[asterisk-commits] lmadsen: tag 1.6.0.18-rc2 r228970 - /tags/1.6.0.18-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Nov 9 11:00:50 CST 2009


Author: lmadsen
Date: Mon Nov  9 11:00:45 2009
New Revision: 228970

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=228970
Log:
Importing files for 1.6.0.18-rc2 release.

Added:
    tags/1.6.0.18-rc2/.lastclean   (with props)
    tags/1.6.0.18-rc2/.version   (with props)
    tags/1.6.0.18-rc2/ChangeLog   (with props)

Added: tags/1.6.0.18-rc2/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.18-rc2/.lastclean?view=auto&rev=228970
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Added: tags/1.6.0.18-rc2/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.18-rc2/ChangeLog?view=auto&rev=228970
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--- tags/1.6.0.18-rc2/ChangeLog (added)
+++ tags/1.6.0.18-rc2/ChangeLog Mon Nov  9 11:00:45 2009
@@ -1,0 +1,54899 @@
+2009-11-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.6.0.18-rc2
+
+2009-11-09 15:39 +0000 [r228898]  Leif Madsen <lmadsen at digium.com>
+
+	* main/channel.c: Merged revisions 228897 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
+	  | 14 lines Merged revisions 228896 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+	  | 6 lines Update WARNING message. Update a WARNING message to
+	  give a suggested fix when encountered. (closes issue #16198)
+	  Reported by: atis Tested by: atis ........ ................
+
+2009-11-09 14:58 +0000 [r228861]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
+	  (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
+	  2009) | 8 lines Perform limited bounds checking when destroying
+	  ast_mutex_t structures to make sure we don't try to use negative
+	  indices. (closes issue #15588) Reported by: zerohalo Patches:
+	  20090820__issue15588.diff.txt uploaded by tilghman (license 14)
+	  Tested by: zerohalo ........ ................
+
+2009-11-06 22:38 +0000 [r228696]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c, /: Merged revisions 228693 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
+	  | 16 lines Merged revisions 228692 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+	  | 9 lines fixes audiohook write crash occuring in chan_spy
+	  whisper mode. After writing to the audiohook list in ast_write(),
+	  frames were being freed incorrectly. Under certain conditions
+	  this resulted in a double free crash. (closes issue #16133)
+	  Reported by: wetwired ........ ................
+
+2009-11-06 20:42 +0000 [r228651]  Matthew Nicholson <mnicholson at digium.com>
+
+	* funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
+	  (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+	  2009) | 8 lines Properly handle '=' while decoding base64
+	  messages and null terminate strings returned from BASE64_DECODE.
+	  (closes issue #15271) Reported by: chappell Patches:
+	  base64_fix.patch uploaded by chappell (license 8) Tested by:
+	  kobaz ........ ................
+
+2009-11-06 18:40 +0000 [r228549]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
+	  11 lines Merged revisions 228547 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
+	  lines Don't overwrite caller ID name on a trunk with the
+	  configured fullname when using users.conf (issue ABE-1989)
+	  ........ ................
+
+2009-11-06  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.6.0.18-rc1
+
+2009-11-06 17:53 +0000 [r228479-228500]  Joshua Colp <jcolp at digium.com>
+
+	* /, doc/tex/localchannel.tex: Merged revisions 228499 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
+	  lines Fix the localchannel.tex file. ........
+
+	* channels/chan_sip.c: Fix a logic flaw I introduced when I was
+	  testing stuff out.
+
+2009-11-06 17:10 +0000 [r228423]  David Vossel <dvossel at digium.com>
+
+	* /, codecs/codec_ilbc.c: Merged revisions 228420 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
+	  | 19 lines Merged revisions 228418 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+	  | 13 lines fixes segfault in iLBC For reasons not yet known, it
+	  appears possible for an ast_frame to have a datalen greater than
+	  zero while the actual data is NULL during Packet Loss
+	  Concealment. Most codecs don't support PLC so this doesn't affect
+	  them. This patch catches the malformed frame and prevents the
+	  crash from occuring. Additional efforts to determine why it is
+	  possible for a frame to look like this are still being
+	  investigated. (issue #16979) ........ ................
+
+2009-11-06 16:56 +0000 [r228411-228415]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Fix a crash caused by freeing a dialog
+	  directly instead of using dialog_unref. (closes issue #16097)
+	  Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej
+	  (license 841)
+
+	* /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
+	  14 lines Merged revisions 228409 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+	  lines Fix a bug caused by a partially invalid frame (from the
+	  jitterbuffer) passing through the Asterisk core. (closes issue
+	  #15560) Reported by: jvandal (closes issue #15709) Reported by:
+	  covici ........ ................
+
+2009-11-06 15:44 +0000 [r228271-228342]  David Vossel <dvossel at digium.com>
+
+	* /, main/astfd.c: Merged revisions 228339 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
+	  | 12 lines Merged revisions 228338 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+	  | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+	  by: slavon ........ ................
+
+	* funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
+	  Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
+	  (closes issue #15394) Reported by: boroda Patches:
+	  bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
+	  Tested by: dbrooks, boroda ........
+
+2009-11-05 21:24 +0000 [r228190]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
+	  jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
+	  Fix the fix for chanspy option o In 224178, I assumed the
+	  uploaded patch was correct as it had received positive feedback.
+	  The flags were being checked in the incorrect location. Upon
+	  testing the fix this time it was also found that the flags from
+	  the dialplan weren't being copied to the
+	  chanspy_translation_helper. (closes issue #16167) Reported by:
+	  marhbere ........
+
+2009-11-05 19:39 +0000 [r228146]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
+	  (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
+	  | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
+	  chan_misdn connection. Patch submitted by gknispel_proformatique,
+	  tested by francesco_r. "I have many crash since i have upgraded
+	  to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
+	  an ast_frame. (closes issue #16041) Reported by: francesco_r
+	  ........ ................
+
+2009-11-05 19:17 +0000 [r228081]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
+	  (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
+	  8 lines Fix crash on VPB exception when no hardware is present.
+	  (closes issue #14970) Reported by: tzafrir Patches:
+	  vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+	  markwaters ........ ................
+
+2009-11-04 23:52 +0000 [r227946]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
+	  | 21 lines Merged revisions 227944 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+	  | 14 lines Fix incorrect filename comparsion after monitor file
+	  change The logic to detect if a requested file is indeed a
+	  different file from the current file was incorrect. The main
+	  issue being confusion of the use of filename_base which was
+	  previously set without pathing information and then compared to
+	  another full path. Robust file comparison logic has been added to
+	  properly check if two files are the same even if symlinks are
+	  used. (closes issue #15313) Reported by: caspy Patches:
+	  20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+	  325) but mostly tilghman's work ........ ................
+
+2009-11-04 21:15 +0000 [r227763-227833]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
+	  2009) | 17 lines Merged revisions 227827 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+	  2009) | 10 lines This patch modifies the Dial application to
+	  monitor the calling channel for hangups while playing back
+	  announcements. (closes issue #16005) Reported by: falves11
+	  Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+	  (license 96) Tested by: mnicholson, falves11 Review:
+	  https://reviewboard.asterisk.org/r/407/ ........ ................
+
+	* channels/chan_sip.c: Modify the SDP parsing code to parse session
+	  and media level items separately. With the new code, media level
+	  proprieties should no longer be confused with session level
+	  proprieties. This change also reorganizes some of the SDP parsing
+	  code which should make it easier to manage in the future. (closes
+	  issue #14994) Reported by: frawd
+
+2009-11-04 19:27 +0000 [r227717-227743]  Joshua Colp <jcolp at digium.com>
+
+	* /, static-http/prototype.js: Merged revisions 227739 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
+	  04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
+	  lines Fix a security issue where it may be possible for someone
+	  to execute a cross-site AJAX request exploit. (AST-2009-009)
+	  ........ ................
+
+	* /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
+	  12 lines Merged revisions 227700 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+	  lines Fix a security issue where sending a REGISTER with a
+	  differing username in the From URI and Authorization header would
+	  reveal whether it was valid or not. (AST-2009-008) ........
+	  ................
+
+2009-11-03 20:00 +0000 [r227373]  Jason Parker <jparker at digium.com>
+
+	* Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
+	  9 lines Fix some build issues on Solaris. (closes issue #14517)
+	  (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
+	  by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
+	  ........
+
+2009-11-03 19:49 +0000 [r227362-227369]  Leif Madsen <lmadsen at digium.com>
+
+	* apps/app_controlplayback.c, /: Merged revisions 227368 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
+	  Nov 2009) | 8 lines Change warning message to debug message.
+	  app_controlplayback outputs a warning, when in fact it is normal.
+	  (closes issue #16071) Reported by: atis Patches:
+	  controlplayback_warning.patch uploaded by atis (license 242)
+	  ........
+
+	* configs/extensions.conf.sample, /: Merged revisions 227361 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
+	  Nov 2009) | 11 lines Additional fixes to the
+	  extensions.conf.sample file. Update the extensions.conf.sample
+	  [stdexten] context so that we use the variable instead of
+	  requiring it to be passed explicitly. Also updated uses of the
+	  [stdexten] context throughout. (closes issue #15858) Reported by:
+	  pprindeville Patches: stdexten-context-update.txt uploaded by
+	  lmadsen (license 10) Tested by: pprindeville ........
+
+2009-11-03 18:05 +0000 [r227278]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+	  | 4 lines Make sure the outgoing flag is cleared if a new channel
+	  fails to get created for outgoing calls. This is the relevant
+	  portion of asterisk/trunk -r226648 ........
+
+2009-11-03 15:37 +0000 [r227168]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
+	  12 lines Merged revisions 227166 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+	  lines Fix a bug where an RPID header could be generated with a
+	  blank username in the URI. (closes issue #15909) Reported by:
+	  kobaz ........ ................
+
+2009-11-03 15:24 +0000 [r227163]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/extensions.conf.sample, /: Merged revisions 227162 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
+	  Nov 2009) | 7 lines Update extensions.conf.sample file to fix
+	  incorrect extensions. (closes issue #15857) Reported by:
+	  pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
+	  (license 347) Tested by: pprindeville ........
+
+2009-11-03 11:21 +0000 [r227102]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15
+	  lines Merged revisions 227088 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+	  lines Use proper response code when violating Contact ACL's.
+	  https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+	  quick review. (EDVX-003) ........ ................
+
+2009-11-02 21:05 +0000 [r226975-226976]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_sip.c: SIP channel name uniqueness SIP channel
+	  names were supposed to be unique by way of a name suffix derived
+	  from the pointer to the channel's private data. Uniqueness was
+	  preserved on 32-bit systems, but not on 64-bit systems. This
+	  patch, as suggested by kpfleming, replaces this suffix with a
+	  simple incremented unsigned int. (closes issue #15152) Reported
+	  by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+	* /: SIP channel name uniqueness SIP channel names were supposed to
+	  be unique by way of a name suffix derived from the pointer to the
+	  channel's private data. Uniqueness was preserved on 32-bit
+	  systems, but not on 64-bit systems. This patch, as suggested by
+	  kpfleming, replaces this suffix with a simple incremented
+	  unsigned int. (closes issue #15152) Reported by: palbrecht
+	  Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:09 +0000 [r226891]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
+	  18 lines Merged revisions 226889 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+	  11 lines Fix a bug where the recorded privacy introduction file
+	  would not get removed if the caller hung up while the called
+	  party had not yet answered. This was fixed by introducing an
+	  argument to the 'n' option which, when enabled, removes the
+	  introduction file under all scenarios. This was done to preserve
+	  the behavior that has existed for quite some time. (closes issue
+	  #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+	  jpeeler (license 325) ........ ................
+
+2009-11-02 17:16 +0000 [r226813]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
+	  (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+	  | 8 lines Don't allow two separate instances of safe_asterisk
+	  when restarting from the init script. (closes issue #14562)
+	  Reported by: davidw Patches: Initially
+	  20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+	  Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+	  (license 780) Tested by: davidw ........ ................
+
+2009-10-29 18:14 +0000 [r226533]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
+	  revisions 226532 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
+	  13 lines Merged revisions 226531 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+	  lines Add an option to enabling passing music on hold start and
+	  stop requests through instead of acting on them in chan_local.
+	  (closes issue #14709) Reported by: dimas ........
+	  ................
+
+2009-10-28 20:17 +0000 [r226381-226387]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/sip.conf.sample: Merged revisions 226384 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
+	  (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
+	  | 9 lines Update documentation in sip.conf.sample. Update the
+	  documentation in sip.conf.sample in order to make it more clear
+	  that directmedia/canreinvite do not cause Asterisk to ignore
+	  reINVITEs. It is only used to stop Asterisk from generating a
+	  reINVITE, but does not stop it from accepting them if necessary.
+	  (closes issue #15644) Reported by: lmadsen ........
+	  ................
+
+	* /, doc/tex/channelvariables.tex: Merged revisions 226378 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
+	  (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+	  | 7 lines Update CALLINGSUBADDR channel variable documentation.
+	  (closes issue #15734) Reported by: alecdavis Patches:
+	  channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+	  Tested by: alecdavis ........ ................
+
+2009-10-28 18:05 +0000 [r226167-226306]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, include/asterisk/linkedlists.h: Merged revisions 226305 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
+	  (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
+	  Oct 2009) | 2 lines Fix documentation (pointed out by
+	  TheDavidFactor on #-dev) ........ ................
+
+	* main/manager.c, /: Merged revisions 226159 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
+	  | 14 lines Merged revisions 226138 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+	  | 7 lines Manager output is not always NULL-terminated, so force
+	  a NULL at the end of the filestream. (closes issue #15495)
+	  Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+	  by tilghman (license 14) Tested by: pdf ........ ................
+
+2009-10-26 23:13 +0000 [r226019]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+	  linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+	  if host_os is linux-gnueabi * When checking if we are Linux,
+	  check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+	  the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+	  sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+	  tested for the value of 'linux-gnu' in one or two places in the
+	  tree. This patch also fixes the check libcap to check for $OSARCH
+	  rather than $host_os . See also:
+	  http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+	  svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+	  Merged revisions 226018 via svnmerge from
+	  http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-26 15:46 +0000 [r225869]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_fax.c: Backport audio handling loop fixes from trunk
+	  version of app_fax. This backport resolves some issues handling
+	  audio frames during FAX processing, and ensures that the FAX
+	  application doesn't accidentally get notified of a T.38
+	  switchover at the end of a successful FAX. (issue #16127)
+
+2009-10-23 14:05 +0000 [r225583]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile, /: Merged revisions 225582 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
+	  2009) | 17 lines Merged revisions 225581 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+	  2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+	  every build. For some reason the menuselect.makeopts file was
+	  listed as PHONY in the Makefile, resulting in 'make' needing to
+	  rebuild it for every build. This then resulted in the embedded
+	  module rules being rebuilt on every build, which can be slow and
+	  is unnecessary. This patch fixes the problem by properly allowing
+	  'make' to know when the menuselect.makeopts file needs to be
+	  rebuilt (defining the proper dependencies). ........
+	  ................
+
+2009-10-22 21:53 +0000 [r225486]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
+	  225485 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
+	  | 19 lines Merged revisions 225484 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+	  | 11 lines Clean valgrind output by suppressing false errors.
+	  Update valgrind.txt documentation and add valgrind.supp file in
+	  order to allow those who are creating valgrind output to have
+	  less false errors in the logfile. (closes issue #16007) Reported
+	  by: atis Patches: valgrind.txt.diff uploaded by atis (license
+	  242) asterisk2.supp uploaded by atis (license 242) Tested by:
+	  atis, amorsen ........ ................
+
+2009-10-22 17:13 +0000 [r225361]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+	  Merged revisions 225360 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
+	  | 11 lines Merged revisions 225105 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+	  | 4 lines Fix documentation for ast_softhangup() and correct the
+	  misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+	  ........ ................
+
+2009-10-21 22:10 +0000 [r225310-225311]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_iax2.c: Merged revisions 225307 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
+	  (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
+	  | 13 lines IAX2: VNAK loop caused by signaling frames with no
+	  destination call number It is possible for the PBX thread to
+	  queue up signaling frames before a destination call number is
+	  received. This can result in signaling frames being sent out with
+	  no destination call number. Since recent versions of Asterisk
+	  require accurate destination callnumbers for all Full Frames,
+	  this can cause a VNAK loop to occur. To resolve this no signaling
+	  frames are sent until a destination callnumber is received, and
+	  destination call numbers are now only required for iax_pvt
+	  matching when the frame is an ACK. Review:
+	  https://reviewboard.asterisk.org/r/413/ ........ ................
+
+	* configs/iax.conf.sample, /, channels/chan_sip.c,
+	  configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
+	  225033 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
+	  | 27 lines Merged revisions 225032 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+	  | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+	  id removes '(', ' ', ')', non-trailing '.', and '-' from the
+	  string. This means values such as 555.5555 and test-test result
+	  in 555555 and testtest. There are instances, such as Skype
+	  integration, where a specific value is passed via caller id that
+	  must be preserved unmodified. This patch makes the shrinking of
+	  caller id optional in chan_sip and chan_iax in order to support
+	  such cases. By default this option is on to preserve previous
+	  expected behavior. (closes issue #15940) Reported by: dimas
+	  Patches: v2-15940.patch uploaded by dimas (license 88)
+	  15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+	  Tested by: dvossel Review:
+	  https://reviewboard.asterisk.org/r/408/ ........ ................
+
+2009-10-21 03:15 +0000 [r224933]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/translate.h, main/dsp.c, main/frame.c, /,
+	  main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
+	  include/asterisk/frame.h: Merged revisions 224932 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r224932 | russell | 2009-10-20 22:09:04 -0500
+	  (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
+	  | 5 lines Isolate frames returned from a DSP instance or codec
+	  translator. The reasoning for these changes are the same as what
+	  I wrote in the commit message for rev 222878. ........
+	  ................
+
+2009-10-20 22:10 +0000 [r224857]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/audiohook.c: Merged revisions 224856 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009)
+	  | 12 lines Merged revisions 224855 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+	  | 5 lines Pay attention to the return value of the manipulate
+	  function. While this looks like an optimization, it prevents a
+	  crash from occurring when used with certain audiohook callbacks
+	  (diagnosed with SVN trunk, backported to 1.4 to keep the source
+	  consistent across versions). ........ ................
+
+2009-10-20 17:48 +0000 [r224775]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/features.c: Merged revisions 224774 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
+	  12 lines Merged revisions 224773 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+	  lines Add support for relaying early media in the features
+	  attended transfer option. (closes issue #14828) Reported by:
+	  licedey ........ ................
+
+2009-10-19 23:50 +0000 [r224672]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/rtp.c, /: Merged revisions 224671 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
+	  2009) | 14 lines Merged revisions 224670 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
+	  2009) | 7 lines Correct timestamp calculations when RTP sample
+	  rates over 8kHz are used. While testing some endpoints that
+	  support 16kHz and 32kHz sample rates, some log messages were
+	  generated due to calc_rxstamp() computing timestamps in a way
+	  that produced odd results, so this patch sanitizes the result of
+	  the computations. ........ ................
+
+2009-10-19 19:50 +0000 [r224568]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
+	  12 lines Merged revisions 224565 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+	  lines Do not attempt early media bridging (ie: direct RTP setup)
+	  if options are enabled that should prevent it. (closes issue
+	  #14763) Reported by: cupotka ........ ................
+
+2009-10-19 00:12 +0000 [r224449]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
+	  | 3 lines Allow ODBC storage to be queried with multiple
+	  mailboxes. This corrects an issue reported on the -users list.
+	  ........
+
+2009-10-17 02:03 +0000 [r224332-224337]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: fix typo, sorry
+
+	* channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
+	  (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+	  | 13 lines Fix stale caller id data from being reported in AMI
+	  NewChannel event The problem here is that chan_dahdi is designed
+	  in such a way to set certain values in the dahdi_pvt only once.
+	  One of those such values is the configured caller id data in
+	  chan_dahdi.conf. For PRI, the configured caller id data could be
+	  overwritten during a call. Instead of saving the data and
+	  restoring, it was decided that for all non-analog channels it was
+	  simply best to not set the configured caller id in the first
+	  place and also clear it at the end of the call. (closes issue
+	  #15883) Reported by: jsmith ........ ................
+
+2009-10-16 20:48 +0000 [r224262]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
+	  (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+	  | 18 lines Never released PRI channels when using Busy() or
+	  Congestion() dialplan apps. When the Busy() or Congestion()
+	  application is used towards ISDN (an ISDN progress is sent), the
+	  responding ISDN Disconnect or Release may contain the ISDN cause
+	  user busy or one of the congestion causes. In chan_dahdi.c these
+	  causes will only set the needbusy or needcongestion flags and not
+	  activate the softhangup procedure. Unfortunately only the latter
+	  can interrupt the endless wait loop of Busy()/Congestion().
+	  Result: PRI channels staying in state busy for the rest of
+	  asterisk life or until the other end times out and forces the
+	  call to clear. (in issue 0014292) Reported by: tomaso Patches:
+	  disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+	  patch is unrelated to the issue.) ........ ................
+
+2009-10-15 15:57 +0000 [r224179]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
+	  jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
+	  Readd removed ability to allow listening to one side of the call
+	  in app_chanspy (Option o) (closes issue #15675) Reported by:
+	  john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
+	  (license 790) Tested by: jgutierrez on users list:
+	  http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+	  ........
+
+2009-10-12 23:50 +0000 [r223833]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
+	  | 15 lines Merged revisions 223804 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+	  | 8 lines Ensure ringing continues for branched calls after
+	  progress is received While waiting for an answer, don't send
+	  progress for branched calls for which ringing was sent. (closes

[... 54213 lines stripped ...]



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