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2010-07-09Merged revisions 275144 via svnmerge from mnicholson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275145 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Merged revisions 275022 via svnmerge from russell1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274164 via svnmerge from mmichelson1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul 2010) | 22 lines Merged revisions 274157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being accepted. A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue #17571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@274168 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-05Merged revisions 273886 via svnmerge from pabelanger1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul 2010) | 15 lines Merged revisions 273884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines Remove extra line breaks from 'core show config mappings' (closes issue #17583) Reported by: pabelanger Patches: issue17583.patch uploaded by pabelanger (license 224) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273888 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Merged revisions 273718 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010) | 15 lines Merged revisions 273717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup. (closes issue #17564) Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273719 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273566 via svnmerge from russell1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010) | 14 lines Merged revisions 273565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines Don't return a partially initialized datastore. If memory allocation fails in ast_strdup(), don't return a partially initialized datastore. Bad things may happen. (related to ABE-2415) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273571 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273352 via svnmerge from mnicholson1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 | mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2 lines Fixed whitespace problems ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273353 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273350 via svnmerge from mnicholson1-4/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 | mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2 lines Altered my comment about TCP_NODELAY ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273351 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-30Merged revisions 273270 via svnmerge from mnicholson1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 | mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2 lines Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This regression was introduced in r48338. AST-359 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273271 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-30Merged revisions 273233 via svnmerge from pabelanger1-6/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 | pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11 lines Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (closes issue #17469) Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273234 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-30Merged revisions 273144 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 | tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines Permission checking for the system application is backwards. (closes issue #17550) Reported by: kenner Patches: manager.c.diff uploaded by kenner (license 1040) Tested by: kenner ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273145 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-30Merged revisions 273142 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 | tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines Don't attempt to proceed if our internal parser indicates an invalid file. (closes issue #17560) Reported by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273143 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 273058 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010) | 11 lines Recorded merge of revisions 273057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines _Really_ skip the channel... don't just retry for another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273059 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 273054 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 | tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11 lines Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it. (closes issue #17504) Reported by: rrb3942 Patches: showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested by: rrb3942 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273056 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 253357 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 | russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines Increase CLI command output timeout for asterisk -rx to 60 seconds. (closes issue #17049) Reported by: russell Tested by: russell Review: https://reviewboard.asterisk.org/r/573/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272972 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Merged revisions 272926 via svnmerge from tilghman1-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010) | 15 lines Merged revisions 272925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines Don't change ownership/group/permissions on run directory, if it already exists. (closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272927 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Merged revisions 272923 via svnmerge from tilghman1-15/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010) | 19 lines Merged revisions 272921-272922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines Change the way that we read include files, to accommodate for changes in GCC 4.4. (closes issue #17472) Reported by: seandarcy Patches: config2.patch uploaded by nivan (license 1066) Tested by: nivan ........ r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim trailing blanks on #includes ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-25Backport unit test API from trunk.russell3-1/+900
Also, update existing test modules that were already in this branch but had been converted to the unit test API in trunk. Review: https://reviewboard.asterisk.org/r/748/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272531 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272252 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 | pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8 lines Correct manager variable 'EventList' case. (closes issue #17520) Reported by: kobaz Patches: manager.patch uploaded by kobaz (license 834) Tested by: lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272253 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271831 via svnmerge from dvossel1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 | dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines fixes attended transfer behavior when both transferee and transferer hung up If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271832 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270974 via svnmerge from mnicholson2-9/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r270974 | mnicholson | 2010-06-16 15:34:31 -0500 (Wed, 16 Jun 2010) | 8 lines Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes. (closes issue #17496) Reported by: ManChicken (closes issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch uploaded by chappell (license 8) Tested by: DennisD, gentlec, damage, wimpy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270975 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Merged revisions 270584 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r270584 | tilghman | 2010-06-15 13:26:26 -0500 (Tue, 15 Jun 2010) | 12 lines Merged revisions 270583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines Variables have always been case-sensitive, so we should not be removing case-insensitive matches. Bug reported via the -dev list. See http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270585 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11Merged revisions 269976 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r269976 | tilghman | 2010-06-11 13:31:14 -0500 (Fri, 11 Jun 2010) | 15 lines Merged revisions 269960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines For SpeeX, 0 bits remaining is valid and does not need an emitted warning. (closes issue #15762) Reported by: nblasgen Patches: issue15672.patch uploaded by pabelanger (license 224) Tested by: nblasgen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269977 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Merged revisions 269822 via svnmerge from mmichelson1-4/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r269822 | mmichelson | 2010-06-10 14:34:03 -0500 (Thu, 10 Jun 2010) | 25 lines Merged revisions 269821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines Fix potential crash when writing raw SLIN audio on a PLC-enabled channel. The issue here was that the frame created when adjusting for PLC had no offset to its audio data. If this frame were translated to another format prior to being sent out an RTP socket, all went well because the translation code would put an appropriate offset into the frame. However, if the SLIN audio were not translated before being sent out the RTP socket, bad things would happen. Specifically, the ast_rtp_raw_write makes the assumption that the frame has at least enough of an offset that it can accommodate an RTP header. This was not the case. As such, data was being written prior to the allocation, likely corrupting the data the memory allocator had written. Thus when the time came to free the data, all hell broke loose. ....Well, Asterisk crashed at least. The fix was just what one would expect. Offset the data in the frame by a reasonable amount. The method I used is a bit odd since the data in the frame is 16 bit integers and not bytes. I left a big ol' comment about it. This can be improved on if someone is interested. I was more interested in getting the crash resolved. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Merged revisions 269636 via svnmerge from tilghman2-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r269636 | tilghman | 2010-06-10 03:15:45 -0500 (Thu, 10 Jun 2010) | 16 lines Merged revisions 269635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines Ensure restartable system calls can restart (BSD signal semantics) This eliminates the annoying <beep> on the console. (closes issue #17477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269637 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269417 via svnmerge from russell1-5/+14
https://origsvn.digium.com/svn/asterisk/trunk ........ r269417 | russell | 2010-06-09 16:11:43 -0500 (Wed, 09 Jun 2010) | 6 lines Resolve an invalid memory read on an event. Valgrind pointed out that attempting to get an IE value from an event that has no IEs produces an invalid memory read past the end of the event. Thanks to mmichelson for pointing the problem out to me and then testing the fix. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269418 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269346 via svnmerge from pabelanger1-6/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r269346 | pabelanger | 2010-06-09 13:32:52 -0400 (Wed, 09 Jun 2010) | 19 lines Merged revisions 269334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines Fix Debian init script to not use -c. When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. We now handle color displays properly. (closes issue #16784) Reported by: pabelanger Patches: 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger, tilghman ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Merged revisions 268653,268731 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r268653 | tilghman | 2010-06-07 12:14:40 -0500 (Mon, 07 Jun 2010) | 2 lines Avoid unloading res_smdi twice. (closes issue #17237) Reported by: pabelanger ........ r268731 | tilghman | 2010-06-07 13:59:27 -0500 (Mon, 07 Jun 2010) | 4 lines Event well was going dry. (issue #17234) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268732 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Merged revisions 268690 via svnmerge from pabelanger1-5/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r268690 | pabelanger | 2010-06-07 13:34:45 -0400 (Mon, 07 Jun 2010) | 11 lines Set threshold for silence detection defaults to 256 (closes issue #15685) Reported by: david_s5 Patches: dsp-silence-threshold-init.diff uploaded by dant (license 670) issue15685.patch.v5 uploaded by pabelanger (license 224) Tested by: danti Review: https://reviewboard.asterisk.org/r/670/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268691 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Merged revisions 268456 via svnmerge from tilghman1-9/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r268456 | tilghman | 2010-06-05 12:55:28 -0500 (Sat, 05 Jun 2010) | 14 lines Fix crash in DTMF detection. What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue #17371) Reported by: alecdavis (closes issue #17474) Reported by: kenner ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268457 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Merged revisions 268454 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r268454 | tilghman | 2010-06-05 12:27:12 -0500 (Sat, 05 Jun 2010) | 5 lines Verify event is not NULL before attempting to lower its usecount. (closes issue #17234) Reported by: mav3rick ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268455 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Get rid of warning when the console is configured without logger levelsrussell1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268204 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267492 via svnmerge from mmichelson1-40/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r267492 | mmichelson | 2010-06-03 12:09:11 -0500 (Thu, 03 Jun 2010) | 6 lines Remove unnecessary code relating to PLC. The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267507 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267303 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r267303 | russell | 2010-06-02 16:41:54 -0500 (Wed, 02 Jun 2010) | 6 lines Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been specified. When ASTCFLAGS was specified with the make command, Makefile.rules was using the specified value from the command line and not the one here, making it so this flag would go missing. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267304 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267138 via svnmerge from russell1-0/+24
https://origsvn.digium.com/svn/asterisk/trunk ........ r267138 | russell | 2010-06-02 13:53:38 -0500 (Wed, 02 Jun 2010) | 4 lines Add a CLI command that blocks until Asterisk has fully booted. Review: https://reviewboard.asterisk.org/r/684/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267139 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267041 via svnmerge from pabelanger1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r267041 | pabelanger | 2010-06-02 13:25:05 -0400 (Wed, 02 Jun 2010) | 14 lines Merged revisions 267009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun 2010) | 7 lines Cleanup error/warning messages in AEL2 parser (closes issue #16684) Reported by: Silmaril Patches: patch_ael2_logmsg.diff uploaded by Silmaril (license 979) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267062 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 266877 via svnmerge from pabelanger1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r266877 | pabelanger | 2010-06-02 09:32:22 -0400 (Wed, 02 Jun 2010) | 10 lines pthread_join to assure the thread is really gone (closes issue #15465) Reported by: fnordian Patches: bridging.patch uploaded by fnordian (license 110) Tested by: lmadsen, fnordian, peterh Review: https://reviewboard.asterisk.org/r/679/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266878 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Merged revisions 266682 via svnmerge from tilghman1-43/+49
https://origsvn.digium.com/svn/asterisk/trunk ........ r266682 | tilghman | 2010-06-01 11:41:00 -0500 (Tue, 01 Jun 2010) | 16 lines Eliminate stale manager events after a set interval, even if AMI clients don't query for them. Actions (or failures to act) by external clients should not cause memory leaks in Asterisk, especially when those continued leaks could cause Asterisk to misbehave later. (closes issue #17234) Reported by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by tilghman (license 14) 20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14) Tested by: mav3rick, davidw (closes issue #17365) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266683 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Merged revisions 266592 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010) | 18 lines Merged revisions 266585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines Prevent CLI prompt from distorting output of lines shorter than the prompt. Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266598 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Merged revisions 266337 via svnmerge from tilghman1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 | tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line Only report swap on platforms which can examine those statistics ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266338 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Merged revisions 266146 via svnmerge from tilghman2-21/+43
https://origsvn.digium.com/svn/asterisk/trunk ................ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010) | 21 lines Merged revisions 266142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266154 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265320,265467 via svnmerge from twilson2-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265521 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265273 via svnmerge from dvossel1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 | dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines fixes segfault when using generic plc ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265364 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265316 via svnmerge from tilghman1-10/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines On systems with a LOT of RAM, a signed integer sometimes printed negative. (closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265318 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 264997 via svnmerge from mmichelson2-27/+57
https://origsvn.digium.com/svn/asterisk/trunk ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264998 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Merged revisions 264828 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010) | 13 lines Merged revisions 264820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines ast_callerid_parse() had a path that left name uninitialized. Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264829 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Merged revisions 264779 via svnmerge from tilghman1-0/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 | tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines Let ExtensionState resolve dynamic hints. (closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264783 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264452 via svnmerge from mmichelson3-4/+110
https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264400 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 | dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines fixes infinite loop during udptl.c's decode_open_type When decode_length returns the length there is a check to see if that length is negative, if so the decode loop breaks as this means the limit has been reached. The problem here is that length is an unsigned int, so length can never be negative. This resulted in an infinite loop. (issue #17352) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264405 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264379 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 | mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4 lines Cast an unsigned int to a signed int when comparing it with 0. (AST-377) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264388 f38db490-d61c-443f-a65b-d21fe96a405b
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