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Author	SHA1	Message	Date
Paul Belanger	6aee8296c1	Merged revisions 267041 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	bc418d569b	Merged revisions 266877 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b3d58878d5	Merged revisions 266682 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ad56e4ba3b	Merged revisions 266592 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	f1b2ab09db	Fix typo in documentation ... (closes issue #17395) Reported by: pabelanger Patches: res_agi.c.patch uploaded by pabelanger (license 224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	218ec57d43	Merged revisions 266438 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500 (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) | 2 lines Reverting patch and reopening issue #16784, as patch breaks color display. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	dfec2260a8	Merged revisions 266337 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c5595b3526	Merged revisions 266292 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines fixes crash when creation of UDPTL fails (closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b44409fd7b	Merged revisions 266146 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9fda8817a9	Merged revisions 266006 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines fixes failed SIP Directed pickup resulting in dead channel (closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@266007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	98dcc855a6	Merged revisions 265923 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500 (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines Not finding rows in the DB does not rise to the level of a warning. (closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7fa20777af	Merged revisions 265894 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 | tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines Construct socket name, according to the Postgres docs, and document as such. (closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bb6e3a4811	Recorded merge of revisions 265842 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, 26 May 2010) | 9 lines Re-enable "always" option for videosupport option in sip.conf. (closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a199f18aa6	Merged revisions 265747 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines Use configure to determine the prefixes and include directories properly. This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9ceb671660	Merged revisions 265698 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 lines Properly use peer's outboundproxy for outbound REGISTERs. The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	60b472c387	fixes build issue with zaptel ... (closes issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	bdb62057a8	Merged revisions 265611 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May 2010) | 15 lines Merged revisions 265610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a2b47e4eda	Merged revisions 265320,265467 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	23c0ccb411	Merged revisions 265451 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, 24 May 2010) | 8 lines Print openh323 log to the Asterisk console. (closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	eecd2c0f44	Merged revisions 265449 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 lines Allow type=user SIP endpoints to be loaded properly from realtime. (closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a7c719e376	Merged revisions 265273 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	0dedf05565	Blocked revisions 265317 via svnmerge ... ........ r265317 | twilson | 2010-05-24 13:21:20 -0500 (Mon, 24 May 2010) | 15 lines Calendaring support for Exchange Server 2007+ via EWS This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	60b18b4ea0	Merged revisions 265316 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	032ae964b5	Fix memory hogging behavior of app_queue. ... From reviewboard: This review request is for the patch on issue 17081. A user reported that he saw increasing numbers of allocations stemming from app_queue.c when he would run the "queue show" CLI command. The user reported that he was using approximately 40 realtime queues and as he ran the CLI command more and more, the memory usage would shoot up. As it turns out, there was a memory leak and a separate usage of memory that, while not really a leak, was very irresponsible. Both memory problems can be attributed to the function init_queue(). When the "queue show" command is run, all realtime queues have the init_queue() function called on the in-memory queue. The idea is to place the queue in its default state and then overwrite options specified in the realtime backend as we read them. The first problem, the memory leak, had to do with the fact that the string field for the name of the first periodic announcement file was being re-created every time init_queue was called. This patch corrects the behavior by only calling ast_str_create if the memory has not already been allocated. The other problem is a bit more complicated. The majority of the strings in the call_queue structure were changed to use the ast_string_fields API for 1.6.0 and beyond. init_queue resets all string fields on the queue to their default values. Then, later in the realtime queue loading process, these string fields are set to their configured values. For those unfamiliar with string fields, frequent resizing of a string like this is not what the string fields API is designed for. The result of this constant resizing is that as the queue gets loaded, eventually space for the string runs out and so a new memory pool, at twice the size of the previously allocated one, is created for the string fields. The reporter of issue 17081 wrote a script that ran the "queue show" CLI command 2100 times. By the end, each of his 40 queues was taking about a megabyte of memory apiece just for their string fields. My fix for this problem is to revert the call_queue structure from using string fields. In my patch here, I have moved the queue back to using fixed-sized buffers. I ran the script provided by the reporter of 17081 and determined that I no longer saw the steadily-increasing memory usage that I had seen before applying the patch. (closes issue #17081) Reported by: wliegel Patches: 17081v2.patch uploaded by mmichelson (license 60) Tested by: wliegel, mmichelson Review: https://reviewboard.asterisk.org/r/651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d3ae642446	Merged revisions 265090 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May 2010) | 15 lines Merged revisions 265089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e91d462a57	Merged revisions 265087 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 lines Be sure to set the sin_family on the proxy when allocating. (closes issue #17157) Reported by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f83214b612	Merged revisions 265000 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines Fix grammatical error in comment. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9a124ea569	Merged revisions 264997 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	2b45d9dcaa	Merged revisions 264828 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a09588ab45	Merged revisions 264779 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e8a5f2af3	Merged revisions 264752 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 | tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines Error message fix. (closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	af727635c5	Merged revisions 264452 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d7828ba909	Merged revisions 264400 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	145ecb6151	Merged revisions 264379 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	33fc0b768a	Merged revisions 264335 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500 (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a7249b8931	Merged revisions 264331 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines fixes crash in check_rtp_timeout During deadlock avoidance the sip dialog pvt is locked and unlocked. When this occurs we have no guarantee the pvt's owner is still valid. We were trying to access the pvt's owner after this without checking to see if it still existed first. (closes issue #17271) Reported by: under Patches: check_rtp_timeout.diff uploaded by under (license 914) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f134c6effe	Merged revisions 264249 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) | 24 lines Merged revisions 264248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cd48e36e76	Merged revisions 264204 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 | tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines Keep track of digit duration, when we're decoding inband to pass DTMF frames. (closes issue #17235) Reported by: frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license 610) 20100518__issue17235.diff.txt uploaded by tilghman (license 14) Tested by: frawd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a1f66d06e0	Merged revisions 264114 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 | dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines fixes crash during dtmf During the processing of Cisco dtmf the dtmf samples were not being calculated correctly. In an attempt to determine what sample rate was being used, a NULL frame was processed which caused a crash. This patch resolves this. (closes issue #17248) Reported by: falves11 Patches: issue_17248.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	60072abde9	Merged revisions 264031 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19 May 2010) | 8 lines fix incorrectly typed indications for [nz] stutter and dialrecall (closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9567b5c7d9	Merged revisions 263950 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010) | 15 lines Merged revisions 263949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c0ee4e0df0	Merged revisions 263904 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 | dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines fixes segfault on logging (closes issue #17331) Reported by: under Patches: utils.diff uploaded by under (license 914) segfault_on_logging.diff uploaded by dvossel (license 671) Tested by: under, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5e55e1f9ca	Merged revisions 263807 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500 (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	22ac519172	Merged revisions 263640 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines Fix logic error when checking for a devstate provider. When using strsep, if one of the list of specified separators is not found, it is the first parameter to strsep which is now NULL, not the pointer returned by strsep. This issue isn't especially severe in that the worst it is likely to do is waste some cycles when a device with no '/' and no ':' is passed to ast_device_state. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ec63586e57	Merged revisions 263589 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010) | 9 lines With IMAP backend, messages in INBOX were counted twice for MWI. (closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1cf0fdd22e	Don't close 'n', just close 'above_n'. ... (closes issue #17345) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	0752f71a31	Merged revisions 263457 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010) | 19 lines Recorded merge of revisions 263456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines Manager cookies are not compatible with RFC2109. The Version field in the cookies we're setting contain quotes around the version number which is not compatible with RFC2109 and breaks some implementations. (closes issue #17231) Reported by: ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559) Tested by: ecarruda, russell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	1dd8823feb	Merged revisions 263375 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) | 8 lines Update link to new version of core sounds. The latest version of the core sounds files 1.4.19 now includes the missing queue-minute sound file which is called by app_queue but which has been missing. (closes issue #17123) Reported by: n8ideas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ca0af2abc8	backport of DAHDI dynamic buffer policy dialstring option ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	bb763dc410	Merged revisions 252488 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 | tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems. (closes issue #17028) Reported by: pabelanger Patches: issue17028_20100315.patch uploaded by seanbright (license 71) 20100315__issue17028.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@263202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago