[asterisk-commits] lmadsen: tag 1.6.1.0-rc5 r189388 - /tags/1.6.1.0-rc5/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Apr 20 13:51:54 CDT 2009


Author: lmadsen
Date: Mon Apr 20 13:51:50 2009
New Revision: 189388

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=189388
Log:
Importing files for 1.6.1.0-rc5 release.

Added:
    tags/1.6.1.0-rc5/.lastclean   (with props)
    tags/1.6.1.0-rc5/.version   (with props)
    tags/1.6.1.0-rc5/ChangeLog   (with props)

Added: tags/1.6.1.0-rc5/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1.0-rc5/.lastclean?view=auto&rev=189388
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Added: tags/1.6.1.0-rc5/ChangeLog
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1.0-rc5/ChangeLog?view=auto&rev=189388
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--- tags/1.6.1.0-rc5/ChangeLog (added)
+++ tags/1.6.1.0-rc5/ChangeLog Mon Apr 20 13:51:50 2009
@@ -1,0 +1,56449 @@
+2009-04-20  Leif Madsen <lmadsen at digium.com>
+
+	* Create Asterisk 1.6.1.0-rc5
+
+2009-04-20 17:08 +0000 [r189352]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+	  file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+	  Fix a bug with non-UDP connections that caused dialogs to not get
+	  freed. This issue crept up because of a reference count issue on
+	  non-UDP based dialogs. The dialog reference count was increased
+	  when transmitting a packet reliably but never decreased. This
+	  caused the dialog structure to hang around despite being unlinked
+	  from the dialogs container. (closes issue #14919) Reported by:
+	  vrban ........
+
+2009-04-20 14:06 +0000 [r189280]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c, /: Merged revisions 189278 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+	  2009) | 18 lines Merged revisions 189277 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+	  2009) | 12 lines Move the check for chan->fdno == -1 to after the
+	  zombie/hangup check. Many users were finding that their hung up
+	  channels were staying up and causing 100% CPU usage. (issue
+	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+	  uploaded by mmichelson (license 60) Tested by: falves11, bamby
+	  ........ ................
+
+2009-04-18 01:38 +0000 [r189206]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+	  (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+	  | 12 lines Fixed autologoff in agents.conf not working when agent
+	  logs in via AgentLogin app An agent logs in by calling an
+	  extension that calls the AgentLogin app. In agents.conf
+	  ackcall=always is set, so when they get a call they have the
+	  choice to either acknowledge it or ignore it. autologoff=10 is
+	  set as well, so if the agent ignores the call over 10sec one may
+	  assume that the agent should be logged out (and in this case
+	  hungup on as well), but this was not happening. (closes issue
+	  #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+	  dvossel (license 671) Review:
+	  http://reviewboard.digium.com/r/225/ ........ ................
+
+2009-04-17 21:55 +0000 [r189139]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+	  revisions 189137 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+	  | 17 lines Merged revisions 188833,189134 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+	  | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+	  Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+	  rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+	  Modifed/added some debug messages. JIRA ABE-1835 ........
+	  ................
+
+2009-04-17 20:21 +0000 [r189103]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+	  mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+	  lines Prevent a crash when SIP blonde transferring an unbridged
+	  call. If one attempts to use the attended transfer button on a
+	  SIP phone to transfer an unbridged call (such as a call to an
+	  IVR) but hangs up while the target of the transfer is still
+	  ringing, we need to not crash. The problem was that ast_hangup
+	  was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:46 +0000 [r189080]  Sean Bright <sean.bright at gmail.com>
+
+	* main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+	  seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+	  line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:33 +0000 [r189069]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 189010 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+	  2009) | 12 lines Merged revisions 189009 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+	  2009) | 5 lines Make Busy() application set the CDR disposition
+	  to BUSY. (closes issue #14306) Reported by: cristiandimache
+	  ........ ................
+
+2009-04-17 14:48 +0000 [r188940-188949]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+	  22 lines Merged revisions 188946 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+	  15 lines Fix a bug where a value used to create the channel name
+	  was bogus. This commit fixes the scenario where an incoming call
+	  is authenticated using a peer entry. Previously the channel name
+	  was created using either the username setting from the sip.conf
+	  entry or the IP address that the call came from. Now the channel
+	  name will be created using the peer name itself. This commit will
+	  not change the way the channel name is generated for users or
+	  friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+	  chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+	  Nick_Lewis, file ........ ................
+
+	* channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+	  17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+	  lines Fix a situation where the DAHDI channel private structure
+	  lock was not unlocked when it should have been. (issue AST-210)
+	  ........ ................
+
+2009-04-16 22:05 +0000 [r188776-188838]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+	  | 14 lines Merged revisions 188835 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+	  | 7 lines Only update realtime, if global option rtupdate !=
+	  false (closes issue #14885) Reported by: deepesh Patches:
+	  20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+	  Tested by: deepesh ........ ................
+
+	* apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+	  (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+	  | 4 lines Umask should not be exported into global namespace.
+	  (closes issue #14912) Reported by: jcapp ........
+	  ................
+
+2009-04-15 22:12 +0000 [r188649]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+	  (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+	  | 12 lines National prefix inserted even when caller ID not
+	  available When the caller ID is restricted, the expected behavior
+	  is for the caller id to be blank. In chan_dahdi, the national
+	  prefix is placed onto the callers number even if its restricted
+	  (empty) causing the caller id to be the national prefix rather
+	  than blank. (closes issue #13207) Reported by: shawkris Patches:
+	  national_prefix.diff uploaded by dvossel (license 671) Review:
+	  http://reviewboard.digium.com/r/220/ ........ ................
+
+2009-04-15 20:20 +0000 [r188473-188596]  Mark Michelson <mmichelson at digium.com>
+
+	* /, main/file.c: Merged revisions 188585 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+	  2009) | 13 lines Merged revisions 188582 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+	  2009) | 7 lines Update ast_readvideo_callback to match
+	  ast_readaudio_callback. This fixes potential refcount errors that
+	  may occur on ast_filestreams. AST-208 ........ ................
+
+	* apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+	  mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+	  lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:43 +0000 [r188254-188415]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c, /: Merged revisions 188413 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+	  file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+	  an incorrect clock rate when sending T140 text. (closes issue
+	  #14029) Reported by: epicac ........
+
+	* /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+	  file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+	  a bug with the change I made yesterday to outbound proxy support.
+	  Per discussion with oej on IRC we need the actual IP address, not
+	  the outbound proxy IP address, in the sa field. Upon further
+	  inspection this should make the behaviour of all other uses of
+	  the outbound proxy in the code. ........
+
+2009-04-14 05:46 +0000 [r188208-188212]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, /: Merged revisions 188210 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+	  tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+	  As suggested by Russell, warn users when their dialplan arguments
+	  contain pipes, but not commas. ........
+
+	* /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+	  tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+	  Application delimiter is ',', not '|'. (closes issue #14881)
+	  Reported by: stegro Patches: smsq.patch uploaded by stegro
+	  (license 752) ........
+
+2009-04-13 19:33 +0000 [r188104]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+	  2009) | 5 lines Fix another crash related to cached realtime
+	  music on hold. This was another off-by-one problem caused by
+	  moh_register. ........
+
+2009-04-13 16:32 +0000 [r188069]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+	  file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+	  Fix a bug where using an outbound proxy would cause the local
+	  address to be 127.0.0.1. Copy the outbound proxy IP address into
+	  the SIP dialog structure as the IP address we will be sending to.
+	  This has to be done because the logic that determines what local
+	  IP address to use in the SIP messages is not aware of an outbound
+	  proxy being in place. It only knows what IP address we are
+	  sending to. (closes issue #12006) Reported by: mnicholson
+	  ........
+
+2009-04-13 14:20 +0000 [r188038]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+	  mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+	  lines Set all queue variables on both the caller and member
+	  channels. This allows for the variables to be accessed if a
+	  member macro is run. Thanks to Grigoriy Puzankin for bringing
+	  this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187914]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/Makefile, /: Merged revisions 187906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+	  jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+	  Fix module embedding for chan_h323. Include libchanh323.a in the
+	  modules.link file so that all the symbols can be resolved at link
+	  time. (closes issue #11966) Reported by: dome Patches:
+	  issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+	  jpeeler ........
+
+2009-04-10 17:30 +0000 [r187767]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/sip-friends.sql,
+	  contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+	  (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+	  Apr 2009) | 2 lines Add lastms column to the contributed table
+	  designs ........ ................
+
+2009-04-10 16:54 +0000 [r187723]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, build_tools/embed_modules.xml: Merged revisions 187721 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+	  Apr 2009) | 5 lines clean up some patterns for files to remove
+	  add embedding support for bridge and test modules ........
+
+2009-04-10 16:03 +0000 [r187678]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+	  tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+	  Ensure pvt is not NULL before dereferencing it. (closes issue
+	  #14784) Reported by: pj ........
+
+2009-04-10 16:00 +0000 [r187676]  Russell Bryant <russell at digium.com>
+
+	* tests/test_heap.c, /: Merged revisions 187675 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 |
+	  russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
+	  Disable test modules by default. ........
+
+2009-04-10 03:56 +0000 [r187600]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c, main/pbx.c, main/manager.c, /,
+	  include/asterisk/linkedlists.h, main/features.c, main/http.c,
+	  main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged
+	  revisions 187599 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 |
+	  tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines
+	  Modify headers and macros, according to Russell's suggestions on
+	  the -dev list ........
+
+2009-04-09 19:14 +0000 [r187495]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr
+	  2009) | 24 lines Merged revisions 187484 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
+	  2009) | 18 lines Handle a SIP race condition (reinvite before an
+	  ACK) properly. RFC 5047 explains the proper course of action to
+	  take if a reINVITE is received before the ACK from a previous
+	  invite transaction. What we are to do is to treat the reINVITE as
+	  if it were both an ACK and a reINVITE and process it normally.
+	  Later, when we receive the ACK we had been expecting, we will
+	  ignore it since its CSeq is less than the current iseqno of the
+	  sip_pvt representing this dialog. (closes issue #13849) Reported
+	  by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
+	  (license 60) Tested by: mmichelson, klaus3000 ........
+	  ................
+
+2009-04-09 18:54 +0000 [r187486]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/manager.c, /, include/asterisk/linkedlists.h,
+	  include/asterisk/lock.h: Merged revisions 187483 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
+	  (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
+	  | 8 lines Race condition between ast_cli_command() and 'module
+	  unload' could cause a deadlock. Add lock timeouts to avoid this
+	  potential deadlock. (closes issue #14705) Reported by: jamessan
+	  Patches: 20090320__bug14705.diff.txt uploaded by tilghman
+	  (license 14) Tested by: jamessan ........ ................
+
+2009-04-09 17:43 +0000 [r187427]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 187421,187424 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
+	  09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
+	  cached realtime moh. The moh_register function links an mohclass
+	  and then immediately unrefs the class since the container now has
+	  a reference. The problem with using realtime music on hold is
+	  that the class is allocated, registered, and started in one fell
+	  swoop. The refcounting logic resulted in the count being off by
+	  one. The same problem did not happen when using a static config
+	  because the allocation and registration of an mohclass is a
+	  separate operation from starting moh. This also did not affect
+	  non-cached realtime moh because the classes are not registered at
+	  all. I also have modified res_musiconhold to use the _t_ variants
+	  of the ao2_ functions so that more info can be gleaned when
+	  attempting to trace the refcounts. I found this to be incredibly
+	  helpful for debugging this issue and there's no good reason to
+	  remove it. (closes issue #14661) Reported by: sum ........
+	  r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
+	  2009) | 3 lines Use safe macro practices even though they really
+	  aren't necessary. ........
+
+2009-04-09 17:22 +0000 [r187305-187388]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
+	  tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
+	  Allow '/' in username portion of register; this is a regression.
+	  (closes issue #14668) Reported by: Netview ........
+
+	* /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
+	  187363 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
+	  | 10 lines Merged revisions 187362 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
+	  | 3 lines Permit zero-length text messages in SIP. (Related to an
+	  issue posted to the -users list, subject "AEL2, BASE64_DECODE and
+	  hexadecimal") ........ ................
+
+	* main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
+	  utils/Makefile, include/asterisk.h, /, main/Makefile,
+	  main/file.c, main/astfd.c (added): Merged revisions 187302 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
+	  (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
+	  | 3 lines Add debugging mode for diagnosing file descriptor
+	  leaks. (Related to issue #14625) ........ r187301 | tilghman |
+	  2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
+	  missed this file in the last commit. ........ ................
+
+2009-04-08 16:53 +0000 [r186987-187048]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
+	  (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
+	  2009) | 10 lines Fix a small logical error when loading moh
+	  classes. We were unconditionally incrementing the number of
+	  mohclasses registered. However, we should actually only increment
+	  if the call to moh_register was successful. While this probably
+	  has never caused problems, I noticed it and decided to fix it
+	  anyway. ........ ................
+
+	* main/channel.c, /: Merged revisions 186985 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
+	  2009) | 30 lines Merged revisions 186984 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
+	  2009) | 24 lines Make a couple of changes with regards to a new
+	  message printed in ast_read(). "ast_read() called with no
+	  recorded file descriptor" is a new message added after a bug was
+	  discovered. Unfortunately, it seems there are a bunch of places
+	  that potentially make such calls to ast_read() and trigger this
+	  error message to be displayed. This commit does two things to
+	  help to make this message appear less. First, the message has
+	  been downgraded to a debug level message if dev mode is not
+	  enabled. The message means a lot more to developers than it does
+	  to end users, and so developers should take an effort to be sure
+	  to call ast_read only when a channel is ready to be read from.
+	  However, since this doesn't actually cause an error in operation
+	  and is not something a user can easily fix, we should not spam
+	  their console with these messages. Second, the message has been
+	  moved to after the check for any pending masquerades. ast_read()
+	  being called with no recorded file descriptor should not
+	  interfere with a masquerade taking place. This could be seen as a
+	  simple way of resolving issue #14723. However, I still want to
+	  try to clear out the existing ways of triggering this message,
+	  since I feel that would be a better resolution for the issue.
+	  ........ ................
+
+2009-04-08 05:07 +0000 [r186900]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
+	  tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
+	  Add lastms to the require API call. ........
+
+2009-04-08 00:10 +0000 [r186835-186844]  Mark Michelson <mmichelson at digium.com>
+
+	* /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
+	  revisions 186842 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
+	  2009) | 14 lines Merged revisions 186841 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
+	  2009) | 8 lines Fix a few typos of the word "frequency." (closes
+	  issue #14842) Reported by: jvandal Patches: frequency-typo.diff
+	  uploaded by jvandal (license 413) ........ ................
+
+	* /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
+	  mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
+	  lines Fix bad merge from fix for issue 13867. (closes issue
+	  #14686) Reported by: davidw ........
+
+	* main/channel.c, /: Merged revisions 186833 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
+	  2009) | 15 lines Merged revisions 186832 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
+	  2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
+	  p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+	  warning sounds will not be properly played to either party of the
+	  bridge. (closes issue #14845) Reported by: adomjan ........
+	  ................
+
+2009-04-07 22:33 +0000 [r186806]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
+	  | 10 lines Merged revisions 186775 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
+	  | 3 lines Fix Macro documentation to match current (and intended)
+	  behavior. (See -dev mailing list) ........ ................
+
+2009-04-07 20:53 +0000 [r186722]  Mark Michelson <mmichelson at digium.com>
+
+	* main/manager.c, /: Merged revisions 186720 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
+	  2009) | 12 lines Merged revisions 186719 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
+	  2009) | 6 lines Ensure that \r\n is printed after the ActionID in
+	  an OriginateResponse. (closes issue #14847) Reported by: kobaz
+	  ........ ................
+
+2009-04-03 20:21 +0000 [r186466]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
+	  (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
+	  2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
+	  properly switch formats when requested Don't offer
+	  AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+	  provide a slight performance benefit, the translation core in
+	  Asterisk has some flaws when a channel driver offers multiple raw
+	  formats. this fix is much simpler than fixing the translation
+	  core to solve that issue (although that will be done later).
+	  ........ ................
+
+2009-04-03 20:04 +0000 [r186448]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+	  revisions 186444,186447 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
+	  | 14 lines Merged revisions 186415 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
+	  | 7 lines Distinguish in a sent email between simple sends and
+	  forwards. (closes issue #11678) Reported by: jamessan Patches:
+	  20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman, lmadsen ........ ................ r186447 |
+	  tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
+	  Merged revisions 186445 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
+	  | 2 lines Found a conflict in the last commit, due to multiple
+	  targets ........ ................
+
+2009-04-03 16:38 +0000 [r186381]  David Vossel <dvossel at digium.com>
+
+	* /, main/audiohook.c: Merged revisions 186379 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
+	  dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines
+	  audio_audiohook_write_list() did not correctly update sample size
+	  after ast_translate. audio_audiohook_write_list() did not take
+	  into account that the sample size may change after translation
+	  depending on if the original frame is is 8khz or 16khz. the
+	  sample size is now updated after translating to reflect this
+	  possibility. This caused the audio on the receiving end to sound
+	  terrible. Thanks to jcolp and mmichelson for helping me work this
+	  out. (issue AST-197) ........
+
+2009-04-03  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.1.0-rc4 released.
+
+2009-04-03 15:54 +0000 [r186323]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/crypto.h, /: Merged revisions 186321 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+	  03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+	  lines Fix a problem with the crypto variable definitions not
+	  actually being defined properly. (closes issue #14804) Reported
+	  by: jvandal ........ ................
+
+2009-04-03 14:33 +0000 [r186288]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+	  2009) | 20 lines Fix the ability to retrieve voicemail messages
+	  from IMAP. A recent change made interactive vm_states no longer
+	  get added to the list of vm_states and instead get stored in
+	  thread-local storage. In trunk and all the 1.6.X branches, the
+	  problem is that when we search for messages in a voicemail box,
+	  we would attempt to update the appropriate vm_state struct by
+	  directly searching in the list of vm_states instead of using the
+	  get_vm_state_by_imap_user function. This meant we could not find
+	  the interactive vm_state that we wanted. (closes issue #14685)
+	  Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+	  (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:06 +0000 [r186232]  Russell Bryant <russell at digium.com>
+
+	* cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+	  | 29 lines Merged revisions 186229 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+	  | 21 lines Fix a memory leak in cdr_radius. I came across this
+	  while doing some testing of my ast_channel_ao2 branch. After
+	  running a test overnight that generated over 5 million calls,
+	  Asterisk had taken up about 1 GB of my system memory. So, I
+	  re-ran the test with MALLOC_DEBUG turned on. However, it showed
+	  no leaks in Asterisk during the test, even though Asterisk was
+	  still consuming it somehow. Instead, I turned to valgrind, which
+	  when run with --leak-check=full, told me exactly where the leak
+	  came from, which was from allocations inside the radiusclient-ng
+	  library. This explains why MALLOC_DEBUG did not report it. After
+	  a bit of analysis, I found that we were leaking a little bit of
+	  memory every time a CDR record was passed to cdr_radius. I don't
+	  actually have a radius server set up to receive CDR records.
+	  However, I always have my development systems compile and install
+	  all modules. In addition to making sure there are not build
+	  errors across modules, always loading modules helps find bugs
+	  like this, too, so it is strongly recommend for all developers.
+	  ........ ................
+
+2009-04-02 21:59 +0000 [r186177]  Mark Michelson <mmichelson at digium.com>
+
+	* configs/features.conf.sample, /: Merged revisions 186175 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+	  2009) | 5 lines Fix instructions in one-step parking comment to
+	  make more sense. Changed a capital K to a lowercase k. ........
+	  ................
+
+2009-04-02 17:27 +0000 [r186108]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+	  Apr 2009) | 3 lines ensure that the buffer passed to
+	  DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186062]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+	  186060 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+	  | 16 lines Merged revisions 186059 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
+	  ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+	  Apr 2009) | 2 lines Fix for AST-2009-003 ........
+	  ................ ................
+
+2009-04-02 13:53 +0000 [r185956]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+	  2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+	  DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+	  do, in fact, read data from userspace as part of their work. due
+	  to this fix, valgrind now reports a number of cases where
+	  chan_dahdi passed an uninitialized (or partially) buffer to these
+	  ioctls, which could lead to unexpected behavior. this patch
+	  corrects chan_dahdi to ensure that buffers passed to these ioctls
+	  are always fully initialized. ........ ................
+
+2009-04-01 19:06 +0000 [r185848]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+	  | 16 lines Merged revisions 185845 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+	  | 10 lines Fixes issue with dropped calles due to re-Invite glare
+	  and re-Invites never executing after a 491 Acknowledgement for
+	  491 responses were never being processed because it didn't match
+	  our pending invite's seqno. Since the ACK was never processed,
+	  the 491 frame would continue to be retransmitted until eventually

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