[asterisk-commits] lmadsen: tag 1.6.1-beta4 r165876 - /tags/1.6.1-beta4/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Dec 18 17:48:31 CST 2008


Author: lmadsen
Date: Thu Dec 18 17:48:30 2008
New Revision: 165876

URL: http://svn.digium.com/view/asterisk?view=rev&rev=165876
Log:
Importing files for 1.6.1-beta4 release

Added:
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    tags/1.6.1-beta4/.version   (with props)
    tags/1.6.1-beta4/ChangeLog   (with props)

Added: tags/1.6.1-beta4/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.6.1-beta4/.lastclean?view=auto&rev=165876
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Added: tags/1.6.1-beta4/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.6.1-beta4/ChangeLog?view=auto&rev=165876
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--- tags/1.6.1-beta4/ChangeLog (added)
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@@ -1,0 +1,50541 @@
+2008-12-18  Leif Madsen <leif at digium.com>
+
+	* Asterisk 1.6.1-beta4 released.
+
+2008-12-18 21:57 +0000 [r165808]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 165797 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600
+	  (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008)
+	  | 8 lines Add mutexes around accesses to the IMAP library
+	  interface. This prevents certain crashes, especially when shared
+	  mailboxes are used. (closes issue #13653) Reported by:
+	  howardwilkinson Patches:
+	  asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+	  howardwilkinson (license 590) Tested by: jpeeler ........
+	  ................
+
+2008-12-18 21:47 +0000 [r165804]  Russell Bryant <russell at digium.com>
+
+	* /, main/utils.c: Merged revisions 165801 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008)
+	  | 19 lines Merged revisions 165796 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
+	  | 11 lines Make ast_carefulwrite() be more careful. This patch
+	  handles some additional cases that could result in partial writes
+	  to the file description. This was done to address complaints
+	  about partial writes on AMI. (issue #13546) (more changes needed
+	  to address potential problems in 1.6) Reported by: srt Tested by:
+	  russell Review: http://reviewboard.digium.com/r/99/ ........
+	  ................
+
+2008-12-18 21:24 +0000 [r165794]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_queue.c, channels/chan_oss.c, channels/chan_dahdi.c,
+	  channels/chan_misdn.c, /, channels/chan_sip.c, pbx/pbx_ael.c:
+	  Merged revisions 165792 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 |
+	  file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
+	  Numerous documentation updates. (closes issue #13970) Reported
+	  by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt
+	  uploaded by blitzrage (license 10) ........
+
+2008-12-18 19:45 +0000 [r165728]  Russell Bryant <russell at digium.com>
+
+	* apps/app_dial.c, main/pbx.c, /, include/asterisk/pbx.h: Merged
+	  revisions 165723 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 |
+	  russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines
+	  Remove the need for AST_PBX_KEEPALIVE with the GoSub option from
+	  Dial. This is part of an effort to completely remove
+	  AST_PBX_KEEPALIVE and other similar return codes from the source.
+	  While this usage was perfectly safe, there are others that are
+	  problematic. Since we know ahead of time that we do not want to
+	  PBX to destroy the channel, the PBX API has been changed so that
+	  information can be provided as an argument, instead, thus
+	  removing the need for the KEEPALIVE return value. Further changes
+	  to get rid of KEEPALIVE and related code is being done by murf.
+	  There is a patch up for that on review 29. Review:
+	  http://reviewboard.digium.com/r/98/ ........
+
+2008-12-18 19:36 +0000 [r165725]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_odbc.c, /: Merged revisions 165724 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165724 |
+	  mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8
+	  lines Fix crashes in res_odbc. The variable "class" was being set
+	  NULL just prior to being dereferenced in an ao2_link call. I have
+	  moved the setting of the variable to NULL until after the
+	  ao2_link call. ........
+
+2008-12-18 18:58 +0000 [r165664]  Russell Bryant <russell at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 165662 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r165662 | russell | 2008-12-18 12:54:47 -0600
+	  (Thu, 18 Dec 2008) | 15 lines Merged revisions 165661 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008)
+	  | 7 lines Set the process group ID on the MOH process so that all
+	  children will get killed (closes issue #14099) Reported by: caspy
+	  Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy
+	  (license 645) ........ ................
+
+2008-12-18 18:47 +0000 [r165660]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 165658 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008)
+	  | 2 lines Fix 2 resource leaks and fix another pipe-to-comma
+	  conversion ........
+
+2008-12-18 17:59 +0000 [r165605-165606]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merge in changes to return chan_sip to
+	  matching based on how it was previously done and how it is done
+	  in trunk. It will do name based for users and friends and IP
+	  based for peers. (closes issue #14107) Reported by: jsmith
+
+	* main/rtp.c, /: Merged revisions 165599 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) |
+	  11 lines Merged revisions 165591 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
+	  lines Only care about a compatible codec for early bridging if we
+	  are actually bridging to another channel. If we are not we
+	  actually want to bring the audio back to us. (closes issue
+	  #13545) Reported by: davidw ........ ................
+
+2008-12-18 16:48 +0000 [r165543]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_odbc.c, /: Merged revisions 165541 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165541 |
+	  tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
+	  Fix reference counts of the class and add an assertion to the
+	  end. ........
+
+2008-12-17 21:48 +0000 [r165332]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_odbc.c, /: Merged revisions 165330 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165330 |
+	  mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3
+	  lines Fix a refcount leak in res_odbc ........
+
+2008-12-17 21:31 +0000 [r165329]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_macro.c: Merged revisions 165325 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 |
+	  tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines
+	  Oops, broke trunk ........
+
+2008-12-17 21:25 +0000 [r165324]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c, /,
+	  res/res_realtime.c: Merged revisions 165318 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec
+	  2008) | 15 lines Merged revisions 165255 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
+	  2008) | 7 lines Fix some memory leaks found while looking at how
+	  realtime configs are handled. Also cleaned up some coding
+	  guidelines violations in app_realtime.c, mostly related to
+	  spacing ........ ................
+
+2008-12-17 21:22 +0000 [r165323]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_macro.c: Merged revisions 165319 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008)
+	  | 11 lines Merged revisions 165317 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
+	  | 4 lines Reverse the fix from issue #6176 and add proper
+	  handling for that issue. (Closes issue #13962, closes issue
+	  #13363) Fixed by myself (license 14) ........ ................
+
+2008-12-17 21:02 +0000 [r165279]  Steve Murphy <murf at digium.com>
+
+	* /, utils/extconf.c: Merged revisions 165254 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165254 |
+	  murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines
+	  This patch is here committed to satisfy the buildbot, who has a
+	  problem with the const. ........
+
+2008-12-17 20:02 +0000 [r165242]  Terry Wilson <twilson at digium.com>
+
+	* /, res/res_phoneprov.c: Merged revisions 165219 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165219 |
+	  twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines
+	  Polycom phones close the connection after reading a little bit of
+	  the firmware files, we should stop sending in that case. Also,
+	  make that case print out a debug statement instead of a scary
+	  WARNING. ........
+
+2008-12-17 19:54 +0000 [r165218]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 165216 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 |
+	  file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
+	  Call proxy_update so that the IP address gets populated. Sending
+	  stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by:
+	  chris-mac ........
+
+2008-12-17 17:56 +0000 [r165146]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 165142-165143 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed,
+	  17 Dec 2008) | 10 lines Use the create_vm_state_from_user
+	  function in a place where it was not being used before. Also,
+	  I've moved the urgent folder check in messagecount() up a bit so
+	  that the flow is a bit better. This was something I noticed while
+	  taking a look at issue #13973, although I don't think this is the
+	  underlying cause of the issue. ........ r165143 | mmichelson |
+	  2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And
+	  actually assign the function to a pointer... ........
+
+2008-12-17 05:53 +0000 [r165093]  Steve Murphy <murf at digium.com>
+
+	* utils/conf2ael.c, pbx/ael/ael-test/ref.ael-vtest13,
+	  utils/check_expr.c, utils/Makefile,
+	  pbx/ael/ael-test/ref.ael-vtest17, /, pbx/pbx_ael.c,
+	  utils/ael_main.c, utils/extconf.c: Merged revisions 165071 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk I
+	  might add here that in I tested the merged fixes from trunk in
+	  both 1.6.0 and 1.6.1 via both 'make' and ./runtests in the ael
+	  regression tests for all but DEBUG_CHANNEL_LOCKS,
+	  DEBUG_SCHEDULER, and CHANNEL_TRACE options. ........ r165071 |
+	  murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines A
+	  possibly "horrible fix" for a "horribly broken" situation. As
+	  stuff shifts around in the asterisk code, the miscellaneous
+	  inclusions from the standalone stuff gets broken. There's no easy
+	  fix for this situation. I made sure that everything in utils
+	  builds without problem ***AND*** that aelparse runs the
+	  regressions correctly with the following make menuselect options
+	  both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS
+	  MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS
+	  DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to
+	  #undef all these features in the various utils native files; I
+	  guess I could do the same for the copied-in files, surrounded by
+	  STANDALONE ifdef. A standalone isn't going to care about threads,
+	  mutexes, etc. ........
+
+2008-12-16 23:07 +0000 [r164980]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 164978 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec
+	  2008) | 15 lines Merged revisions 164977 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
+	  2008) | 7 lines After looking through SIP registration code most
+	  of the day, this is one of the few things I could find that was
+	  just plain wrong. Even though it probably isn't possible for it
+	  to happen, it seems weird to have code that checks if a pointer
+	  is NULL and then immediately dereferences that pointer if it was
+	  NULL. ........ ................
+
+2008-12-16 22:52 +0000 [r164960]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, apps/app_record.c: Merged revisions 164942 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 |
+	  jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines
+	  (closes issue #13669) Reported by: pj Delete file recording if
+	  recording terminated from a hangup. ........
+
+2008-12-16 21:40 +0000 [r164813-164884]  Russell Bryant <russell at digium.com>
+
+	* /, main/utils.c: Merged revisions 164882 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008)
+	  | 17 lines Merged revisions 164881 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
+	  | 9 lines Fix an issue where DEBUG_THREADS may erroneously report
+	  that a thread is exiting while holding a lock. If the last lock
+	  attempt was a trylock, and it failed, it will still be in the
+	  list of locks so that it can be reported. (closes issue #13219)
+	  Reported by: pj ........ ................
+
+	* /, apps/app_macro.c: Merged revisions 164877 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008)
+	  | 14 lines Merged revisions 164876 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
+	  | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
+	  been returned. This is a bug I noticed while looking at the code
+	  for app_macro. This return code means that another thread has
+	  assumed ownership of the channel and it can no longer be touched.
+	  (I hate this return code with a passion, by the way.) ........
+	  ................
+
+	* main/manager.c, /: Merged revisions 164807 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008)
+	  | 17 lines Merged revisions 164806 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
+	  | 9 lines Add "restart gracefully" to the AMI blacklist of CLI
+	  commands. "module unload" was already identified as a command
+	  that can not be used from the AMI. "restart gracefully"
+	  effectively unloads all modules, and will run in to the same
+	  problems. (closes issue #13894) Reported by: kernelsensei
+	  ........ ................
+
+2008-12-16 20:18 +0000 [r164805]  Steve Murphy <murf at digium.com>
+
+	* main/pbx.c, /: Merged revisions 164801 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 |
+	  murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines
+	  (closes issue #14076) Reported by: toc Tested by: murf OK, Well
+	  this issue has had its share of flip-flopping. I found the
+	  following: 1. the code in question, in ext_cmp1 in pbx.c, would
+	  not allow two extensions that vary only by any dashes contained
+	  within them, to be defined in the same context. 2. for input
+	  dialstrings, dashes are NOT ignored. So, skipping them when
+	  sorting patterns seemed a bit silly. Thus, you might declare ext
+	  891 in a context, but if you try dialing 8-9-1, it will NOT match
+	  891. So, I proposed to remove the code from ext_cmp1 to skip the
+	  spaces and dashes. Just kept us from declaring 891 and 8-9-1 in
+	  the same context, forcing users to generate otherwise uselessly
+	  obfuscated dialplan code to get the same effect. Then, I tried
+	  out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the
+	  same context! 2. You can't define 891, and have 8-9-1 match it!
+	  Nor can you define 8-9-1, and have 891 match it! So, it appears
+	  that my proposal simply restores the pbx to behaving as it did in
+	  1.4. ........
+
+2008-12-16 19:54 +0000 [r164799]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/safe_asterisk, /: Merged revisions 164798 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16
+	  Dec 2008) | 4 lines Set up umask as a possible configuration
+	  option. (closes issue #13753) Reported by: irroot ........
+
+2008-12-16 17:18 +0000 [r164677-164739]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/threadstorage.h, /, main/threadstorage.c: Merged
+	  revisions 164737 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008)
+	  | 22 lines Merged revisions 164736 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
+	  | 14 lines Fix memory leak and invalid reporting issues with
+	  DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
+	  being used within the context of the thread local data
+	  destructors. We would go off and allocate more thread local data
+	  while the pthread lib was in the middle of destroying it all.
+	  This led to a memory leak. Another issue was an invalid argument
+	  being provided to the the object_add API call. (closes issue
+	  #13678) Reported by: ys Tested by: Russell ........
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 164675 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008)
+	  | 19 lines Merged revisions 164672 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
+	  | 11 lines Fix a memory leak related to the use of the "setvar"
+	  configuration option. The problem was that these variables were
+	  being appended to the list of vars on the sip_pvt every time a
+	  re-registration or re-subscription came in. Since it's just a
+	  waste of memory to put them there unless the request was an
+	  INVITE, then the fix is to check the request type before copying
+	  the vars. (closes issue #14037) Reported by: marvinek Tested by:
+	  russell ........ ................
+
+2008-12-16 15:47 +0000 [r164662]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 164659 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 |
+	  file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines
+	  When using externhost make sure the port gets set to the bindaddr
+	  port if one was not specified in the externhost value itself.
+	  (closes issue #13634) Reported by: performer ........
+
+2008-12-16 15:42 +0000 [r164658]  Steve Murphy <murf at digium.com>
+
+	* main/pbx.c, /: Merged revisions 164648 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) |
+	  13 lines Merged revisions 164634 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
+	  lines I added a sentence to clarify why - and ' ' are ignored in
+	  patterns as per bug 14076. Leif says he'll put some stuff about
+	  it in the extensions.conf sample, etc. ........ ................
+
+2008-12-16 15:02 +0000 [r164521-164625]  Russell Bryant <russell at digium.com>
+
+	* /, apps/app_minivm.c: Merged revisions 164623 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 |
+	  russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines
+	  Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable
+	  that was not needed. (closes issue #14081) Reported by: pkempgen
+	  ........
+
+	* /, res/res_musiconhold.c: Merged revisions 164606 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r164606 | russell | 2008-12-16 08:31:02 -0600
+	  (Tue, 16 Dec 2008) | 13 lines Merged revisions 164605 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008)
+	  | 5 lines Don't try to change working directory if a directory
+	  was not configured. (closes issue #14089) Reported by: caspy
+	  ........ ................
+
+	* channels/chan_dahdi.c, /: Merged revisions 164602 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008)
+	  | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue
+	  #14090) Reported by: alecdavis Patches:
+	  chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
+	  585) ........
+
+	* channels/chan_iax2.c, /: Merged revisions 164525 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008)
+	  | 6 lines Open a timer before loading configuration so that the
+	  trunking configuration option will take effect. (closes issue
+	  #14082) Reported by: seandarcy ........
+
+	* channels/chan_iax2.c, /: Merged revisions 164522 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008)
+	  | 4 lines Fix log message to refer to the generic timing
+	  interface, not DAHDI specifically (inspired by issue #14082)
+	  ........
+
+	* main/frame.c, /: Merged revisions 164519 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164519 |
+	  russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines
+	  Make sure we handle a uint32_t payload in ast_frdup() (closes
+	  issue #14080) Reported by: fnordian Patches: frame.patch uploaded
+	  by fnordian (license 110) ........
+
+2008-12-15 19:54 +0000 [r164421-164425]  Mark Michelson <mmichelson at digium.com>
+
+	* /, include/asterisk/pbx.h: Merged revisions 164423 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600
+	  (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec
+	  2008) | 3 lines Add the deadlock note to ast_spawn_extension as
+	  well ........ ................
+
+	* /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
+	  revisions 164419 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec
+	  2008) | 12 lines Merged revisions 164416 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
+	  2008) | 4 lines Add notes to autoservice and pbx doxygen
+	  regarding a potential deadlock scenario so that it is avoided in
+	  the future ........ ................
+
+2008-12-15 18:27 +0000 [r164355]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, cdr/cdr_pgsql.c: Merged revisions 164349 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164349 |
+	  tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines
+	  When querying for the structure of the CDR table, remove the
+	  schema, if it exists. (Closes issue #14058) ........
+
+2008-12-15 18:14 +0000 [r164314-164353]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 164351 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) |
+	  13 lines Merged revisions 164350 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
+	  lines Do not try to unlock a non-existant channel if the transfer
+	  fails. (closes issue #13800) Reported by: dwagner Patches:
+	  asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
+	  608) ........ ................
+
+	* /, main/file.c: Merged revisions 164312 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164312 |
+	  file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines Use
+	  ast_seekstream to return the file stream back to the beginning
+	  instead of directly seeking to zero. This is because some audio
+	  formats have headers at the front that need to be skipped, which
+	  will be done by the format module. (closes issue #14079) Reported
+	  by: elguero ........
+
+2008-12-15 16:32 +0000 [r164276-164300]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /, main/features.c: Merged revisions 164203 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r164203 | russell | 2008-12-15 08:40:24 -0600
+	  (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
+	  | 31 lines Handle a case where a call can be bridged to a channel
+	  that is still ringing. The issue that was reported was about a
+	  case where a RINGING channel got redirected to an extension to
+	  pick up a call from parking. Once the parked call got taken out
+	  of parking, it heard silence until the other side answered.
+	  Ideally, the caller that was parked would get a ringing
+	  indication. This patch fixes this case so that the caller
+	  receives ringback once it comes out of parking until the other
+	  side answers. The fixes are: - Make sure we remember that a
+	  channel was an outgoing channel when doing a masquerade. This
+	  prevents an erroneous ast_answer() call on the channel, which
+	  causes a bogus 200 OK to be sent in the case of SIP. - Add some
+	  additional comments to explain related parts of code. - Update
+	  the handling of the ast_channel visible_indication field. Storing
+	  values that are not stateful is pointless. Control frames that
+	  are events or commands should be ignored. - When a bridge first
+	  starts, check to see if the peer channel needs to be given
+	  ringing indication because the calling side is still ringing. -
+	  Rework ast_indicate_data() a bit for the sake of readability.
+	  (closes issue #13747) Reported by: davidw Tested by: russell
+	  Review: http://reviewboard.digium.com/r/90/ ........
+	  ................
+
+	* /, pbx/pbx_dundi.c: Merged revisions 164272 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164272 |
+	  russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines
+	  When a reload is issued, always process the configuration for
+	  dundi.conf. The reason is that a reload can be used to refresh
+	  DNS lookups for defined peers. Even if the config file hasn't
+	  changed, we want to process it for that purpose. (closes issue
+	  #13776) Reported by: kombjuder ........
+
+2008-12-15 16:18 +0000 [r164273-164274]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 164270 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 |
+	  mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4
+	  lines Fix a compile warning and a logic error that could have
+	  been bad for non-realtime queues ........
+
+	* apps/app_queue.c, /: Merged revisions 164268 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 |
+	  mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17
+	  lines Fix up a few issues with regards to queues * Fix reference
+	  counting used in the __queues_show function * Add code to be sure
+	  that the "queue show" command does not print information for a
+	  realtime queue which has been deleted from the backend * Add a
+	  missing unref to the realtime queue loading function for the case
+	  where a queue is in the module's container but has been deleted
+	  from the realtime backend (closes issue #14033) Reported by:
+	  cristiandimache Patches: 14033.patch uploaded by putnopvut
+	  (license 60) Tested by: cristiandimache ........
+
+2008-12-15 15:50 +0000 [r164266]  Joshua Colp <jcolp at digium.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
+	  configure.ac: Merged revisions 164257 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 |
+	  file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines
+	  Make app_fax compatible with newer versions of spandsp. This
+	  remains backwards compatible with earlier versions though so do
+	  not fret. (closes issue #14073) Reported by: seandarcy ........
+
+2008-12-13 01:01 +0000 [r163914]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 163912 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 |
+	  file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines
+	  Only detach and destroy the whisper audiohooks if they are
+	  actually in use. ........
+
+2008-12-13 00:08 +0000 [r163875]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 163873 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 |
+	  twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines
+	  When using realtime queues, app_queue wasn't updating the
+	  strategy if it was changed in the realtime backend. This patch
+	  resolves the issue for almost all situations. It is currently not
+	  supported to switch to the linear strategy via realtime since the
+	  ao2_container for members will have been set to have multiple
+	  buckets and therefore the members would be unordered. (closes
+	  issue #14034) Reported by: cristiandimache Tested by:
+	  otherwiseguy, cristiandimache ........
+
+2008-12-12 23:08 +0000 [r163830]  Russell Bryant <russell at digium.com>
+
+	* /: Merged revisions 163829 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ ........
+
+2008-12-12 22:05 +0000 [r163764]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, main/editline/read.c, /: Merged revisions 163762
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r163762 | tilghman | 2008-12-12 16:04:26 -0600
+	  (Fri, 12 Dec 2008) | 14 lines Merged revisions 163761 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
+	  | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
+	  but also add a pointer inside editline to look back to
+	  asterisk.c, so others don't spend as much time as I did looking
+	  (in the wrong place) for the appropriate function. Reported by:
+	  ZX81, via the #asterisk-users channel Fixed by: me (license 14)
+	  ........ ................
+
+2008-12-12 19:58 +0000 [r163715]  Steve Murphy <murf at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 163675 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1
+	  line demote always-appearing debug message (for certain boards)
+	  to ast_debug lev 3 msg instead ........
+
+2008-12-12 18:53 +0000 [r163656-163672]  Russell Bryant <russell at digium.com>
+
+	* main/tcptls.c, /, channels/chan_sip.c: Merged revisions 163670
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12
+	  Dec 2008) | 6 lines Rename a number of tcptls_session variables.
+	  There are no functional changes here. The name "ser" was used in
+	  a lot of places. However, it is a relic from when the struct was
+	  a server_instance, not a session_instance. It was renamed since
+	  it represents both a server or client connection. ........
+
+	* /, channels/chan_sip.c: Merged revisions 163667 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 |
+	  russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines
+	  Fix a small race condition in sip_tcp_locate(). We must increase
+	  the reference count on the tcptls_session _before_ unlocking the
+	  thread list. ........
+
+	* /, channels/chan_sip.c: Merged revisions 163642 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 |
+	  russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines
+	  Resolve crashes when using SIP TCP/TLS with qualify. The problem
+	  was a reference count error on the tcptls_session structure.
+	  (closes issue #13989) Reported by: Nugget ........
+
+2008-12-12 18:19 +0000 [r163640]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 163629 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 |
+	  file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
+	  When a device registers we need to unlink them (if linked) from
+	  the peers_by_ip container and link them back in since their IP
+	  address has changed. This would have manifested itself if you
+	  configured a new device (as type=peer), registered, and then
+	  tried to place a call from the device. Since the peer was not
+	  linked into the peers_by_ip container it would have never been
+	  found. (closes issue #13811) Reported by: pj ........
+
+2008-12-12 17:26 +0000 [r163624]  Michiel van Baak <michiel at vanbaak.info>
+
+	* res/res_monitor.c, /: Merged revisions 163612 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r163612 |
+	  mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines
+	  Document default Monitor file location. (closes issue #14065)
+	  Reported by: kshumard Patches:
+	  res_monitor.documentation.patch.txt uploaded by kshumard (license
+	  92) ........
+
+2008-12-12 16:57 +0000 [r163581]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c, /, channels/chan_sip.c: Merged revisions 163579
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec
+	  2008) | 4 lines Since chan_sip is callback devicestate driven do
+	  not pass in actual states, pass in unknown so we get asked.
+	  Additionally do not pass in an actual device state value in
+	  ast_setstate since the channel may be callback driven. (closes
+	  issue #13525) Reported by: pj ........
+
+2008-12-12 14:48 +0000 [r163514-163515]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, main/autoservice.c, /,
+	  include/asterisk/channel.h: Merged revisions 163449 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r163449 | russell | 2008-12-12 07:55:30 -0600
+	  (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008)
+	  | 26 lines Resolve issues that could cause DTMF to be processed
+	  out of order. These changes come from team/russell/issue_12658 1)
+	  Change autoservice to put digits on the head of the channel's
+	  frame readq instead of the tail. If there were frames on the
+	  readq that autoservice had not yet read, the previous code would
+	  have resulted in out of order processing. This required a new API
+	  call to queue a frame to the head of the queue instead of the
+	  tail. 2) Change up the processing of DTMF in ast_read(). Some of
+	  the problems were the result of having two sources of pending
+	  DTMF frames. There was the dtmfq and the more generic readq. Both
+	  were used for pending DTMF in various scenarios. Simplifying
+	  things to only use the frame readq avoids some of the problems.
+	  3) Fix a bug where a DTMF END frame could get passed through when
+	  it shouldn't have. If code set END_DTMF_ONLY in the middle of
+	  digit emulation, and a digit arrived before emulation was
+	  complete, digits would get processed out of order. (closes issue
+	  #12658) Reported by: dimas Tested by: russell, file Review:
+	  http://reviewboard.digium.com/r/85/ ........ ................
+
+	* /, pbx/pbx_dundi.c: Merged revisions 163512 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008)
+	  | 13 lines Merged revisions 163511 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)

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