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2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 305923 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24Merged revisions 303549 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301504 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Some scheduler API cleanup and improvements.Russell Bryant
Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-07Merged revisions 297825 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296230 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296084 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296002 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-22Merged revisions 295866 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-16Merged revisions 295282 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294466 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) | 1 line Allow ast_do_masquerade() failure to be reported again. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294349 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08Merged revisions 294278 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-22Merged revisions 292704 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-13Merged revisions 291581 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-29Merged revisions 289340 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289340 | qwell | 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) | 22 lines Merged revisions 289339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines Allow a manager originate to succeed on forwarded devices. The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-29Merged revisions 289268 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep 2010) | 5 lines Update the CDR record when ast_channel_set_caller_event() is called (related to issue #17569) Reported by: tbelder ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-29Merged revisions 289253 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010) | 1 line Make development error message indicate which channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-29Merged revisions 289179 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289179 | mnicholson | 2010-09-29 10:04:56 -0500 (Wed, 29 Sep 2010) | 22 lines Merged revisions 289178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500 (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines Set the caller id on CDRs when it is set on the parent channel. (closes issue #17569) Reported by: tbelder Patches: 17569.diff uploaded by tbelder (license 618) Tested by: tbelder ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-28Merged revisions 289099 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289099 | bbryant | 2010-09-28 14:18:02 -0400 (Tue, 28 Sep 2010) | 28 lines Merged revisions 289095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines Fixes an issue with the Newchannel AMI event during the Masquerading process. Fixes an issue with the Newchannel AMI event during the Masquerading process, where no Newchannel AMI event was generated for the psuedo channel used during the masquerading process. (closes issue #17987) Reported by: RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish (license 1122) Tested by: RadicAlish Review: https://reviewboard.asterisk.org/r/937/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 288079-288080 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code. ........ r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines Simplify locking code for REDIRECTING interception macro when forwarding a call. Simplified the locking code by using a local copy of the redirecting party information in app_dial.c:do_forward() and app_queue.c:wait_for_answer() for launching the REDIRECTING interception macro when a call is forwarded. Reduced the lock time of the 'o->chan' and 'in' channels. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 288007 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288007 | bbryant | 2010-09-21 15:48:53 -0400 (Tue, 21 Sep 2010) | 21 lines Merged revisions 288006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400 (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate memory on the first frame being queued in ast_queue_frame. (closes issue #17882) Reported by: seanbright Tested by: seanbright ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287931 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010) | 2 lines Revert change in favor of a more targeted fix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287833 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010) | 3 lines Don't generate connected line buffer twice for comparison ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287757 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010) | 7 lines Avoid infinite loop with certain local channel connected line updates Compare connected line data before sending a connected line indication to avoid possible loops. Review: https://reviewboard.asterisk.org/r/932/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-20Merged revisions 287685 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep 2010) | 18 lines ast_channel_masquerade: Avoid recursive masquerades. Check all 4 combinations of (original/clonechan) * (masq/masqr). Initially original->masq and clonechan->masqr were only checked. It's possible with multiple masq's planned - and not yet executed, that the 'original' chan could already have another masq'd into it - thus original->masqr would be set, that masqr would lost. Likewise for the clonechan->masq. (closes issue #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-20Merged revisions 287661 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep 2010) | 14 lines ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels. Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run. End result would be corrupted memory, and the frequent report 'Bad Magic Number'. (closes issue #17801,#17710) Reported by: notthematrix Patches: Based on bug17801.diff1.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/928 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-20Merged revisions 287647 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-14Merged revisions 286682 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines Merged revisions 286681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines Only drop duplicate answer frames if the channel is bridged. Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-09Merged revisions 285745 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285745 | qwell | 2010-09-09 15:11:06 -0500 (Thu, 09 Sep 2010) | 23 lines Merged revisions 285744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-16Merged revisions 282468 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-12Merged revisions 281913 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines Ensure SSRC is changed when media source is changed to resolve audio delay. This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-29Merged revisions 280450 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500 (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-29Merged revisions 280307 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-27Merged revisions 279949 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-27Merged revisions 279636,279815 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines Ignore a control subclass of -1 in ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in addition to "channel" in chan_dahdi.conf. Review: https://reviewboard.asterisk.org/r/804 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-22Merged revisions 278618 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines Allow PLC to function properly when channels use SLIN for audio. If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Print f->subclass.integer instead of f->subclass.Matthew Nicholson
(fix build breakage introduced in r277250) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Merged revisions 277247 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Merged revisions 276652 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Expand the caller ANI field to an ast_party_idRichard Mudgett
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14ast_callerid restructuringRichard Mudgett
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-12Add which ITU spec specifies the numbering plan.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-12Merged revisions 275665 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines Change ast_write to not stop generator when called from ast_prod. For SIP channels configured with the progressinband option on, the ringback was being immediately stopped. This problem was due to ast_prod being moved for a deadlock fix in 259858. Prodding the channel after setting up the generator triggered the check in ast_write to stop the generator. The fix here should write the frame the same as was done before the call to ast_prod was moved. (closes issue #17372) Reported by: tech_admin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08Generate a correct AstData string for ast_callerid.cid_tonRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08Fix trunk compile.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,Eliel C. Sardanons
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-09 06:09:36 +0000