[svn-commits] lmadsen: tag 1.6.1.0-rc5 r189388 - /tags/1.6.1.0-rc5/
SVN commits to the Digium repositories
svn-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
==============================================================================
--- tags/1.6.1.0-rc5/.lastclean (added)
+++ tags/1.6.1.0-rc5/.lastclean Mon Apr 20 13:51:50 2009
@@ -1,0 +1,1 @@
+36
Propchange: tags/1.6.1.0-rc5/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.6.1.0-rc5/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.6.1.0-rc5/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.6.1.0-rc5/.version
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1.0-rc5/.version?view=auto&rev=189388
==============================================================================
--- tags/1.6.1.0-rc5/.version (added)
+++ tags/1.6.1.0-rc5/.version Mon Apr 20 13:51:50 2009
@@ -1,0 +1,1 @@
+1.6.1.0-rc5
Propchange: tags/1.6.1.0-rc5/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.6.1.0-rc5/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.6.1.0-rc5/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
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
==============================================================================
--- 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
[... 55773 lines stripped ...]
More information about the svn-commits
mailing list