[asterisk-commits] russell: tag 1.4.15 r90168 - in /tags/1.4.15: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Nov 29 14:50:59 CST 2007
Author: russell
Date: Thu Nov 29 14:50:58 2007
New Revision: 90168
URL: http://svn.digium.com/view/asterisk?view=rev&rev=90168
Log:
importing files for 1.4.15 release
Added:
tags/1.4.15/.lastclean (with props)
tags/1.4.15/.version (with props)
tags/1.4.15/ChangeLog (with props)
Added: tags/1.4.15/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.15/.lastclean?view=auto&rev=90168
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--- tags/1.4.15/ChangeLog (added)
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+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
+
+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
+
+ * 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>
+
+ * pbx/pbx_config.c: Use ast_free to free memory, or else we shall
+ implode if MALLOC_DEBUG is enabled. (closes issue #11347)
+ Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
+ (license 281)
+
+ * apps/app_mixmonitor.c: Close the audio file before sending it to
+ the post processing application. (closes issue #11357) Reported
+ by: reformed Patches: mixmonitor.patch uploaded by reformed
+ (license 330)
+
+2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/app.c: when parsing application options that take arguments,
+ don't indicate that the option was supplied unless a
+ non-zero-length argument was found for it
+
+2007-11-26 15:48 +0000 [r89580] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Revert vmu->email back to an empty string
+ if it was empty when imap_store_file was called. This prevents
+ sending a duplicate e-mail. (closes issue #11204, reported by
+ spditner, patched by me)
+
+2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: If channel allocation fails because the alert
+ pipe could not be created also free the scheduler context.
+ (closes issue #11355) Reported by: eliel Patches:
+ main.channel.c.patch uploaded by eliel (license 64)
+
+ * apps/app_meetme.c: When unloading app_meetme destroy any auto
+ created contexts created by SLA. (closes issue #11367) Reported
+ by: eliel
+
+2007-11-25 17:17 +0000 [r89559] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
+ attempted to use the ESCAPE clause to set the escape delimiter to
+ a backslash. Unfortunately, this does not universally work on all
+ databases, since on databases which natively use the backslash as
+ a delimiter, the backslash itself needs to be delimited, but on
+ other databases that have no delimiter, backslashing the
+ backslash causes an error. So the only solution that I can come
+ up with is to create an option in res_odbc that explicitly
+ specifies whether or not backslash is a native delimiter. If it
+ is, we use it natively; if not, we use the ESCAPE clause to make
+ it one. Reported by: elguero Patch by: tilghman (Closes issue
+ #11364)
+
+2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_adsi.c: Free some frames that would otherwise leak on
+ error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
+ issue #11351)
+
+ * apps/app_voicemail.c, main/app.c: Currently, zero-length
+ voicemail messages cause a hangup in VoicemailMain. This change
+ fixes the problem, with a multi-faceted approach. First, we do
+ our best to avoid these messages from being created in the first
+ place, and second, if that fails, we detect when the voicemail
+ message is zero-length and avoid exiting at that point. Reported
+ by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)
+
+ * main/manager.c: Up until this point, the XML output of the
+ manager has been technically invalid, due to the repetition of
+ certain parameters in a single event. This caused various issues
+ for XML parsers, some of which refused to parse at all, given the
+ invalidity of the rendered XML. So this commit fixes the XML
+ output, ensuring that each entity parameter has a unique name,
+ thus ensuring valid XML. Reported by: msetim Patch by: tilghman
+ (Closes issue #10220)
+
+ * res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
+ not just 2nd-Nth parameters. Reported by: apsaras Patch by:
+ tilghman (Closes issue #11353)
+
+2007-11-22 17:29 +0000 [r89527] Russell Bryant <russell at digium.com>
+
+ * configs/agents.conf.sample: mvanbaak pointed out a spelling error
+ in this sample configuration file. While I was at it, I went
+ ahead and tweaked it a little bit more.
+
+2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix a small error I made in my previous commit
+
+ * apps/app_queue.c: Changing an inaccurate debug message to be less
+ inaccurate. Under the circumstances, this message would always
+ report that there were 0 members available, even though that may
+ not be true.
+
+2007-11-21 18:59 +0000 [r89491] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: If a channel gets masqueraded in the middle
+ of a park, don't play the announcement to the masqueraded
+ channel, and dial back to the original channel on timeout.
+
+2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/module.h: re-doxygen some comments
+
+ * main/loader.c, include/asterisk/module.h,
+ build_tools/make_buildopts_h: bring back compile-option checking
+ when loading modules, only this time use a string-based storage
+ and comparison mechanism because it is easier to support on other
+ platforms
+
+2007-11-20 17:50 +0000 [r89457] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: According to comments in main/pbx.c, it is essential
+ that if we are going to lock the conlock as well as the hints
+ lock, it must be locked in that respective order. In order to
+ prevent a potential deadlock, we need to lock the conlock prior
+ to locking the hints lock in ast_hint_state_changed (see the call
+ stack example on issue #11323 for how this can happen). (closes
+ issue #11323, reported by eelcob, suggestion for patch by eelcob,
+ patch by me)
+
+2007-11-20 15:22 +0000 [r89450] Steve Murphy <murf at digium.com>
+
+ * doc/queues-with-callback-members.txt: closes issue #11324; break
+ statements missing in switch cases.
+
+2007-11-20 13:40 +0000 [r89445] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: added RR patch from iroot #10908, thanks.
+
+2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Print out the correct filename
+ (features.conf) in the log message when parkpos options are
+ incorrect. (closes issue #11295) Reported by: Laureano Patches:
+ res_features.c.patch uploaded by Laureano (license 265)
+
+ * doc/localchannel.txt: Clarify documentation a bit, include that a
+ frame has to pass through the core in order for the Local channel
+ optimization to happen. (closes issue #11246) Reported by: jon
+
+2007-11-16 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.14 released.
+
+2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell at digium.com>
+
+ * main/loader.c, include/asterisk/module.h,
+ build_tools/make_buildopts_h: Temporarily revert revision 89325,
+ which added md5 magic for keeping track of what build options
+ were used. We agreed that we should remove this before making a
+ 1.4 release, and then we can put it back in. Then, we can take a
+ month or so to play around with it to get it how we want it.
+
+2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/loader.c, include/asterisk/module.h,
+ build_tools/make_buildopts_h: To help combat problems where
+ people build external modules (asterisk-addons or others) and
+ then change the build options of the Asterisk build in a way that
+ makes the incompatible without warning, this commit introduces an
+ MD5 signature of the important build-time options and includes
+ that signature into modules when they are built. When the loader
+ loads one of these modules and notices the problem, it will emit
+ a warning to console and refuse to initialize the module, as
+ doing so could cause the system to be unstable or even crash. If
+ you upgrade to this version of Asterisk, you must rebuild *all*
+ of your modules that came from other sources before trying to run
+ this version. If you are using Digium's G.729 binary codec
+ module, you will need v33 or newer.
+
+2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make realtime queues accessible from the
+ QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
+ patched by atis, with small modifications from me)
+
+2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Start Asterisk in Debian at a more reasonable time
+ (since zaptel is at level 20)
+
+ * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
+ by valgrind
+
+ * channels/chan_iax2.c: Yet another memory corruption issue.
+ Reported by: atis Patch by: tilghman Fixes issue #10923
+
+2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Update the SLAStation application to account
+ for the case where the SLA thread has a call out to the station,
+ but the user has pressed a line button to answer the call instead
+ of picking up the handset. If they do, the phone sends out a new
+ INVITE. So, the SLAStation app must check to see if it is picking
+ up a ringing trunk, and ensure that the other stations stop
+ ringing. (reported internally, patched by me, tested by mogorman)
+
+2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Undoing previous commit since I realize it was
+ wrong
+
+ * main/manager.c: Adding a missing mutex unlock. (closes issue
+ 11256, reported and patched by ys)
+
+2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't send re-invites during pending INVITE
+ transactions. Patch by one47 - thanks! Closes issue #9305
+
+ * channels/chan_sip.c: Improve support for multipart messages. Code
+ by gasparz, changes by me (mostly formatting). Thanks, gasparz!
+ Closes issue #10947
+
+2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher at digium.com>
+
+ * main/app.c: When a recording ends with '#', we are improperly
+ trimming an extra 200ms from the recording. Reported by: sim
+ Patch by: tilghman Closes issue #11247
+
+2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp at digium.com>
+
+ * main/srv.c: Return the proper value when the srv_callback
+ function executes properly. (closes issue #11240) Reported by:
+ jtodd
+
+2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
+ systems which require a third arg to open() when using O_CREAT.
+ Issue 11238, reported by puzzled.
+
+ * res/res_features.c: Revert change from revision 67064. It is
+ documented behavior that if a parking extension already exists
+ while using PARKINGEXTEN, dialplan execution will continue. If
+ blind transferring to a Park with PARKINGEXTEN, you must keep
+ this in mind, and handle the failure yourself. Issue 11237,
+ reported by jon.
+
+2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: If we set a value for qualify, we should
+ actually pay attention to it, instead of overriding the value
+
+2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_mixmonitor.c: Reverting commit made in revision 89205
+ since it is unnecessary. Thanks to Kevin for pointing this out
+
+2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher at digium.com>
+
+ * main/utils.c: Debugging is running into the 16-lock limit.
+ Increase to avoid. (This define is only effective when debugging
+ is turned on, so there's no effect for most installations.)
+
+2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
+ only 1 argument is given, then the args.options and
+ args.post_process strings are uninitialized and could contain
+ garbage. This change handles this situation properly by only
+ using arguments that we have parsed.
+
+2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker at digium.com>
+
+ * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev
+
+2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher at digium.com>
+
+ * main/config.c: If two config writes collide, file corruption
+ could result. Use a mkstemp() file, instead. Reported by:
+ paravoid Patch by: tilghman Closes issue #10781
+
+ * main/channel.c, channels/chan_sip.c: Fix two cases of memory
+ corruption caused by background threads. Reported by: atis Patch
+ by: tilghman Fixes issue #10923
+
+2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
+ no number was dialed and overlapdial is set, we wait for the ISDN
+ timeout instead of starting our own timer. added a comment for
+ the misdn.conf.sample for the overlapdial config option.
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
+ channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
+ restart all interfaces Restart_Indicator, to automatically send a
+ RESTART after the L2 of a PTP Port comes up. Also fixed some
+ places where we have send a RELEASE without need for it.
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
+ state/event issue with overlapdial=yes when no extension matched.
+ removed the general sending of a RELEASE_COMPLETE when we receive
+ a RELEASE, this is done by mISDNuser/mISDN. This makes it
+ possible to use asterisk-1.4 with mISDN trunk, but requires users
+ of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
+ (when using the NT mode at all)
+
+ * channels/misdn/isdn_lib.c: fixed the support for CW and therefore
+ for the reject_cause option.
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
+ aded ntkeepcalls option, to avoid droÃpping calls when the L2
+ goes down on a PTP link. There are some pbx which do turn off the
+ L1 for a very short while and restart it immediately. normally
+ T310 should be started and after 10 seconds or so the calls
+ should be dropped, this is a simple fix wihtout this timer.
+
+2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker at digium.com>
+
+ * main/say.c: Properly say the seconds here.. Issue 11203, fix
+ described by vma.
+
+2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Rework of the commit I made yesterday to use
+ the already built-in ast_uri_decode function as opposed to my
+ home-rolled one. Also added comments. Thanks to oej for pointing
+ me in the right direction
+
+2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker at digium.com>
+
+ * configs/res_odbc.conf.sample: Avoid warnings on load when using
+ sample configuration files. Issue 11195, patch by eliel.
+
+2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: I made this same adjustment in trunk to fix
+ a bug, and it makes sense to do it in 1.4 as well. If an
+ imapfolder is specified in voicemail.conf, don't ever explicitly
+ connect to INBOX since it may not exist.
+
+2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/srv.c: fix a glaring bug in the new SRV record handling that
+ would cause incorrect weight sorting
+
+2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/valgrind.txt: Typo
+
+2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Do not add a sip: to the beginning of the To
+ URI unless needed. (closes issue #10756) Reported by: goestelecom
+
+ * channels/chan_sip.c: Improve the devicestate logic for multiple
+ devices. If any are available then the extension is considered
+ available. (closes issue #10164) Reported by: nic_bellamy
+ Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
+ (license 299)
+
+ * channels/chan_sip.c: Add support for allowing one outgoing
+ transaction. This means if a response comes back out of order
+ chan_sip will still handle it. I dream of a chan_sip with real
+ transaction support. (closes issue #10946) Reported by: flefoll
+ (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
+ Reported by: atca_pres
+
+ * channels/chan_sip.c: If callerid is configured in sip.conf use
+ that for checking the presence of an extension in the dialplan.
+ (closes issue #11185) Reported by: spditner
+
+2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: The member refcount must be incremented, to
+ avoid using it after deallocation. A huge thanks go to lvl- for
+ patiently providing the necessary valgrind output that was
+ necessary to finding this problem of memory corruption. Reported
+ by: lvl- Patch by: tilghman Closes issue #11174
+
+2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: This patch makes it possible for SIP phones
+ to dial extensions defined with '#' characters in extensions.conf
+ AND maintain their escaped characters when forming URI's (closes
+ issue #10681, reported by cahen, patched by me, code review by
+ file)
+
+2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf at digium.com>
+
+ * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
+ 10578, I just ran 1.4 thru valgrind; some of the config leakage
+ I've already fixed, but it doesn't hurt to double check. I found
+ and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
+ tho.
+
+2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Fixing a segfault in the manager "core show
+ channels concise" command. (closes issue #11183, reported by arnd
+ and patched by ys)
+
+2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.ael.sample: Suppress AEL warnings on load.
+ Reported by: eliel Patch by: eliel Closes issue #11178
+
+2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Fix init_classes() so that classes that
+ actually do have files loaded aren't treated as empty, and
+ immediately destroyed ...
+
+2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker at digium.com>
+
+ * codecs/codec_zap.c: Correctly set the total number of channels
+ from a zaptel transcoder board. SPD-49, patch by Matthew
+ Nicholson.
+
+2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/lock.h: We went to the trouble of creating a
+ method of tracking failed trylocks, then never turned it on
+ (oops).
+
+2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej at edvina.net>
+
+ * main/tdd.c: Bug fixes to tdd support in zaptel.
+
+2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: If someone were to delete the files used
+ by an existing MOH class, and then issue a reload, further use of
+ that class could result in a crash due to dividing by zero. This
+ set of changes fixes up some places to prevent this from
+ happening. (closes issue #10948) Reported by: jcomellas Patches:
+ res_musiconhold_division_by_zero.patch uploaded by jcomellas
+ (license 282) Additional changes added by me.
+
+2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf at digium.com>
+
+ * main/config.c: closes issue #8786 - where the [catname](!) and
+ [catname](othercat1,othercat2,...) notation gets dropped across a
+ ConfigUpdate (or any other thing that would cause a config file
+ to be written). While I was at it, I also cleaned up some of the
+ destroy routines to free up comments, which was not being done.
+ Made sure the new struct I introduced is also cleaned up properly
+ at destruction time. My code handles multiple template
+ inclusions. Many thanks to ssokol for his patch, which, while not
+ literally used in the final merge, served as a foundation for the
+ fix.
+
+2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Make it so that if a peer is determined to
+ be unreachable using qualify their devicestate will report back
+ unavailable. (closes issue #11006) Reported by: pj
+
+ * channels/chan_zap.c: Fix improbable but possible memory leaks in
+ chan_zap. (closes issue #11166) Reported by: eliel Patches:
+ chan_zap.c.patch uploaded by eliel (license 64)
+
+2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: Remove some checks to see if locks are
+ initialized from the non-DEBUG_THREADS versions of the lock
+ routines. These are incorrect for a number of reasons: - It
+ breaks the build on mac. - If there is a problem with locks not
+ getting initialized, then the proper fix is to find that place
+ and fix the code so that it does get initialized. - If additional
+ debug code is needed to help find the problem areas, then this
+ type of things should _only_ be put in the DEBUG_THREADS
+ wrappers.
+
+2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/srv.h: update comment to match the state of the
+ code
+
+2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Reworked deadlock avoidance in __ast_read.
+ Restored audio to callback agents. (closes issue #11071, reported
+ by callguy, patched by me, tested by callguy and Ted Brown)
+
+2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
+ to channel variables, I went looking around at the ways that
+ channel variables are handled. In general, they were not handled
+ in a thread-safe way. The channel _must_ be locked when reading
+ or writing from/to the channel variable list. What I have done to
+ improve this situation is to make pbx_builtin_setvar_helper() and
+ friends lock the channel when doing their thing. Asterisk API
+ calls almost all lock the channel for you as necessary, but this
+ family of functions did not. (closes issue #10923, reported by
+ atis) (closes issue #11159, reported by 850t)
+
+ * channels/chan_sip.c: When traversing the list of channel
+ variables here in transmit_invite(), the asterisk channel must be
+ locked, as this data may change at any time. (I have seen
+ numerous reports of crashes related to the handling of channel
+ variables. There are a couple of issues on the bug tracker
+ related to it, but it has also been noted on IRC and mailing
+ lists. So, I am finding and fixing some places where channel
+ variables are handled improperly.)
+
+ * channels/chan_sip.c: Fix up some indentation.
+
+ * main/srv.c, include/asterisk/srv.h: Merge changes from
+ asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
+ record support in Asterisk was broken. There was no guarantee on
+ what record Asterisk would choose to actually use. This set of
+ changes improves the situation by ensuring that Asterisk will
+ choose the highest priority record.
+
+ * main/channel.c: Merge the last bit of changes from
+ asterisk/team/russell/readq-1.4 The issue here is that the
+ channel frame readq handling got broken when the code was
+ converted to use the linked list macros. It caused corruption of
+ the list head and tail pointers. So, I fixed up the usage of the
+ linked list macros and in passing, simplified the code. I also
+ documented what the code is doing, as it was a bit difficult to
+ figure out at first. This bug showed itself with crashes showing
+ messed up head/tail pointers for the readq. However, there are a
+ couple of crashes that aren't quite as obvious, but I think may
+ be related. So, if your bug gets closed by this commit, but you
+ still have a problem, please reopen or create a new bug report.
+ (closes issue #10936) (closes issue #10595) (closes issue #10368)
+ (closes issue #11084) (closes issue #10040) (closes issue #10840)
+
+2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: If a SIP channel is put on hold multiple
+ times do not keep incrementing the onHold value. (closes issue
+ #11085) Reported by: francesco_r Tested by: blitzrage (closes
+ issue #10474) Reported by: acennami
+
+2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Fix up datastore handling in ast_do_masquerade().
+ The code is intended to move any channel datastores from the old
+ channel to the new one. However, it did not use the linked list
+ macros properly to accomplish the task. The existing code would
+ only work if there was only a single datastore on the old
+ channel.
+
+2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: Make sure we destroy the config structure on
+ configuration failure. Issue 11163, patch by eliel.
+
+2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c: Don't check used pooled connections for
+ connection status, as it will cause issues for prepared queries.
+ Reported by: Nick Gorham (via -dev list) Patch by: tilghman
+
+2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo at icir.org>
+
+ * include/asterisk/stringfields.h, main/channel.c,
+ apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
+ Rename ast_string_field_free_pool to
+ ast_string_field_free_memory, and ast_string_field_free_all to
+ ast_string_field_reset_all to avoid misuse (due to too similar
+ names and an error in documentation). Fix two related memory
+ leaks in app_meetme. No need to merge to trunk, different fix
+ already applied there. Not applicable to 1.2
+
+2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Make subscribecontext behave as advertised.
+ It will now look for the presence of a hint in the given context
+ (be it subscribecontext or context). (closes issue #10702)
+ Reported by: slavon
+
+ * channels/chan_sip.c: If an INFO request within a dialog is
+ received with a content length of 0 simply send back a 200 OK. It
+ is valid to do this and the remote side is probably using it to
[... 12954 lines stripped ...]
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