[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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Added: tags/1.6.2.0-beta3/ChangeLog
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--- tags/1.6.2.0-beta3/ChangeLog (added)
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@@ -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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