[asterisk-commits] lmadsen: tag 1.6.2.9-rc1 r266650 - /tags/1.6.2.9-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 1 11:02:56 CDT 2010
Author: lmadsen
Date: Tue Jun 1 11:02:54 2010
New Revision: 266650
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=266650
Log:
Importing files for 1.6.2.9-rc1 release.
Added:
tags/1.6.2.9-rc1/.lastclean (with props)
tags/1.6.2.9-rc1/.version (with props)
tags/1.6.2.9-rc1/ChangeLog (with props)
Added: tags/1.6.2.9-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.9-rc1/.lastclean?view=auto&rev=266650
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--- tags/1.6.2.9-rc1/ChangeLog (added)
+++ tags/1.6.2.9-rc1/ChangeLog Tue Jun 1 11:02:54 2010
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+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.9-rc1 Released.
+
+2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: 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 ........ ................
+
+2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * res/res_agi.c: Fix typo in documentation (closes issue #17395)
+ Reported by: pabelanger Patches: res_agi.c.patch uploaded by
+ pabelanger (license 224)
+
+2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: 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. ........ ................
+
+2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: 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
+ ........
+
+2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher at digium.com>
+
+ * utils/extconf.c, main/asterisk.c, /, main/logger.c: 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 ........ ................
+
+2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c, /: 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)
+ ........ ................
+
+ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: 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 ........
+
+2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: 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 ........
+
+2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: 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. ........
+
+2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c, /: 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 ........ ................
+
+2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: 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 ........
+
+2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/h323/ast_h323.cxx: 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) ........
+
+ * /, channels/chan_sip.c: 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
+ ........
+
+2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, /: 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 ........
+
+2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: 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)
+ ........
+
+2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: 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/
+
+ * apps/app_queue.c, include/asterisk/file.h, /: 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 ........ ................
+
+ * /, channels/chan_sip.c: 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 ........
+
+ * /, include/asterisk/channel.h: 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. ........
+ ................
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: 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/
+ ........ ................
+
+2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/callerid.c: 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. ........
+ ................
+
+2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /: 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 ........
+
+ * apps/app_stack.c, /: 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)
+ ........
+
+2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/_private.h, include/asterisk/options.h,
+ main/asterisk.c, main/loader.c, main/channel.c, /,
+ channels/chan_sip.c: 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/ ........
+
+2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel at digium.com>
+
+ * main/udptl.c, /: 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) ........
+
+2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/udptl.c, /: 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) ........
+
+ * apps/app_speech_utils.c, /: 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 ........ ................
+
+2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/options.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: 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 ........ ................
+
+ * main/dsp.c, /: 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 ........
+
+2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel at digium.com>
+
+ * main/rtp.c, /: 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) ........
+
+2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, configs/indications.conf.sample: 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) ........
+
+2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher at digium.com>
+
+ * main/dsp.c, /: 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 ........ ................
+
+2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel at digium.com>
+
+ * main/strings.c, /: 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 ........
+
+2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_directory.c, /: 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 ........ ................
+
+2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/devicestate.c: 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. ........ ................
+
+2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: 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 ........
+
+ * main/app.c: Don't close 'n', just close 'above_n'. (closes issue
+ #17345) Reported by: wdoekes
+
+2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen at digium.com>
+
+ * main/manager.c, /: 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 ........
+ ................
+
+ * sounds/Makefile, /: 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
+ ........ ................
+
+2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel at digium.com>
+
+ * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
+ policy dialstring option
+
+2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * /, codecs/gsm/Makefile: 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 ........
+
+2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
+ | 1 line Fix inverted logic in cli command: ss7 set debug on/off
+ ........
+
+2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell at digium.com>
+
+ * channels/chan_console.c, /: Merged revisions 262897 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
[... 24356 lines stripped ...]
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