua.c, configs/extensions.lua.sample,
include/asterisk/hashtab.h: 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: (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!
2008-09-25 01:46 +0000 [r144357] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 144356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r144356 | tilghman | 2008-09-24 20:44:47 -0500 (Wed, 24 Sep 2008)
| 6 lines Backport Hebrew language to voicemail. (closes issue
#13155) Reported by: greenfieldtech Patches:
voicemail-hebrew-patch-1.4-SVN.c.patch uploaded by greenfieldtech
(license 369) ........
2008-09-24 22:05 +0000 [r144314] Doug Bailey <dbailey@digium.com>
* res/res_phoneprov.c: Blanch the 404 error message for those with
no sense of humor
2008-09-24 08:42 +0000 [r144257] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 144238 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24
Sep 2008) | 1 line improved helptext of misdn_set_opt. ........
2008-09-24 06:43 +0000 [r144199] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: Create a 'hashcompat' option that permits the
results of a CURL() able to be passed directly into the HASH()
function. Requested via the -users list, and committed at
Astricon in the Code Zone.
2008-09-23 23:33 +0000 [r144149] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a conflict in flag values
2008-09-23 16:52 +0000 [r144067] Steve Murphy <murf@digium.com>
* /, main/features.c: 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 15:37 +0000 [r144025] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: 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 22:50 +0000 [r143904] Sean Bright <sean.bright@gmail.com>
* /, formats/format_pcm.c: 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 09:53 +0000 [r143799-143843] Michiel van Baak <michiel@vanbaak.info>
* doc/tex/privacy.tex: fix privacymanager example so it shows how
to use the PRIVACYMRGSTATUS variable
* doc/tex/privacy.tex: document the new context argument for
privacymanager so people can do pattern matching on the input
* doc/tex/privacy.tex: fix privacy documentation. We no longer do
priority jumping +101
* channels/chan_skinny.c: make 'module unload chan_skinny.so'
actually work. (closes issue #13524) Reported by: wedhorn
Patches: unload.diff uploaded by wedhorn (license 30) With small
tweak by me to prevent a crash
2008-09-20 00:52 +0000 [r143737] Sean Bright <sean.bright@gmail.com>
* /, contrib/scripts/vmail.cgi: 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 21:41 +0000 [r143697] Steve Murphy <murf@digium.com>
* /: This blocks 143674 from trunk; it appears to already done in
trunk, since ast_odbc_direct_execute creates a new stmt for each
attempt.
2008-09-19 15:43 +0000 [r143609] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: 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 23:41 +0000 [r143559] Steve Murphy <murf@digium.com>
* /, channels/chan_sip.c: 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 20:57 +0000 [r143405] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: 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 20:25 +0000 [r143340-143400] Mark Michelson <mmichelson@digium.com>
* main/astmm.c: If attempting to free a NULL pointer when
MALLOC_DEBUG is set, don't bother searching for a region to free,
just immediately exit. This has the dual benefit of suppressing a
warning message about freeing memory at (nil) and of optimizing
the free() replacement by not having to do any futile searching
for the proper region to free. (closes issue #13498) Reported by:
pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested
by: pj
* /, main/rtp.c: 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 21:31 +0000 [r143141] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: 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 22:16 +0000 [r143082] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: 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 14:15 +0000 [r143034] Sean Bright <sean.bright@gmail.com>
* apps/app_osplookup.c: Everytime a compile fails, a puppy dies.
2008-09-13 13:54 +0000 [r142992-143031] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, channels/chan_iax2.c, channels/iax2-parser.c:
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
* channels/chan_unistim.c, main/udptl.c, apps/app_meetme.c,
res/res_snmp.c, codecs/codec_adpcm.c, res/res_phoneprov.c,
codecs/codec_gsm.c, apps/app_alarmreceiver.c,
channels/chan_gtalk.c, res/res_http_post.c,
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
res/res_jabber.c, main/enum.c, res/res_config_sqlite.c,
main/config.c, main/loader.c, main/cdr.c, channels/chan_dahdi.c,
channels/chan_phone.c, res/res_smdi.c, main/manager.c,
funcs/func_config.c, apps/app_osplookup.c,
channels/chan_skinny.c, funcs/func_odbc.c, main/features.c,
apps/app_minivm.c, main/http.c, channels/chan_alsa.c,
apps/app_amd.c, apps/app_directory.c, res/res_config_ldap.c,
apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c,
res/res_config_pgsql.c, main/dnsmgr.c, codecs/codec_g722.c,
channels/chan_sip.c, apps/app_festival.c, codecs/codec_speex.c,
codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h,
channels/chan_agent.c, codecs/codec_g726.c,
channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c,
main/rtp.c, apps/app_playback.c, channels/chan_jingle.c,
channels/chan_h323.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
res/res_indications.c, main/asterisk.c, res/res_odbc.c,
main/dsp.c, apps/app_voicemail.c: Create a new config file
status, CONFIG_STATUS_FILEINVALID for differentiating when a file
is invalid from when a file is missing. This is most important
when we have two configuration files. Consider the following
example: Old system: sip.conf users.conf Old result New result
======== ========== ========== ========== Missing Missing SIP
doesn't load SIP doesn't load Missing OK SIP doesn't load SIP
doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK
Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid
SIP loads incompletely SIP doesn't load Invalid Missing SIP
doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP
doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So
in the case when users.conf doesn't load because there's a typo
that disrupts the syntax, we may only partially load users,
instead of failing with an error, which may cause some calls not
to get processed. Worse yet, the old system would do this with no
indication that anything was even wrong. (closes issue #10690)
Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded
by Corydon76 (license 14)
2008-09-12 22:24 +0000 [r142929] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: 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 20:49 +0000 [r142866] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: 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 ........
2008-09-12 18:22 +0000 [r142808] Michiel van Baak <michiel@vanbaak.info>
* /: Recorded merge of revisions 142807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r142807 | mvanbaak | 2008-09-12 19:59:25 +0200 (Fri, 12 Sep 2008)
| 2 lines fix copyright year range ........
2008-09-12 16:54 +0000 [r142741-142748] Tilghman Lesher <tlesher@digium.com>
* main/app.c: When checking for an encoded character, make sure the
string isn't blank, first. (Closes issue #13470)
* /, apps/app_voicemail.c: 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 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 04:50 +0000 [r142676] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/pbx.c, /, main/features.c,
include/asterisk/channel.h, apps/app_queue.c: 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.
........
2008-09-12 00:49 +0000 [r142635] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_adaptive_odbc.c: Build under dev-mode
2008-09-11 23:12 +0000 [r142576] Steve Murphy <murf@digium.com>
* /, main/features.c: 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.
........
2008-09-11 21:45 +0000 [r142536] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
Add usegmtime, as per the recent -users list discussion, and also
add my explanation to the file, since that additional text helps
people understand the concept.
2008-09-10 22:11 +0000 [r142475] Steve Murphy <murf@digium.com>
* /, main/features.c: 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 19:09 +0000 [r142417] Sean Bright <sean.bright@gmail.com>
* /, configure, acinclude.m4: 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 16:55 +0000 [r142359] Tilghman Lesher <tlesher@digium.com>
* /, sounds/Makefile: 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 16:41 +0000 [r142318-142355] Russell Bryant <russell@digium.com>
* /, main/sched.c: 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) ........
* CHANGES: Move last change to CHANGES up to the 1.6.2 section
2008-09-09 22:08 +0000 [r142280] Philippe Sultan <philippe.sultan@gmail.com>
* configs/jabber.conf.sample, CHANGES, res/res_jabber.c: Disable
autoprune by default. (closes issue #13411) Reported by: caio1982
Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license
22) Tested by: caio1982
2008-09-09 19:16 +0000 [r142219] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: 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 ........
2008-09-09 17:30 +0000 [r142181] Richard Mudgett <rmudgett@digium.com>
* main/callerid.c: Cleaned up comment
2008-09-09 17:15 +0000 [r142080-142146] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This is the trunk version of the patch to close
issue 12979. The difference between this and the 1.6.0 and 1.6.1
versions is that this is a much more invasive change. With this,
we completely get rid of the interfaces list, along with all its
helper functions. Let me take a moment to say that this change
personally excites me since it may mean huge steps forward
regarding proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also results
in a huge reduction in lines of code and complexity. Way to go
Brett! (closes issue #12979) Reported by: sigxcpu Patches:
20080710_issue12979_queue_custom_state_interface_trunk_2.diff
uploaded by bbryant (license 36) Tested by: sigxcpu, putnopvut
* /, channels/chan_sip.c: 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 15:44 +0000 [r142064] Russell Bryant <russell@digium.com>
* /, main/features.c: 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 [r142000] Bradley Latus <brad.latus@gmail.com>
* include/asterisk/astobj2.h: Minor fix to doco
2008-09-09 12:32 +0000 [r141995-141998] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: 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: 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 01:47 +0000 [r141949] Russell Bryant <russell@digium.com>
* main/channel.c: 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 23:00 +0000 [r141810-141906] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Optimization: The only reason we should check
member status is if the queue has a joinempty or a leavewhenempty
setting which could cause the caller to not join the queue or
exit the queue. Prior to this patch, we could potentially
traverse the entire queue's member list for no reason since even
if the members are currently not available in some way we're
going to let the caller join the queue anyway.
* channels/chan_sip.c: Um, apparently I didn't actually finish
merging before committing. Bad bad bad
* /, channels/chan_sip.c: 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 21:05 +0000 [r141807] Russell Bryant <russell@digium.com>
* main/pbx.c, /: Merged revisions 141806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
| 7 lines When doing an async goto, detect if the channel is
already in the middle of a masquerade. This can happen when
chan_local is trying to optimize itself out. If this happens,
fail the async goto instead of bursting into flames. (closes
issue #13435) Reported by: geoff2010 ........
2008-09-08 20:18 +0000 [r141745] Jason Parker <jparker@digium.com>
* Makefile, /, redhat (removed): 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 17:13 +0000 [r141682] Sean Bright <sean.bright@gmail.com>
* build_tools/make_buildopts_h: 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
2008-09-07 00:04 +0000 [r141626] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_curl.c: make func_curl.c compile under devmode.
2008-09-06 20:19 +0000 [r141566] Steve Murphy <murf@digium.com>
* /, channels/chan_sip.c: Merged revisions 141565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
line This fix comes from Joshua Colp The Brilliant, who, given
the trace, came up with a solution. This will most likely will
close 13235 and 13409. I'll wait till Monday to verify, and then
close these bugs. ........
2008-09-06 15:40 +0000 [r141504-141507] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: Get rid of the casts that cause warnings on
OpenBSD. The compiler is errantly detecting warnings when we
redefine a structure each time it is used, even though the
structure is identical. Reported by: mvanbaak, via #asterisk-dev
* /, res/res_agi.c: 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
........
2008-09-06 12:03 +0000 [r141464] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_sip.c, channels/chan_iax2.c, main/cli.c: Some fixes
to autocompletion in some commands. Changes applied by this
patch: - Fix autocompletion in 'sip prune realtime', sip peers
where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers Also I have
modified the syntax in the usage, was wrong... - Pass
ast_cli_args->argv and ast_cli_args->argc while running
autocompletion on CLI commands (CLI_GENERATE). With this we avoid
comparisons on ast_cli_args->line like this: strcasestr(a->line,
" description") strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on.. Making the code
more confusing (check the spaces in description!). The only thing
we must be sure is to first check a->pos or a->argc. - Fix 'iax2
prune realtime' autocompletion, now we autocomplete this command
with 'all' & 'iax2 peers', check a look that iax2 peers where all
the peers, now only the ones in the cache.. (closes issue #13133)
Reported by: eliel Patches: clichanges.patch uploaded by eliel
(license 64)
2008-09-05 22:03 +0000 [r141367-141425] Mark Michelson <mmichelson@digium.com>
* funcs/func_curl.c: Fix func_curl compilation
* /, channels/chan_agent.c: 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 19:12 +0000 [r141328] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c, CHANGES: Add the CURLOPT dialplan function,
which permits setting various options for use with the CURL
dialplan function. (closes issue #12920) Reported by: davevg
Patches: 20080904__bug12920.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76, davevg
2008-09-05 14:18 +0000 [r141115-141157] Steve Murphy <murf@digium.com>
* main/channel.c, /: Merged revisions 141156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
line A small change to prevent double-posting of CDR's; thanks to
Daniel Ferrer for bringing it to our attention ........
* pbx/ael/ael-test/ref.ael-vtest25 (added), /,
pbx/ael/ael-test/ael-vtest25/extensions.ael,
pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
revisions 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 17:27 +0000 [r141039] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c, res/res_agi.c: 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 20:16 +0000 [r140975] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix some locking order issues in app_queue.
This was brought up by atis on IRC a while ago.
2008-09-03 18:06 +0000 [r140938] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c, CHANGES: Added 'skinny show lines
verbose' This will print the subs and their status for every line
(if any). wedhorn did most of the work with his patch which
introduced 'skinny show debug' but a discussion on IRC stated
that it should be added to 'skinny show lines' Input on the
output format by Qwell on IRC. (closes issue #13344) Reported by:
wedhorn
2008-09-03 14:41 +0000 [r140860-140887] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix compilation
* /, apps/app_voicemail.c: 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 14:01 +0000 [r140824] 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: 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.
2008-09-03 13:48 +0000 [r140821] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_sqlite.c: Move some duplicated code into a separate
function. Also try to do some wacky stuff in the commit message,
like: a newline \n a bell \a a tab \t a format specification %p
That is all.
2008-09-03 13:41 +0000 [r140817-140820] Russell Bryant <russell@digium.com>
* main/pbx.c: Formatting change to test something on the svn server
* /, main/poll.c: 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 23:48 +0000 [r140752] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: 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 23:44 +0000 [r140691-140749] Steve Murphy <murf@digium.com>
* main/cdr.c, /: 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 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 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 18:15 +0000 [r140606] Sean Bright <sean.bright@gmail.com>
* /, channels/chan_iax2.c: 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 15:11 +0000 [r140563-140566] Russell Bryant <russell@digium.com>
* codecs/codec_resample.c, apps/app_jack.c: Update instructions for
getting libresample
* res/ais/lck.c (removed), res/ais/ckpt.c (removed), res/ais/amf.c
(removed): I'm not sure how these files got to trunk (probably my
fault), but they should not be here
2008-09-02 14:41 +0000 [r140559] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: When a call is rejected because of
call-limit, the channel driver is behaving as expected, so we
shouldn't report it as an error. Change to LOG_NOTICE instead.
2008-08-29 17:53 +0000 [r140491] Jeff Peeler <jpeeler@digium.com>
* main/features.c, CHANGES: 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 17:47 +0000 [r140418-140489] Mark Michelson <mmichelson@digium.com>
* 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, apps/app_queue.c,
channels/chan_iax2.c, main/config.c: 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: 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 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 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 23:23 +0000 [r140355] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: Oops
2008-08-27 20:11 +0000 [r140301] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: 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 21:59 +0000 [r140246] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Move the mwi send thread functionality
back into the do_monitor thread so that it is easier to manage
CID spill resources when do_monitor needs to be killed. (closes
issue #13213) Reported by: bbryant
2008-08-26 18:48 +0000 [r140205] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: 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 18:46 +0000 [r140201] Tilghman Lesher <tlesher@digium.com>
* apps/app_followme.c: OpenBSD compat fix (reminded by mvanbaak on
#asterisk-dev)
2008-08-26 18:11 +0000 [r140169] Russell Bryant <russell@digium.com>
* Makefile: Fix building menuselect-tree with PRINT_DIR set. We
_must_ use the --quiet flag here, or else some arbitrary text
will end up in the resulting menuselect-tree file and things will
explode.
2008-08-26 18:05 +0000 [r140167] Tilghman Lesher <tlesher@digium.com>
* configs/followme.conf.sample, apps/app_followme.c: Standardize
the option names for consistency (but continue to work with the
existing names for backwards compatibility). (closes issue
#13370) Reported by: jsturtevant
2008-08-26 16:10 +0000 [r140061] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: 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. ........
2008-08-26 15:57 +0000 [r140057] Steve Murphy <murf@digium.com>
* main/cdr.c, configs/cdr.conf.sample, CHANGES,
include/asterisk/options.h: (closes issue #13366) Reported by:
erousseau This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it could only be
applied to trunk. Basically, for accounting where "initiated"
seconds are billed for, if the microseconds field on the end time
is greater than the microseconds field for the answer time, add
one second to the billsec field. The implementation was requested
by erousseau, and I've implemented it as requested. I've updated
the CHANGES, the cdr.conf.sample, and the .h files accordingly,
to accept and set a flag for the corresponding new option. cdr.c
adds in the extra second based on the usec fields if the option
is set. Tested, seems to be working fine.
2008-08-26 15:29 +0000 [r140053] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: 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 23:13 +0000 [r139981] Tilghman Lesher <tlesher@digium.com>
* Makefile, doc/asterisk.8, include/asterisk/options.h,
main/asterisk.c, main/term.c: Optional light colored background,
for those who use black on white terminals. (closes issue #13306)
Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt
uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen
2008-08-25 21:48 +0000 [r139928] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: 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 21:32 +0000 [r139915] Sean Bright <sean.bright@gmail.com>
* build_tools/get_moduleinfo, /, build_tools/get_makeopts: 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 20:59 +0000 [r139870] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: 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 17:24 +0000 [r139832] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add output of variables to AgentRingNoAnswer
manager event if eventwhencalled is set to "vars" in queues.conf.
Yay for consistency. (closes issue #13369) Reported by: srt
Patches: 13369_agentringnoanswer_variables.diff uploaded by srt
(license 378)
2008-08-25 16:02 +0000 [r139775] Tilghman Lesher <tlesher@digium.com>
* doc/followme.txt (added), apps/app_followme.c: Realtime
capabilities for the Find-Me-Follow-Me application. (closes issue
#13295) Reported by: Corydon76 Patches:
20080813__followme_realtime_enabled.diff.txt uploaded by
Corydon76 (license 14) Tested by: dferrer
2008-08-25 15:54 +0000 [r139770] Steve Murphy <murf@digium.com>
* main/pbx.c, /, main/features.c: 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 16:26 +0000 [r139704-139707] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: Memory leak
* cdr/cdr_pgsql.c: Eliminate open coding of ast_str
2008-08-22 22:32 +0000 [r139627-139662] Steve Murphy <murf@digium.com>
* /, main/features.c: 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
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 21:57 +0000 [r139622-139624] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: 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: remove extra comma typo
2008-08-22 20:20 +0000 [r139457-139563] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: 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: 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 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 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 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 23:41 +0000 [r139391] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 139387 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21
Aug 2008) | 3 lines Fixes loop that could possibly never exit in
the event of a channel never being able to be opened or specify
after a restart. (closes issue #11017) ........
2008-08-21 23:00 +0000 [r139345-139346] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* apps/app_receivefax.c (removed), apps/app_sendfax.c (removed):
oops
* apps/app_receivefax.c (added), apps/app_sendfax.c (added):
initiate T38 negotiation in FaxSend; use channel variables; other
stuff too
2008-08-21 09:55 +0000 [r139281] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Fix two memory leaks in chan_gtalk, thanks
Eliel! (closes issue #13310) Reported by: eliel Patches:
chan_gtalk.c.patch uploaded by eliel (license 64)
2008-08-20 22:16 +0000 [r139215] Russell Bryant <russell@digium.com>
* /, apps/app_chanspy.c: 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 22:06 +0000 [r139210] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: 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 20:03 +0000 [r139153-139154] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Remove extraneous debugging messages.
* codecs/codec_dahdi.c: Fix bug where the samples were not accurate
when in G723 mode, which would cause the timestamp field of the
RTP header to be invalid.
2008-08-20 17:25 +0000 [r139083] Steve Murphy <murf@digium.com>
* main/cdr.c, /: 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 16:16 +0000 [r139020] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: fix unholding phones after hangup on
older cisco phones. Patch by wedhorn.
2008-08-20 15:38 +0000 [r138887-139016] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: 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 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 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 16:56 +0000 [r138851] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: chan_skinny now respects callwaiting=no
(closes issue #12691) Reported by: sbisker Patches:
callwaitingv1.diff uploaded by wedhorn (license 30) Tested by:
wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with
latest firmware
2008-08-19 16:31 +0000 [r138815-138845] Steve Murphy <murf@digium.com>
* res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h,
utils/ael_main.c, utils/conf2ael.c: 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:
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 15:58 +0000 [r138814] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Mention JID rather than SreenName in help
messages
2008-08-19 00:10 +0000 [r138775-138780] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: Let it compile now, too (woops)
* channels/chan_sip.c: And remove code we don't need anymore.
* channels/chan_sip.c: While we're at it, make this machine
parseable too.
* channels/chan_sip.c: Change event header to RegistrationTime to
be more consistent (and avoid breaking existing frameworks).
Pointed out by Laureano on #asterisk-dev.
2008-08-18 21:07 +0000 [r138738] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
doc/tex/misdn.tex, channels/chan_misdn.c,
configs/misdn.conf.sample, channels/misdn/isdn_lib.c,
channels/misdn_config.c: channels/chan_misdn.c * Made
bearer2str() use allowed_bearers_array[] * Made use the causes.h
defines instead of hardcoded numbers. * Made use Asterisk
presentation indicator values if either of the mISDN presentation
or screen options are negative. * Updated the misdn_set_opt
application option descriptions. * Renamed the awkward Caller ID
presentation misdn_set_opt application option value not_screened
to restricted. Deprecated the not_screened option value.
channels/misdn/isdn_lib.c * Made use the causes.h defines instead
of hardcoded numbers. * Fixed some spelling errors and typos. *
Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h * Added doxygen comments to struct
misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
comments to struct misdn_stack. channels/misdn_config.c
configs/misdn.conf.sample * Updated the mISDN presentation and
screen parameter descriptions. doc/tex/misdn.tex * Updated the
misdn_set_opt application option descriptions. * Fixed some
spelling errors and typos.
2008-08-18 20:23 +0000 [r138687-138694] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, apps/app_queue.c: 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 138685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
2008) | 10 lines Change the inequalities used in app_queue with
regards to timeouts from being strict to non-strict for more
accuracy. (closes issue #13239) Reported by: atis Patches:
app_queue_timeouts_v2.patch uploaded by atis (license 242)
........
2008-08-18 15:54 +0000 [r138631] Jason Parker <jparker@digium.com>
* Makefile: Remove option that isn't valid here.
2008-08-18 02:13 +0000 [r138518] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: add missing define for SS7 in
dahdi_restart
2008-08-17 14:12 +0000 [r138442-138482] Sean Bright <sean.bright@gmail.com>
* main/features.c: 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: 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: 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: 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: Since it's introduction in revision
3497, cdr_tds has *never* read the port configuration option from
cdr_tds.conf. So go ahead and remove it from the sample config.
2008-08-16 13:07 +0000 [r138409-138412] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Fix compilation warnings (found with
dev-mode)
* main/pbx.c: Also make sure hinting won't crash on reload. (Closes
issue #13312)
2008-08-16 01:13 +0000 [r138311-138361] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: 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 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 22:54 +0000 [r138206-138260] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: 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 ........
* main/pbx.c, configs/extensions.conf.sample: Remove deprecated
syntax from sample config file (closes issue #13314) Reported by:
kue
2008-08-15 20:12 +0000 [r138155] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
dfd to match 1.4 (left over from DAHDI transition)
2008-08-15 19:36 +0000 [r138086-138148] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Change free to ast_free_ptr, too
* main/pbx.c: e->data can be NULL, so use the safe version of
ast_strdup() (closes issue #13312) Reported by: pj
* channels/chan_sip.c: regseconds is actually stored as the epoch
time, not registration length
2008-08-15 15:09 +0000 [r138028] Russell Bryant <russell@digium.com>
* main/autoservice.c, /: Merged revisions 138027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
| 9 lines Ensure that when a hangup occurs in autoservice, that a
hangup frame gets properly deferred to be read from the channel
owner when it gets taken out of autoservice. (closes issue
#12874) Reported by: dimas Patches: v1-12874.patch uploaded by
dimas (license 88) ........
2008-08-15 15:03 +0000 [r138024] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: 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 [r137987] Russell Bryant <russell@digium.com>
* doc/tex/Makefile: Fix a bashism that causes an error when trying
to build the pdf on ubuntu
2008-08-14 18:47 +0000 [r137933] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_sqlite3_custom.c: 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 18:12 +0000 [r137901] Russell Bryant <russell@digium.com>
* CHANGES: Prepare for adding 1.6.2 changes
2008-08-14 16:52 +0000 [r137848] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, /: 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 15:32 +0000 [r137812] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: 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 15:03 +0000 [r137780] Sean Bright <sean.bright@gmail.com>
* cdr/cdr_tds.c: 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 14:15 +0000 [r137732] Russell Bryant <russell@digium.com>
* /, configs/sip.conf.sample: 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 02:03 +0000 [r137680] Kevin P. Fleming <kpfleming@digium.com>
* /, Zaptel-to-DAHDI.txt: 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 ........
|