[asterisk-commits] lmadsen: tag 1.6.2.0-beta3 r200106 - /tags/1.6.2.0-beta3/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Jun 11 07:41:11 CDT 2009
Author: lmadsen
Date: Thu Jun 11 07:41:07 2009
New Revision: 200106
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=200106
Log:
Importing files for 1.6.2.0-beta3 release.
Added:
tags/1.6.2.0-beta3/.lastclean (with props)
tags/1.6.2.0-beta3/.version (with props)
tags/1.6.2.0-beta3/ChangeLog (with props)
Added: tags/1.6.2.0-beta3/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.2.0-beta3/.lastclean?view=auto&rev=200106
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--- tags/1.6.2.0-beta3/ChangeLog (added)
+++ tags/1.6.2.0-beta3/ChangeLog Thu Jun 11 07:41:07 2009
@@ -1,0 +1,14735 @@
+2009-06-11 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.2.0-beta3
+
+2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen at digium.com>
+
+ * build_tools/make_version_h, /, build_tools/make_version_c: Merged
+ revisions 200039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+ lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+ Fix path for .flavor and .version (issue #14737) Reported by:
+ davidw Patches: flavor.patch uploaded by davidw (license 780)
+ Tested by: davidw ........
+
+2009-06-10 20:37 +0000 [r199998] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c, /: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:18 +0000 [r199966] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
+ mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
+ lines Only try to use the invite_branch on outgoing INVITEs with
+ auth credentials. I have added a comment to the code to help ease
+ understanding of the logic here as well. ........
+
+2009-06-10 16:13 +0000 [r199860] Sean Bright <sean.bright at gmail.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+ (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+ ................
+
+2009-06-09 20:48 +0000 [r199744-199819] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+ dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+ CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+ only used UDP rather than copying the transport type from the
+ peer. (closes issue #15283) Reported by: jthurman Patches:
+ sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+ Tested by: jthurman, dvossel ........
+
+ * main/loader.c, /, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009)
+ | 11 lines module load priority This patch adds the option to
+ give a module a load priority. The value represents the order in
+ which a module's load() function is initialized. The lower the
+ value, the higher the priority. The value is only checked if the
+ AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
+ flag is not set, the value will never be read and the module will
+ be given the lowest possible priority on load. Since some modules
+ are reliant on a timing interface, the timing modules have been
+ given a high load priorty. (closes issue #15191) Reported by:
+ alecdavis Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/262/ ........
+
+2009-06-08 19:39 +0000 [r199634] Sean Bright <sean.bright at gmail.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+ (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........ ................
+
+2009-06-08 17:42 +0000 [r199591] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 199588 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
+ 08 Jun 2009) | 9 lines Fix a deadlock that could occur when
+ setting rtp stats on SIP calls. (closes issue #15143) Reported
+ by: cristiandimache Patches: 15143.patch uploaded by mmichelson
+ (license 60) Tested by: cristiandimache ........
+
+2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell at digium.com>
+
+ * Makefile, /: Merged revisions 199368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
+ russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
+ Switch from "echo -n" to printf. On my mac, the -n was just
+ getting printed out. ........
+
+2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c: Merged
+ revisions 199298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+ | 21 lines Merged revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:52 +0000 [r199230] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+ 2009) | 14 lines Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:15 +0000 [r199140] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+ (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+ ................
+
+2009-06-04 14:53 +0000 [r199054] Sean Bright <sean.bright at gmail.com>
+
+ * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
+ Merged revisions 199051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
+ 2009) | 47 lines Merged revisions 199022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+ 2009) | 40 lines Safely handle AMI connections/reload requests
+ that occur during startup. During asterisk startup, a lock on the
+ list of modules is obtained by the primary thread while each
+ module is initialized. Issue 13778 pointed out a problem with
+ this approach, however. Because the AMI is loaded before other
+ modules, it is possible for a module reload to be issued by a
+ connected client (via Action: Command), causing a deadlock. The
+ resolution for 13778 was to move initialization of the manager to
+ happen after the other modules had already been lodaded. While
+ this fixed this particular issue, it caused a problem for users
+ (like FreePBX) who call AMI scripts via an #exec in a
+ configuration file (See issue 15189). The solution I have come up
+ with is to defer any reload requests that come in until after the
+ server is fully booted. When a call comes in to ast_module_reload
+ (from wherever) before we are fully booted, the request is added
+ to a queue of pending requests. Once we are done booting up, we
+ then execute these deferred requests in turn. Note that I have
+ tried to make this a bit more intelligent in that it will not
+ queue up more than 1 request for the same module to be reloaded,
+ and if a general reload request comes in ('module reload') the
+ queue is flushed and we only issue a single deferred reload for
+ the entire system. As for how this will impact existing
+ installations - Before 13778, a reload issued before module
+ initialization was completed would result in a deadlock. After
+ 13778, you simply couldn't connect to the manager during startup
+ (which causes problems with #exec-that-calls-AMI configuration
+ files). I believe this is a good general purpose solution that
+ won't negatively impact existing installations. (closes issue
+ #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/ ........ ................
+
+2009-06-03 15:24 +0000 [r198827-198886] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 198856 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
+ dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/ ........
+
+ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
+ | 8 lines fixes issue with channels not going down after transfer
+ Iax2 currently does not support native bridging if the timeoutms
+ value is set. We check for that in iax2_bridge, but then set
+ timeoutms to 0 by default. If the timeoutms is not provided it is
+ set to -1. By setting timeoutms to 0 it is processed causing a
+ bridging retry loop. (closes issue #15216) Reported by: oxymoron
+ Tested by: dvossel ........
+
+2009-06-02 13:51 +0000 [r198794] Joshua Colp <jcolp at digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 198791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
+ file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
+ Correct documentation for the register line, specifically where
+ the domain should be specified. (closes issue #14367) Reported
+ by: Nick_Lewis ........
+
+2009-06-01 21:04 +0000 [r198730] Russell Bryant <russell at digium.com>
+
+ * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
+ | 2 lines Tell the IAX2 parser about more control frame types.
+ ........
+
+2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher at digium.com>
+
+ * /, contrib/scripts/meetme.sql: Merged revisions 198626 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
+ Jun 2009) | 2 lines Add information for new meetme realtime
+ fields ........
+
+2009-05-31 17:53 +0000 [r198471] Tilghman Lesher <tlesher at digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
+ | 2 lines Fix documentation for FIELDQTY. ........
+
+2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels at gmail.com>
+
+ * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
+ 11 lines Avoid a crash when res_timing_dahdi is unloaded but
+ wasn't properly loaded. if dahdi_test_timer() fails,
+ timing_funcs_handle remains NULL causing a crash when calling
+ ast_unregister_timing_interface() with a NULL pointer. (closes
+ issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
+ uploaded by eliel (license 64) ........
+
+2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell at digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
+ | 12 lines Merged revisions 198311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
+ | 5 lines Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28 ........
+ ................
+
+2009-05-30 20:22 +0000 [r198297-198397] Sean Bright <sean.bright at gmail.com>
+
+ * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
+ seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
+ lines Properly terminate the receive buffer before sending to
+ iksemel. aji_io_recv takes the maximum number of bytes to read
+ (instead of the total buffer size), so we have to subtract 1 from
+ our buffer size. Without this, when we receive packets that are
+ larger than our buffer, iksemel will choke and things get wonky.
+ (closes issue #15232) Reported by: lp0 Patches:
+ 05302009_res_jabber.c.patch uploaded by seanbright (license 71)
+ Tested by: seanbright, lp0 ........
+
+ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
+ 2009) | 19 lines Merged revisions 198370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
+ 2009) | 12 lines Properly terminate AMI JabberSend response
+ messages. The response message (either Error or Success) needs an
+ extra trailing \r\n after the fields to inform the client that
+ the message is complete. (closes issue #14876) Reported by: srt
+ Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+ (license 71) asterisk_14876.patch uploaded by srt (license 378)
+ trunk-14876-2.diff uploaded by phsultan (license 73) ........
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 198251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
+ 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
+ treat a missing one. (closes issue #15056) Reported by:
+ p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+ seanbright (license 71) Tested by: p_lindheimer ........
+ ................
+
+2009-05-30 02:35 +0000 [r198250] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
+ file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
+ When removing all packets from a dialog we also need to free the
+ data if present. ........
+
+2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant <russell at digium.com>
+
+ * /, configs/modules.conf.sample: Merged revisions 198186 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
+ May 2009) | 2 lines Suggesting that only a single timing module
+ be loaded is no longer necessary. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
+ | 2 lines Improve handling of trying to ACK too many timer
+ expirations. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
+ | 38 lines Resolve issues with choppy sound when using
+ res_timing_pthread. The situation that caused this problem was
+ when continuous mode was being turned on and off while a rate was
+ set for a timing interface. A very easy way to replicate this bug
+ was to do a Playback() from behind a Local channel. In this
+ scenario, a rate gets set on the channel for doing file playback.
+ At the same time, continuous mode gets turned on and off about
+ every 20 ms as frames get queued on to the PBX side channel from
+ the other side of the Local channel. Essentially, this module
+ treated continuous mode and a set rate as mutually exclusive
+ states for the timer to be in. When I dug deep enough, I observed
+ the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
+ almost 20 ms ... 3) Continuous mode gets turned on for a queued
+ up frame 4) Continuous mode gets turned off 5) The timer goes
+ back to its tick per 20 ms. state but starts counting at 0 ms. 6)
+ Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
+ and produce a timer tick, but not most of the time. This is what
+ produced the choppy sound (or sometimes no sound at all). Now,
+ the module treats continuous mode and a set rate as completely
+ independent timer modes. They can be enabled and disabled
+ independently of each other and things work as expected. (closes
+ issue #14412) Reported by: dome Patches: issue14412.diff.txt
+ uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
+ uploaded by russell (license 2) Tested by: DennisD, russell
+ ........
+
+2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons <eliels at gmail.com>
+
+ * CREDITS, /: Merged revisions 198083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
+ eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
+ Apply anti-spam obfuscation to an email address. ........
+
+2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
+ revisions 198072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
+ 2009) | 21 lines Merged revisions 198068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
+ 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
+ the default CDR disposition. This change also involves the
+ addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
+ originated channels to distinguish: them from dialed channels.
+ (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
+ uploaded by mnicholson (license 96) Tested by: mnicholson,
+ dbrooks (closes issue #15122) Reported by: sum Tested by: sum
+ ........ ................
+
+2009-05-29 18:40 +0000 [r198066] Joshua Colp <jcolp at digium.com>
+
+ * /, main/file.c: Merged revisions 198064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
+ file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
+ a memory leak of the write buffer when writing a file. ........
+
+2009-05-29 18:18 +0000 [r198008] Sean Bright <sean.bright at gmail.com>
+
+ * Makefile, /: Merged revisions 198000 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 197998 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
+ 2009) | 8 lines Fix 'make config' target for Slackware. There was
+ a missing semi-colon after the echo statement in the Makefile
+ that was causing problems for some users. Fix suggested by
+ reporter. (closes issue #15225) Reported by: pdavis ........
+ ................
+
+2009-05-29 16:29 +0000 [r197994] Russell Bryant <russell at digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
+ | 2 lines Trim trailing whitespace so that I can work on this bug
+ without it bothering me. :-) ........
+
+2009-05-28 23:54 +0000 [r197894] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
+ | 8 lines Update documentation in MixMonitor. Updated the
+ MixMonitor documentation for the 'b' option so that it is more
+ obvious that you must not optimize away the Local channel when
+ using this option. (closes issue #14829) Reported by: licedey
+ Tested by: mmichelson, licedey, lmadsen ........
+
+2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
+ lines Fix a bug where the trunkmtu setting was not set to the
+ default value of 1240 on load but was on reload. ........
+
+2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons <eliels at gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
+ 19 lines Merged revisions 197562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
+ 13 lines Use the address we already know when reloading a peer
+ with nat=yes. If we already have an address for a peer, and we
+ are reloading the sip configuration, try to use that address to
+ contact the peer, instead of getting it from the Contact. (closes
+ issue #15194) Reported by: ibc Patches: sip.patch uploaded by
+ eliel (license 64) Tested by: manwe ........ ................
+
+2009-05-28 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 197606 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
+ 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
+ 28 May 2009) | 16 lines Allow for media to arrive from an
+ alternate source when responding to a reinvite with 491. When we
+ receive a SIP reinvite, it is possible that we may not be able to
+ process the reinvite immediately since we have also sent a
+ reinvite out ourselves. The problem is that whoever sent us the
+ reinvite may have also sent a reinvite out to another party, and
+ that reinvite may have succeeded. As a result, even though we are
+ not going to accept the reinvite we just received, it is
+ important for us to not have problems if we suddenly start
+ receiving RTP from a new source. The fix for this is to grab the
+ media source information from the SDP of the reinvite that we
+ receive. This information is passed to the RTP layer so that it
+ will know about the alternate source for media. Review:
+ https://reviewboard.asterisk.org/r/252 ........ ................
+
+ * main/audiohook.c, apps/app_chanspy.c, /,
+ include/asterisk/audiohook.h: Merged revisions 197543 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500
+ (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
+ 2009) | 21 lines Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+ the read and write slinfactories on the audiohook do not skew
+ beyond a certain tolerance. In addition, there is a new audiohook
+ flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+ we do not allow for a slinfactory to build up a substantial
+ amount of audio before flushing it. For this particular issue,
+ this means that the person spying on the call will hear the
+ conversations in real time with very little delay in the audio.
+ (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+ uploaded by mmichelson (license 60) Tested by: snblitz ........
+ ................
+
+2009-05-28 14:56 +0000 [r197471-197542] Joshua Colp <jcolp at digium.com>
+
+ * /, main/utils.c: Merged revisions 197538 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
+ file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
+ a bug in stringfields where it did not actually free the pools of
+ memory. (closes issue #15074) Reported by: pj ........
+
+ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
+ 15 lines Merged revisions 197466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
+ lines Fix a bug where the flag indicating the presence of rport
+ would get overwritten by the nat setting. The presence of rport
+ is now stored as a separate flag. Once the dialog is setup and
+ authenticated (or it passes through unauthenticated) the proper
+ nat flag is set. (closes issue #13823) Reported by: dimas
+ ........ ................
+
+2009-05-28 11:40 +0000 [r197441] Gavin Henry <ghenry at suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, doc/ldap.txt,
+ configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
+ trunk
+
+2009-05-27 23:49 +0000 [r197375] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/xml.c: Merged revisions 197374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
+ tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
+ Revert commit 192032. This define is needed on Mac OS X. ........
+
+2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
+ 2009) | 3 lines Ensure that this header includes xmldoc.h, since
+ it depends on it. ........
+
+2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright at gmail.com>
+
+ * Makefile, /: Merged revisions 197260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
+ seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
+ lines Use bash explicitly when calling build_tools/mkpkgconfig
+ from the Makefile. Since we use bashisms in
+ build_tools/mkpkgconfig, we should call on bash explicitly when
+ running from the Makefile, otherwise we get errors during a 'make
+ install.' (closes issue #15209) Reported by: seandarcy ........
+
+2009-05-27 19:30 +0000 [r197247] Tilghman Lesher <tlesher at digium.com>
+
+ * /, funcs/func_cut.c: Recorded merge of revisions 197209 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
+ (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
+ | 5 lines Use a different determinator on whether to print the
+ delimiter, since leading fields may be blank. (closes issue
+ #15208) Reported by: ramonpeek Patch by me, though inspired in
+ part by a patch from ramonpeek ........ ................
+
+2009-05-27 17:28 +0000 [r197176] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, include/asterisk/channel.h: Fix broken attended
+ transfers The bridge was terminating immediately after the
+ attended transfer was completed. The problem was because upon
+ reentering ast_channel_bridge nexteventts was checked to see if
+ it was set and if so could possibly return AST_BRIDGE_COMPLETE.
+ (closes issue #15183) Reported by: andrebarbosa Tested by:
+ andrebarbosa, tootai, loloski
+
+2009-05-27 16:12 +0000 [r196950-197092] Sean Bright <sean.bright at gmail.com>
+
+ * configs/smdi.conf.sample, configs/extensions.conf.sample,
+ configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
+ configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
+ 2009) | 6 lines Fix references to /etc/dahdi/system.conf and
+ /etc/asterisk/chan_dahdi.conf in the sample configuration files.
+ (closes issue #15207) Reported by: seandarcy ........
+
+ * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
+ 2009) | 9 lines Display an error message when chan_alsa fails to
+ load due to a missing or inaccessible configuration file. Before
+ this change, when chan_alsa failed to load due to a missing or
+ inaccessible configuration file, no message would be displayed.
+ With this change, when chan_alsa fails to load due to a missing
+ or inaccessible configuration file, a message will be displayed.
+ (closes issue #14760) Reported by: Nick_Lewis Patches:
+ chan_alsa.c-confload.patch uploaded by Nick (license 657)
+ ........
+
+ * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
+ seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
+ lines Reset the terminal to the correct fg/bg after XML
+ documenation is rendered. (closes issue #15200) Reported by:
+ ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
+ (license 71) Tested by: ajohnson ........
+
+ * main/manager.c, /: Merged revisions 196945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
+ seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
+ lines Add ActionID to CoreShowChannel event. There is
+ inconsistency in how we handle manager responses that are lists
+ of items and, unfortunately, third parties have come to rely on
+ ActionID being on every event within those lists instead of just
+ keeping track of the ActionID for the current response. This
+ change makes CoreShowChannels include the ActionID with each
+ CoreShowChannel event generated as a result of it being called.
+ (closes issue #15001) Reported by: sum Patches:
+ patchactionid2.patch uploaded by sum (license 766) ........
+
+2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell at digium.com>
+
+ * /, autoconf/ast_check_osptk.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 196946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
+ russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
+ Update configure script to check for OSP toolkit 3.5.0. (closes
+ issue #14988) Reported by: tzafrir Patches: configure.ac.diff
+ uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
+ by homesick (license 91) ........
+
+ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
+ | 16 lines Merged revisions 196826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
+ | 9 lines Resolve a file handle leak. The frames here should have
+ always been freed. However, out of luck, there was never any
+ memory leaked. However, after file streams became reference
+ counted, this code would leak the file stream for the file being
+ read. (closes issue #15181) Reported by: jkroon ........
+ ................
+
+2009-05-26 16:39 +0000 [r196793] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
+ seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
+ lines Add a missing unref for queues in handle_statechange.
+ ........
+
+2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
+ file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
+ a bug where the sip unregister CLI command did not completely
+ unregister the peer. (closes issue #15118) Reported by: alecdavis
+ Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
+ (license 585) ........
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
+ 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
+ lines Remove some bash specific stuff from safe_asterisk. (closes
+ issue #10812) Reported by: paravoid Patches:
+ safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva at gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
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