[svn-commits] lmadsen: tag 1.6.1-beta4 r165876 - /tags/1.6.1-beta4/
SVN commits to the Digium repositories
svn-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:
tags/1.6.1-beta4/.lastclean (with props)
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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--- 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)
[... 49855 lines stripped ...]
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