[asterisk-commits] russell: tag 1.4.16 r93736 - in /tags/1.4.16: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Dec 18 13:19:27 CST 2007
Author: russell
Date: Tue Dec 18 13:19:26 2007
New Revision: 93736
URL: http://svn.digium.com/view/asterisk?view=rev&rev=93736
Log:
importing files for 1.4.16 release
Added:
tags/1.4.16/.lastclean (with props)
tags/1.4.16/.version (with props)
tags/1.4.16/ChangeLog (with props)
Added: tags/1.4.16/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.16/.lastclean?view=auto&rev=93736
==============================================================================
--- tags/1.4.16/.lastclean (added)
+++ tags/1.4.16/.lastclean Tue Dec 18 13:19:26 2007
@@ -1,0 +1,1 @@
+30
Propchange: tags/1.4.16/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.16/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.16/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.16/.version
URL: http://svn.digium.com/view/asterisk/tags/1.4.16/.version?view=auto&rev=93736
==============================================================================
--- tags/1.4.16/.version (added)
+++ tags/1.4.16/.version Tue Dec 18 13:19:26 2007
@@ -1,0 +1,1 @@
+1.4.16
Propchange: tags/1.4.16/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.16/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.16/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.16/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.4.16/ChangeLog?view=auto&rev=93736
==============================================================================
--- tags/1.4.16/ChangeLog (added)
+++ tags/1.4.16/ChangeLog Tue Dec 18 13:19:26 2007
@@ -1,0 +1,14298 @@
+2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 93667 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
+ | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........
+
+2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Rework deadlock avoidance used in ast_write,
+ since it meant that agent channels which were being monitored had
+ one audio file recorded and one empty audio file saved. (closes
+ issue #11529, reported by atis patched by me)
+
+2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker at digium.com>
+
+ * main/translate.c: What was I thinking when I wrote this
+ masterpiece? -1 + 1 = 0.. who woulda thunk it?.
+
+2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp at digium.com>
+
+ * main/utils.c: Do not try to access information about a lock when
+ printing out a trylock attempt. It is possible for the lock that
+ it references to no longer be valid. This would have caused
+ segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
+ Reported by: callguy (closes issue #11100) Reported by: callguy
+
+2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/time.h: Today is tomorrow's yesterday, and
+ yesterday's tomorrow is today, and tomorrow's tomorrow is the day
+ after tomorrow, so who cares if you recycle anyway? If this
+ confuses you, that's nothing compared to what this fixes. ;-)
+
+2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: We need to create the directory for a
+ voicemail user even if they are using IMAP storage since
+ greetings are stored in the filesystem. (closes issue #11388,
+ reported by spditner, patch by me inspired by a patch by
+ spditner)
+
+2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_zap.c: If a call is received with a called number
+ IE containing nothing go to the 's' extension. (closes issue
+ #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming at digium.com>
+
+ * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
+ res/Makefile, channels/Makefile, formats/Makefile: fix some
+ copy-and-paste leftovers
+
+2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
+ apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
+ dependencies on res_monitor and res_features. I wonder if
+ Asterisk can run at all without res_features. My guess is that
+ there's propably a lot of more modules and the core that depends
+ on it. Reported by: caio1982 (closes issue #11574)
+
+2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming at digium.com>
+
+ * formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
+ configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
+ formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
+ include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
+ configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
+ (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
+ http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
+ rizzo brought up some issues related to the way that the metadata
+ required for menuselect and the rest of the build system is
+ extracted from the source files. Since I had a few hours to kill
+ on an airplane today, I decided to improve this situation... so
+ now the system caches the extracted metadata and uses it to build
+ the menuselect 'tree' as much as it can. The result of this is
+ that when a single source file is changed, only the metadata for
+ that file needs to be extracted again, and the rest is used from
+ the cache files. I also reduced the number of forked processes
+ required to do the metadata extraction; it was actually possible
+ to do most of what we needed in the Makefiles themselves without
+ using any shell scripts at all! On my laptop, these changes
+ resulted in an 80% decrease in the time required for the
+ 'menuselect.makeopts' automatic check to occur after editing a
+ single source file. While doing this work I also cleaned up a few
+ minor things in the Makefiles, adding a check for 'awk' to the
+ configure script and changed all remaining places we use 'grep'
+ or 'awk' to use the ones found by the configure script, and
+ changed the 'prep_tarball' script to build the menuselect
+ metadata so that tarballs of Asterisk will include it and won't
+ require the user to wait while it is extracted after unpacking.
+
+2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell at digium.com>
+
+ * main/config.c: There are a lot of existing systems that #include
+ non-existent files. So, to make the transition to treating this
+ as an error a bit less painless, just issue a huge error message
+ for now. Then, later, we can reinstate the code that treats it as
+ a failure. (Thanks to philippel for the feedback)
+
+2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Up the length of the format on the SIP
+ channel since it can now be rather long. (closes issue #11552)
+ Reported by: francesco_r
+
+2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
+ state setting and overlap_task thread starting.
+
+2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Change help documentation to match actual behavior
+ (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
+ (Closes issue #11548)
+
+2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
+ don't spam the CLI with messages about possible deadlocks.
+ Instead just print the intended single message every five
+ seconds. (closes issue 11537, reported and patched by dimas)
+
+2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_zap.c: Properly initialize polarity statuses, so
+ that they are detected properly. Reported by: julianjm Patch by:
+ julianjm (Closes issue #10238)
+
+2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker at digium.com>
+
+ * main/pbx.c: Make application help text a little more clear about
+ the use of extensions in a filename.
+
+2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Prevent another potential fd leak
+
+ * apps/app_voicemail.c: Prevent a possible fd leak.
+
+2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker at digium.com>
+
+ * main/config.c, channels/chan_sip.c, channels/chan_h323.c,
+ channels/chan_iax2.c: If a typo is found in a config file, we
+ previous continued on with what was already loaded. We do not
+ want to do this (see bug below for details). This makes it so
+ that if a [ is found without a ], the entire config will fail,
+ and nothing in it will be loaded. Isue #10690.
+
+2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: emit a warning message when we drop a G.729B
+ CNG frame destined for the transcoder
+
+2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker at digium.com>
+
+ * apps/app_meetme.c: Don't increment user count until after name
+ has been recorded (if enabled). Issue 11048, tested by pep.
+
+2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: resolve compiler warning
+
+2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_features.c: Correctly detect where a dynamic feature was
+ activated. Before this patch, the channel which initiated the
+ bridge was always assumed to have been the one which activated
+ the dynamic feature. This patch corrects this. (closes issue
+ #11529, reported and patched by nic_bellamy)
+
+2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Test
+ directly for the API that fixed AST-2007-026, to ensure that
+ older versions of PostgreSQL are no longer acceptable. (Closes
+ issue #11526)
+
+2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Removing an unused variable.
+
+2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp at digium.com>
+
+ * main/global_datastores.c: Fix potential memory leak with the
+ dialed interfaces list if another memory allocation fails.
+ (closes issue #11507) Reported by: eliel Patches:
+ global_datastores.c.patch uploaded by eliel (license 64)
+
+2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fixing autofill to be more accurate.
+ Specifically, if calls ahead of the current caller were ringing
+ members (but not yet bridged) there could be available members
+ and waiting callers who would not get matched up. The member
+ availability checker was correctly determining the number of
+ available members in this scenario, but the queue itself did not
+ parallelly reflect this status on the pending calls. This commit
+ corrects the issue. (closes issue #11459, reported by
+ equissoftware, patched by me)
+
+2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Add G729A as another possible payload name for G729.
+ Some devices use this instead of G729, which is perfectly normal
+ since the payload number itself is defined and can't be used by
+ anything else so the name doesn't matter that much. (closes issue
+ #11483) Reported by: revolution Patches: rtp.diff uploaded by
+ revolution (license 346)
+
+2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: If there are no members in a queue, then the
+ loop where the datastore for detecting duplicate dialed numbers
+ will be skipped, meaning the datastore isn't created. This means
+ that when we try to free it, there's a crash. This stops that
+ crash from occurring. (closes issue #11499, reported by slavon,
+ patched by eliel)
+
+2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: It is possible for nativeformats to contain
+ more then one codec, so print out multiple ones. (closes issue
+ #11366) Reported by: ovi
+
+2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Avoid reinvite race situations with two
+ Asterisks trying to reinvite each other in 1.4 and trunk. This
+ patch implements support for the 491 error code that Asterisk 1.4
+ generates on situations where we get an incoming INVITE and
+ already has one in progress. Thanks to mavetju for reporting and
+ to Raj Jain for an excellent explanation of the problem. Patch by
+ myself. Tested with 8 Asterisk servers connected to each other in
+ a training network. Closes issue #10481
+
+2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker at digium.com>
+
+ * main/dsp.c: We need to make sure we free the input frame if we
+ return a different frame in ast_dsp_process. Issue 11273, pointed
+ out by dimas, with a patch by eliel.
+
+2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: even though Asterisk explicitly requests that
+ endpoints using G.729 do *not* use Annex B (silence detection and
+ comfort noise generation) some do anyway; the transcoder card
+ interface does not currently work properly with CNG frames, so
+ trim off the CNG before sending the data
+
+2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell at digium.com>
+
+ * main/utils.c: Make the lock protecting each thread's list of
+ locks it currently holds recursive. I think that this will fix
+ the situation where some people have said that "core show locks"
+ locks up the CLI. (related to issue #11080)
+
+ * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
+ code. The ast_mutex_init() function had the mutex attribute
+ object marked as static. This means that multiple threads
+ initializing locks at the same time could step on each other and
+ end up with improperly initialized locks. (found when tracking
+ down locking issues related to issue #11080)
+
+ * include/asterisk/lock.h: I love fixing lock related errors in the
+ lock debugging code. That's about as ironic as it gets in
+ Asterisk programming land. Anyway, I spotted this bug while
+ trying to track down why systems are locking up and acting weird
+ in issue #11080. The mutex attribute object was marked as static
+ in this function when it should not have been.
+
+ * apps/app_dial.c: * Add channel locking around datastore
+ operations that expect the channel to be locked. * Document why
+ we don't record Local channels in the dialed interfaces list. *
+ Remove the dialed variable as it isn't needed. * Restructure some
+ code for clarity and coding guidelines stuff
+
+ * apps/app_queue.c: * Add channel locking around datastore
+ operations that expect the channel to be locked. * Document why
+ we don't record Local channels in the dialed interfaces list. *
+ Handle memory allocation failure. * Remove the dialed variable,
+ as it wasn't actually needed. * Tweak some formatting to conform
+ to coding guidelines.
+
+ * main/autoservice.c: * Add a bit more of a verbose comment as to
+ why a hangup frame needs to be queued up if autoservice gets a
+ NULL return from ast_read(). * Make the process of queueing the
+ hangup frame more efficient by putting the frame where it is
+ going to end up and avoiding some locking and extra memory
+ allocations and freeing.
+
+2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson at digium.com>
+
+ * main/autoservice.c: Hangups that happen during autoservice were
+ not processed appropriately. This is because a hangup actually
+ causes a NULL frame to be received, not a hangup frame. Queueing
+ a hangup if we receive a NULL frame during autoservice corrects
+ this problem (closes issue #11467, reported by jmls, patched by
+ me)
+
+2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell at digium.com>
+
+ * apps/app_dial.c: Don't unlock the dialed_interfaces list until
+ we're done messing with the iterator.
+
+ * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
+ from Queue() and Dial() again. There was a slight flaw in the
+ code to prevent call forwards from looping that caused this
+ problem. (related to issue #11486)
+
+ * apps/app_queue.c: Fix in an issue in the call forwarding handling
+ code that was causing crashes on every call into a queue. I'm not
+ entirely sure about the logic in this part of the code, so I want
+ to look at it some more tomorrow. However, this makes it safe and
+ keeps it from crashing. (closes issue #11486, reported by adamg,
+ patched by me)
+
+2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher at digium.com>
+
+ * main/rtp.c: At the end of a call, when we're reporting, RTCP may
+ already be partially torn down, so check for NULL dereference
+ Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
+
+2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: IMAP storage did not honor the maxmsg
+ setting in voicemail.conf, and it also had the possibility of
+ crashing if a user had more than 256 messages in their voicemail.
+ This patch kills two birds with one stone by adding maxmsg
+ support and also setting a hard limit on the number of messages
+ at 255 so that the crashes cannot happen. (closes issue #11101,
+ reported by Skavin, patched by me)
+
+2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell at digium.com>
+
+ * main/loader.c, include/asterisk/module.h: Add a new module flag
+ to indicate that a build sum is present. Modules built against
+ older Asterisk 1.4 headers will now load properly with just a
+ warning indicating that they are old and may cause problems.
+ (patch by paravoid)
+
+2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp at digium.com>
+
+ * main/udptl.c: Fix various in the udptl implementation. It could
+ return empty modem frames, have an incorrect sequence number on
+ packets, and display the wrong sequence number in the debug
+ messages. (closes issue #11228) Reported by: Cache Patches:
+ udptl-4.patch uploaded by dimas (license 88)
+
+ * channels/chan_sip.c: Add support for accepting and sending T.38
+ in the initial INVITE. (closes issue #9402) Reported by: thdei
+
+2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej at edvina.net>
+
+ * main/loader.c, include/asterisk/logger.h, main/logger.c: Make
+ sure logger is reloaded at general reload in the cli. (Discovered
+ during Asterisk training in Portugal)
+
+2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Reverting extra stuff I didn't mean to
+ commit
+
+ * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
+ did not properly handle the case where only a label was provided.
+ This was due to the fact that the answering channel did not have
+ an extension set, so ast_parseable_goto would fail. This fix
+ eliminates the call to ast_parseable_goto on the answering
+ channel since it is a wasteful call. The answering channel and
+ the calling channel are both directed to the same extension and
+ context, just different priorities, so we can just copy the
+ values from the calling channel to the answering channel and
+ increment the answering channel's priority. (closes issue #11382,
+ reported by jon, patch by me with correction by jon)
+
+2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher at digium.com>
+
+ * sounds/Makefile: Upgrade to the latest version of extra sounds
+
+2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell at digium.com>
+
+ * main/threadstorage.c: Make the lock in the threadstorage
+ debugging code untracked to avoid a deadlock on thread
+ destruction. (closes issue #11207) Reported by: ys Patches:
+ threadstorage.c.diff uploaded by ys (license 281) Also fixes an
+ open bug report: (closes issue #11446)
+
+ * main/utils.c: When DEBUG_THREADS is enabled, we only have the
+ details about who is holding a lock that we are waiting on for a
+ mutex, not rwlocks. This should fix the problem where people have
+ reported "core show locks" crashing sometimes.
+
+ * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
+ reported as happening on Mac systems. It turns out that the
+ problem was the Mac version of the ast_atomic_fetchadd_int()
+ function. The Mac atomic add function returns the _new_ value,
+ while this function is supposed to return the old value. So, the
+ crashes happened on unreferencing objects. If the reference count
+ was decreased to 1, ao2_ref() thought that it had been decreased
+ to zero, and called the destructor. However, there was still an
+ outstanding reference around. (closes issue #11176) (closes issue
+ #11289)
+
+ * include/asterisk/file.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/compiler.h: Modify file.h to maintain API
+ compatibility with earlier versions. If a recent compiler is
+ being used, then a warning will show up for any modules still
+ using the old name "private" instead of "_private". (patch
+ suggested by paravoid)
+
+ * main/pbx.c: Make some changes to some additions I made recently
+ for doing channel autoservice when looking up extensions. This
+ code was added to handle the case where a dialplan switch was in
+ use that could block for a long time. However, the way that I
+ added it, it did this for all extension lookups. However, lookups
+ in the in-memory tree of extensions should _not_ take long enough
+ to matter. So, move the autoservice stuff to be only around
+ executing a switch.
+
+2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker at digium.com>
+
+ * main/channel.c: If we fail to create a channel after allocating a
+ timing fd, we need to make sure to close it. Issue 11454, patch
+ by eliel.
+
+2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Fix build issue on the build cluster.
+
+2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/compat.h: Solaris requires the inclusion of
+ sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
+ snuffy,tilghman (Closes issue #11430)
+
+ * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
+ NULL deref could result Reported by: xrg Patch by: tilghman
+ (Closes issue #11387)
+
+2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c, main/channel.c, main/global_datastores.c
+ (added), channels/chan_local.c, main/Makefile,
+ include/asterisk/channel.h, include/asterisk/global_datastores.h
+ (added), apps/app_queue.c: A big one... This is the merge of the
+ forward-loop branch. The main change here is that call-forwards
+ can no longer loop. This is accomplished by creating a datastore
+ on the calling channel which has a linked list of all devices
+ dialed. If a forward happens, then the local channel which is
+ created inherits the datastore. If, through this progression of
+ forwards and datastore inheritance, a device is attempted to be
+ dialed a second time, it will simply be skipped and a warning
+ message will be printed to the CLI. After the dialing has been
+ completed, the datastore is detached from the channel and
+ destroyed. This change also introduces some side effects to the
+ code which I shall enumerate here: 1. Datastore inheritance has
+ been backported from trunk into 1.4 2. A large chunk of code has
+ been removed from app_dial. This chunk is the section of code
+ which handles the call forward case after the channel has been
+ requested but before it has been called. This was removed because
+ call-forwarding still works fine without it, it makes the code
+ less error-prone should it need changing, and it made this set of
+ changes much less painful to just have the forwarding handled in
+ one place in each module. 3. Two new files, global_datastores.h
+ and .c have been added. These are necessary since the datastore
+ which is attached to the channel may be created and attached in
+ either app_dial or app_queue, so they need a common place to find
+ the datastore info. This approach was taken in case similar
+ datastores are needed in the future, there will be a common place
+ to add them.
+
+2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker at digium.com>
+
+ * apps/app_meetme.c: Make sure we always close the conference fd if
+ we have an open one. Issue 11383, reported by markmhy, patch by
+ eliel.
+
+2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_mgcp.c: Changing some bad logic when calculating
+ the interdigit timeout. (closes issue #11402, reported and
+ patched by eferro)
+
+2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker at digium.com>
+
+ * res/res_features.c: Fix crash in ParkAndAnnounce application.
+ Issue #11436, reported by lytledd, patch by eliel.
+
+2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Do not create a smoother for G723.1 frames, they need
+ to be left alone to their native 20/24 byte size.
+
+ * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
+ the indication currently playing on a channel when a masquerade
+ operation happens. (issue #BE-88)
+
+2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker at digium.com>
+
+ * channels/chan_iax2.c: Only log debug messages if debug is
+ enabled. Closes issue #11416, patch by casper.
+
+2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c: The other day when I went through making
+ changes as a result of the ao2_link() change, I added some code
+ to set pointers to NULL after they were unreferenced. This
+ pointed out that in this place, the object was unreferenced
+ before the code was done using it. So, move the unref down a
+ little bit. (crash reported by jmls on IRC)
+
+2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher at digium.com>
+
+ * main/autoservice.c: Clarify the return value on autoservice.
+ Specifically, if you started autoservice and autoservice was
+ already on, it would erroneously return an error. Reported by:
+ adiemus Patch by: dimas (Closes issue #11433)
+
+2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
+ apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
+ ao2_link(). Previously, in inherited a reference. Now, it
+ automatically increases the reference count to reflect the
+ reference that is now held by the container. This was done to be
+ more consistent with ao2_unlink(), which automatically releases
+ the reference held by the container. It also makes it so it is no
+ longer possible for a pointer to be invalid after ao2_link()
+ returns.
+
+ * include/asterisk/astobj2.h: Add some notes on the behavior of
+ ao2_unlink() after a discussion with Tilghman
+
+2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix locking issues under one legged replaces
+ scenarios. (closes issue #11420) Reported by: irroot Patches:
+ chan_sip_oneleg.patch uploaded by irroot (license 52)
+
+2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_mgcp.c: Clear the DTMF buffer if the call times
+ out. (closes issue #11418, reported and patched by eferro)
+
+2007-11-29 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.15 released.
+
+2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
+ we use thread-safe escaping (Fixes AST-2007-026)
+
+2007-11-29 19:38 +0000 [r90163] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: This patch handles the case where a queue
+ member with a negative penalty is added via the manager. If a
+ negative value is submitted for a member penalty, we set it to 0.
+ (closes issue #11411, reported and patched by Laureano)
+
+2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c: Properly escape input buffers (Fixes
+ AST-2007-025)
+
+ * formats/format_g726.c, include/asterisk/file.h,
+ formats/format_wav.c, formats/format_pcm.c,
+ formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
+ formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
+ as a field name in a header file messes with C++ projects
+ Reported by: chewbacca Patch by: casper (Closes issue #11401)
+
+ * sounds/Makefile: Upgrade the core sounds release version
+
+2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant <russell at digium.com>
+
+ * funcs/func_callerid.c: fix some formatting i accidentally changed
+
+ * funcs/func_callerid.c, main/channel.c,
+ include/asterisk/channel.h: This set of changes is to make some
+ callerID handling thread-safe. The ast_set_callerid() function
+ needed to lock the channel. Also, the handlers for the CALLERID()
+ dialplan function needed to lock the channel when reading or
+ writing callerid values directly on the channel structure.
+
+ * include/asterisk/file.h, main/file.c: Merge a change from
+ team/russell/chan_refcount ... This makes ast_stopstream()
+ thread-safe.
+
+2007-11-28 22:59 +0000 [r90101] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
+ Reported by: eliel Patches: load_realtime.patch uploaded by eliel
+ (license 64)
+
+2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configs/users.conf.sample, main/manager.c: it is impossible to
+ set permissions for manager accounts created by users.conf
+ (reported internally, patched by me)
+
+2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: Removing some seemingly pointless code. This sets a
+ channel variable for every priority executed in the dialplan if
+ you have debug set to anything non-zero. This seems pointless due
+ to the fact that these channel variables are not referenced
+ anywhere else in the code and their names are esoteric enough
+ that they would not be practical to reference in the dialplan.
+ Plus the fact that this behavior isn't documented anywhere means
+ that the change is not likely to cause any disruption. If
+ anything, this may actually cause a slight performance increase
+ if running with debug on. The motivating influence for this code
+ change is the eventwhencalled option for queues. If set to vars,
+ all channel variables will be output to the manager. These
+ unnecessary channel variables make the output a lot more
+ difficult to deal with.
+
+ * apps/app_voicemail.c: Recording greetings when using IMAP storage
+ was causing zero-length files to be stored. Since greetings are
+ not retrieved from IMAP anyway, it is pointless to attempt
+ storing them there. (closes issue #11359, reported by spditner,
+ patched by me)
+
+2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, include/asterisk/pbx.h: - update documentation for
+ some of the goto functions to note that they handle locking the
+ channel as needed - update ast_explicit_goto() to lock the
+ channel as needed
+
+ * main/autoservice.c: Don't do frame processing if ast_read()
+ returned NULL.
+
+ * apps/app_queue.c: Instead of depending on the return value of
+ ast_true(), explicitly set the eventwhencalled variable to 1.
+
+ * main/pbx.c: Don't start/stop autoservice in
+ pbx_extension_helper() unless a channel exists
+
+2007-11-27 23:10 +0000 [r89837] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Two changes with regards to the
+ 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
+ unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
+ did exactly the same thing. Thus the sign change of the ast_true
+ call. 2) The vars2manager function overwrote a \n for every
+ channel variable it parsed, resulting in bizarre output for the
+ channel variables. This patch remedies this. (related to issue
+ #11385, however I'm not sure if this will actually be enough to
+ close it)
+
+2007-11-27 21:45 +0000 [r89790] Russell Bryant <russell at digium.com>
+
+ * main/autoservice.c, main/pbx.c: Merge changes from
+ team/russell/autoservice_1.4 This set of changes fixes an issue
+ that was reported to me on IRC yesterday. The user, d1mas, was
+ using chan_zap for incoming calls and was having DTMF recognition
+ issues in some situations. Specifically, he noticed that the
+ problem occurred when using DISA or WaitExten. He also noticed
+ that when using Read, the problem did not occur. His system also
+ used DUNDi for dialplan lookups. So, he theorized that if the
+ DUNDi lookups blocked for some period of time, that audio from
+ the zap channel could get lost. If the audio got lost, then it
+ wouldn't be run through the DTMF detector, and digits could get
+ lost. He was correct, and the following set of changes fixes the
+ problem. However, the changes go a little bit further than what
+ was necessary to fix this exact problem. 1) I updated
+ pbx_extension_helper() to autoservice the associated channel to
+ handle cases where extension lookups may take a long time. This
+ would normally be a dialplan switch that does some lookup over
+ the network, such as the DUNDi or IAX2 switches. This ensures
+ that even while a DUNDi lookup is blocking, the channel will be
+ continuously serviced. 2) I made a change to the autoservice
+ code. This is actually something that has bothered me for a long
+ time. When a channel is in autoservice, _all_ frames get thrown
+ away. However, some frames really shouldn't be thrown away. The
+ most notable examples are signalling (CONTROL) frames, and DTMF.
+ So, this patch queues up important frames while a channel is in
+ autoservice. When autoservice is stopped on the channel, the
+ queued up frames get stuck back on the channel so that they can
+ get processed instead of thrown away. 3) I made another change to
+ the autoservice code to handle the case where autoservice is
+ started on channels recursively. Previously, you could call
+ ast_autoservice_start() multiple times on a channel, and it would
+ stop the first time ast_autoservice_stop() gets called. Now, it
+ will ensure that autoservice doesn't actually stop until the
+ final call to ast_autoservice_stop().
+
+2007-11-27 20:22 +0000 [r89727] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_config_pgsql.c: Changing some calls from free() to
+ ast_free() since they were allocated with ast_calloc(). (closes
+ issue #11390, reported and patched by Laureano)
+
+2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/app.c: on second thought... revert all the other changes
+ i've made in app options parsing leaving only one: if an empty
+ argument is supplied for an option, set that argument pointer to
+ point to an empty string rather than NULL, so that the
+ application can do normal checks on it without worrying about it
+ being NULL
+
+ * main/app.c: generate a warning when an application option that
+ requires an argument is ignored due to lack of an argument
+
+2007-11-27 16:12 +0000 [r89634] Russell Bryant <russell at digium.com>
+
+ * configs/voicemail.conf.sample: Add a note to the sample voicemail
+ config noting that when using IMAP storage, only the first format
+ specified will be attached to the message.
+
+2007-11-27 15:38 +0000 [r89631] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_env.c: Default result of STAT should be "0" not "".
+ Reported via the -users mailing list, fixed by me.
+
+2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson <oej at edvina.net>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
+ get a codec offer using a well-known payload type, but using it
+ for another codec that we don't know, Asterisk did not remove
+ that codec from the list. With this patch, we remove the codec
+ from audio and video rtp objects and deny it ever existed. Thanks
+ to lasse for testing. (closes issue #11376) Reported by: lasse
+ Patches: bug11376.txt uploaded by oej (license 306) Tested by:
+ lasse
+
+ * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
+ #11304) Reported by: pj
+
+2007-11-27 06:24 +0000 [r89622] Steve Murphy <murf at digium.com>
+
+ * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
+ include/asterisk/cdr.h: closes issue #11379; OK, this is an
+ attempt to make both sides happy. To the cdr.conf file, I added
+ the option 'unanswered', which defaults to 'no'. In this mode,
+ you will see a cdr for a call, whether it was answered or not.
+ The disposition will be NO ANSWER or ANSWERED, as appropriate.
+ The src is as you'd expect, the destination channel will be one
+ of the channels from the Dial() call, usually the last in the
+ list if more than one chan was specified. With unanswered set to
+ 'yes', you will still see this cdr entry in both cases. But in
+ the case where the dial timed out, you will also see a cdr for
+ each line attempted, marked NO ANSWER, with no destination
+ channel name. The new option defaults to 'no', so you don't see
+ the pesky extra cdr's by default, and you will not see the
+ irritating 'not posted' messages.
+
+2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_playback.c: After issuing a "say load new", if a caller
+ hangs up during the middle of playback of a number, app_playback
+ will continue to try to play the remaining files. With this
+ change, no more files will be played back upon hangup. (closes
+ issue #11345, reported and patched by IgorG)
+
+ * apps/app_playback.c: After issuing a "say load new" tons of
+ warning messages are printed out to the CLI every time do_say in
+ app_playback is called. Removing these warnings
+
+2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp <jcolp at digium.com>
+
+ * main/dial.c: Fix issues with async dialing with an application
+ executing. The application has to be terminated and control
+ returned to the thread before hanging things up. (issue #BE-252)
+
+ * res/res_features.c: Add module counting removal for error
+ conditions. (closes issue #11333) Reported by: Laureano Patches:
+ res_features_v2.c.patch uploaded by Laureano (license 265)
+
+2007-11-26 17:41 +0000 [r89594] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Add channel locking to a function that needed to be
+ doing it. This is just a little something I noticed while working
+ on a completely unrelated issue.
+
+2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp <jcolp at digium.com>
[... 13526 lines stripped ...]
More information about the asterisk-commits
mailing list