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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-29 21:21:18 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-29 21:21:18 +0000
commit8a38926d0279dd39625452d2f89bb96f8bc8b3fa (patch)
tree579a560cc8d0a0a0407249d0007f0d1e9bee5571
parent1e0a801a8deac8b9f3d86fed10a7c4b7addaee05 (diff)
parent108402b0085e4e7fff2b469aee2e7b0ad44efff8 (diff)
Creating tag for the release of asterisk-1.4.15
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.15@90172 f38db490-d61c-443f-a65b-d21fe96a405b
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-rw-r--r--.version1
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-1.4.15
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-2007-11-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.15 released.
-
-2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@beronet.com>
-
- * channels/chan_misdn.c: added RR patch from iroot #10908, thanks.
-
-2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp@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@digium.com>
-
- * Asterisk 1.4.14 released.
-
-2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
-
- * doc/valgrind.txt: Typo
-
-2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp@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@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@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@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@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@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@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@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@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@edvina.net>
-
- * main/tdd.c: Bug fixes to tdd support in zaptel.
-
-2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell@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@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@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@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@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@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@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@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@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@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@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@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@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
- make sure the signalling is still alive. (closes issue #5747)
- Reported by: chandi Patches: infofix-81430-1.patch uploaded by
- IgorG (license 20)
-
-2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker@digium.com>
-
- * main/say.c: We need to make sure to specify a language to
- ast_fileexists, otherwise it may fail for anything besides en
- Issue 11147, fix discovered by both citats and myself
- (independently), with input from Corydon76
-
-2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
- Patch by: ys Closes issue #11143
-
- * doc/valgrind.txt (added): Add some notes on using valgrind
-
-2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Make sure we set the poll fds to NULL after
- free()ing it. Part of issue 11017, patch by tzafrir.
-
-2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Fix up commit for my Zap channel with spies in
- Meetme fix. (thanks Tony Mountifield!)
-
- * apps/app_meetme.c: If a Zap channel contains a spy or a spy is
- added take it out of the conference in kernel space and make it
- go through Asterisk so the spy gets audio from both sides.
- (closes issue #10060) Reported by: mparker
-
-2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Make sure we free some allocated memory before
- returning. Issue 11131, patch by eliel.
-
- * channels/chan_gtalk.c: Don't try to allocate memory that we're
- just going to re-allocate later anyways. Issue 11130, patch by
- eliel.
-
-2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Create samples for ALL of the available options in
- asterisk.conf
-
-2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
- save' cli command saves a file where the semicolon is not
- escaped. Fixed this; User also wanted comments to be preserved
- across dialplan save, but this is impossible at this point in
- time, because comments are not stored in the dialplan. They are
- 'compiled' out of extensions.conf. The only way to preserve those
- comments is to use the config file reader/writer that the GUI
- uses to allow online user edits. extensions.conf is first and
- foremost, a config file, and is read in by the normal config-file
- reading routines. Then, it is processed into a dialplan
- (context/exten structs).
-
- * pbx/pbx_ael.c: Included some verbage in the check_includes func,
- to inform the user that included contexts that have no match in
- the AEL, might be OK, as AEL cannot check in the extensions.conf
- or the in-memory contexts, as they may not be there at the time
- of the check.
-
-2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
- Reported by: ys Fixed by: ys Closes issue #11116
-
-2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Merge the changes from
- team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
- race condition related to the handling of POKEing peers.
- Essentially, a reference to a peer is held by the scheduler when
- there are pending callbacks, but the reference count didn't
- reflect it. So, it was possible for a peer to hit a reference
- count of zero and have its destructor begin to be called at the
- same time that the scheduler thread ran a POKE related callback.
- If that happened, a crash would likely occur. (closes issue
- #11082, closes issue #11094)
-
-2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker@digium.com>
-
- * channels/Makefile: Only try to clean out h323/ if the
- h323/Makefile exists.
-
-2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Add two more checks before printing out a
- warning message about bridging. If either channel has hungup of
- course the bridge will have failed. (closes issue #10009)
- Reported by: dimas
-
-2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker@digium.com>
-
- * main/editline/np/vis.c: Fix build of editline on Solaris. Issue
- 11113, patch by snuffy.
-
-2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp@digium.com>
-
- * apps/app_followme.c: Return 1.4 to a state where it builds.
- Changing the arguments to a function and not changing where they
- are used is bad, mmmk?
-
-2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix issue where the recorded name wasn't
- getting removed correctly. (closes issue #11115) Reported by:
- davevg Patches: followme-v3.diff
-
-2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/gsm: missed one directory
-
- * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
- codecs/lpc10, main/db1-ast, main/editline, main,
- codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
- codecs/lpc10/Makefile, utils, codecs, agi,
- main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
- clean up (and ignore) assembler and preprocessor intermediate
- files if any are created during the build
-
- * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
- already there (used when debugging preprocessor issues) because
- the compiler will whine about each compile command
-
-2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a completely unnecessary quota
- check from IMAP code.
-
-2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Add some more details to
- the output of "core show locks". When a thread is waiting for a
- lock, this will now show the details about who currently has it
- locked. (inspired by issue #11100)
-
- * main/astmm.c: Remove a lock that doesn't make any sense. The
- regions lock needs to be held when traversing the list of
- allocated chunks so that they can be printed out to the CLI.
- (Thanks to eliel on #asterisk-dev for pointing this out!)
-
-2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where if both sides of the dialog
- cancelled the dialog at the same time chan_sip could kepe
- retransmitting a response for no reason. (closes issue #9566)
- Reported by: atca_pres Patches: bug9566.patch uploaded by oej
-
-2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker@digium.com>
-
- * funcs/func_realtime.c, funcs/func_cut.c: Allow some function
- modules to compile under dev mode. Issue 11104, patch by andrew.
-
-2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Fix issue with ast_unescape_semicolon going into an
- endless loop. (closes issue #10550) Reported by: ramonpeek
- Patches: unescape-85177-1.patch uploaded by IgorG (license 20)
-
-2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
- funcs/func_cut.c: Add autoservice to several more functions which
- might delay in their responses. Also, make sure that func_odbc
- functions have a channel on which to set variables. Reported by
- russell Fixed by tilghman Closes issue #11099
-
-2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
- utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
- pbx/ael/ael.flex: closes issue #11086 where a user complains that
- references to following contexts report a problem; The problem
- was REALLy that he was referring to empty contexts, which were
- being ignored. Reporter stated that empty contexts should be OK.
- I checked it out against extensions.conf, and sure enough, empty
- contexts ARE ok. So, I removed the restriction from AEL. This,
- though, highlighted a problem with multiple contexts of the same
- name. This should be OK, also. So, I added the extend keyword to
- AEL, and it can preceed the 'context' keyword (mixed with
- 'abstract', if nec.). This will turn off the warnings in AEL if
- the same context name is used 2 or more times. Also, I now call
- ast_context_find_or_create for contexts now, instead of just
- ast_context_create; I did this because pbx_config does this. The
- 'extend' keyword thus becomes a statement of intent. AEL can now
- duplicate the behavior of pbx_config,
-
-2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c: The addition of autoservice to func_curl
- additionally made func_curl dependent on the existence of a
- channel, with no real reason. This should make func_curl once
- again work without a channel. Reported by jmls. Fixed by
- tilghman. Closes issue #11090
-
-2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, include/asterisk/linkedlists.h: appending one
- list to another should leave the first list empty, and not
- require the user to do that
-
-2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_cut.c: Backport alternate encoding of newline
- delimiters from trunk to 1.4, as approved by Russell Reported by
- blitzrage Closes issue #10903
-
-2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Correctly respect hidecalleridname
- configuration option. Simplify code slightly in the process.
- Issue 11079, reported by ddv2005
-
-2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
- to specify app:spec in hint arguments
-
- * funcs/func_logic.c: closes issue #11052 -- where nothing after
- the ? will allow un-initialized variable values to corrupt and
- crash asterisk on 64-bit platforms
-
- * main/Makefile: this update to Makefile corrects how ast_expr2f.c
- should be generated
-
- * main/ast_expr2f.c: This should get rid of a really, really
- irritating warning generated by some 64-bit platforms from libc,
- where free(0) is frowned upon
-
-2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: If lock tracking is not enabled, then we
- can not attempt to log any mutex failures. If so, we could end up
- in infinite recursion. The only lock that is affected by this is
- a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
- issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
- (license 281)
-
-2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher@digium.com>
-
- * main/astmm.c: Minor FreeBSD build fix
-
-2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: After reading online I have confirmed that
- Record-Route headers should be copied to 1xx responses as well.
- (closes issue #10113) Reported by: makoto
-
- * apps/app_controlplayback.c: Make sure res is a positive value
- before performing the check to determine whether the user stopped
- it or not. (closes issue #11023) Reported by: cfc
-
-2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't leak a frame in the case that an END frame
- is received and the time since the BEGIN is less than that of the
- defined minimum DTMF duration. (closes issue #11051) Reported by:
- casper Patches: channel.c.86664.diff uploaded by casper (license
- 55)
-
- * include/asterisk/lock.h: Update the static mutex initializer to
- include the initialization of the internal mutex used to protect
- the lock debugging data. (closes issue #11044, patch suggested by
- Ivan)
-
-2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Account for the fact that sometimes headers
- may be terminated with \r\n instead of just \n (closes issue
- #11043, reported by yehavi)
-
-2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Move log message to before the frame it
- references is freed. (closes issue #11050) Reported by: slavon
- Patches: channel.c.86662.diff uploaded by casper (license 55)
-
- * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
- issue #11041) Reported by: jsmith Patches:
- asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * main/loader.c: Fixes for building under OpenSolaris. (closes
- issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
- uploaded by snuffy (license 35)
-
-2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
- DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
- does not match after an overlap call. Also added out_cause=1
-
-2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp@digium.com>
-
- * main/app.c: When returning a DTMF digit from
- ast_control_streamfile cast it as a char so that 0 does not
- overlap with the success return code. (closes issue #11023)
- Reported by: cfc
-
- * channels/chan_sip.c: Fix two issues with domains and transfers.
- If a port was given in the hostname it was treated as part of the
- hostname. If domains were configured but external domains were
- not enabled all transfers would be considered remote. (closes
- issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
- uploaded by ramonpeek (license 266)
-
- * channels/chan_sip.c: Set port number in received as information
- for registrations as well. (closes issue #11028) Reported by:
- brad-x
-
-2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: Fixed OSP module did not report
- source/devinfo IP in correct format.
-
-2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker@digium.com>
-
- * Makefile: Correct documentation. I removed the wrong line..
-
- * Makefile: Add documentation for options in asterisk.conf Issue
- 11029, patch by eserra
-
-2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell@digium.com>
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
- commit.
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Add support for
- setting the maximum trunk size for IAX2 trunking
-
- * main/channel.c, include/asterisk/channel.h: The channel needs to
- stay locked while running timer callbacks, as they access and
- modify channel data that may change elsewhere. I went through
- every timer callback in the source tree to make sure that none of
- them did any additional locking that could introduce deadlocks,
- and all is well. (closes issue #10765) Reported by: Ivan Patches:
- ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
- 229)
-
-2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: If a non-existent file is specified to be
- played either as a periodic announcement or as a hold/position
- announcement, the caller would be kicked out of the queue. No
- longer does this happen.
-
-2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell@digium.com>
-
- * codecs/codec_zap.c: Execute the RELEASE operation on transcoder
- channels in the destroy callback. (patch from jsloan)
-
- * main/utils.c: Revert a change that I made for issue #10979 which,
- as has been pointed out to me in issue #11018, doesn't really
- make sense. There is no reason to have the base64 decode function
- force a '\0' terminated buffer, when the result is almost always
- binary, anyway. In fact, this caused some breakage, as some code
- in res_crypto passed in a buffer exactly the right size to get
- its binary result, which got stomped on by this patch. (closes
- issue #11018, reported by dimas)
-
-2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changing the strategy field of the call_queue
- struct to be signed instead of unsigned, since the code attempts
- to set the strategy to -1 if you specify a bogus strategy. While
- this isn't a huge issue in 1.4, it could be a problem for someone
- who, say, tries to use the roundrobin strategy in trunk (despite
- all the deprecation warnings in 1.4).
-
-2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: If Asterisk is in the middle of shutting
- down, respond to OPTIONS with 503 Unavailable. (closes issue
- #10994) Reported by: eserra Patches: sip-options-503.patch
- uploaded by eserra (license 45)
-
-2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Whoops, forgot to remove the original
- sip_scheddestroy. (closes issue #11010) Reported by: vadim
-
-2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: When runuser/rungroup is specified, a remote
- console could only be attained by root (Closes issue #9999)
-
-2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't schedule dialog destruction if a
- MESSAGE is received using an existing dialog. (closes issue
- #11010) Reported by: vadim
-
-2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Since monitor-join is deprecated now,
- remove the example from the sample queues.conf file
-
- * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
- the monitor-join queue option
-
- * apps/app_queue.c: Adding deprecated warning to monitor-join
- option, since the plan is to no longer support this in favor of
- monitor-type = mixmonitor (related to issue #10885)
-
-2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: really picky formatting tweak ...
-
- * include/asterisk/lock.h: Some locking errors exposed the fact
- that the lock debugging code itself was not thread safe. How
- ironic! Anyway, these changes ensure that the code that is
- accessing the lock debugging data is thread-safe. Many thanks to
- Ivan for finding and fixing the core issue here, and also thanks
- to those that tested the patch and provided test results. (closes
- issue #10571) (closes issue #10886) (closes issue #10875) (might
- close some others, as well ...) Patches: (from issue #10571)
- ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
- 229) - a few small changes by me
-
-2007-10-16 21:14 +0000 [r85958] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Trying to remove a non-dynamic queue member via
- dynamic means can lead to some interesting (read nasty)
- situations. This patch clears up the issue by making only dynamic
- queue members removable via dynamic methods.
-
-2007-10-16 19:41 +0000 [r85921] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/localtime.c: Also set up gmtoff (this is used in the
- %z gnu extension to strftime) Reported and fixed by jcmoore
- Closes issue #11002
-
-2007-10-16 19:10 +0000 [r85896] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Remove a pointless lock.
-
-2007-10-16 15:21 +0000 [r85852] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing a double free which happens in the
- statechange thread. (closes issue #10987, reported by andrew)
-
-2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp <jcolp@digium.com>
-
- * apps/app_hasnewvoicemail.c: Check to make sure a value has been
- given to the VMCOUNT dialplan function. (closes issue #10996)
- Reported by: marsosa
-
- * main/threadstorage.c: Fix memory allocation issue in
- threadstorage. (closes issue #10995) Reported by: snuffy Patches:
- new-patch.diff uploaded by snuffy (license 35)
-
-2007-10-16 10:46 +0000 [r85800] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Fix the output for this channel help CLI
- command
-
-2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Ensure that no pending state changes are leaked
- when the device state change thread gets stopped on module
- unload.
-
- * apps/app_queue.c: Previously, app_queue created a thread to
- handle every single device state change. I changed this a while
- ago in trunk for performance reasons. However, bug 8407 points
- out that it is actually a race condition, causing device state
- changes to get processed in random order. So, I backported my
- changes from trunk to 1.4. (closes issue #8407, patch provided by
- tim_ringenbach, committed patch by me)
-
-2007-10-15 20:29 +0000 [r85687] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Don't execute a gosub if the arguments is
- zero-len (not just NULL) Reported by davevg Fixed by me Closes
- issue #10985
-
-2007-10-15 20:21 +0000 [r85686] Russell Bryant <russell@digium.com>
-
- * main/say.c: Add a small fix for the tw version of saying dates.
- (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
- uploaded by sharkey (license 172)
-
-2007-10-15 20:15 +0000 [r85684] Jason Parker <jparker@digium.com>
-
- * Makefile: Properly use DESTDIR in 'config' target. Do not try to
- run chkconfig or similar if using DESTDIR. Issue 10938, patch by
- cabal95.
-
-2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Be pedantic about handling memory allocation
- failure.
-
- * main/utils.c: The loop in the handler for the "core show locks"
- could potentially block for some amount of time. Be a little bit
- more careful and prepare all of the output in an intermediary
- buffer while holding a global resource. Then, after releasing it,
- send the output to ast_cli().
-
- * channels/chan_sip.c: Make the default for the srvlookup option to
- be yes. It doesn't really make sense for it to default to off.
- The default configuration file has it on, and proper RFC
- behavior, as indicated by a comment in the code, is for it to be
- on. So, let's have it on by default to make lives easier. (closes
- issue #10954, suggested by jtodd)
-
-2007-10-15 16:39 +0000 [r85571] Joshua Colp <jcolp@digium.com>
-
- * configs/features.conf.sample: Document that DTMF based features
- only work when two channels are bridged together. (closes issue
- #10773) Reported by: pbayley
-
-2007-10-15 16:34 +0000 [r85561] Russell Bryant <russell@digium.com>
-
- * include/asterisk/strings.h: Make a few changes so that characters
- in the upper half of the ISO-8859-1 character set don't get
- stripped when reading configuration. (closes issue #10982,
- dandre)
-
-2007-10-15 16:22 +0000 [r85559] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Bring both DTMF begin and end frames up through to
- the core for DTMF feature handling. (closes issue #10826)
- Reported by: dimas
-
-2007-10-15 15:40 +0000 [r85556] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Ensure the buffer passed to
- ast_canmatch_extension() is properly initialized so that it is
- null terminated. (issue #10977) Reported by: dimas Patches:
- pbxdundi.patch uploaded by dimas (license 88) - small mods by me
-
-2007-10-15 14:55 +0000 [r85552] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If Monitor or a spy was added to a P2P or native
- bridged channel bring the channel back to the generic bridging
- core so the monitor or spy operations work. (closes issue #10943)
- Reported by: julianjm
-
-2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant <russell@digium.com>
-
- * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
- (closes issue #10980) Reported by: casper Patches:
- db.c.84633.diff uploaded by casper (license 55)
-
- * main/asterisk.c: Make sure remote consoles unmute themselves
- again after reconnecting. (closes issue #10847) Reported by: atis
- Patches: console_unmute_on_reconnect.patch uploaded by atis
- (license 242)
-
- * main/utils.c: Make sure that the base64 decoder returns a
- terminated string. (closes issue #10979) Reported by: ys Patches:
- util.c.diff uploaded by ys (license 281) - small mods by me
-
- * pbx/pbx_config.c: Don't create the context for users in
- users.conf until we know at least one user exists. (closes issue
- #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
- dimas (license 88)
-
-2007-10-13 15:26 +0000 [r85536] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.ael.sample: Remove deprecated syntax from
- sample ael file Reported and patched by: dimas Closes issue
- #10967
-
-2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
- issue with console verbosity when running asterisk -rx to execute
- a command and retrieve its output. The issue was that there was
- no way for the main Asterisk process to know that the remote
- console was connecting in the -rx mode. The way that James has
- fixed this is to have all remote consoles muted by default. Then,
- regular remote consoles automatically execute a CLI command to
- unmute themselves when they first start up. (closes issue #10847)
- Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
- by jamesgolovich (license 176)
-
- * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
- handle the case where read() may return the text for more than
- one CLI command at once for a remote console. (closes issue
- #10888) Reported by: jamesgolovich Patches:
- asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
- 176)
-
-2007-10-12 18:30 +0000 [r85523] Tilghman Lesher <tlesher@digium.com>
-
- * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address
-
-2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
- SDMI. (closes issue #10959)
-
- * pbx/pbx_realtime.c: Fix the potential use of an uninitialized
- buffer in a log message. (closes issue #10958) Reported by: dimas
- Patches: realtime.patch uploaded by dimas (license 88)
-
-2007-10-11 15:26 +0000 [r85397] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: When creating a new packet don't try to stop
- retransmission of it. It was just allocated/created so it's
- impossible for it to have already been scheduled. (closes issue
- #10945) Reported by: flefoll Patches:
- chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
- (license 244)
-
-2007-10-11 04:35 +0000 [r85356] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: A dollar sign by itself, not indicating a start of a
- variable or expression prematurely ends substitution (closes
- issue #10939)
-
-2007-10-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.13 released.
-
-2007-10-10 15:56 +0000 [r85316] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h: I introduced a new member to the
- ast_filestream struct in 1.4.12, but put it in the middle of the
- struct, instead of at the end. One of the Debian folks, paravoid,
- pointed out that this breaks binary compatability with modules
- compiled against older headers. So, I'm moving the new member to
- the end of the struct to resolve the situation.
-
-2007-10-10 15:51 +0000 [r85315] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: The thread ID should be unsigned.
-
-2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If devicestate is passed a port number strip
- it out. (closes issue #10930) Reported by: ibc
-
- * channels/chan_sip.c: Add support for handling a 182 Queued
- response. (closes issue #10924) Reported by: ramonpeek Patches:
- queued-182.diff uploaded by ramonpeek (license 266)
-
-2007-10-10 14:26 +0000 [r85276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A bunch of changes from sprintf to
- snprintf. See security advisory AST-2002-022
-
-2007-10-10 14:14 +0000 [r85242] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Close voicemail message description file if
- duration did not meet the minimum, or else we will eventually run
- out of file descriptors. (closes issue #10918) Reported by:
- brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
- 279)
-
-2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h: use a macro instead of an inline
- function, so that backtraces will report the caller of
- ast_frame_free() properly
-
-2007-10-09 21:55 +0000 [r85158] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, main/utils.c, include/asterisk/lock.h: This
- commit fixes the following issues: - Deadlock in ast_write (issue
- #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
- initialization error in lock.h (issue #10571)
-
-2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't perform a reinvite if a transfer is in
- progress. (issue #10915) Reported by: ramonpeek
-
- * main/rtp.c: Only update codec information if the channel has a
- technology private structure. (issue #10915) Reported by:
- ramonpeek
-
- * main/rtp.c: Update codec information as well as address when
- doing hold reinvites. (issue #10868) Reported by: mavince
-
- * main/channel.c: Don't keep trying to native bridge if either of
- the channels are involved in a masquerade operation to be done.
- (closes issue #10696) Reported by: tbelder
-
-2007-10-08 03:28 +0000 [r84957] Russell Bryant <russell@digium.com>
-
- * Makefile.rules: Enable file dependency tracking for _all_ builds,
- and not just for builds with dev-mode enabled. I have seen enough
- problems caused by this that I don't think it's worth keeping. I
- want to continue to encourage anybody that is interested to
- continue to run Asterisk from svn. Furthermore, I do not want
- their systems to break when we change a structure definition in a
- header file. :)
-
-2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Presence packets from a client who's connected
- with our Jabber ID are valid, therefore, those clients must be
- considered as buddies. The resource string helps us make the
- distinction between clients. Closes issue #10707, reported by
- yusufmotiwala.
-
- * res/res_jabber.c: Prevent Asterisk from crashing when receiving a
- presence packet without resource from a buddy that is known to
- have a resource list. Revert a change I previously made, where
- Asterisk could point to a freed memory location.
-
-2007-10-05 19:42 +0000 [r84851] Tilghman Lesher <tlesher@digium.com>
-
- * main/db.c: Log exactly why we can't open the database, if we fail
- (closes issue #10887)
-
-2007-10-05 18:55 +0000 [r84818] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Update the remembered RTP peer information when
- putting an endpoint on hold or taking it off hold so that the RTP
- stack does not initiate a needless reinvite. (closes issue
- #10868) Reported by: mavince
-
-2007-10-05 16:44 +0000 [r84783] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Do deadlock avoidance in a couple more
- places. You can't lock two channels at the same time without
- doing extra work to make sure it succeeds. (closes issue #10895,
- patch by me)
-
-2007-10-05 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12.1 released. (This is mainly to include the
- app_queue fix for a memory leak on reload, but includes a couple
- of other bug fixes, as well.)
-
-2007-10-05 01:39 +0000 [r84742] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a copy/paste error in the description of
- UpdateConfig that was pointed out by JerJer on #asterisk-dev
-
-2007-10-04 21:57 +0000 [r84692] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Don't allocate space for queue members unless
- it's needed. You end up deleting dynamic members on a reload. Not
- good. closes issue (#10879, reported by dazza76, patched by me)
-
-2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: callers of sig2str already add the word
- 'signalling' in the appropriate place, so don't duplicate it
-
-2007-10-04 14:51 +0000 [r84637] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Create a duplicate of the channel's member name
- as the tab completion stuff will free it. (closes issue #10884)
- Reported by: adamg
-
-2007-10-03 22:59 +0000 [r84581] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: When an RFC 2833 event is sent that we don't
- recognize, ignore it, don't queue a NULL digit (closes issue
- #10877)
-
-2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
- in a switch expr doesn't execute correctly, because the commas in
- the function args are not converted to vertbars before the func
- is called. I modified just the switch code to convert the commas
- to vertbars if there, but if more of these sort of probs are
- found, I may have to resort to something a little more
- fundamental. We'll see, I guess.
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
- null input to a switch statement results in a hangup; since
- switch is implemented with extensions, and the default case is
- implemented with a '.', and the '.' matches 1 or more remaining
- characters, the case where 0 characters exist isn't matched, and
- the extension isn't matched, and the goto fails, and a hangup
- occurs. Now, when a default case is generated, it also generates
- a single fixed extension that will match a null input. That
- extension just does a goto to the default extension for that
- switch. I played with an alternate solution, where I just tack an
- extra char onto all the patterns and the goto, but not the
- default case's pattern. Then even a null input will still have at
- least one char in it. But it made me nervous, having that extra
- char in , even if that's a pretty secret and low-level issue.
-
-2007-10-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12 released.
-
-2007-10-02 20:06 +0000 [r84474] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/prep_tarball: * Don't build the
- menuselect-tree for the tarball, as it requires running the
- configure script first * Change the Makefile to note that
- menuselect-tree depends on the configure script.
-
-2007-10-02 19:01 +0000 [r84410-84437] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix some odd formatting I missed..
-
- * res/res_features.c: Finish up on transferee channel before return
- on failure. Issue 10821, patch by Ivan
-
-2007-10-02 14:12 +0000 [r84370] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Use snprintf instead of sprintf in one
- place. There is no vulnerability here due to various buffer sizes
- around the code, but I still didn't like seeing a non
- length-limited copy of data coming off of the wire into a stack
- buffer, as this would be a problem in the future if buffer sizes
- elsewhere got changed or size limitations removed ...
-
-2007-10-02 09:48 +0000 [r84345] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: terminate USERUSER String with 0
-
-2007-10-01 21:52 +0000 [r84291] Jason Parker <jparker@digium.com>
-
- * Makefile, Makefile.rules, channels/Makefile: Add dist-clean
- support for subdirs. Change h323 to only remove the Makefile on a
- dist-clean, rather than a clean. This fixes a bug I found with
- trying to run make after a make clean
-
-2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/channel.c, main/manager.c, channels/chan_agent.c: moved
- get_base_channel() code from action_redirect to
- ast_channel_masquerade() for issue 7706 and BE-160
-
-2007-10-01 21:18 +0000 [r84273] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...
-
-2007-10-01 21:07 +0000 [r84271] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Fulfull a feature request
- from Qwell on the "core show locks" output. It will now note the
- lock type for each lock that a thread holds. (mutex, rdlock, or
- wrlock)
-
-2007-10-01 20:27 +0000 [r84239] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
- #10777 -- by returning a null for the parse tree when there's
- really nothing there, and making sure we don't try to do checking
- on a null tree.
-
-2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Add another sanity check in the AGI read loop. We
- really don't care about EAGAIN unless we didn't read an entire
- line. If there is a newline at the end if the read buffer, break,
- because we got the whole thing. (reported and patched by bmd)
-
- * include/asterisk/lock.h: Show rwlocks in the "core show locks"
- output. Before, it only showed mutexes.
-
- * channels/Makefile: Remove another file in "make clean". (closes
- issue #10814, paravoid)
-
- * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.
-
-2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp <jcolp@digium.com>
-
- * configs/usbradio.conf.sample (removed): Remove chan_usbradio
- config file from tree, it is not present in here. (closes issue
- #10839) Reported by: casper
-
- * res/res_musiconhold.c: Fix randomness. save_pos was being set to
- 0 initially instead of -1, causing it to jump to position 0 when
- moh started. (closes issue #10859) Reported by: jamesgolovich
- Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * apps/app_dial.c: Only attempt early bridging if the options given
- to Dial() permit it. (closes issue #10861) Reported by: peekyb
-
-2007-09-30 20:02 +0000 [r84146] Russell Bryant <russell@digium.com>
-
- * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
- modules. The load and reload parameters were in the wrong place.
- (closes issue #10846, alebm)
-
-2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
- (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
- pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
- pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
- pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest22/qq.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1 (added),
- pbx/ael/ael-test/ael-ntest22/t2 (added),
- pbx/ael/ael-test/ael-ntest22/t3 (added),
- pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
- pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
- that matches what I did in 84134 for AEL regressions.
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
- 10786; All config files support #include with globbing (you know,
- *,[chars],?,{list,list},etc), so I've updated the AEL system to
- support this also.
-
-2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Correct pronunciations of numbers for .nl (Closes
- issue #10837)
-
- * main/channel.c: Avoid a deadlock with ALL of the locks in the
- masquerade function, not just the pairs of channels. (Closes
- issue #10406)
-
-2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/manager.c, channels/chan_agent.c,
- include/asterisk/channel.h: if an Agent is redirected, the base
- channel should actually be redirected. This was causing multiple
- issues, especially issue 7706 and BE-160
-
-2007-09-27 00:01 +0000 [r83976] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: remove a todo item that has been completed
-
-2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_alsa.c: avoid the weird usage of assert() in the
- ALSA header files that gcc 4.2 wants to complain about
-
-2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: I changed my mind ... I think this should be
- a LOG_NOTICE.
-
- * channels/chan_sip.c: Add a log message that was requested by the
- masses in the developer tutorial session at Astricon. chan_sip
- did not output any message when a call was rejected because the
- extension was not found. This adds a verbose message (at verbose
- level 3) to note when this happens.
-
- * channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
- (please run the configure script with --enable-dev-mode so this
- doesn't happen again ...)
-
-2007-09-26 18:35 +0000 [r83879] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Remove unused 4k of memory on the program
- stack (closes issue #10827)
-
-2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: jmls pointed out that unsetting the group and setting
- the group to the blank string aren't quite the same.
-
- * build_tools/make_defaults_h: In the source, keys are relative to
- the datadir, not varlib (which is the same in most cases, but
- it's good to be accurate). Closes issue #10811
-
- * doc/realtime.txt: Oops. Removed the unworkable workaround. This
- note should never have been in the release.
-
- * main/app.c: Making change to group splitting, as discussed on the
- -dev list. The main effect of this will be to permit
- Set(GROUP([cat])=), i.e. unsetting a group.
-
-2007-09-24 07:54 +0000 [r83620] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed round_robin group dial method, this
- never worked well on BRI Ports (2 channels)
-
-2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
- are made here for the first arg in the for(arg1; arg2; arg3) {}
- statement, as were done for the 3rd arg. It can now be an
- assignment that will embedded in a Set() app, or a macro call, or
- an app call.
-
- * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
- statement is now wrapped in Set() only if there's an '=' in that
- string. Otherwise, if it begins with '&', then a Macro call is
- generated; otherwise it is made into an app call. A bit more
- accomodating, keeps the new guys happy, and the guys with ael-1
- code should be happy, too
-
-2007-09-21 14:37 +0000 [r83432] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
- channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
- main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
- pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
- channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
- main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
- main/db1-ast/hash/hash.c, include/asterisk/channel.h,
- channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
- with cosnt pointers. This set of changes gets all of Asterisk
- (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
- #10774, patch from qwell)
-
-2007-09-21 13:34 +0000 [r83400] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix video under certain circumstances. It
- would have been possible for the formats on the channel to not
- contain the video format. (closes issue #10782) Reported by:
- cwhuang
-
-2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: When daemonizing, don't change working directory
- to "/". It makes it not be able to do a core dump when not
- running as uid=root. (closes issue #10766, xrg)
-
- * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
- ask for /bin/bash, as it uses bashisms. (closes issue #10772,
- reported by culrich)
-
-2007-09-20 17:09 +0000 [r83246] Jason Parker <jparker@digium.com>
-
- * apps/app_disa.c: If # is pressed after dialing an extension in
- DISA, stop trying to collect more digits. (issue #10754) Reported
- by: atis Patches: app_disa.c.branch.patch uploaded by atis
- (license 242) app_disa.c.trunk.patch uploaded by atis (license
- 242)
-
-2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make sure the minimum T1 timer value is
- obeyed in all cases. (closes issue #10768) Reported by: flefoll
- Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
- (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
- flefoll (license 244)
-
- * channels/chan_sip.c: Fix a minor spelling error. (closes issue
- #10769) Reported by: flefoll Patches:
- chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
- 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
- (license 244)
-
-2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant <russell@digium.com>
-
- * apps/app_system.c: The System() and TrySystem() applications can
- take a substantial amount of time to execute while not servicing
- the channel. So, put the channel in autoservice while the command
- is being executed. (closes issue #10726, reported by mnicholson)
-
- * funcs/func_curl.c: Using curl can take a substantial amount of
- time, so the channel should be autoserviced while waiting for it
- to complete. (closes issue #10725, reported by mnicholson)
-
- * channels/chan_iax2.c: When handling a reload of chan_iax2, don't
- use an ao2_callback() to POKE all peers. Instead, use an
- iterator. By using an iterator, the peers container is not locked
- while the POKE is being done. It can cause a deadlock if the
- peers container is locked because poking a peer will try to lock
- pvt structs, while there is a lot of other code that will hold a
- pvt lock when trying to go lock the peers container. (reported to
- me directly by Loic Didelot. Thank you for the debug info!)
-
- * main/manager.c: Fix up another potential race condition. Do the
- loop decrementing use count on events with the eventq protected
- from being changed. (reported on IRC by Ivan)
-
-2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Protect the CDR record from modification by
- pbx_exec so that the application data contains the Queue data.
- (closes issue #10761) Reported by: snar Patches:
- app-queue-mixmonitor.patch uploaded by snar (license 245)
-
- * channels/chan_sip.c: (closes issue #10760) Reported by: dimas
- Patches: chan_sip.patch uploaded by dimas (license 88) Read in
- subscribecontext option in general to be the default.
-
-2007-09-19 09:32 +0000 [r83023-83024] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: removed comment which violates the coding
- guidelines.
-
- * channels/misdn_config.c, channels/chan_misdn.c,
- channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
- allow configuring the asterisk dtmf detector instead of the
- mISDN_dsp ones. also added the patch from irroot #10190, so that
- dtmf tones detected by the asterisk detector are passed outofband
- to asterisk, to make any use of dtmf tones at all.
-
-2007-09-19 00:19 +0000 [r82992] Russell Bryant <russell@digium.com>
-
- * apps/app_flash.c: Change the description of app_flash to note how
- it can be a useful tool instead of just saying that it is
- generally a worthless feature. (Thanks to Jim Van Meggelen for
- pointing it out and providing the proposed text)
-
-2007-09-18 23:41 +0000 [r82961] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Initialize a variable to NULL to make the world
- happy.
-
-2007-09-18 22:42 +0000 [r82929] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
- interrupting the fgets() call when using FastAGI. This version of
- the patch maintains the original behavior of the code when not
- using FastAGI. (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-4.patch uploaded by juggie (license 24)
- res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
- mods by me Tested by: juggie, festr
-
-2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey <dbailey@digium.com>
-
- * main/manager.c: Corrected patch applied in revision r82887.
-
- * main/manager.c: Fixed a bug where http manager sessions prevented
- the eventq from being cleaned out because http manager sessions
- do not have a valid file descriptor.
-
-2007-09-18 20:56 +0000 [r82867] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a memory leak that can occur on systems under
- higher load. The issue is that when events are appended to the
- master event queue, they use the number of active sessions as a
- use count so it will know when all active sessions at the time
- the event happened have consumed it. However, the handling of the
- number of sessions was not properly synchronized, so the use
- count was not always correct, causing an event to disappear
- early, or get stuck in the event queue for forever. (closes issue
- #9238, reported by bweschke, patch from Ivan, modified by me)
-
-2007-09-18 20:09 +0000 [r82865] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Moving the logic for handling an empty
- membername to the create_member function so that there is a
- common place where this occurs instead of being spread out to
- several different places.
-
-2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: there is no need for conditional logic to
- select ->interface or ->membername, snince ->membername will
- always be populated
-
-2007-09-18 16:31 +0000 [r82802] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: When copying the contents from the wildcard
- peer, do a deep copy instead of shallow copy so that it doesn't
- crash when beging destroyed. (closes issue #10546, patch by me)
-
-2007-09-18 15:28 +0000 [r82751] Jason Parker <jparker@digium.com>
-
- * configs/sip.conf.sample: Correct the allowexternaldomains option
- in SIP sample config. Issue 10753
-
-2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
- ast_localtime() instead of a couple places in app_voicemail to
- prevent the problem everywhere instead of just a couple of
- places. (related to issue #10746)
-
- * apps/app_voicemail.c: Initialize some memory to fix crashes when
- leaving voicemail. This problem was fixed by running Asterisk
- under valgrind. (closes issue #10746, reported by arcivanov,
- patched by me) *** IMPORTANT NOTE: We need to check to see if
- this same bug exists elsewhere.
-
- * res/res_features.c: Handle the case where there are multiple
- dynamic features with the same digit mapping, but won't always
- match the activated on/by access controls. In that case, the code
- needs to keep trying features for a match. (reported by Atis on
- the asterisk-dev list, patched by me)
-
-2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: revert a change that wasn't supposed to be
- committed... doh!
-
- * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
- where a logical member name (if specified) was not used, but
- instead the direct interface was listed
-
-2007-09-17 02:00 +0000 [r82514] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
- of passing a NULL pointer into snprintf pass "". It makes Solaris
- much happier.
-
-2007-09-14 21:19 +0000 [r82444] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
- had to leave out the bit about ending the previous cdr in the
- fork; it would destroy current implementations.
-
-2007-09-14 21:17 +0000 [r82435] Russell Bryant <russell@digium.com>
-
- * configs/zapata.conf.sample: Add a note to help clarify the value
- set with the echocancel option. (inspired by Malcolm's blog post
- on blogs.digium.com about HPEC)
-
-2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Crap, I broke the build. Fixed.
-
- * apps/app_queue.c: Adding member name field to manager events
- where they were missing before (closes issue #10721, reported by
- snar)
-
-2007-09-14 17:48 +0000 [r82394] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: If a channel does not have an owner, do not
- try to set a channel variable. This will end up making the
- channel variable global, which is not right. Closes issue #10720,
- patch by flefoll.
-
-2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant <russell@digium.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- checking for libusb here, so nobody has to deal with conflicts in
- the chan_usbradio-1.4 branch every time the configure script gets
- changed
-
- * channels/chan_usbradio.c (removed), channels/xpmr (removed),
- channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
- It can't live here because we have a strict policy to not include
- new features in release branches. However, I'm going to merge it
- into trunk, and I also have a special 1.4 based branch that
- includes this module. svn co
- http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4
-
-2007-09-14 14:42 +0000 [r82376] Mark Michelson <mmichelson@digium.com>
-
- * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
- (closes issue #10717, reported and patched by leedm777)
-
-2007-09-14 01:24 +0000 [r82368] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c: Fixed problem with changes made to cdr
- functionality
-
-2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_usbradio.c: this new driver may not live in this
- branch for long (since it is a new feature), but it definitely
- should not be built by default
-
-2007-09-14 00:34 +0000 [r82366] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
- channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
- channels/xpmr (added), channels/xpmr/LICENSE (added),
- channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
- (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
- channel driver for USB Radio device and support thereof.
-
-2007-09-13 23:11 +0000 [r82358] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime
-
-2007-09-13 20:16 +0000 [r82346] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Preemptively fixing a possible segfault. It is
- possible that queuename is NULL (meaning pause ALL queues), so
- use q->name instead.
-
-2007-09-13 20:11 +0000 [r82344] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
- mutliple threads tried to close the same file. Do we actually
- need the locking here? What happens if you open the same file
- twice, and two threads try to write to it at the same time? Is
- fputs() going to write out the entire line at once? I suspect
- that it could be possible for the second fopen to run during the
- first fputs, so the position could be in the middle of the
- previously written line... Issue 10347, initial patch by
- explidous (but I removed all of the paranoia stuff..)
-
-2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c: resolve a warning when not building under dev
- mode
-
- * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
- in tracking astobj2 statistics if dev-mode is enabled. Also, when
- dev mode is enabled, register the CLI command that can be used to
- run the astobj2 test and print out statistics.
-
-2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 82334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
- | 2 lines clarify the OpenSSL and OpenH323 license exceptions
- ........
-
-2007-09-13 16:25 +0000 [r82326] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added logic to handle the unlikely case that
- someone has two queues with the same name. Asterisk will log a
- warning message letting the user know that one was already
- defined with that name and is it skipping all further instances.
- This also will work for realtime queues but in order for that to
- happen, the user would have to trigger a perfectly timed reload
- as a realtime queue is being looked up, which is highly unlikely
- (but taken care of nonetheless).
-
-2007-09-13 11:47 +0000 [r82309] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #9401, reported and patched
- by irrot, with slight modifications by me. Handle DTMF sent by
- Asterisk properly.
-
-2007-09-12 21:56 +0000 [r82296] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Fix a check of the wrong pointer, as pointed out
- by an XXX comment left in the code. The problem was harmless,
- however.
-
-2007-09-12 21:28 +0000 [r82291] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
- files
-
-2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_meetme.c: remove a race condition for the creation of
- recordthread's, and fix a small memory leak. This closes issue#
- 10636
-
-2007-09-12 20:12 +0000 [r82285] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/private.h, main/stdtime/tzfile.h,
- include/asterisk/localtime.h, main/stdtime/localtime.c: Working
- on issue #10531 exposed a rather nasty 64-bit issue on
- ast_mktime, so we updated the localtime.c file from source. Next
- we'll have to write ast_strptime to match.
-
-2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Clean up the output of "asterisk -h". This
- tweaks the wording and wraps lines at 80 characters. (closes
- issue #10699, seanbright)
-
- * res/res_agi.c: revert patch from issue #10553, as someone not
- using fastagi reported that this broke their system.
-
-2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Accidentally committed changes to
- app_voicemail which do NOT need to be in the 1.4 branch yet.
- reverting...
-
- * apps/app_voicemail.c, apps/app_queue.c: We should only initialize
- a realtime queue when it is allocated, not every time we access
- it. This prevents the members ao2_container from being
- reallocated every time the queue is accessed. I also removed a
- debug message I had accidentally left in on a previous commit.
-
-2007-09-11 22:37 +0000 [r82267] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
- these calls was reading bogus memory ...
-
-2007-09-11 21:41 +0000 [r82265] Joshua Colp <jcolp@digium.com>
-
- * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
- #10679) Reported by: andrew Build under dev mode when K6OPTS is
- enabled.
-
-2007-09-11 20:49 +0000 [r82263] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix another missing unref of member objects.
- This one was pointed out by Marta. When building the outgoing
- list in try_calling(), a member reference is stored in each
- outgoing entry. However, when this list got destroyed, the
- reference was not released.
-
-2007-09-11 20:36 +0000 [r82261] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: this change should fix issue # 10659 -- what I worry
- about is how many other bug reports it may generate. Hopefully,
- we can please the/a majority. Hopefully. We shall see. Calls not
- marked ANSWERED and with only one channel name will not be
- posted. This should eliminate the double CDR's.
-
-2007-09-11 16:05 +0000 [r82252] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: All instances of ao2_iterators which were just
- named 'i' have been renamed to 'mem_iter' so that when refcounted
- queues are merged into trunk, there will be little confusion
- regarding iterator names, especially when a queue and member
- iterator are used in the same function.
-
-2007-09-11 16:03 +0000 [r82250] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: The sample dundi.conf claims support for a
- wildcard peer entry - [*], but the code did not support it. This
- patch makes it work. (closes issue #10546, patch by dds, with
- some changes by me)
-
-2007-09-11 16:01 +0000 [r82249] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
- hold/retrieve issue.
-
-2007-09-11 15:26 +0000 [r82245] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
- juggie When using fastagi, fgets() can return before a full line
- is read. Add explicit handling for the case where it gets
- interrupted.
-
-2007-09-11 14:56 +0000 [r82243] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
- Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
- (license 176) Don't leak memory when unloading DUNDi.
-
-2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Add a couple more missing unrefs of queue
- member objects
-
- * apps/app_queue.c: Add a missing unref of a queue member in an
- error handling block
-
- * apps/app_queue.c: Document why membercount can not simply be
- replaced by ao2_container_count()
-
- * main/astobj2.c: backport astobj2 race condition fix. This
- function is the exact same as trunk so it applies here as well.
-
-2007-09-10 18:02 +0000 [r82155] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: Convert struct member to use refcounts (closes
- issue #10199)
-
-2007-09-10 15:02 +0000 [r82091] Mark Michelson <mmichelson@digium.com>
-
- * configs/misdn.conf.sample: Removing non-existent options from
- misdn configuration sample. (closes issue #10678, reported and
- patched by IgorG)
-
-2007-09-09 02:35 +0000 [r82028] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix inline compiles on really old
- compilers (who uses gcc 2.7 anymore, really?)
-
-2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
- did not free the entry it removed.
-
- * .cleancount: (closes issue #10672) Bump the cleancount so that a
- "make clean" will be forced. This is needed because my fix in
- revision 81599 made a change to a data structure in file.h, and
- since file dependency tracking is only on with dev-mode enabled,
- file format modules that don't get rebuilt may crash, as is the
- case with this issue. This makes me wonder - how much faster does
- the code build without the file dependency tracking enabled? If
- it doesn't make much of a difference, then it may be worth just
- keeping it on all of the time, or perhaps just not in release
- tarballs, so that this type of issue is avoided.
-
-2007-09-07 19:48 +0000 [r81923] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
- as the mixmonitor filename. This moves the setting of the
- MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
- by sim.
-
-2007-09-07 15:25 +0000 [r81886] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Moving the explanation for joinempty
- to a more appropriate place
-
-2007-09-06 22:28 +0000 [r81832] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: (closes issue #9724, closes issue #10374)
- Reported by: kenw Patches: 9724.txt uploaded by russell (license
- 2) Tested by: kenw, russell Resolve a deadlock that occurs when
- doing a SIP transfer to parking. I come across this type of
- deadlock fairly often it seems. It is very important to mind the
- boundary between the channel driver and the core in respect to
- the channel lock and the channel-pvt lock. Channel drivers lock
- to lock the pvt and then the channel once it calls into the core,
- while the core will do it in the opposite order. The way this is
- avoided is by having channel drivers either release their pvt
- lock while calling into the core, or such as in this case,
- unlocking the pvt just long enough to acquire the channel lock.
-
-2007-09-06 22:05 +0000 [r81778-81826] Jason Parker <jparker@digium.com>
-
- * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
- for ASTLDFLAGS. This adds LDOPTS
-
- * include/asterisk/astobj2.h: This should fix a build issue that
- people building against uClibc were seeing with the addition of
- astobj2
-
-2007-09-06 19:40 +0000 [r81776] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: (closes issue #10122) Reported by:
- stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
- qwell (license 4) Tested by: stevefeinstein After looking over
- the code I agree with Qwell. Setting the file descriptor to
- conference each time just causes a fight back and forth.
-
-2007-09-06 16:56 +0000 [r81743] Philippe Sultan <philippe.sultan@gmail.com>
-
- * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
- length fixes. Removed an unused variable in aji_client structure
- (context)
-
-2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixes an issue where valid DTMF had to be
- pressed twice to exit a queue if a member's phone was ringing.
- (closes issue #10655, reported by strider2k, patched by me)
-
- * res/res_features.c: Fixes a memory leak (closes issue #10658,
- reported and patched by Ivan)
-
-2007-09-06 14:20 +0000 [r81650] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
- and Google's XMPP server complaint, if set, the 'from' attribute
- must be set to the user's full JID.
-
-2007-09-05 20:53 +0000 [r81599] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, main/say.c, res/res_features.c,
- main/file.c, include/asterisk/channel.h: Fix an issue that can
- occur when you do an attended transfer to parking. If you
- complete the transfer before the announcement of the parking spot
- finishes, then the channel being parked will hear the remainder
- of the announcement. These changes make it so that will not
- happen anymore. Basically, res_features sets a flag on the
- channel is playing the announcement to so that the file streaming
- core knows that it needs to watch out for a channel masquerade,
- and if it occurs, to abort the announcement. (closes BE-182)
-
-2007-09-05 17:18 +0000 [r81569] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Solaris x86 compatibility fix
-
-2007-09-05 15:19 +0000 [r81525] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing the build...
-
-2007-09-05 15:14 +0000 [r81523] Jason Parker <jparker@digium.com>
-
- * channels/chan_phone.c: Do not try to unregister a NULL channel
- tech. Also changed load_module function to use defines rather
- than numbers for return values. Issue 10651, patch by
- rbraun_proformatique, with additions by me.
-
-2007-09-05 15:03 +0000 [r81520] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
- only count members who are logged in and available. (related to
- issue #10652, reported by wuwu)
-
-2007-09-05 13:11 +0000 [r81492] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10650) Reported by: tacvbo Only
- print out that the spy was removed while holding the spy lock.
-
-2007-09-04 20:54 +0000 [r81453-81455] Jason Parker <jparker@digium.com>
-
- * apps/app_followme.c: Rather than attempt to play a file, we can
- just check whether it exists. Issue 10634, patch by me, testing
- by pabelanger, sanity checked by bweschke
-
- * configs/followme.conf.sample: Change default followme config file
- to point to the correct files. Issue 10644, patch by pabelanger
-
-2007-09-04 18:37 +0000 [r81448] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- Remove the typedefs on ao2_container and ao2_iterator. This is
- simply because we don't typedef objects anywhere else in
- Asterisk, so we might as well make this follow the same
- convention.
-
-2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: there is no point in sending 401
- Unauthorized to a UAS that sent us a properly-formatted
- Authentication header with the expected username and nonce but an
- incorrect response (which indicates the shared secret does not
- match)... instead, let's send 403 Forbidden so that the UAS
- doesn't retry with the same authentication credentials repeatedly
-
-2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: (closes issue #10632) Reported by:
- jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
- by jamesgolovich (license 176) Fix memory leak when unloading
- chan_iax2. The firmware files were not being freed.
-
- * main/channel.c: (closes issue #10476) Reported by: mdu113 Only
- look for the end of a digit when waiting for a digit. This in
- turn disables emulation in the core.
-
- * main/dns.c: (closes issue #10610) Reported by: john Patches:
- dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
- Don't return a match if no SRV record actually exists.
-
-2007-09-03 18:57 +0000 [r81433] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove a couple of calls to
- ast_string_field_free_pools() on peers in error handling blocks
- in the code for building peers. The peer object destructor does
- this and doing it twice will cause a crash. (closes issue #10625,
- reported by and patched by pnlarsson)
-
-2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed a comment to be more accurate. (really
- this is just a test to make sure I can commit properly from home)
-
- * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
- into ao2_match_by_addr and making it available everywhere since
- it could be a handy callback to have
-
-2007-08-31 21:27 +0000 [r81418] Russell Bryant <russell@digium.com>
-
- * include/asterisk/astobj2.h: Remove references to a debugging
- parameter that does not exist
-
-2007-08-31 19:48 +0000 [r81416] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixed broken behavior of a reload on realtime
- queues. Prior to this patch, if a reload was issued and a
- realtime queue had callers waiting in it, then the queue would be
- removed from the queue list, but it would not actually be freed
- (in fact, a debug message warning about a memory leak would come
- up). With this patch, reloads do not touch realtime queues at
- all.
-
-2007-08-31 19:16 +0000 [r81415] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_logic.c: The IF() function was not allowing true
- values that had embedded colons (closes issue #10613)
-
-2007-08-31 18:44 +0000 [r81412] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Re-order dial options to be in line with the
- existing alpha order. Issue 10621, initial patch by junky
-
-2007-08-31 17:38 +0000 [r81410] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
- available. Closes issue 10548, reported by keepitcool.
-
-2007-08-31 15:53 +0000 [r81406] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c: Make it the engine's responsible to check for
- the presence of results.
-
-2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: add missing "transcoder show" (and deprecated
- "show transcoder") CLI commands that were in 1.2 but never added
- to 1.4
-
-2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10618) Reported by: dimas
- Don't pass through the stopped sounds frame.... just drop it.
-
- * res/res_features.c: (closes issue #10009) Reported by: dimas
- Don't output a bridge failed warning message if it failed because
- one of the channels was part of the masquerade process. That is
- perfectly normal.
-
-2007-08-30 22:05 +0000 [r81397] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing an extraneous (and possibly
- misleading) log message. Firstly, if the announce file isn't
- found, the streaming functions will report it. Secondly, not all
- non-zero returns from play_file mean that the announce file
- wasn't found. Positive return values simply mean that a digit was
- pressed (most likely to skip through the announcement). (closes
- issue #10612, reported and patched by dimas)
-
-2007-08-30 21:23 +0000 [r81395] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10514) Reported by: casper
- Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
- Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
- for it to ever be that value.
-
-2007-08-30 21:11 +0000 [r81392] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: via issue 10599, where 'CDR already initialized'
- messages are being generated. Since all channels will have an
- init'd CDR attached at creation time, this message is now
- particularly useless. Removed.
-
-2007-08-30 15:38 +0000 [r81383] Russell Bryant <russell@digium.com>
-
- * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
- define. (closes issue #10559, paravoid)
-
-2007-08-30 15:35 +0000 [r81381] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed some manager event messages to reflect
- whether a queue member is a realtime member or not
-
-2007-08-30 15:33 +0000 [r81379] Russell Bryant <russell@digium.com>
-
- * configs/modem.conf.sample (removed), configs/enum.conf.sample,
- configs/extensions.ael.sample: Fix a typo, update a reload
- command, and remove an unused configuration file. (closes issue
- #10606, casper)
-
-2007-08-30 14:53 +0000 [r81375] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
- pbx.diff uploaded by jmls (license 141) Backport changes from
- 81372. Add REASON dialplan variable for when an originated call
- fails and the failed extension is executed.
-
-2007-08-30 14:43 +0000 [r81373] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: Fixed some warnings.
-
-2007-08-30 14:23 +0000 [r81369] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (issue #10599) Reported by: dimas Handle the
- -1 control subclass during feature dialing (it indicates to stop
- sounds).
-
-2007-08-30 08:31 +0000 [r81367] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
- issue where a misdn_read would lock the channel, but read would
- not return because it blocks. later chan_misdn would try to queue
- a frame like a AST_CONTROL_ANSWER which could result in a
- deadlock situation. misdn_read will now not block forever
- anymore, and we don't queue the ANSWER frame at all when we
- already was called with misdn_answer -> answer would be called
- twice. Also we don't explicitly send a RELEASE_COMPLETE on
- receiption of a RELEASE anymore, because mISDN does that for us,
- this resulted in a problem on some switches, which would block
- our port after some calls for a short while.
-
-2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch, in essence, will correctly pause a
- realtime queue member and reflect those changes in the realtime
- engine. (issue #10424, reported by irroot, patch by me) This
- patch creates a new function called update_realtime_member_field,
- which is a generic function which will allow any one field of a
- realtime queue member to be updated. This patch only uses this
- function to update the paused status of a queue member, but it
- lays the foundation for persisting the state of a realtime member
- the same way that static members' state is maintained when using
- the persistentmembers setting
-
- * apps/app_queue.c: Changed some tabs to spaces
-
-2007-08-29 15:57 +0000 [r81342] Russell Bryant <russell@digium.com>
-
- * main/Makefile: If chan_h323 is not being built, don't use g++ to
- do the final link of Asterisk. (in response to a question on the
- asterisk-dev list)
-
-2007-08-29 15:52 +0000 [r81340] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This fix creates a more accurate way of
- detecting whether realtime members were deleted. (closes issue
- 10541, reported by Alric, patched by me) The REALLY nice things
- about this patch is that queue members now have a "realtime"
- field which will be true if the member is a realtime member. This
- means we can check this value prior to certain processing if it
- should ONLY be done for realtime members.
-
-2007-08-29 14:13 +0000 [r81331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
- rtp timeouts work even if two RTP streams are directly bridged in
- the RTP stack.
-
-2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change the message about receiving a
- mini-frame before the first full voice frame to a DEBUG message.
-
- * pbx/pbx_dundi.c: revert unintentional changes in rev 81226
-
- * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
- tones. (closes issue #7953, hanabana)
-
-2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson <mmichelson@digium.com>
-
- * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
- res_config_mysql (closes issue #10573, reported by chrisvaughan,
- patch suggested by chrisvaughan as well)
-
- * apps/app_queue.c: Resolve a potential deadlock. In this case, a
- single queue is locked, then the queue list. In changethread(),
- the queue list is locked, and then each individual queue is
- locked. Under the right circumstances, this could deadlock. As
- such, I have unlocked the individual queue before locking the
- queue list, and then locked the queue back after the queue list
- is unlocked.
-
- * channels/chan_agent.c: DTMF begin frames should be ignored so
- that when an agent acks a call with the '#' key, he doesn't cause
- a queue's announce file to be interrupted. Also went ahead and
- did the same for the '*' key and for ending a call. (closes issue
- #10528, reported by deskhack, patched by me)
-
-2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
- of the memory it allocates on unload and should be fixed (when
- I'm not in the middle of something else ...).
-
- * pbx/pbx_dundi.c: explicity define a variable as a boolean
-
- * res/res_musiconhold.c: (closes issue #10419) Reported by:
- mustardman Patches: asterisk-mohposition.diff.txt uploaded by
- jamesgolovich (license 176) This patch fixes a few problems with
- music on hold. * Fix issues with starting at the beginning of a
- file when it shouldn't. * Fix the inuse counter to be decremented
- even if the class had not been set to be deleted when not in use
- anymore * Don't arbitrarily limit the number of MOH files to 255
-
-2007-08-27 15:01 +0000 [r81012] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
- Patches: chan_sip-ChannelReload-20080825.patch uploaded by
- jesselang (license 202) Remove an extra \r\n to make the
- ChannelReload event conform with every other event.
-
-2007-08-27 14:55 +0000 [r81010] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Found a case where the queue's membercount is
- off. It does not take into account dynamic members on a reload.
-
-2007-08-27 13:20 +0000 [r80974] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
- jitter value output in the CLI to be as expected.
-
-2007-08-26 18:11 +0000 [r80932] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove an extra signal_condition() for the
- scheduler thread. (closes issue #10564, patch from casper)
-
-2007-08-25 17:37 +0000 [r80895] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some issues with the handling of the
- scheduler in chan_iax2. Most of the places that scheduled items
- to be executed by the scheduler thread did not signal the
- scheduler thread to wake up so that it could recalculate the time
- until the next action. These changes will make the scheduler
- thread more responsive and ensure that actions get executed as
- close to when intended as possible instead of it being possible
- for very long delays.
-
-2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_zapateller.c: An empty string is an empty callerid ...
- so zap it. This closes issue #10502, which was pointed out by
- dswartz. Thank you, and may the swartz be with you
-
-2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
- available when Asterisk first starts, then don't give up on
- poking peers. Allow the poke to get rescheduled so that it will
- work once the dnsmgr is able to resolve the host. (closes issue
- #10521, patch by jamesgolovich)
-
- * main/dsp.c: Improve the debouncing logic in the DTMF detector to
- fix some reliability issues. Previously, this code used a shift
- register of hits and non-hits. However, if the start of the digit
- isn't clean, it is possible for the leading edge detector to miss
- the digit. These changes replace the flawed shift register logic
- and also does the debouncing on the trailing edge as well.
- (closes issue #10535, many thanks to softins for the patch)
-
-2007-08-24 19:52 +0000 [r80818] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: A minor correction to the available logic of
- autofill. If a queue member is paused, they're not really
- "available" so don't count them as such. Somewhat related to
- issue #10155
-
-2007-08-24 18:52 +0000 [r80789] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: From a complaint by jmls, I realize that the message
- in cdr_disposition is unnecessary. To get failure disposition,
- just return -1; no use having more than one case do that.
-
-2007-08-24 15:51 +0000 [r80750] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.
-
-2007-08-24 15:41 +0000 [r80747] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
- the code, instead of only in documentation (closes issue #10549)
-
-2007-08-24 15:28 +0000 [r80722] Russell Bryant <russell@digium.com>
-
- * utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
- I think this would have caused a "*" to get in the
- menuselect-tree file.
-
-2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy <murf@digium.com>
-
- * utils/ael_main.c: This change addresses JerJer's complaint that
- aelparse builds and installs even if pbx_ael is unchecked in the
- menuselect stuff.
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
- backport of 80649, a fix to an unreported problem in the ael
- parser, that results in a crash on a 64bit machine
-
-2007-08-24 11:42 +0000 [r80661] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
- answered too early, which results in CDRs wrongly stating that a
- call was ANSWERED when the calling party cancelled a call before
- before being established. We must not answer the call upon
- reception of a 'transport-accept' iq packet, but this packet
- still needs to be acknowledged, otherwise the remote peer would
- close the call (like in #8970).
-
-2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
- warnings
-
- * channels/chan_misdn.c: make chan_misdn compile without warnings
-
-2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant <russell@digium.com>
-
- * include/asterisk/features.h, res/res_features.c: When executing a
- dynamic feature, don't look it up a second time by digit pattern
- after we already looked it up by name. This causes broken
- behavior if there is more than one feature defined with the same
- digit pattern. (closes issue #10539, reported by bungalow, patch
- by me)
-
- * funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
- none of these values take ms so I don't know what I was thinking
-
- * funcs/func_timeout.c: Fix func_timeout to take values in floating
- point so 1.5 actually means 1.5 seconds instead of being rounded.
- (closes issue #10540, reported by spendergrass, patch by me)
-
-2007-08-23 17:14 +0000 [r80505-80507] Jason Parker <jparker@digium.com>
-
- * /: *sigh*
-
- * /: use autotagged externals
-
-2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: report the actual channel number that was
- unregistered, instead of assuming that the interface list
- consists of channels 1 through <x> with no gaps in the sequence
-
-2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some code where it was possible for a
- reference to a peer to not get released when it should. Thank you
- to Marta Carbone for pointing this out!
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- This is a hack to maintain old behavior of chan_iax2. This
- ensures that if the peers and users are being stored in a linked
- list, that they go in the list in the same order that the older
- code used. This is necessary to maintain the behavior of which
- peers and users get matched when traversing the container.
-
- * res/res_agi.c: Revert res_agi fix that didn't quite work until we
- get it right ...
-
- * include/asterisk/astobj2.h: Add some more documentation on
- iterating ao2 containers. The documentation implies that is
- possible to miss an object or see an object twice while
- iterating. After looking through the code and talking with
- mmichelson, I have documented the exact conditions under which
- this can happen (which are rare and harmless in most cases).
-
- * main/astobj2.c: When converting this code to use the list macros,
- I changed it so objects are added to the head of a bucket instead
- of the tail. However, while looking over code with mmichelson, we
- noticed that the algorithm used in ao2_iterator_next requires
- that items are added to the tail. This wouldn't have caused any
- huge problem, but it wasn't correct. It meant that if an object
- was added to a container while you were iterating it, and it was
- added to the same bucket that the current element is in, then the
- new object would be returned by ao2_iterator_next, and any other
- objects in the bucket would be bypassed in the traversal.
-
- * channels/chan_sip.c: Don't crash when using realtime in chan_sip
- without an insecure setting in the database. (closes issue
- #10348, reported by link55, fixed by me)
-
- * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
- (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
- channels/chan_iax2.c: Merge changes from
- team/russell/iax_refcount. This set of changes fixes problems
- with the handling of iax2_user and iax2_peer objects. It was very
- possible for a thread to still hold a reference to one of these
- objects while a reload operation tries to delete them. The fix
- here is to ensure that all references to these objects are
- tracked so that they can't go away while still in use. To
- accomplish this, I used the astobj2 reference counted object
- model. This code has been in one of Luigi Rizzo's branches for a
- long time and was primarily developed by one of his students,
- Marta Carbone. I wanted to go ahead and bring this in to 1.4
- because there are other problems similar to the ones fixed by
- these changes, so we might as well go ahead and use the new
- astobj if we're going to go through all of the work necessary to
- fix the problems. As a nice side benefit of these changes, peer
- and user handling got more efficient. Using astobj2 lets us not
- hold the container lock for peers or users nearly as long while
- iterating. Also, by changing a define at the top of chan_iax2.c,
- the objects will be distributed in a hash table, drastically
- increasing lookup speed in these containers, which will have a
- very big impact on systems that have a large number of users or
- peers. The use of the hash table will be made the default in
- trunk. It is not the default in 1.4 because it changes the
- behavior slightly. Previously, since peers and users were stored
- in memory in the same order they were specified in the
- configuration file, you could influence peer and user matching
- order based on the order they are specified in the configuration.
- The hash table does not guarantee any order in the container, so
- this behavior will be going away. It just means that you have to
- be a little more careful ensuring that peers and users are
- matched explicitly and not forcing chan_iax2 to have to guess
- which user is the right one based on secret, host, and access
- list settings, instead of simply using the username. If you have
- any questions, feel free to ask on the asterisk-dev list.
-
- * res/res_agi.c: Juggie in #asterisk-dev was reporting problems
- where fgets would return without reading the whole line when
- using fastagi. When this happens, errno was set to EINTR or
- EAGAIN. This patch accounts for the possibility and lets fgets
- continue in that case.
-
-2007-08-22 18:53 +0000 [r80302-80330] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
- build_tools/strip_nonapi, build_tools/prep_moduledeps,
- build_tools/make_buildopts_h: Fix a few build issues in Solaris
- (and likely others). Use GREP and ID variables from autoconf.
- Reported to me in #asterisk-dev I forgot who reported this -
- sorry. :(
-
- * Makefile: Change a syntax that the GNU make in Solaris dislikes.
-
- * build_tools/make_version: Fix a bashism (we explicitly request
- /bin/sh). Remove some oddly placed quotes I found in passing.
-
-2007-08-22 16:21 +0000 [r80257] Russell Bryant <russell@digium.com>
-
- * Makefile: Honor the contents of the COPTS variable as custom
- target CFLAGS. Apparently this is what openwrt does. (reported by
- Brian Capouch on the asterisk-dev list, patch by me)
-
-2007-08-22 16:14 +0000 [r80255] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10526) Reported by: sinistermidget
- Revert commit from issue #10355 and return timestamp skew to 640.
-
-2007-08-21 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.11 released.
-
-2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Don't record SIP dialog history if it's not
- turned on. Also, put an upper limit on how many history entires
- will be stored for each SIP dialog. It is currently set to 50,
- but can be increased if deemed necessary. (closes issue #10421,
- closes issue #10418, patches suggested by jmoldenhauer, patches
- updated by me) (Security implications documented in AST-2007-020)
-
-2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy <murf@digium.com>
-
- * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
- the diffs from ulaw.h and alaw.h for now; accidentally added them
- in 80166
-
- * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
- This patch solves problem 1 in 8126; it should not slow down the
- alaw codec, but should prevent signal degradation via multiple
- trips thru the codec. Fossil estimates the twice thru this codec
- will prevent fax from working. 4-6 times thru would result
- hearable, noticeable, voice degradation.
-
-2007-08-21 15:22 +0000 [r80132] Russell Bryant <russell@digium.com>
-
- * channels/chan_mgcp.c: Don't try to dereference the owner channel
- when it may not exist (issue #10507, maxper)
-
-2007-08-21 15:03 +0000 [r80130] Jason Parker <jparker@digium.com>
-
- * configs/cdr.conf.sample: (issue #10510) Reported by: casper
- Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
- errors in sample cdr config file.
-
-2007-08-20 21:57 +0000 [r80088] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix the build of app_queue
-
-2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: After a discussion on #asterisk-dev, it was
- decided that this should be in 1.4 as well. (issue #10424,
- reported and patched by irroot)
-
- * apps/app_queue.c: Found a pointless ternary if. member->dynamic
- was set to 1 and has no opportunity to change between then and
- this line, so "dynamic" will ALWAYS be output.
-
-2007-08-20 16:08 +0000 [r80047] Jason Parker <jparker@digium.com>
-
- * configs/extensions.conf.sample: (issue #10499) Reported by:
- casper Patches: extensions.conf.sample.diff uploaded by casper
- (license 55) Update CLI examples in extensions.conf.sample to
- reflect command changes.
-
-2007-08-20 15:34 +0000 [r80044] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Ukrainian language voicemail support.
- (closes issue #10458, reported and patched by Oleh)
-
-2007-08-20 02:42 +0000 [r79998] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
- snuffy via IRC)
-
-2007-08-18 14:30 +0000 [r79947] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Don't allocate vmu for messagecount when we
- could just use the stack instead (closes issue #10490) Also,
- remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
-
-2007-08-17 21:01 +0000 [r79912] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Avoid a crash in the handling of DTMF based
- Caller ID. It is valid for ast_read to return NULL in the case
- that the channel has been hung up. (crash reported by
- anonymouz666 on IRC in #asterisk-dev)
-
-2007-08-17 19:14 +0000 [r79906] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Patch allows for more seamless transition
- from file storage voicemail to ODBC storage voicemail. If a
- retrieval of a greeting from the database fails, but the file is
- found on the file system, then we go ahead an insert the greeting
- into the database. The result of this is that people who switch
- from file storage to ODBC storage do not need to rerecord their
- voicemail greetings.
-
-2007-08-17 19:12 +0000 [r79902-79904] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
- Don't send a semicolon over the wire in sip notify messages.
- Caused by fix for issue 9938. I basically took the code that
- existed before 9938 was fixed, and copied it into a new function
- - ast_unescape_semicolon There should be very few places this
- will be needed (pbx_config does NOT need this (see issue 9938 for
- details)) Issue 10430, patch by me, with help/ideas from murf
- (thanks murf).
-
- * channels/chan_local.c: Re-add the setting of callerid name and
- number. Issue 10485, reported by and fix explained by paradise.
-
-2007-08-17 13:37 +0000 [r79857] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix some crashes in chan_sip. This patch
- changes various places that add items to the scheduler to ensure
- that they don't overwrite the ID of a previously scheduled item.
- If there is one, it should be removed. (closes issue #10391,
- closes issue #10256, probably others, patch by me)
-
-2007-08-17 08:22 +0000 [r79833] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: sometimes we don't need to signal dtmf
- tones to asterisk, we just want them to go through as inband.
- Otherwise they might be generated by the other channel partner
- and then there is a double tone.
-
-2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Fix a little race condition that could
- cause a crash if two channels had MOH stopped at the same time
- that were using a class that had been marked for deletion when
- its use count hits zero.
-
- * res/res_musiconhold.c: This patch fixes a bug where reloading the
- module with "module reload" did not delete classes from memory
- that were no longer in the config. This patch fixes that problem
- as well as another one. Previously, if you reloaded MOH using the
- "moh reload" CLI command, which behaved differently than "module
- reload ...", MOH had to be stopped on every channel and started
- again immediately. However, there was no way to tell what class
- was being used, so they would all fall back to the default class.
- (closes issue #10139) Reported by: blitzrage Patches:
- asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
- (license 176) Tested by: jamesgolovich
-
- * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
- introduced by making frame handling and scheduling
- multi-threaded. Unfortunately, we have to do some expensive
- deadlock avoidance when queueing frames on to the ast_channel
- owner of the IAX2 pvt struct. This was already handled for
- regular frames, but ast_queue_hangup and ast_queue_control were
- still used directly. Making these changes introduced even more
- places where the IAX2 pvt struct can disappear in the context of
- a function holding its lock due to calling a function that has to
- unlock/lock it to avoid deadlocks. I went through and fixed all
- of these places to account for this possibility. (issue #10362,
- patch by me)
-
-2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Fixes a problem where agents would get
- stuck busy due to their wrapuptime being longer than the queue's
- wrapuptime and ringinuse=no for the queue. (closes issue #10215,
- reported by Doug, repaired by me) Special thanks to fkasumovic
- for pointing out the source of the problem and to bweschke for
- helping to come up with a solution!
-
- * apps/app_voicemail.c: base_encode is not trying to open a log
- file, so we should not call it a log file in the warning.
- (related to issue #10452, reported by bcnit)
-
-2007-08-16 09:37 +0000 [r79665] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: A fix for two critical problems detected while
- working with Daniel McKeehan in issue #10184. Upon priority
- change, the resource list is not NULL terminated when moving an
- item to the end of the list. This makes Asterisk endlessy loop
- whenever it needs to read the list. Jids with different resource
- and priority values, like in Gmail's and GoogleTalk's jabber
- clients put that problem in evidence. Upon reception of a 'from'
- attribute with an empty resource string, Asterisk crashes when
- trying to access the found->cap pointer if the resource list for
- the given buddy is not empty. This situation is perfectly valid
- and must be handled. The Gizmoproject's jabber client put that
- problem in evidence. Also added a few comments in the code as
- well as a handle for the capabilities from Gmail's jabber client,
- which are stored in a caps:c tag rather than the usual c tag.
- Closes issue #10184.
-
-2007-08-16 08:21 +0000 [r79642] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
- we want to send IA5 Chars.
-
-2007-08-15 14:40 +0000 [r79553] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10440) Reported by: irroot (closes
- issue #10454) Reported by: flo_turc Increase maximum timestamp
- skew to 120. 20 was apparently far too low.
-
-2007-08-15 14:26 +0000 [r79527] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed an error in the Russian language
- voicemail intro. (issue #10458, reported and patched by Oleh)
-
-2007-08-15 14:18 +0000 [r79523] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10456) Reported by: irroot
- Patches: sip_timeout.patch uploaded by irroot (license 52) Change
- hardcoded timer value to defined value. I'm doing this in 1.4 as
- well so if it needs to be changed in the future this place would
- not have been forgotten.
-
-2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another spot where an iax2_peer would
- be leaked if realtime was in use.
-
- * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
- related to the use of realtime. I found these while working on
- iax2_peer object reference tracking.
-
-2007-08-14 15:27 +0000 [r79397] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10415) Reported by: atis
- Revert fix for #10327 as it causes more issues then it solves.
-
-2007-08-13 22:40 +0000 [r79363] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: memset really, really needs to be used here.
-
-2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Instead of accepting a single DTMF
- character accept a full string.
-
-2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Don't call find_peer in
- registry_authrequest with the pvt lock held to avoid a deadlock.
-
- * channels/chan_iax2.c: Release the pvt lock before calling
- find_peer in register_verify to avoid a deadlock. Also, remove
- some unnecessary locking in auth_fail that was only done
- recursively.
-
- * channels/chan_iax2.c: Don't call find_peer within update_registry
- with a pvt lock held. This can cause a deadlock as the code will
- eventually call find_callno.
-
- * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
- have tracked them down to a single core issue. You can not call
- find_callno() while holding a pvt lock as this function has to
- lock another (every) other pvt lock. Doing so can lead to a
- classic deadlock. So, I am tracking down all of the code paths
- where this can happen and fixing them. The fix I committed
- earlier today was along the same theme. This patch fixes some
- code down the path of authenticate_reply.
-
-2007-08-13 17:49 +0000 [r79255] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
- pbx/ael/ael-test/ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
- fixes bug 10411. I added a new regression test, some regression
- test cleanups
-
-2007-08-13 15:28 +0000 [r79214] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a potential deadlock in socket_process.
- check_provisioning can eventually call find_callno. You can't
- hold a pvt lock while calling find_callno because it goes through
- and locks every single one looking for a match.
-
-2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Add an API call to allow the engine to
- know that DTMF was received.
-
- * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
- channels/chan_local.c, channels/chan_misdn.c,
- channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_h323.c, channels/chan_gtalk.c,
- channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
- Don't set the callerid name and number a second time on a newly
- created channel. ast_channel_alloc itself already sets it and
- setting it twice would cause a memory leak.
-
-2007-08-11 05:23 +0000 [r79142] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Ensure the connection gets marked as used at
- allocation time (closes issue #10429, report and fix by
- mnicholson)
-
-2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy <murf@digium.com>
-
- * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
- a user complaint on #asterisk, I have forced pbx_spool to explain
- what reason codes mean, when they are logged
-
- * main/cdr.c: Re bug behavior mentioned in #asterisk, made this
- tweak to code, to prevent hundreds of log messages from being
- generated
-
- * main/cdr.c: This will help debug; from a question asked on
- #asterisk
-
-2007-08-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.10.1 released.
-
-2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: The last set of changes that I made to
- "core show locks" made it not able to track mutexes unless they
- were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
- with ast_mutex_init() were not tracked. It should work now.
-
-2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp <jcolp@digium.com>
-
- * main/file.c: Don't bother having the core pass through or emulate
- begin DTMF frames when in an ast_waitstream. It only cares about
- the end of DTMF.
-
- * configs/queues.conf.sample: (closes issue #10422) Reported by:
- bhowell Add note to sample configuration about module load order
- and how it can cause perfectly good queue members to be marked as
- invalid.
-
-2007-08-10 13:24 +0000 [r78936] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
- information element. We send them now also in the disconnect
- message.
-
-2007-08-09 23:47 +0000 [r78907] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Improved a bit of logic regarding
- comma-separated mailboxes in has_voicemail. Also added some
- braces to some compound if statements since unbraced if
- statements scare me in general.
-
-2007-08-09 23:10 +0000 [r78891] Steve Murphy <murf@digium.com>
-
- * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
- output
-
-2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing some extra debug code I left in my
- last commit
-
- * apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
- 1. instead of using inboxcount as the core message counting
- function, we use messagecount instead. This makes it possible to
- count messages in folders besides just INBOX and Old. 2.
- inboxcount and hasvoicemail now use messagecount as their means
- of determining return values. 3. Added a copy_message function
- for IMAP storage. Unfortunately I don't have the means to test
- it, but it seems like a pretty straightforward function. 4.
- Removed a #ifndef IMAP_STORAGE and matching #endif from
- leave_voicemail for a couple of reasons. One, we want to support
- copying mail to multiple IMAP boxes, and two, IMAP was broken
- because a STORE macro had been moved into this section of code.
-
- * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
- put some new code in the wrong place and so I've reverted the
- canreinvite section to how it was and put my new code where it
- should be.
-
-2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: add a comment to indicate that inboxcount
- for ODBC_STORAGE needs to be fixed to support multiple mailboxes
-
- * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
- ODBC_STORAGE. Also, leave a comment for this to be fixed for
- IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
- working on this code right now for another reason. This is broken
- even worse in trunk, but for a different reason. The fact that
- the mailbox option supported multiple mailboxes is completely not
- obvious from the code in the channel drivers. Anyway, I will fix
- that in another commit ...
-
- * apps/app_meetme.c: Fix a problem with the combination of the 'F'
- option to pass DTMF through a conference and options that use
- DTMF to activate various features. The problem was that the BEGIN
- frame would be passed through, but the END frame would get
- intercepted to activate a feature. Then, the other conference
- members would hear DTMF for forever, which they didn't seem to
- like very much. (closes issue #10400, reported by stevefeinstein,
- fixed by me)
-
-2007-08-08 19:29 +0000 [r78646] Jason Parker <jparker@digium.com>
-
- * doc/jabber.txt: Fix mogs email address.
-
-2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed some compiler warnings so that
- compiling with dev-mode and IMAP storage would not have any
- errors. This section of code may get changed again shortly since
- my change uncovers a rather silly bit of logic.
-
- * apps/app_queue.c: Changing a bit of logic so that someone will
- NEVER exit the queue on timeout unless they have enabled the 'n'
- option. This commit relates to issue #10320. Thanks to
- jfitzgibbon for detailing the idea behind this code change.
-
-2007-08-08 13:51 +0000 [r78569] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample: (closes issue #10335) Reported by:
- adamgundy Update sip.conf to include another scenario where
- directrtpsetup will fail.
-
-2007-08-07 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.10 released.
-
-2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c: Fix the build of this module on 64-bit
- platforms
-
-2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: The logic behind inboxcount's return value
- was reversed in has_voicemail and message_count. (closes issue
- #10401, reported by st1710, patched by me)
-
-2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Don't free the environment handle when the
- connection fails, because other connections might be depending
- upon it
-
-2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Allow chan_sip to build in devmode
-
-2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, res/res_config_odbc.c,
- apps/app_directory.c: Reconnection doesn't happen automatically
- when a DB goes down (fixes issue #9389)
-
-2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Properly check the capabilities count to
- avoid a segfault. (ASA-2007-019)
-
-2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
- 4 lines Revert patch committed for issue #9660. It broke E&M
- trunks. (closes issue #10360) (closes issue #10364) ........
-
-2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Add additional DTMF log messages to help when
- debugging issues.
-
-2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix an issue where dynamic threads can get
- free'd, but still exist in the dynamic thread list. (closes issue
- #10392, patch from Mihai, with credit to his colleague, Pete)
-
- * include/asterisk/linkedlists.h: Fix the return value of
- AST_LIST_REMOVE(). This shouldn't be causing any problems,
- though, because the only code that uses the return value only
- checks to see if it is NULL. (closes issue #10390, pointed out by
- mihai)
-
-2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for a transfer to occur
- before the remote device has our tag in which case they send none
- in the transfer. In this case we need to not fail the transfer
- dialog lookup.
-
-2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker@digium.com>
-
- * main/config.c: Fix an issue with using UpdateConfig (manager
- action) where escaped semicolons in a config would be converted
- to just semicolons (\; to ;) Issue 9938
-
-2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
- we pass through RTP timestamp information we need to make the
- allowed timestamp skew considerably less. There are situations
- where a source may change and due to the timestamp difference the
- receiver will experience an audio gap since we did not indicate
- by setting the marker bit that the source changed.
-
- * configure, configure.ac: (closes issue #10383) Reported by: rizzo
- Include stdlib.h so NULL gets defined for gethostbyname_r checks.
-
-2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fixed a mistake I made in realtime_peer
- which caused it to return NULL every time. Thanks to Jon Fealy
- for emailing me the correction.
-
-2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
- bug #10382)
-
-2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Fix compilation failure when
- MALLOC_DEBUG is enabled, but DEBUG_THREADS is not
-
-2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: If peer is not found, the error message is
- misleading (should be peer not found, not ACL failure)
-
-2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c, channels/chan_sip.c, include/asterisk/config.h:
- Changed the behavior of sip's realtime_peer function to match the
- corresponding way of matching for non-realtime peers. Now matches
- are made on both the IP address and port number, or if the
- insecure setting is set to "port" then just match on the IP
- address. In order to accomplish this, I also added a new API
- call, ast_category_root, which returns the first variable of an
- ast_category struct
-
-2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: (closes issue #10194) Reported by:
- blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
- blitzrage Fix a problem when you call Voicemail() with multiple
- mailboxes specified and ODBC_STORAGE is in use. The audio part of
- the message was only given to the first mailbox specified.
-
- * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
- improvements to lock debugging. These changes take effect with
- DEBUG_THREADS enabled and provide the following: * This will keep
- track of which locks are held by which thread as well as which
- lock a thread is waiting for in a thread-local data structure. A
- reference to this structure is available on the stack in the
- dummy_start() function, which is the common entry point for all
- threads. This information can be easily retrieved using gdb if
- you switch to the dummy_start() stack frame of any thread and
- print the contents of the lock_info variable. * All of the
- thread-local structures for keeping track of this lock
- information are also stored in a list so that the information can
- be dumped to the CLI using the "core show locks" CLI command.
- This introduces a little bit of a performance hit as it requires
- additional underlying locking operations inside of every
- lock/unlock on an ast_mutex. However, the benefits of having this
- information available at the CLI is huge, especially considering
- this is only done in DEBUG_THREADS mode. It means that in most
- cases where we debug deadlocks, we no longer have to request
- access to the machine to analyze the contents of ast_mutex_t
- structures. We can now just ask them to get the output of "core
- show locks", which gives us all of the information we needed in
- most cases. I also had to make some additional changes to astmm.c
- to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
- enabled. I disabled tracking of one of the locks in astmm.c
- because it gets used inside the replacement memory allocation
- routines, and the lock tracking code allocates memory. This
- caused infinite recursion.
-
- * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
- frames when the mohinterpret option is set to "passthrough". This
- was pointed out by Kevin in the middle of a training session.
-
- * channels/chan_iax2.c: Don't reuse the timespec that was set to 0
- in the previous timedwait as it will just return immediately.
- Also, fix some logic so the thread's lock isn't unlocked twice in
- the weird case of dynamic threads getting acquired right after a
- timeout. (pointed out by SteveK)
-
-2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
- actually allow 6 chars to be sent. Also make note of the "A"
- option of date format. Issue 9779, modifications by DEA, wedhorn,
- and myself.
-
- * channels/chan_skinny.c: If a device disconnects, the session will
- go away. If this happens during call setup, we need to give up.
- Issue 10325.
-
-2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix the case where a dynamic thread times
- out waiting for something to do during the first time it runs.
- This shouldn't ever happen, but we should account for it anyway.
- (pointed out by pete, who works with mihai)
-
-2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure we clear the prompt status
- message on a hangup. Also rearrange messages to better fit with
- what a wireshark trace shows it should be. Issue 10299, initial
- patch and solution by sbisker, modified by me to fit with
- wireshark trace.
-
-2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf@digium.com>
-
- * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
- line This patch hopefully solves 10141; The user is running with
- it, and it doesn't appear to harm asterisk's operation, and may
- prevent a crash. I'll store it in 1.2, as we have shut down
- support on 1.2, but since I developed the patch before support
- finished, and it might affect 1.4 and trunk, I'm going ahead with
- it. ........
-
-2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another race condition in the handling
- of dynamic threads. If the dynamic thread timed out waiting for
- something to do, but was acquired to perform an action
- immediately afterwords, then wait on the condition again to give
- the other thread a chance to finish setting up the data for what
- action this thread should perform. Otherwise, if it immediately
- continues, it will perform the wrong action. (reported on IRC by
- mihai, patch by me) (related to issue #10289)
-
- * channels/chan_iax2.c: Add another sanity check to
- vnak_retransmit(). This check ensures that frames that have
- already been marked for deletion don't get retransmitted. (closes
- issue #10361, patch from mihai)
-
-2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure that we show the correct
- extension if dialed from a macro "From: 5555" rather than "From:
- s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
- tested by sbisker
-
- * channels/chan_skinny.c: Put in some additional debug information
- for softkey/stimulus messages. Issue 10291, patch by DEA.
-
-2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some race conditions which have been
- causing weird problems in chan_iax2. The most notable problem is
- that people have been seeing storms of VNAK frames being sent due
- to really old frames mysteriously being in the retransmission
- queue and never getting removed. It was possible that a dynamic
- thread got created, but did not acquire its lock before the
- thread that created it signals it to perform an action. When this
- happens, the thread will sleep until it hits a timeout, and then
- get destroyed. So, the action never gets performed and in some
- cases, means a frame doesn't get transmitted and never gets freed
- since the scheduler never gets a chance to reschedule
- transmission. Another less severe race condition is in the
- handling of a timeout for a dynamic thread. It was possible for
- it to be acquired to perform at action at the same time that it
- hit a timeout. When this occurs, whatever action it was acquired
- for would never get performed. (patch contributed by Mihai and
- SteveK) (closes issue #10289) (closes issue #10248) (closes issue
- #10232) (possibly related to issue #10359)
-
-2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
- not compile cleanly (missing def)
-
-2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue that caused one-way audio on
- some newer devices (specifically the 7921), due to sending
- packets in the wrong order during hangup. Also make sure we clear
- tones/messages on the correct line/instance. Issue 10291, patch
- by DEA, tested by sbisker and myself.
-
-2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: (closes issue #10351) Reported by: ftarz Some
- platforms don't like it when you pass NULL to vsnprintf so pass
- "" instead.
-
- * include/asterisk/threadstorage.h, channels/chan_mgcp.c,
- apps/app_voicemail.c, main/acl.c, utils/smsq.c,
- channels/chan_iax2.c: Add some fixes for building on Solaris.
-
- * main/utils.c: Whoops, I meant R_5 not R5.
-
- * configure, configure.ac: And for my last trick... make sure that
- if gethostbyname_r is exported by a library that it is used.
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/utils.c: Extend autoconf logic to determine which version of
- gethostbyname_r is on the system.
-
-2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixes an issue I introduced to queues wherein a
- queue with joinempty=yes would kick people out of the queue
- because of erroneously thinking the 'n' option was in use.
- (closes issue #10320, reported by jfitzgibbon, patched by me,
- tested by blitzrage and me) Thank you blitzrage for all the
- testing you've done lately with queues! It's much appreciated!
-
- * apps/app_queue.c: If a queue uses dynamic realtime members, then
- the member list should be updated after each attempt to call the
- queue. This fixes an issue where if a caller calls into a queue
- where no one is logged in, they would wait forever even if a
- member logged in at some point. (closes issue #10346, reported by
- and tested by blitzrage, patched by me)
-
-2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c: Much newer version, 0.70 with much additions
-
- * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
- receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
- logic in TONE_VERIFY/RELAX mode in chan_zap.
-
-2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf@digium.com>
-
- * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
- line This probably isn't super-general, but it's a first stab at
- using kill -11 to generate a core file instead of gcore. ........
-
-2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, include/asterisk/speech.h: Add a flag to the
- speech API that allows an engine to set whether it received
- results or not.
-
-2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
- DEBUG_THREADS or it does nothing
-
-2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This patch makes Asterisk send 100 Trying
- provisional responses upon receipt of re-invites. This makes it
- so that if there are two or more Asterisk servers between
- endpoints, the Asterisk servers will not keep retransmitting the
- re-invites. (closes issue #10274, reported by cstadlmann, patched
- by me with approval from file)
-
-2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker@digium.com>
-
- * main/say.c: Applications like SayAlpha() should not hang up the
- channel if you request an "unknown" character such as a comma.
- Instead, skip the character and move on. Issue 10083, initial
- patch by jsmith, modified by me.
-
-2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix an issue that could potentially cause
- corruption of the global iax frame queue. In the network_thread()
- loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
- macro. However, to remove an element of the list within this
- loop, it used AST_LIST_REMOVE, instead of
- AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
- internal variables of the SAFE macro invalid. Mihai says that he
- already made this change in his local copy and it didn't help his
- VNAK storm issues, but I still think it's wrong. :)
-
- * res/res_agi.c: (closes issue #10279) Reported by: seanbright
- Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
- seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
- uploaded by seanbright (license 71) Allow the "agi_network: yes"
- line to be printed out in the AGI debug output. Also, allow
- partial writes to be handled when writing out this line just like
- it is for all of the others.
-
- * main/channel.c: file and I both committed changes for issue
- #10301. Remove a duplicated assignment to restore the original
- value of the previous channel.
-
-2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, res/res_agi.c: Merged revisions 77782 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
- | 2 lines Revert change in revision 71656, even though it fixed a
- bug, because many people were depending upon the (broken)
- behavior. ........
-
-2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell@digium.com>
-
- * main/channel.c: (closes issue #10301) Reported by: fnordian
- Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
- (license 110) Additional changes by me Fix some problems in
- channel_find_locked() which can cause an infinite loop. The
- reference to the previous channel is set to NULL in some cases.
- These changes ensure that the reference to the previous channel
- gets restored before needing it again. I'm not convinced that the
- code that is setting it to NULL is really the right thing to do.
- However, I am making these changes to fix the obvious problem and
- just leaving an XXX comment that it needs a better explanation
- that what is there now.
-
-2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10327) Reported by: kkiely
- Instead of directly mucking with the extension/context/priority
- of the channel we are transferring when it has a PBX simply call
- ast_async_goto on it. This will ensure that the channel gets
- handled properly and sent to the right place.
-
- * main/channel.c: (closes issue #10301) Reported by: fnordian
- Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
- (license 110) Restore previous behavior where if we failed to
- lock the channel we wanted we would return to exactly the same
- point as if we had just reentered the function.
-
- * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
- lines (closes issue #10334) Reported by: ramonpeek Pass through
- the return value from macro_exec through the MacroIf application.
- ........
-
-2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Missing newline
-
-2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
- Patches: cdr_pgsql.patch uploaded by prashant (license 114)
- Finish the Postgresql connection after the log messages are
- printed so we don't access invalid memory.
-
- * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
- Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
- julianjm (license 99) Clear ONHOLD flag when decrementing the
- onHold peer count. If we did not do this the count may keep
- decreasing.
-
-2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: "re-invite" was misspelled
-
-2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10302) Reported by: litnialex If a
- DTMF end frame comes from a channel without a begin and it is
- going to a technology that only accepts end frames (aka INFO)
- then use the minimum DTMF duration if one is not in the frame
- already.
-
-2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/mp3.txt: change protocol for downloads as well
-
- * doc/mp3.txt, sounds/Makefile: use new canonical name for download
- server
-
-2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
- in trunk, but not in 1.4. When Asterisk is compiled under dev
- mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
- define it in the same way that trunk does. Also, revert the
- change that added this define in the Makefile The advantage to
- doing it this way is that buildopts.h gets installed when you
- install Asterisk. Then, when building any out of tree modules, or
- building asterisk-addons, these modules know which options the
- rest of Asterisk was built with.
-
-2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com>
-
- * Makefile, main/logger.c: Fixes to get ast_backtrace working
- properly. The AST_DEVMODE macro was never defined so the majority
- of ast_backtrace never attempted compilation. The makefile now
- defines AST_DEVMODE if configure was run with --enable-dev-mode.
- Also, changes were made to acccomodate 64 bit systems in
- ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
- their roles in allowing me to get this committed
-
-2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/logger.c: Missed one
-
- * main/logger.c: Oops, that builtin define should be all-lowercase.
-
-2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com>
-
- * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
- meaning free gets called on the same variable twice. This patch
- sets the connection to NULL after calls to PQfinish so that the
- problem does not occur. Also in this patch, prashant_jois
- informed me that it is safe to pass a null pointer to PQfinish,
- so I have removed the check for conn's existence from
- my_unload_module. (closes issue 10295, reported by junky, patched
- by me with input from prashant_jois)
-
-2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com>
-
- * apps/app_meetme.c: This fix solves problem with intense squelch
- noise when someone joins conf in bug 9430; We repro'd the problem
- with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
- I'm applying it. It looks like playing the recorded username will
- louse up the next thing played into the channel. Josh rearranged
- the code so as to start things over before playing data directly
- into the conference.
-
-2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
- Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
- digit to terminate a DTMF string with. If none is specified then
- no terminator will be used.
-
-2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
- not a signed int, so use %u instead of %d in the format string
-
-2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com>
-
- * /: so are my fingers...
-
- * /: autotagexternals script is still obviously misbehaving...
-
- * /: use autotagged externals
-
-2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com>
-
- * configure, acinclude.m4: Fix autoconf logic for finding OpenH323
- when it is not in the first place searched (/usr/share/openh323).
-
-2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: set the sequence number in a frame for all frame
- types
-
-2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
- line this fixes bug 10293, where the error message because
- defaultzone or loadzone was not defined was confusing ........
-
-2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
- Jul 2007) | 2 lines Oops, res contains the error code, not errno.
- I was wondering why a mutex was reporting "No such file or
- directory"... ........
-
- * main/app.c: Found another place where we should be using the
- umask (thanks jcmoore)
-
-2007-07-24 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.4.9 released.
-
-2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker@digium.com>
-
- * /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell |
- 2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't
- create the Asterisk channel until we are starting the PBX on it.
- (ASA-2007-018) ........
-
- * channels/chan_iax2.c: Don't create the Asterisk channel until we
- are starting the PBX on it. (ASA-2007-018)
-
-2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added a membercount variable to call_queue
- struct which keeps track of the number of logged in members in a
- particular queue. This makes it so that the 'n' option for
- Queue() can act properly depending on which strategy is used. If
- the strategy is roundrobin, rrmemory, or ringall, we want to ring
- each phone once before moving on in the dialplan. However, if any
- other strategy is used, we will only ring one phone since it
- cannot be guaranteed that a different phone will ring on
- subsequent attempts to ring a phone. As a side effect of this,
- the QUEUE_MEMBER_COUNT dialplan function now just reads the
- membercount variable instead of traversing through the member
- list to figure out how many members there are. Special thanks to
- blitzrage for helping to test this out. (closes issue #10127,
- reported by bcnit, patched by me, tested by blitzrage)
-
-2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: It was our stated intention for 1.4 that
- files created in app_voicemail should depend upon the umask.
- Unfortunately, mkstemp() creates files with mode 0600, regardless
- of the umask. This corrects that deficiency.
-
-2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix some incorrect softkey labels in
- messages. Don't try to play dialtone in some unimplemented
- features.
-
-2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
- 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
- force channel format changes when a generator is present. The
- generator may have changed the formats itself and changing them
- back would cause issues. ........
-
-2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Don't try to queue up hold/unhold frames
- on a non-existent channel. Issue 10276.
-
-2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com>
-
- * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
- by putting the value of the digit char in an int.
-
- * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
- lines (closes issue #10236) Reported by: homesick Patches:
- rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
- Remote Party ID on guest calls. ........
-
- * channels/chan_skinny.c: (closes issue #10268) Reported by:
- mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
- (license 7) Add another OS that has to use the Macros for byte
- ordering.
-
-2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use a signed integer for storing the number
- of bytes in the packet read from the network. Using an unsigned
- value here made it impossible to handle an error returned from
- recvfrom(). Furthermore, in the case that recvfrom() did return
- an error, this would cause a crash due to a heap overflow.
- (closes issue #10265, reported by and fix suggested by
- timrobbins)
-
-2007-07-22 21:42 +0000 [r76410] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /: Blocked revisions 76409 via svnmerge ........ r76409 |
- tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines
- We should not use C++ reserved words in API headers (closes issue
- #10266) ........
-
-2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
- 4 lines Backport a fix for a memory leak that was fixed in trunk
- in reivision 76221 by rizzo. The memory used for the localaddr
- list was not freed during a configuration reload. ........
-
-2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com>
-
- * sounds/Makefile: This patch from 10249 is worth applying! It
- prevents downloading sound files if they are already downloaded.
- Darn Practical, if you ask me
-
-2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Allow getting a call from an existing
- "sub" channel. Cancel ringing if endpoint hangs up before
- answering. Fixes were backported from trunk (there was apparently
- a bit of confusion during merge of a previous patch). (closes
- issue #10241)
-
- * main/manager.c: Eliminate a compiler warning with gcc 4.2 by
- constifying a char *
-
- * channels/chan_skinny.c: It's possible for sub->owner to be NULL
- here if you cancel the call immediately after/during sending a
- digit.
-
-2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_directory.c: When using users.conf for the entries in
- the directory, if multiple users had the same last name, only the
- first user listed would be available in the directory. (closes
- issue #10200, reported by mrskippy, patched by me)
-
-2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Use the define that specifies the default length
- of an artificially created DTMF digit in the ast_senddigit()
- function. The define is set to 100ms by default, which is the
- same thing that this function was using. But, using the define
- lets changes take effect in this case, as well as the others
- where it was already used.
-
-2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
- lines (closes issue #10247) Reported by: fkasumovic Patches:
- chan_sip.patch uploaded by fkasumovic (license #101) Drop any
- peer realm authentication entries when reloading so multiple
- entries do not get added to the peer. ........
-
- * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
- Patches: res_conver.patch uploaded by fkasumovic (license #101)
- Use the last occurance of . to find the extension, not the first
- occurance.
-
- * apps/app_queue.c: Move makeannouncement variable declaration to
- proper place.
-
-2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Remove some duplicate code.
-
-2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: The diff on this looks pretty big but all I did
- was remove a pointless if statement (always evaluates true).
-
- * apps/app_queue.c: Changes in handling return values of several
- functions in app_queue. This all started as a fix for issue
- #10008 but now includes all of the following changes: 1.
- Simplifying the code to handle positive return values from ast
- API calls. 2. Removing the background_file function. 3. The fix
- for issue #10008 (closes issue #10008, reported and patched by
- dimas)
-
-2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
- 6 lines When processing full frames, take sequence number
- wraparound into account when deciding whether or not we need to
- request retransmissions by sending a VNAK. This code could cause
- VNAKs to be sent erroneously in some cases, and to not be sent in
- other cases when it should have been. (closes issue #10237,
- reported and patched by mihai) ........
-
-2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Need to make sure we set milliseconds and
- timestamp - pointed out by the recent ast_ time stuff from
- Tilghman
-
-2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
- 5 lines When traversing the queue of frames for possible
- retransmission after receiving a VNAK, handle sequence number
- wraparound so that all frames that should be retransmitted
- actually do get retransmitted. (issue #10227, reported and
- patched by mihai) ........
-
-2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
- | 2 lines Store prior to copy (closes issue #10193) ........
-
-2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Umm, why are we transmitting dialtone on
- cfwdall?
-
-2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
- pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
- Asterisk won't build under devmode using GCC 4.2.
-
-2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
- Issue 9245, patch by slimey.
-
- * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
- 9887, patches by DEA
-
-2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
- | 1 line removed the word 'pissed' from ast_log(...) function
- call for BE-90 ........
-
-2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Few more places that needs to check for
- onhold state.
-
- * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
- It is possible for hold status to exist without call limits set,
- so we need to ensure update_call_counter is executed regardless.
-
- * channels/chan_h323.c: Don't bother reloading chan_h323 if it did
- not load successfully in the first place. This would otherwise
- cause a crash.
-
- * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
- the threadid of each running thread before shutting them down as
- the thread themselves may change the value.
-
-2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_meetme.c: Using a freed frame causes crashes (closes
- issue #9317)
-
-2007-07-17 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.8 released.
-
-2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
- Jul 2007) | 3 lines Properly check for the length in the skinny
- packet to prevent an invalid memcpy. (ASA-2007-016) ........
-
- * main/rtp.c: cast arguments to ast_log so that it builds without
- warnings for me
-
- * channels/iax2-parser.c, channels/iax2-parser.h, /,
- channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
- 5 lines Ensure that when encoding the contents of an ast_frame
- into an iax_frame, that the size of the destination buffer is
- known in the iax_frame so that code won't write past the end of
- the allocated buffer when sending outgoing frames. (ASA-2007-014)
- ........
-
- * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
- 4 lines After parsing information elements in IAX frames, set the
- data length to zero, so that code later on does not think it has
- data to copy. (ASA-2007-015) ........
-
-2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Ensure that the pointer to STUN data does not go to
- unaccessible memory. (ASA-2007-017)
-
-2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (issue #10210) Reported by: juggie Patches:
- 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
- juggie, blitzrage Log a warning if someone uses DeadAGI on a live
- channel.
-
-2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
- can return -1 on failure, so I need to be sure that we only enter
- the if statement if it is successful. Related to my fix to issue
- #10186
-
-2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
- 10209-trunk-2.patch uploaded by juggie Tested by: juggie,
- blitzrage In ast_pbx_run(), mark a channel as hung up after an
- application returned -1, or when it runs out of extensions to
- execute. This is so that code can detect that this channel has
- been hung up for things like making sure DeadAGI is used on
- actual dead channels, and is beneficial for other things, like
- making sure someone doesn't try to start spying on a channel that
- is about to go away.
-
- * res/res_agi.c: Remove a duplicated newline character in AGI debug
- output. (closes issue #10207, patch by seanbright)
-
-2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dns.c, /: Merged revisions 75304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
- | 3 lines provide proper copyright/license attribution for this
- structure that was copied from a BSD-licensed header file long,
- long ago... ........
-
- * /: another fix that is not needed here (finishing up 75251)
-
-2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Restoring functionality from 1.2 wherein
- Retrydial will not exit if there is no announce file specified.
- This change makes it so that if there is no announce file
- specified, the application will continue until finished (or
- caller hangs up). If a bogus announce file is specified, then a
- warning message will be printed saying that the file could not be
- found, but execution will still continue. (closes issue #10186,
- reported by jon, patched by me)
-
-2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block change that is not relevant here
-
-2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
- Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
- load_moh_classes() could return without destroying the loaded
- configuration. ........
-
-2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
- 2007) | 5 lines Fixed an issue where chanspy flags were
- uninitialized if no options were passed. What triggered this
- investigation was an IRC chat where some people's quiet flags
- were set while others' weren't even though none of them had
- specified the q option. ........
-
-2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
- Jul 2007) | 6 lines Ensure that adding a user to the list of
- users of a specific music on hold class is not done at the same
- time as any of the other operations on this list to prevent list
- corruption. Using the global moh_data lock for this is not ideal,
- but it is what is used to protect these lists everywhere else in
- the module, and I am only changing what is necessary to fix the
- bug. ........
-
- * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
- 12 lines (closes issue #9660) Reported by: mmacvicar Patches
- submitted by: bbryant, russell Tested by: mmacvicar, marco,
- arcivanov, jmhunter, explidous When using a TDM400P (and probably
- other analog cards) there was a chance that you could hang up and
- pick the phone back up where it has been long enough to be not
- considered a flash hook, but too soon such that the device
- reports that it is busy and the person on the phone will only
- hear silence. This patch makes chan_zap more tolerant of this and
- gives the device a couple of seconds to succeed so the person on
- the phone happily gets their dialtone. ........
-
-2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Change to my previous fix regarding agent
- logoff soft. Now uses deferlogoff instead of loginstart since
- loginstart is used after logoff. Thanks to makoto for pointing
- this out and suggesting the fix. (closes issue #10178, reported
- and patched by makoto, with modification by me)
-
-2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This patch resolves 10143; thanks to irroot
- for the patch; looked acceptable. Let the community decide if it
- messes things up
-
-2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Whoops... didn't want this to be returned to 0
- each iteration.
-
- * main/channel.c: When waiting for a digit ensure that a begin
- frame was received with it, not just an end frame. (issue #10084
- reported by rushowr)
-
-2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: It helps if I actually add this stuff for
- the 7921 too - otherwise it won't actually do much of anything.
-
- * channels/chan_skinny.c: Add device ID for 7921 wireless skinny
- phone
-
- * channels/chan_skinny.c: Fix dialing in skinny that was broken in
- some cases. Issue 10136, fix provided by DEA.
-
-2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
- 2007) | 2 lines Only print out a warning for situations where it
- is actually helpful. (issue #10187 reported by denke) ........
-
-2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
- 5 lines The function make_trunk() can fail and return -1 instead
- of a valid new call number. Fix the uses of this function to
- handle this instead of treating it as the new call number. This
- would cause a deadlock and memory corruption. (possible cause of
- issue #9614 and others, patch by me) ........
-
-2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
- Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
- did not work...at all. Now it does. (closes issue #10178,
- reported and patched by makoto, with slight modification for 1.4
- and trunk by me) ........
-
-2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
- Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
- the condition that uses LIKE. This fixes realtime extensions with
- ODBC. (closes issue #10175, reported by stuarth, patch by me)
- ........
-
-2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com>
-
- * Makefile: This fixes 10172, where the entire man8 dir gets
- removed during an uninstall of asterisk
-
- * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
- further reversion of previously applied floating point stuff for
- expr2
-
-2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 74587 via svnmerge ........ r74587 | file |
- 2007-07-11 14:15:11 -0300 (Wed, 11 Jul 2007) | 2 lines Use some
- Makefile magic to determine if linux/compiler.h is present.
- (issue #10174 reported by francesco_r) ........
-
- * channels/chan_phone.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Instead of
- figuring out kernel versions that have compiler.h and not...
- let's just use autoconf to check for it's presence. (issue #10174
- reported by francesco_r)
-
- * channels/chan_phone.c: Only check if we need to do a SIGMA based
- tone generation if we have a card. (issue #10179 reported by
- mikowhy)
-
-2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Forwarding a message with IMAP storage was
- storing the message in the sender's box instead of the forwarded
- mailbox. (closes issue #10138, reported and patched by jaroth)
-
-2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
- lines Fix an issue where it was possible to have a service level
- of over 100% Between the time recalc_holdtime and update_queue
- was called, it was possible that the call could have been hungup.
- Move both additions to the same place, so this won't happen.
- Issue 10158, initial patch by makoto, modified by me. ........
-
- * main/dns.c: Don't use #if to check if something is defined - use
- #ifdef instead. Pointed out by kpfleming
-
- * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
- 2007) | 4 lines Fix an issue with wrapuptime not working when
- using AgentLogin. Issue 10169, patch by makoto, with a minor mod
- by me to not re-break issue 9618 ........
-
- * main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Merged revisions 74373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
- lines Use res_ndestroy on systems that have it. Otherwise, use
- res_nclose. This prevents a memleak on NetBSD - and possibly
- others. Issue 10133, patch by me, reported and tested by scw
- ........
-
-2007-07-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7.1 released.
-
-2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: fix an uninitialized variable
-
-2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
- lines Fix a small typo in description in of Voicemail()
- application. Issue 10170, patch by casper. ........
-
-2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
- Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
- #10075, this part reported by jmls on IRC, patch by me) ........
-
-2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com>
-
- * /, main/app.c: Merged revisions 74264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
- lines Ensure the group information category exists before trying
- to do a string comparison with it. (issue #10171 reported by
- mlegas) ........
-
- * channels/chan_sip.c: Only spit out an inringing warning message
- when it is applicable. Since call limits are already toast in
- realtime let's not scare the user if they are using it. (issue
- #10166 reported by bcnit)
-
-2007-07-09 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7 released.
-
-2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script to check for
- a required function that is not present in the 1.2 version of
- libpri. This will prevent the configure script from thinking that
- it has compatible libpri support for Asterisk 1.4, when it
- actually does not because the installed version is from 1.2.
-
- * /: Blocked revisions 74165 via svnmerge ........ r74165 | russell
- | 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the
- specified class isn't found, properly fall back to the channel's
- music class or the default. (issue #10123, reported by blitzrage,
- patches from juggie, qwell, and me) ........
-
- * res/res_musiconhold.c: (closes issue #10123) Reported by:
- blitzrage Patches submitted by: juggie, qwell, me Tested by:
- blitzrage When trying to find a music on hold class to use, try
- all of the options, instead of only the first one that is set.
- Also, change the MusicOnHold applications to not hang up on the
- channel when a class can not be found.
-
-2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
- lines Several chan_zap options were not working on reload because
- they were arbitrarily disallowed when reloading some/most PRI
- options (such as signalling) was disallowed. Options such as
- polarityonanswerdelay and answeronpolarityswitch can safely be
- changed on a reload. This corrects that behavior. Issue 9186,
- patch by tzafrir. ........
-
-2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot to get rid of an extraneous debug
- message.
-
- * apps/app_queue.c: The n option for Queue should make the queue
- exit immediately after failure to reach any members and should
- not be dependent on the timeout value passed to Queue (closes
- issue #10127, reported by bcnit, repaired by me)
-
-2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c: Only destroy the scheduler context if it
- was allocated. (issue #10124 reported by gzero)
-
-2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
- Pass the mailbox instead of the context to inbox_count when the
- context is "default." (closes issue #10135, reported by yannj,
- repaired by me)
-
-2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
- synchronization tweaks. (issue #10124 reported by gzero)
-
- * configure, acinclude.m4: Use AC_CHECK_HEADER to check for
- ptlib/openh323 to allow for cross compiling. (issue #9675
- reported by zandbelt)
-
-2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
- 'make progdocs'. (Closes issue #10104)
-
-2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Give Agent channel names priority when doing CDR
- merging. (issue #10011 reported by krtorio)
-
- * pbx/pbx_config.c: Add a few sanity checks when writing out the
- dialplan. (issue #10157 reported by dome)
-
-2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: While tracking down a bug, I need some more
- history. Dumphistory is very useful, indeed.
-
-2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
- 4 lines If a sip_pvt struct has already registered an extension
- state callback, remove the old one before adding a new one. If
- this isn't done, Asterisk will crash. (issue #10120) ........
-
-2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing a rare case which causes voicemail
- to crash when compiled with IMAP storage. inboxcount has the
- possibility of finding an "interactive" vm_state when no
- persistent "non-interactive" vm_state exists for that mailbox. If
- this should happen when someone attempts to leave a message, it
- results in a crash. This patch, along with my commit in revision
- 72670 fix issue 10053, reported by jaroth. closes issue #10053
-
-2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
- Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
- Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
- with MSSQL 2005 by explicitly stating that '\' is being used as
- an escape character. ........
-
- * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
- 7 lines (closes issue #10125) Reported by: makoto Patches
- submitted by: makoto This fixes a crash in chan_sip that happens
- when the bindaddr setting is not valid on Asterisk startup, gets
- fixed, and then a reload gets issued. ........
-
-2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
- Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
- (issue 9618, reported by jiddings, patched by moi) closes issue
- #9618 ........
-
-2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com>
-
- * BUGS: fix a little spelling error
-
- * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
- the monitor thread if it was never started. (closes issue #10124,
- reported by gzero, fixed by me)
-
- * channels/chan_iax2.c: copy from the correct buffer when deferring
- a full frame (related to issue #9937)
-
- * channels/chan_iax2.c: * Store the call number that a thread is
- processing without the full frame bit set to ease debugging *
- When deferring a full frame for processing, stick it into the
- queue for the thread that is processing frames for that call, not
- the one that read the current frame and is about to go back into
- the idle list (related to issue #9937)
-
-2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
- | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
- like we don't support it for G.729 ........
-
-2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Pass HOLD and UNHOLD frames to the other
- channel when they are returned from a native bridge function.
- This fixes a problem where when two zap channels are natively
- bridged and one does a flash hook, the other channel did not
- receive music on hold. (Reported to me directly by Doug Bailey at
- Digium)
-
-2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
- lines Copy language information to the dialog structure when
- calling a peer for situations where a PBX may be started on the
- dialed channel. (issue #10121 reported by clegall_proformatique)
- ........
-
-2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Correcting a minor CLI bug I found. When
- issuing the queue show command, if you type queue show and then
- press tab, you can continue pressing tab and it will keep
- auto-completing queue names even though only 1 queue can be used
- as an argument.
-
-2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com>
-
- * channels/chan_vpb.cc, channels/Makefile: Make this module build
- for me in dev-mode
-
-2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
- lines Tweak spy locking. (issue #9951 reported by welles)
- ........
-
- * channels/chan_local.c, /: Merged revisions 73318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
- 2007) | 2 lines Actually check to make sure a PBX was started on
- one of the Local channels instead of blindly assuming it was.
- (issue #10112 reported by makoto) ........
-
- * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
- lines Reset ServicelevelPerf variable back to 0 if we are unable
- to calculate it each time... otherwise we will get previous
- values. (issue #10117 reported by noriyuki) ........
-
-2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
- Jul 2007) | 1 line bchannel configurations like echocancel and
- volume control, need to be setuped on inbound calls too. ........
-
- * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
- Jul 2007) | 1 line bad bug in overlapdial case, we called
- start_pbx multiple times, because the state wasn't changed..
- ........
-
-2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
- main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
- expr floating patch from 1.4; too much of a behavior change. If
- you want this fix, try trunk instead. bug 9508.
-
-2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com>
-
- * /: What the heck. This should not have happened.
-
- * /: use autotagged externals
-
-2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
- | 2 lines RetryDial should accept a 0 argument, but it does not,
- because atoi does not distinguish between 0 and error (closes
- issue #10106) ........
-
-2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
- Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
- be called from mISDN Source channels.. #9449 ........
-
-2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
- main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
- main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
- for floating point numbers added to ast_expr2 $\[...\] exprs.
- Fixes bug 9508, where the expr code fails with fp numbers. The
- MATH function returns fp numbers by default, so this fix is
- considered necessary.
-
-2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Remove a bogus comment and add proper locking to
- the handler function for the CLI command to show information on
- manager actions.
-
-2007-07-02 17:59 +0000 [r72925] Jason Parker <jparker@digium.com>
-
- * /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell |
- 2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an
- issue with playing "oclock" multiple times in French with 24 hour
- time format. Issue 10101 ........
-
-2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Added additional DTMF debug messages for when
- emulation occurs.
-
-2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 72585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
- 1 line check if the bchannel stack id is already used, if so
- don't use it a second time. Also added a release_chan lock, so
- that the same chan_list object cannot be freed twice. chan_misdn
- does not crash anymore on heavy load with these changes. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 72099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified generation for dummy bchannels, also we mark
- them as dummies, so they are not used later as real-bchannels,
- optimized the RESTART mechanisms, we block a channel now on
- cause:44, and send out a RESTART automatically, then on reception
- of RESTART_ACKNOWLEDGE we unblock the channel again. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
- revisions 72087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified channel finding and locking a lot. removed
- unnecessary #ifdefed areas. ........
-
-2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
- 5 lines When appending lines to call files to keep track of
- retries, write a leading newline just in case the original call
- file did not have a newline at the end. This fix is in response
- to a problem I saw reported on the asterisk-users mailing list.
- ........
-
-2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Tweak the configure script so that error
- output isn't spewed to the console when searching for GTK2 libs,
- and they aren't found.
-
- * formats/format_pcm.c: give format_pcm a more concise destription
-
-2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
- absence of the function. This was already done in trunk.
-
-2007-06-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.6 released.
-
-2007-06-29 16:31 +0000 [r72630] Russell Bryant <russell@digium.com>
-
- * /: Blocked revisions 72629 via svnmerge ........ r72629 | russell
- | 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport
- changes that make chan_iax2 not start the PBX on an incoming
- channel until the three-way call setup is completed. These
- changes are already in 1.4 and trunk. ........
-
-2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Minor change for older GCC versions.
-
- * Makefile, configure, configure.ac, makeopts.in: Backport fix for
- GCC versions without support for declaration-after-statement.
-
-2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/manager.c: Issue 10055 - Change memory allocation to use the
- heap for a command, since the output has the potential to
- overflow the stack (as it did here)
-
- * res/res_jabber.c: Fix 1.4 breakage
-
-2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: regenerate the
- configure script for rizzo
-
-2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac: add a check for gethostbyname_r so we can simplify
- the handling e.g. in utils.c Also add comments on a couple of
- features which are not working on FreeBSD. All the above has been
- already done in trunk so the merge must be blocked. Can someone
- please regenerate ./configure ?
-
- * Makefile, channels/chan_zap.c, main/say.c: Add
- -Wdeclaration-after-statement to AST_DEVMODE flags to catch
- variable declarations in the middle of a block. Fix the few
- instances of the above spotted out by the compiler. All of this
- has been already done or is not applicable in trunk, so the merge
- of this change will be blocked.
-
- * apps/app_meetme.c: cast a time_t so that it does not conflict
- with the print format. This change was already done on trunk so
- this change needs to be blocked from merging.
-
-2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
- 3 lines Reinstating patch. This actually fixes the problem,
- however I was running a development branch without it and
- mistakenly thought it wasn't fixed. Fixes issue #10010, and
- #9654: 100% CPU usage caused by an asterisk console losing it's
- controlling terminal. ........
-
-2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
- 2007) | 2 lines Update documentation to clarify variable usage
- with MixMonitor. (issue #9494 reported by netoguy) ........
-
-2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72333 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
- 2 lines Reverted changes for earlier revisions 72259 to 72261.
- Issue #9654, #10010 ........
-
-2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
- our list. Since these are dynamic payloads the other side
- shouldn't care. (issue #9426 reported by irroot)
-
- * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
- lines Fix issue where queue log events might be missing. (issue
- #7765 reported by mtryfoss) ........
-
-2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
- 5 lines Fix a minor issue with parsing the priority number. You
- could have as much whitespace as you want around a numeric
- priority, but you couldn't have any whitespace around a special
- priority like "n" or "hint". (issue #10039, reported by mitheloc,
- fixed by me) ........
-
-2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
- 4 lines Fixes 100% load when controlling terminal disappears.
- Issue #9654, #10010 ........
-
-2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
- lines I may possibly get shot for doing this... but... defer CDR
- processing until after the channel has been dealt with. This
- should eliminate all of the issues with channels going funky
- (SIP/PRI) when you are posting CDRs to a database that is either
- slow or unavailable and do not want to enable batching. ........
-
-2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: use the proper type for storing group number
- bits so that if someone specifies 'group=42' it will actually
- work instead of being silently ignored
-
-2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <jparker@digium.com>
-
- * /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell |
- 2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix
- another problem in voicemail with missing symbols. Issue 10074,
- patch by kryptolus, extended to include #if 0'd blocks (just in
- case) ........
-
- * apps/app_voicemail.c: Fix another problem in voicemail with
- missing symbols. Issue 10074, patch by kryptolus, extended to
- include #if 0'd blocks (just in case)
-
-2007-06-27 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Make the ast_read_noaudio API call behave better
- under circumstances where DTMF emulation was happening and a
- generator was setup. (issue #10065 reported by stevefeinstein)
-
-2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Don't modify a variable that we don't want
- modified. Make a copy of it instead. Issue 10029, patch by
- phsultan with slight modifications by me (to remove needless
- casts).
-
-2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Only output debug information related to RTCP
- timestamps when RTCP debug is turned on (issue #10066, patch by
- me)
-
-2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
- 1 line for inbound TE calls, we setup the bchannel when we get
- the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
- removed some #if 0 areas which weren't used anymore. ........
- r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
- 1 line isdn_lib.c didn't compile ........
-
-2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.
-
-2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a pointless line. This variable
- was already set earlier and between then and this line, there is
- no way that the values on the right side of the assignment could
- have changed.
-
-2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Don't dereference a pointer that may be NULL here.
- Issue 10017.
-
-2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A few changes, the ultimate goal of which
- is to keep better track of the number of messages that a mailbox
- currently has. A description of the changes: 1. Changed the
- "updated" field of the vm_state struct to act more as a binary
- semaphore than a counting semaphore, since its current
- implementation made the inboxcount function not work properly.
- This change falls in line with a change made by UPenn with their
- IMAP setup and helps to sync our changes with theirs. 2.
- Eliminated some redundant calls to get_vm_state_by_mailbox inside
- leave_voicemail 3. Use the play_folder variable to keep track of
- the number of old and new messages in a mailbox as the messages
- are deleted 4. Added an increment to the number of new messages
- that was not there previously in the leave_voicemail function
-
-2007-06-26 17:49 +0000 [r71848] Jason Parker <jparker@digium.com>
-
- * /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell |
- 2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try
- to install an init script that doesn't exist. Reported to me on
- #asterisk on Freenode IRC. ........
-
-2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing bug where the authuser was
- mistakenly pulled from the mailbox string instead of the IMAP
- user. (closes issue 10054, reported and patched by jaroth)
-
-2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
- | 2 lines Issue 10062 - Trying to move a message without
- selecting one first results in memory corruption ........
-
- * /, res/res_agi.c: Merged revisions 71656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
- | 2 lines Issue 10035 - handle_exec returns a result inconsistent
- with all of the other AGI commands ........
-
-2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Build a peer as well when hash323 is
- enabled in users.conf (issue #9599 reported by asagage)
-
- * channels/chan_agent.c: Minor tweak for queueing up the unhold
- frame... this will teach me to do bugs while half asleep. (issue
- #10046 reported by dimas)
-
-2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com>
-
- * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
- #10048, Matti)
-
-2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
- lines Ignore other URIs after the first in a 300 Multiple Choice
- response. (issue #10041 reported by homesick) ........
-
- * main/cdr.c: Fix it so 1.4 actually compiles on my box.
-
- * channels/chan_agent.c: Check to make sure the channel pointer is
- present before queueing up an unhold frame on it. (issue #10046
- reported by dimas)
-
-2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com>
-
- * build_tools/prep_tarball: Include the menuselect-tree file in
- tarballs to make builds from tarballs a little bit faster
-
- * main/asterisk.c, /: Merged revisions 71358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
- 2 lines Revert the patch from issue 9654 due to an unexpected
- side effect ........
-
-2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
- peer->cdr is what we really wanted to use
-
- * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
- Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
- be able to set variables to the empty string. ........
-
-2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
- where clid and src are lost via bridging.
-
-2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
- Jun 2007) | 1 line fixed a bug that was introduced by copy and
- paste in the last commit ..bchannels weren't cleaned properly.
- ........
-
-2007-06-22 16:05 +0000 [r71128] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 71124 via svnmerge ........ r71124 | file |
- 2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an
- unhold indication when going off hold. (issue #10036 reported by
- speedy) ........
-
-2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 70672 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
- 1 line we activate the bchannels in TE mode on incoming calls
- only when we want to connect the call. ........
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
- Jun 2007) | 1 line forgot one place .. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
- Jun 2007) | 1 line on receiption of cause:44 we mark the channel
- as in use and inform the user about the situation, we need to
- test the RESTART stuff then. Also shuffled the
- empty_chan_in_stack function after the bchannel cleaning
- functions, to avoid race conditions. ........
-
- * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
- Jun 2007) | 1 line when we send out a SETUP, but get no response,
- we should cleanup everything after reception of a hangup.
- ........
-
- * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
- 1 line restart indicator 0x80 is correct, at least that's what
- libpri does. ........
-
- * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
- Jun 2007) | 1 line if the bridged partner is mISDN too we should
- not send dtmf tones, they are transmitted inband always ........
-
- * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
- Jun 2007) | 1 line if we have already some digits, we just stop
- the tones. ........
-
-2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com>
-
- * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
- revisions 71065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
- lines Fix a few silly usages of ast_playstream() - it only ever
- returns 0... Issue 10035 ........
-
-2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 71064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
- 10 lines Fixed infinite loop when controlling terminal was lost
- and return value of input function wasn't checked for errors.
- This would cause 100% cpu to be taken up. (closes issue #9654,
- issue #10010) Reported by: mnicholson, and eserra Idea for the
- patch from mnicholson, patched by me ........
-
-2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: My conditions for merging amaflags info was naive;
- DOCUMENTATION is the default, although null is possible; theft of
- user-settable fields is not good. Just copy them, leave them
- alone.
-
-2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a small typo which ... well ...
- completely broke chan_iax2. oops! (issue #9937, patch by me)
-
-2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 70948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
- line This little fix is in response to bug 10016, but may not
- cure it. The code is wrong, clearly. In a situation where you set
- the CDR's amaflags, and then ForkCDR, and then set the new CDR's
- amaflags to some other value, you will see that all CDRs have had
- their amaflags changed. This is not good. So I fixed it. ........
-
-2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
- lines Don't explode if the gain option is specified without a
- value. (issue #9274 reported by mfarver) ........
-
-2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Put the thread reading from the socket back
- in the idle list if it deferred the processing of a full frame to
- another thread
-
- * channels/chan_iax2.c: If a full frame is received while one of
- the iax2 threads is in the middle of handling a full frame for
- the same call, queue it up for processing by that same thread
- later instead of dropping it. (issue #9937, patch by me)
-
-2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
- line it was pointed out that the cdr_custom config load could get
- a lock, and under certain circumstances, would never release it.
- I also noted that the situation where more than one mapping spec
- was warned about, but did not ignore further mappings as it had
- promised. I think I have fixed both situations. ........
-
-2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: When volgain is used don't leave a
- temporary file behind. (Closes Issue 8514, Reported and patched
- by ulogic, code reviewed by Jason Parker)
-
-2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Do not Packet2Packet bridge if packetization settings
- do not allow it. (issue #9117 reported by phsultan)
-
-2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Remove a couple of duplicate unlocks
-
-2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix building with ODBC storage enabled.
- (issue #10025 reported by denisgalvao)
-
-2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Via complaints aired in asterisk-users, I submit
- these changes, which allow cdr updates to see macro
- context/exten, whether hung up or not
-
-2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
- Issue 10020, patch by my, with credit to prashant_jois for
- pointing out the problem.
-
- * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
- condition fix
-
- * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
- when reloading the module. Issue 10022, patch by me, with credit
- to prashant_jois for finding the bug.
-
-2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
- lines Don't overwrite the configured username setting upon a
- REGISTER. (issue #8565 reported by jsmith) ........
-
-2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure we clear the previously dialed
- number if it did not exist. Issue 9958.
-
-2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
- | 2 lines Issue 9997 - Timelimit times out the wrong channel
- ........
-
-2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
- 5 lines Fix a problem where an established call would not be
- properly disconnected when a PRI disconnect is received depending
- on which cause code was received. (issue #9588, original patch by
- softins, updated patch from jtexter3, and some additional
- feedback from mhardeman) ........
-
-2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, main/frame.c: Put the speex packetization values back
- in but disable it when setting up the smoother.
-
- * main/frame.c: Don't do packetization/smoother stuff with speex,
- it doesn't work.
-
-2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com>
-
- * contrib/scripts/ast_grab_core: don't delete the backtrace in
- ast_grab_core
-
- * channels/chan_gtalk.c: Only attempt to queue a hangup on the
- owner channel if it actually exists. (issue #9795, patch from
- zandbelt)
-
-2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com>
-
- * main/channel.c, /: Merged revisions 70053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
- line This fixes 9246, where channel variables are not available
- in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
- consolidate the channel variables during a masquerade, and then
- copy the merged variables back onto the clone, so the zombie has
- the same vars that the 'original' has. ........
-
-2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 69992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
- lines Handle the CC field in the RTP header. (issue #9384
- reported by DoodleHu) ........
-
-2007-06-19 16:46 +0000 [r69991] Russell Bryant <russell@digium.com>
-
- * /: Blocked revisions 69990 via svnmerge ........ r69990 | russell
- | 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines
- Backport fix for crashes related to subscriptions from 1.4 ...
- Fix a crash that could occur when handing device state changes.
- When the state of a device changes, the device state thread tells
- the extension state handling code that it changed. Then, the
- extension state code calls the callback in chan_sip so that it
- can update subscriptions to that extension. A pointer to a
- sip_pvt structure is passed to this function as the call which
- needs a NOTIFY sent. However, there was no locking done to ensure
- that the pvt struct didn't disappear during this process. (issue
- #9946, reported by tdonahue, patch by me, patch updated to trunk
- to use the sip_pvt lock wrappers by eliel) ........
-
-2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 69986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
- lines Update BRIDGEPEER variable if set to the new channel name
- when a masquerade happens. (issue #9699 reported by dimas)
- ........
-
-2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix a crash that could occur when handing
- device state changes. When the state of a device changes, the
- device state thread tells the extension state handling code that
- it changed. Then, the extension state code calls the callback in
- chan_sip so that it can update subscriptions to that extension. A
- pointer to a sip_pvt structure is passed to this function as the
- call which needs a NOTIFY sent. However, there was no locking
- done to ensure that the pvt struct didn't disappear during this
- process. (issue #9946, reported by tdonahue, patch by me, patch
- updated to trunk to use the sip_pvt lock wrappers by eliel)
-
-2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
- lines Perform an extra hangup check just in case. (issue #9589
- reported by bcnit) ........
-
- * /, res/res_features.c: Merged revisions 69846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
- lines Add parked call extension AFTER the parking slot has been
- announced, otherwise two threads will try to handle the same
- channel and it will go kaboom. (issue #9191 reported by japple)
- ........
-
- * main/callerid.c: Fix for building on PowerPC under Linux.
-
-2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Issue 10005 - Segfault with missing
- arguments, plus fix a missing define for SIP INFO channels
-
-2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't count RTP timeout when involved in a
- T38 fax session. (issue #9222 reported by ivoc)
-
- * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
- lines Set the peer name on the dialog to the one configured in
- sip.conf and NOT the username to be used for authentication
- attempts. (issue #9967 reported by achauvin) ........
-
-2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
- | 2 lines Issue 9998 - Remove SIG prefix, since it's not
- supported by ksh ........
-
-2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com>
-
- * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
- for the first time so that it does not needlessly spit out
- changed messages when the host really didn't change.
-
-2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com>
-
- * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
- build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
- include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
- To prevent 92138749238754 more reports of "I have unixodbc
- installed, but still can't build *_odbc.so!", check for ltdl
- directly, instead of just listing it as another library to
- include in the unixodbc check in the configure script. This also
- makes ltdl show up as a dependency in menuselect so people know
- what to go install. (related to issue #9989, patch by me)
-
- * build_tools/prep_moduledeps: Change the use of "echo -e" to
- "printf". On systems where /bin/sh is not bash, most of the lines
- in menuselect-tree were getting a "-e" at the beginning of every
- line. I'm surprised nobody noticed this, but I think the XML
- parser was being very nice and ignoring them.
-
-2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't defer the BYE till later on a transfer
- when the transfer itself goes kaboom and has no hope of working.
-
- * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
- something we never locked, and better handle a specific scenario
- with doing an attended transfer between two non-bridged calls.
-
-2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com>
-
- * Makefile: Tweak paths for BSD systems (issue #10001, stuarth)
-
-2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where it would be possible for the
- negotiated codecs to get set back to nothing. (issue #9992
- reported by yehavi)
-
-2007-06-15 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.5 released.
-
-2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Fix a silly deadlock in res_features that I
- found while debugging on one of blitzrage's test machines. It was
- one of the situations where he was seeing hung channels, and may
- be the cause of some of the reports from other people. (related
- to issue #9235)
-
-2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Add support for setting the maximum
- length of acceptable DTMF in SpeechBackground.
-
-2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
- "SUCCESS" for both an answer of the incoming call on the trunk,
- or if the trunk reached its ring timeout. This patch changes the
- variable to say "RINGTIMEOUT" in that case. (issue #9973,
- reported by n00dle, patch by me)
-
-2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com>
-
- * main/config.c, /: Merged revisions 69469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
- lines Fix an issue where the line number in an unterminated
- comment block error message would show the wrong line number.
- "Reported" to me on #asterisk (somebody posted an error message,
- and I happened to catch it) ........
-
- * sounds/Makefile: Update to latest versions of sound files.
-
-2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com>
-
- * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
- cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
- main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
- apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
- main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
- channels/chan_iax2.c: use ast_localtime() in every place
- localtime_r() was being used
-
-2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com>
-
- * main/say.c: Fix some problems with saying dates and times for the
- "tw" langauge (issue #9964, ljmid)
-
-2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
- 2007) | 4 lines Change a quite broken while loop to a for loop,
- so "continue;" works as expected instead of eating 99% CPU...
- Issue 9966, patch by me. ........
-
-2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Whoops...
-
- * channels/chan_iax2.c: Let's make chan_iax2 media only native
- transfers actually work. (issue #9376 reported by simone
- cittadini)
-
- * channels/iax2-parser.c: Add TXMEDIA to list so that it is
- properly displayed during iax2 packet output.
-
-2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Move the logic for destroying a call when no
- response is received to a BYE outside of the block that checks
- for FLAG_FATAL to be set. This flag is only set when the packet
- is transmitted with the reliability set to XMIT_CRITICAL when the
- original packet is transmitted. A BYE is always sent with it set
- to XMIT_RELIABLE, meaning this code could never be encountered.
- This resulted in seeing some SIP channels that would never go
- away with the last packet sent being a BYE. (part of issue #9235,
- patch from jcmoore)
-
-2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Contains a patch for fixing an encoding
- problem when using Outlook to view voicemail emails and
- attachments. This fix has also been tested on Thunderbird,
- Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
- patched by mutterc)
-
-2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Really ignore NULL frames and check whether
- the channel hungup or not. (issue #9912 reported by junky)
-
- * /, main/app.c: Merged revisions 69127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
- lines Return group counting to previous behavior where you could
- only have one group per category. (issue #9711 reported by
- irroot) ........
-
-2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Clarify a bit of logic. This doesn't change
- behavior in any way, but it is helpful when following the logic
- to debug problems like 9235.
-
- * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
- was accessed without the lock held. This issue was reported to me
- via email by Dmitry Mishchenko. Thanks!
-
- * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
- in #asterisk-bugs. PQclear() was not called on the result
- structure after doing a PQexec(). Also, fix up some formatting in
- passing.
-
-2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Change the full frame dropping log message
- to debug to avoid future bug reports.
-
- * channels/chan_iax2.c: Schedule the sending of a PING packet a
- second later than previously so that it does not collide with the
- LAGRQ.
-
-2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com>
-
- * main/channel.c: In ast_channel_make_compatible(), just return if
- the channels' read and write formats already match up. There are
- code paths that call this function on a pair of channels multiple
- times. This made calls fail that were using g729 in some cases.
- The reason is that codec_g729a will unregister itself from the
- list of available translators will all licenses are in use. So,
- the first time the function got called, the right translation
- path was allocated. However, the second time it got called, the
- code would not find a translation path to/from g729 and make the
- call fail, even if the channel actually already had a g729
- translation path allocated. (SPD-32)
-
-2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 68921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
- lines Bring RTP back to Asterisk at the end of a native bridge no
- matter what. ........
-
-2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com>
-
- * include/asterisk/time.h: Solaris 10 sometimes (?) needs this
- include in order to have NULL defined.
-
-2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
- used
-
-2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 68732 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
- 1 line added check for NULL Pointer when calling misdn_new.
- Asterisk does not allow us to create channels anymore when stop
- gracefully is used :). also modified the restart_indicator to 0
- ........
-
-2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 68682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
- lines Improve deadlock handling of the channel list. (issue #8376
- reported by one47) ........
-
-2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
- 1 line fixed problem that the dummybc chanels had no lock,
- checking for the lock now. Also fixed the channel restart stuff,
- we can now specify and restart particular channels too. ........
-
-2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * pbx/pbx_config.c: "dialplan save" produced garbage in the config
- file
-
-2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com>
-
- * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
- 4 lines Don't automatically hang up after running Dictate so that
- callers can exit cleanly using '#' (closes issue #9577, patch
- from Thomas Andrews) ........
-
-2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: actually remember the type/subclass of full
- frames that are in process
-
-2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com>
-
- * /, main/say.c: Merged revisions 68397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
- lines Don't call ast_waitstream_full when the control file
- descriptor and audio file descriptor are not set, simply call
- ast_waitstream! (issue #8530 reported by rickead2000) ........
-
- * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
- lines Do a DNS lookup immediately upon calling the dnsmgr
- function, don't wait until a refresh happens. (issue #9097
- reported by plack) ........
-
-2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com>
-
- * /, main/say.c: Merged revisions 68351 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
- 3 lines Fix a problem where saying a character wouldn't properly
- break out when the caller pressed '#' (issue #8113, reported by
- patbaker82, patch from jamesgolovich (hey, long time no see!) and
- patbaker82) ........
-
-2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix incorrect French syntax of "old
- messages". Request for feedback was sent to asterisk-dev mailing
- list, with little response. Issue 9118, patch by junky.
-
-2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: some improvements to the IAX2 full frame
- dropping logic recently added: - use inaddrcmp(), since we have
- it - output the type of frame and subclass being dropped, and the
- type/subclass that is already being processed (which caused the
- drop)
-
-2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
- queue members when using realtime configuration for queues. Also,
- remove an unneeded leading slash for the astdb family. (issue
- #9911, patch by atis)
-
-2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue with newer phones which
- require packets be padded out to the correct length. Issue 9887,
- patch by DEA.
-
- * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
- lines Don't try to save voicemail greetings unless the user
- presses '1' to accept/save. Issue 9904, patch by me. ........
-
-2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
- check to make sure that greetings get stored properly. (Issue
- 8016, reported by edhorton, patched by alamantia with
- modification by me. Thanks to Jason Parker for the advice on
- this).
-
-2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net>
-
- * channels/chan_features.c: Disable chan_features by default in
- menuselect
-
-2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com>
-
- * main/strcompat.c: Include stdarg.h for build issues on Solaris
- (issue #9381)
-
-2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Fix logic when doing a name based channel search
- for a structure when you want to start from a specific point in
- the channel list. (issue #9324 reported by slavon)
-
- * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
- lines Allow the 'g' option to work if used with the 'S' option.
- (issue #9888 reported by gasparz) ........
-
-2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
- forget.
-
- * res/res_jabber.c: Ok, we found out that this is not about if you
- have any *active* clients using TLS, but if you have initialized
- TLS at all during the lifetime of the module. So if you reload to
- disable TLS, it won't help.
-
- * res/res_jabber.c: If you have a jabber client that uses TLS,
- refuse unload. Bad fix, but will prevent crashes while we are
- trying to find a workaround. Iksemel development seems to have
- stalled and we might have to stop using the TCP/TLS connections
- in that library and use our own, which would scale better from a
- poll/select perspective I guess. It would also make it easier to
- migrate to OpenSSL and stop Asterisk from depending on both
- OpenSSL and GnuTLS.
-
- * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
- sure we can unload res_jabber. Patch by phsultan - thanks! Due to
- a bug in the iksemel library, this will not work if you are using
- GTLS in the connection. That's being investigated. If you figure
- out a way to handle that without us having to patch iksemel, let
- us know in the bug report. Thanks.
-
-2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
- lines Only notify the devicestate system of a peer state change
- when the peer is built from the config file. (issue #9900
- reported by arkadia) ........
-
- * main/file.c: Properly handle cases where a stream can't be
- written to. (issue #9757 reported by junky)
-
-2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com>
-
- * res/res_snmp.c: Disable reload functionality in res_snmp. It is
- not possible to initialize the snmp library more than once
- without completely unloading the module and loading it again.
- (issue #9571, reported by hristo, additional helpful debug
- information from festr, patch from me)
-
- * channels/chan_sip.c: Fix a crash when doing call pickups with SIP
- phones. The code unlocked the channel when it should not have.
- (issue #9652, reported by corruptor, fixed by me)
-
-2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
- under a specific set of circumstances: 1. VoiceMailMain was
- configured in the dialplan with an extension as its argument 2. A
- message was left for this mailbox 3. Tried to call VoiceMailMain
- but hung up before entering password. This was fixed by checking
- that a pointer was non-null prior to trying to dereference it.
- (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
- modifications by me).
-
-2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/linkedlists.h: Merged
- revisions 67715 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
- 5 lines We have some bug reports showing crashes due to a double
- free of a channel. Add a sanity check to ast_channel_free() to
- make sure we don't go on trying to free a channel that wasn't
- found in the channel list. (issue #8850, and others...) ........
-
-2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 67649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
- lines Reinvite the RTP back to the Asterisk machine when the
- timeout happens. (issue #9888 reported by gasparz) ........
-
- * main/translate.c: Fix plc_samples warning when registering a
- translator. (issue #9897 reported by xylome)
-
- * apps/app_directed_pickup.c: Include macroexten while searching
- for a channel to pick up in case they are in a macro. (issue
- #9491 reported by jamesb63)
-
- * res/res_agi.c: Make the new "agi debug off" CLI command work.
- (issue #9890 reported by eliel)
-
- * /, main/devicestate.c: Merged revisions 67593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
- lines Revert channel name splitting fix for Zap. The moral of the
- story is don't use - in your user/peer names. (issue #9668
- reported by stevedavies) ........
-
-2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Fix some crashes related to the use of the
- "meetme" CLI command. The code for this command was not locking
- the conference list at all. (issue #9351, reported by and patch
- submitted by Junk-Y, committed patch is different and by me)
-
-2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
- 9883, wherein macros were not allowing the includes construct.
- fixed and tested, looks OK. Now includes can serve as an adjunct
- to catch.
-
-2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com>
-
- * include/asterisk/linkedlists.h: This bug has been hanging over my
- head ever since I wrote this SLA code. Every time I tried to go
- debug it by adding some debug output, the behavior would change.
- It turns out I wasn't crazy. I had the following piece of code:
- if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
- AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
- conditional statement didn't do much good at all. It always ran
- at least all of the macro minus the first statement, so I was
- seeing list entries magically disappear when they weren't
- supposed to. After many hours of debugging, I have come to this
- extremely irritating fix. :) (issues #9581, #9497)
-
- * channels/chan_zap.c: Suppress a bunch of debug output unless
- option_debug is on
-
-2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
- saved to IMAP storage using extensions other than gsm were unable
- to be played over the phone. (Issue 9786, reporter:
- xmarksthespot, Patched by xmarksthe spot with revisions by me,
- reviewed by Russell Bryant).
-
-2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Correctly update date/time on devices
- throughout the life of the device, instead of just at
- registration. Issue 9152, yet another patch by DEA.
-
-2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Added code to automatically add a default case to
- switches that don't have one. In some cases, rather than fall
- thru, it results in a goto with -1 result, which terminates the
- extension; a sort of dialplan seqfault, sort of. This was
- required to fix bug reported in 9881
-
-2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Handle a failure in malloc() in
- ast_safe_string_alloc()
-
- * main/channel.c: Fix a problem that showed itself by causing Zap
- channel names to be completely bogus on my machine.
- ast_safe_string_alloc() was broken. It called vsnprintf() on a
- va_args list twice without re-initializing it. After the first
- usage, va_end() and va_start() must be called again.
-
-2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
- 1 line briding is a bool, fixed copy and paste issue. ........
-
- * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
- Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
- cb_events function a lot. Commented the different possibilities a
- bit and made functions of shared code. When the dialed extension
- does not exist in the extensions.conf we'll jump into the 'i'
- extension if this does exist, else we disconnect the call with
- the cause:1 = No Route to Destination. ........
-
-2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/loader.c, include/asterisk/module.h: When
- shutting down "gracefully", go through and run the unload()
- callbacks for all of the modules. "stop now" is considered a
- non-graceful shutdown and will not go through this process.
- (issue #9804, reported by chrisost, patch by me)
-
-2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Only muck with the thread structure if an
- idle one was found/created.
-
-2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: ensure that a burst of full frames
- (AST_FRAME_DTMF being the prime example) will not be processed
- out of order... this is a brute force fix, but seems to be the
- safest fix for now (thanks to the Digium PQ department for
- finding this bug)
-
-2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
- 1 line added possibility to deactivate bridging per port ........
-
-2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
- | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
- ........
-
-2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
- structure can disappear and the code did not account for it and
- crashes. (issues #9642, #9569, #9666, probably others ... based
- on the work by stevedavies and mihai, with additional changes
- from me)
-
-2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix for skinny keepalives. If there is no
- traffic from the phone for (keep_alive * 1100) ms (arbitrarily
- adding 10% for network issues, etc), unregister the device. Issue
- 8394, patch by DEA.
-
- * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
- to the recent fix for chan_skinny) Issue 9855, patch by DEA.
-
-2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add comments for two functions that get
- called with the appropriate call locked, but perform operations
- that could result in the pvt structure getting destroyed before
- returning again, causing numerous seg faults all over the module.
- (inspired by issues #9642, #9569, and #9666, and the work done by
- stevedavies and mihai)
-
-2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: This typo has been here since 1.4 forked. It has been
- the source of heartburn to many a dialplan/CDR programmer.
-
-2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)
-
-2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Better handle SIP devices that say they have
- SDP content... but really don't. (issue #9398 reported by
- mthomasslo)
-
- * apps/app_dial.c: Initialize cidname variable to nothing since it
- may be used without having been touched. (issue #9661 reported by
- dimas)
-
- * res/res_features.c: Returning a value that indicates the parking
- of a call was a success when it really wasn't (because the
- parking slot selected was in use) is the wrong thing to do.
- (issue #9723 reported by mdu113)
-
-2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.mandrake.zaptel,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
- | 2 lines Add revision Id tags (by request of tzafrir) ........
-
-2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Change the configure script to build a
- test program against libcurl to make sure the results from
- curl-config can be used to compile successfully. This is intended
- to help prevent a situation where you are cross compiling, and
- the configure script finds the curl library installed on the
- host. (issue #9865, reported and patched by zandbelt)
-
-2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash
-
-2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
- handling an implicit ACK to a frame that was marked as the final
- transmission for a call, don't call iax2_destroy() for that call
- while the global frame queue is still locked. There is a very
- nice explanation of the deadlock in the report. (issue #9663,
- thorough report and patch from stevedavies, additional positive
- test reports from mihai and joff_oconnell)
-
- * include/asterisk/stringfields.h: Fix some compiler warnings in
- C++ modules. (issue #9866, reported by osk, patch by Corydon76)
-
-2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_odbc.c: On some drivers, deallocating the statement
- handle isn't enough. We also have to clear the cursor (nice,
- Oracle)
-
-2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing extraneous debugging lines from
- revision 66897. Sorry :)
-
- * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
- storage. Attachments with format specified as gsm were duplicated
- (i.e. two attachments) were left. Thank you very much to
- xmarksthespot for submitting the patch that fixed this. (Issues
- 9787 and 8873, Reported by xmarksthespot and jerjer, patched by
- xmarksthespot)
-
-2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com>
-
- * channels/chan_skinny.c: Changes to the way DTMF is handled in the
- core broke dialing in chan_skinny. This patch makes chan_skinny
- usable again. I did not end up testing this, but there are
- multiple positive test reports listed in the bug report. (issue
- #9596, reported by pj, testing by pj and mvanbaak, and the fix
- was written by DEA)
-
- * apps/app_page.c: List app_meetme as a module that app_page
- depends on.
-
-2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * doc/asterisk.8: Issue 9850 - update preferred command line syntax
-
-2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com>
-
- * res/res_speech.c, include/asterisk/app.h,
- include/asterisk/speech.h: Change a couple of header files to not
- use "new", which is a reserved keyword in C++. (issue #9830,
- reported by osk)
-
-2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
- | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
- Issue 8329 will remain unfixed for pbx_realtime, but only because
- we lack core API to do it. ........
-
-2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
- lines It is now possible for this path of execution to have the
- frame pointer be NULL, therefore we need to check for it before
- trying to access it. (issue #9836 reported by barthpbx) ........
-
-2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed seg-faults when recording greetings
- in voicemail with IMAP enabled. (Issue No. 9735, reported by
- xmarksthespot, patched by me)
-
-2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Silly me for having out of date source! Oh
- well... I'm still leaving my comment.
-
- * channels/chan_sip.c: When calling some peer/host that may not
- exist/reply back... don't keep the dialog in memory for all of
- eternity.
-
- * channels/chan_zap.c, channels/chan_features.c: Change how channel
- names are generated a bit. (issue #9825 reported by eldadran)
-
-2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
- | 2 lines If the value of a variable passed to FIELDQTY is blank,
- then FIELDQTY should return 0, not 1. ........
-
-2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Properly handle 408 request timeout -
- according to the RFC, the dialog dies if a request in a dialog
- gets this response.
-
- * channels/chan_sip.c: Don't issue hangup on hangup on hangup on
- hangup (for jcmoore)
-
-2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Handle cases where a frame may have no data. (issue
- #9519 reported by dmb)
-
-2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't reset hangupcause if we already have
- one
-
- * channels/chan_sip.c: Tracking down hanging channels, killing them
- one by one. Issue #9235 and related
-
-2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com>
-
- * doc/datastores.txt: Update datastores documentation. (issue #9801
- reported by mnicholson)
-
-2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
- lines Issue #9802 - Change inuse counter on CANCEL ........
-
-2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Make the usedistinctiveringdetection option
- work again. (issue #9823 reported by premeau)
-
-2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: I don't know what this was trying to do, but
- it's clearly incorrect. Issues 9808 and 9809.
-
-2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: have to check for OSP toolkit _after_
- checking for OpenSSL
-
-2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/say.c: Merged revisions 66127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
- | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
- ........
-
-2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
- res/res_jabber.c: handle the GNUTLS library properly in the
- configure script and build system don't build in OSP support
- unless we have found and are allowed to use SSL support
-
-2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com>
-
- * main/channel.c: if the string field init fails, clean up the
- stuff that was allocated already
-
-2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Fix slinfactory logic when dealing with
- frames coming in that may already be in the signed linear format.
-
-2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Check the result of ast_string_field_init() in
- ast_channel_alloc()
-
- * main/rtp.c: Make 1.4 build on my machine, too..
-
-2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com>
-
- * configure: Rebuild configure script for previous ar fix.
-
- * configure.ac: Following moving strip to AC_PATH_TOOL, we need to
- do something similar for ar.
-
-2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Checking for the strip application needs to be done with
- AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
- compilation environments.
-
- * Makefile: Clear CFLAGS before running make for menuselect. (issue
- #9784, reported by ovi, patch by me)
-
-2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_gtalk.c: oops, use #ifdef instead of #if
-
- * channels/chan_gtalk.c: don't reference GnuTLS headers and
- functions unless the configure script found it
-
- * main/rtp.c: don't use uninitialized variables
-
-2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Add the ability to blacklist certain commands
- from being executed using the Command AMI action. (issue #9240
- reported by junky)
-
-2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
- core dump since the GnuTLS interface did not support
- multithreading correctly.
-
- * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
- Patch by phsultan. Thanks!
-
-2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com>
-
- * .cleancount: Update cleancount for that last commit - just for
- good measure.
-
- * include/asterisk/translate.h, codecs/codec_speex.c,
- main/translate.c, codecs/codec_ilbc.c: Fix handling of
- zero-length frames when a codec is capable of native PLC. Issue
- 9183, patch by Mihai.
-
-2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * funcs/func_math.c: merged qwell's func_math patch for issue 9507
-
-2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: I like it when the RTP stack compiles myself...
-
-2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
- when calling from gtalk to SIP over nat.
-
-2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com>
-
- * apps/app_festival.c: Ensure that frames are fully initialized.
- This will probably fix getting weird timestamp log messages in
- logs when using the Festival app. (issue #9781, patch by me)
-
- * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
- code would result in oscillating and incorrect data.
- Additionally, the RTT would sometimes report negative values due
- to incorrect calculations. (issue #9601, patch from davetroy)
-
-2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
- jingle
-
-2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
- lines Allow RFC2833 to be negotiated when an INVITE comes in
- without SDP and is not matched to a user or peer. (issue #9546
- reported by mcrawford) ........
-
-2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
- Fix "login" as component to jabber server. ...and, by accident,
- fix a bug in chan_sip for stopping a loop on retransmits of BYE
- requests.
-
-2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
- Mai 2007) | 1 line we should only activate the generator in
- chan_misdn, when asterisk hask not yet taken the call
- (WAITING4DIGS state). Alerting audio will be generated fomr
- asterisk for example. ........
-
-2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: start the delayed PBX when receive voice or
- video full frames as well, and comment this delayed-PBX activity
-
- * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
- | 2 lines ensure that variables are set on a newly created
- channel before we start a PBX on it ........
-
- * channels/chan_iax2.c: clear the 'delay PBX' flag when we are
- ready to start the PBX
-
- * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
- channel until we have some sort of response to our ACCEPT (ACK or
- anything else)
-
- * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
- | 2 lines if we are going to set variables on a newly created
- channel, it should be done *before* we start the PBX on it
- ........
-
-2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
- 3 lines Revert revision 62417 as someone reported problems with
- it to Mark. This was related to issue #9588. ........
-
-2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_version: when building a version string for a
- developer branch, include the base branch in the version string
-
-2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
- dependency for app_voicemail and chan_zap (Thanks to mnicholson
- for pointing it out)
-
-2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Remove a double const.
-
-2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix a problem with flag recognition.
-
-2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
- 4 lines Fix a memory leak that I just noticed in the device state
- handling in app_queue. On most device state changes, it would
- leak roughly 8 to 64 bytes (the length of the name of the
- device). ........
-
-2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
- Mai 2007) | 1 line we stop the tones only when we're in the
- pre-call phase, otherwise e.g. when in CONNECTED state we should
- not stop tones when we receive an Information Message ........
-
-2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: res_agi needs to export two symbols
- (ast_agi_register and ast_agi_unregister) for usage by others.
- (issue #9755 reported by mnicholson)
-
-2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
- to main/cdr.c, and neither did I. This is the remainder of the
- 9717 patch, the fix for the run-away FAIL status for a call
-
- * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
- 65172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
- line This update will fix the situation that occurs as described
- by 9717, where when several targets are specified for a dial, if
- any one them reports FAIL, the whole call gets FAIL, even though
- others were ringing OK. I rearranged the priorities, so that a
- new disposition, NULL, is at the lowest level, and the
- disposition get init'd to NULL. Then, next up is FAIL, and next
- up is BUSY, then NOANSWER, then ANSWERED. All the related set
- routines will only do so if the disposition value to be set to is
- greater than what's already there. This gives the intended
- effect. So, if all the targets are busy, you'd get BUSY for the
- call disposition. If all get BUSY, but one, and that one rings is
- not answered, you get NOANSWER. If by some freak of nature, the
- NULL value doesn't get overridden, then the disp2str routine will
- report NOANSWER as before. ........
-
-2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
- lines Not getting an ACK to a 200 OK in the initial invite is
- critical to the call. ........
-
- * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
- lines Issue 9235 - part of the problem, maybe not all. Please
- retry with this patch (and no other patch) if you have problems
- with hanging SIP channels. Thank you. A special Thank You to
- WeBRainstorm that gave me access to his system. ........
-
- * channels/chan_sip.c: - Adding support for putting calls OFF hold
- with a re-invite with blank SDP. This was a bug found while doing
- tests at SIPit in Antwerp. - In order to not duplicate code, I
- restructured some of the code for putting calls on/off hold.
- Thanks DEA for reminding me. This fix has been asleep in the
- videocaps branch until now.
-
-2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 65007 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
- 1 line fixed a warning regarding Keypad encoding. encode the IE
- sending_complete at the right position. ........
-
-2007-05-18 10:37 +0000 [r64974] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
- call
-
-2007-05-18 08:58 +0000 [r64904] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
- Mai 2007) | 1 line we *need* to send a PROCEEDING when
- sending_complete is set, even if need_more_infos is requested.
- ........
-
-2007-05-18 02:48 +0000 [r64868] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix a small bug I noticed while working on
- something else. app_queue did not unregister its device state
- monitoring callback in unload_module(). So, this would make
- Asterisk crash on the first device state change after you unload
- the module.
-
-2007-05-17 21:19 +0000 [r64820] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 64819 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
- | 2 lines How is it that we never caught that this is returning
- the opposite of our documentation, until now? ........
-
-2007-05-17 16:53 +0000 [r64761] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
- lines If we have a negative current message, we shouldn't go back
- even further... Issue 9727. ........
-
-2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant <russell@digium.com>
-
- * contrib/scripts/astxs (removed): Remove script that is no longer
- functional since the build system was redone. (issue #9340,
- reported by junky)
-
- * apps/app_dial.c: Increase the size of a buffer to support longer
- dial strings for channels. (issue #9291, reported and fix
- suggested by meni)
-
-2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Even more direct RTP setup fixes! Don't
- allow a codec that isn't supported to creep into the SDP of
- either side. (issue #9446 reported by marcelbarbulescu)
-
- * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
- by xmarksthespot)
-
-2007-05-17 06:13 +0000 [r64686] Russell Bryant <russell@digium.com>
-
- * README: Update the main README to reflect the new build process
- for 1.4 and above. (issue #9725, patch by eliel)
-
-2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson <oej@edvina.net>
-
- * /: Blocking patch already in this code
-
- * channels/chan_sip.c: Fix auth on BYE. (Different patch than for
- 1.2)
-
- * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE
-
- * channels/chan_sip.c: Final part of issue #9483 - fixing
- transfer() of sip calls in the dial plan (twilson)
-
- * channels/chan_sip.c: Issue #9439 - properly handle username
- parameters in SIP uri.
-
- * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
- lines Support SIP uri's starting with SIP: and sip: (reported by
- Tony Mountfield on the mailing list. Thanks!) ........
-
- * /, channels/chan_sip.c: Merged following patch with a lot of
- changes for 1.4 ------ Merged revisions 64514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
- lines Issue #9726 - rlister - Better logging for ACL denials
- While at it, also added better logging and handling of peers that
- are not supposed to register. My patch, stole the issue report
- from Russell. My apologies, Russell :-) ........
-
-2007-05-16 08:44 +0000 [r64515] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
- Mai 2007) | 1 line in the case immediate=yes, we directly jump
- into the dialplan, where people can use PlayTones to indicate a
- Dialtone, so we don't need to to that by ourself. also we should
- not do a dialtone_indicate for incoming calls on a TE port in
- overlapdialmode. ........
-
-2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Properly fix a problem that occurs when you
- set PARKINGEXTEN to an exten where a call is already parked.
- (issue #9723, patch by me)
-
- * res/res_features.c: When someone requests a specific parking
- space using the PARKINGEXTEN variable, ensure that no other
- caller is already there. (issue #9723, reported by mdu113, patch
- by me)
-
-2007-05-14 19:26 +0000 [r64324] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
- more
-
-2007-05-14 19:13 +0000 [r64306] Russell Bryant <russell@digium.com>
-
- * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
- just ignoring it. An unknown indication will trigger an error and
- cause sounds to stop, which in this case, is ringing.
-
-2007-05-14 18:52 +0000 [r64280] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Handle network errors, like host or network
- unreachable, in a better way. This means that calls to hosts or
- qualify (OPTION) messages will fail quicker if the TCP/IP stack
- tells us that there is an issue. Since this is an unconnected UDP
- socket, we will not get error messages directly in most cases,
- but maybe on the second and third try. This is already
- implemented in trunk.
-
-2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp <jcolp@digium.com>
-
- * codecs/codec_speex.c: Properly set datalen field when doing PLC
- in codec_speex. (issue #9722 reported by mihai)
-
- * /, main/devicestate.c: Merged revisions 64275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
- lines Only perform stripping of - strings from the channel name
- for Zap channels. Anywhere else we might remove a legitimate part
- of a device name. (issue #9668 reported by stevedavies) ........
-
- * main/channel.c: Fix scenario where if a phone that simply called
- Echo() put itself on hold it could never get off hold.
-
-2007-05-14 13:58 +0000 [r64193] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
- worrisome CDR warnings have been removed, that are either not
- helpful, or not relevant.
-
-2007-05-14 10:39 +0000 [r64157] Olle Johansson <oej@edvina.net>
-
- * main/channel.c: Add hangupcause when we lack codecs for
- transcoding
-
-2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: This concludes my final adventure with
- bitmasks and the onhold flag. Would anyone care for some peanuts?
-
- * channels/chan_sip.c: Tweak hold flags some more. They can be of
- three states when active: active, inactive, one direction.
-
- * channels/chan_sip.c: Ensure the onhold flag is set no matter what
- when being put on hold.
-
-2007-05-11 20:16 +0000 [r63982] Jason Parker <jparker@digium.com>
-
- * main/manager.c: Hide manager password from "manager show user
- foo". I realize that there are other ways to get this, but we
- really don't need to just show it in plain text so easily. Issue
- 9273, patch by junky
-
-2007-05-11 16:35 +0000 [r63905] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
- 63903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
- | 2 lines Issue 9121 - fixups for safe_asterisk script ........
-
-2007-05-11 16:05 +0000 [r63886] Russell Bryant <russell@digium.com>
-
- * main/manager.c: When MD5 authentication is not possible because
- there is no challenge present, either because the Challenge
- action was never issued, or some other reason, give a proper
- error message and return an error instead of claiming that the
- user wasn't found. (reported by jsmith on IRC)
-
-2007-05-11 15:43 +0000 [r63872] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Make the PARKINGEXTEN feature of parking
- actually work. (issue #9708 reported by mdu113)
-
-2007-05-10 23:15 +0000 [r63830] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
- lines Fix an issue with trying to kill a thread before it gets
- created. Issue 9709, patch by nic_bellamy. ........
-
-2007-05-10 22:23 +0000 [r63804] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Strip terminal escape sequences from CLI command
- output that is going to be sent out over the manager interface.
- (issue #9659, reported by pari, fixed by me)
-
-2007-05-10 20:48 +0000 [r63750] Doug Bailey <dbailey@digium.com>
-
- * main/callerid.c: Add test for negative offsets in cid data to
- prevent infinite loops.
-
-2007-05-10 20:46 +0000 [r63749] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
- lines Do not allocate SIP pvt's for PEERs we can not reach. This
- was seen as a lot of dialogs being created then immediately
- destroyed at reload/restart of the SIP channel. ........
-
-2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Use the DTMF frame on the channel when returning
- a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.
-
- * channels/chan_sip.c: Do not prematurely go on hold if sendonly
- was not actually set.
-
-2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
- lines Make sure we only create a DSP if it's requested on
- SUB_REAL ........
-
-2007-05-09 16:55 +0000 [r63612] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Modify ast_senddigit_begin() to use the same
- assumptions used elsewhere in the code in that if a channel does
- not have a send_digit_begin() callback, it only cares about DTMF
- END events. (pointed out by Michael Neuhauser on the asterisk-dev
- list)
-
-2007-05-09 16:54 +0000 [r63611] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
- lines Properly handle hints that point to multiple devices in
- chan_sip. Why chan_sip is even doing this I have no idea but I
- would rather not go into a rant. (issue #9536 reported by
- rlister) ........
-
-2007-05-09 16:43 +0000 [r63608] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Only call ast_senddigit_begin() in
- ast_senddigit() if the channel has a send_digit_begin() callback.
- Checking the END_DTMF_ONLY flag was the wrong thing to do,
- because that flag indicates that a *bridged* channel only wants
- DTMF END events coming from this channel.
-
-2007-05-09 14:50 +0000 [r63566] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
- | 2 lines Replicate fix from 51158 (app_voicemail) to
- app_directory (Issue 9224) ........
-
-2007-05-09 13:24 +0000 [r63535] Russell Bryant <russell@digium.com>
-
- * Makefile: I have seen multiple people post questions trying to
- figure out what the message "The configure script must be
- executed before running 'make'" means. So, add another like that
- says to specifically run ./configure. If this isn't obvious
- enough, then they should be using something like AsteriskNOW and
- not installing from source.
-
-2007-05-09 13:17 +0000 [r63534] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c: Merged revisions
- 62945,63402,63519 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
- 1 line when we're in state WAITING4DIGS, we use the asterisk
- tone-generator which prods us, so we can't just return -1 in
- misdn_write in this case. Added a MISDN_KEYPAD channel variable,
- and fixed the sending of keypad. this enables us to modify the
- call forward parameters in the switch. ........ r63402 | crichter
- | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
- application misdn_check_l2l1 which tries to pull up the L1/L2 on
- all ports that have the layers down in a group. It waits then for
- a timeout. This helps for scenarios where multiple PMP BRIs are
- grouped together, or where a provider has a faulty PTP
- Implementation, that looses the L2 after a while. ........ r63519
- | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
- release_chan frees ch, so we should never touch ch after
- release_chan, this may cause segfaults. ........
-
-2007-05-09 13:04 +0000 [r63532] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
- (Reported on asterisk-users)
-
-2007-05-08 22:38 +0000 [r63478] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
- | 2 lines Issue 9602 - segfault in app_macro ........
-
-2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: I mixed up the use of the find_feature()
- function, so I renamed it find_dynamic_feature, and changed the
- code to use the correct lock when using it.
-
- * res/res_features.c: Use a read/write lock when accessing the
- built-in features.
-
- * contrib/scripts/realtime_pgsql.sql (added),
- contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
- contrib/scripts to be with the rest of the sql examples. (issue
- #9676, suretec)
-
-2007-05-08 06:22 +0000 [r63360] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
- | 2 lines Issue 9527 - upon entering a folder, no message is
- selected (curmsg == -1), so deleting causes memory corruption
- (beyond bounds) ........
-
-2007-05-07 22:28 +0000 [r63329] Russell Bryant <russell@digium.com>
-
- * configs/res_pgsql.conf.sample (added),
- configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
- (added): Add a sample configuration file and example tables for
- use with res_config_pgsql. (issue #9676, suretec)
-
-2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
- revisions 63285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
- lines Properly handle what happens during a masquerade in
- relation to group counting. (issue #9657 reported by ramonpeek)
- ........
-
- * channels/chan_sip.c: Minor backport of revision 59083 in trunk.
- Don't queue an unhold frame up if the call was never on hold to
- begin with.
-
-2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson <oej@edvina.net>
-
- * main/config.c: Don't remove configuration from memory just
- because one section failed.
-
- * /: Guess svnmerge doesn't handle files that move around. Blocking
- patch to ./config.c
-
-2007-05-06 12:28 +0000 [r63152] Olle Johansson <oej@edvina.net>
-
- * main/file.c: Stop the video stream when you stop playback of all
- streams for a call
-
-2007-05-04 20:03 +0000 [r63099] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Fix a crash when checking version attribute in
- an incoming XML caps element. Issue 9667, patch by phsultan.
-
-2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni <paripurnachand@digium.com>
-
- * configs/manager.conf.sample: explanation for httptimeout in
- manager.conf
-
-2007-05-03 16:44 +0000 [r62989] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
- lines When a peer is seeded or built tell the devicestate core to
- update it's status. This is easier then having chan_sip load
- before pbx_config. (issue #9658 reported by dlynes) ........
-
-2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c: improve loader a bit, by avoiding trying to
- initialize embedded modules twice and avoiding trying to load
- modules from disk when they have been loaded already during the
- 'preload' pass (reported by blitzrage on IRC, patch by me)
-
-2007-05-03 15:23 +0000 [r62942] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
- Bryant, 2007, TM, Patent Pending). This set of changes came from
- a debugging session I had with Dwayne Hubbard. When he called
- into his home FXO, ran the Echo application, and pressed a digit,
- the digit would be echoed back and would never end. This is
- fixed, along with a couple other little improvements. * When
- chan_zap is in the middle of playing a digit to a channel, it
- feeds back null frames, not voice frames. So, I have modified
- ast_read to check the timing on emulated DTMF when it receives
- null frames, in addition to where it was doing this on voice
- frames. * Make a tweak to setting the duration on emulated DTMF
- digits. If there was no duration specified, it set it to be the
- minimum, instead of the default. * Instead of timing the emulated
- digits off of the number of samples in audio frames that pass
- through, just use time values. Now there is no code in this
- section that assumes 8kHz audio.
-
-2007-05-03 14:41 +0000 [r62913] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
- (added), pbx/ael/ael-test/ael-test18/extensions.ael,
- pbx/ael/ael-test/ael-test19/extensions.ael (added),
- pbx/ael/ael-test/ael-test19 (added),
- pbx/ael/ael-test/ref.ael-test20 (added),
- pbx/ael/ael-test/ael-test20/extensions.ael (added),
- pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
- to match what's in trunk
-
-2007-05-03 14:36 +0000 [r62912] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 61357,61770,62885 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
- 1 line some fixes for PMP Hold/Retrieve, it should work now, when
- briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
- (Di, 24 Apr 2007) | 1 line added lock for sending messages to
- avoid double sending. shuffled some empty_chans after the
- cb_event calls, this avoids that a release_complete from a quite
- different call releases a fresh created setup by accident.
- ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
- Mai 2007) | 1 line fixed the problem that misdn_write did not
- return -1 when called with 0 samples in a frame this resultet in
- a deadlock in some circumstances, when the call ended because of
- a busy extension. added encoding of keypad. ........
-
-2007-05-03 13:54 +0000 [r62883] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test18 (added),
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ael-test18/extensions.ael (added),
- pbx/ael/ael-test/ael-test18 (added),
- pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
- These mods fix bug 9623, where an '@' in the eswitch contents
- causes a syntax error. I also updated the regressions.
-
-2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming <kpfleming@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
- May 2007) | 2 lines doh... initializing the pointer variable will
- work just a bit better ........
-
- * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
- May 2007) | 7 lines increase reliability and efficiency of static
- Realtime config loading via ODBC: don't request fields we aren't
- going to use don't request sorting on fields that are pointless
- to sort on explicitly request the fields we want, because we
- can't expect the database to always return them in the order they
- were created (reported by blitzrage in person (!), patch by me)
- ........
-
- * res/res_config_pgsql.c: improve static Realtime config loading
- from PostgreSQL: don't request sorting on fields that are
- pointless to sort on use ast_build_string() instead of snprintf()
- don't request the list of fieldnames that resulted from the query
- when we both knew what they were before we ran the query _AND_ we
- aren't going to do anything with them anyway (patch by me,
- inspired by blitzrage's bug report about res_config_odbc)
-
-2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Merge changes from team/russell/inband_dtmf ...
- Fix some issues related to generating inband DTMF. There are two
- changes here: 1) The list of DTMF tones in the senddigit_begin()
- function explicitly specified 100ms of the tone followed by 100ms
- of silence. This really broke things with the way that Asterisk
- now wants complete control over when the digit begins and ends.
- So, regardless of what Asterisk really wanted to do, this was
- going to play out the tone at the length it wanted to. This
- caused various problems like DTMF translation to inband to be
- extremely unreliable. The list of tones has been changed so that
- the correct DTMF tone is played indefinitely until Asterisk tells
- it to stop. 2) ast_write() had to be modified to let a DTMF_END
- frame get processed even when a generator is present. This is how
- the tone will finally get stopped. (issues #8944, #9250, #9348,
- maybe others. Thanks to mdu113 from #8944 for the testing and
- feedback!)
-
- * main/manager.c: Backport the change that only went in to trunk
- that fixes the command manager action over http. (reported
- internally by pari and bkruse)
-
-2007-05-02 20:46 +0000 [r62738] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
- 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
- in 'h' extension louses up the dst field ........
-
-2007-05-02 17:43 +0000 [r62692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
- | 4 lines Issue 9638 - if a text frame is sent with no
- terminating NULL through a bridged IAX connection, the remote end
- will receive garbage characters tacked onto the end. ........
-
-2007-05-02 17:10 +0000 [r62689] Steve Murphy <murf@digium.com>
-
- * configs/extensions.conf.sample, main/channel.c, main/pbx.c,
- channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
- clid, src fields in channel_alloc call. b)in the channel_alloc
- func, set the cid_num and name fields from the arglist[blush]. c)
- don't update the channel app & app data fields if you are in the
- 'h' extension. d)the load_module func in cdr_radius needs to
- return DECLINE, SUCCESS.
-
-2007-05-02 06:15 +0000 [r62624] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't unlock a channel that we already know
- does not exist (propably isue 8228)
-
-2007-05-01 21:57 +0000 [r62548] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 62547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
- 4 lines Remove an unnecessary check that makes it so if you hang
- up after doing an attended transfer before the target extension
- answers the channel, the transfer is not successful. (issue
- #9338, patch by svanlund) ........
-
-2007-05-01 21:34 +0000 [r62545] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
- (found by rayjay, different fixes by me)
-
-2007-05-01 16:26 +0000 [r62497] Russell Bryant <russell@digium.com>
-
- * /, configs/indications.conf.sample: Merged revisions 62496 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
- 3 lines Add indications.conf information for the Philippines.
- (issue #9525, reported and patched by loloski) ........
-
-2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
- 4 lines This patch fixes an issue where depending on the cause
- code, when the network sends a PRI disconnect, the call may not
- be properly hung up. (issue #9588, reported and patched by
- softins) ........
-
- * include/asterisk/http.h, main/http.c: When serving dynamic
- content, include a Cache-Control header to instruct the browsers
- to not store the resulting content. (issue #9621, reported by
- Pari, patch by me)
-
-2007-04-30 14:52 +0000 [r62371] Jason Parker <jparker@digium.com>
-
- * configs/iax.conf.sample: Remove unused (and potentially
- confusing) jitterbuffer options from sample config.
-
-2007-04-30 14:36 +0000 [r62369] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c, /: Merged revisions 62368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
- lines Update copyright notice. It's now the year 2007! ........
-
-2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Fix a bug that made the "language" setting
- in zapata.conf not functional. (issue #9626, reported and fixed
- by sergee)
-
- * apps/app_meetme.c: Note that the "talker optimization" option
- will be enabled by default in 1.6
-
-2007-04-27 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.4 released.
-
-2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Fix a weird problem where when a caller
- talking to someone sitting behind an agent channel sent a digit,
- the digit would be played to the agent for forever. This is
- because chan_agent always returned -1 from its send_digit_begin
- and _end callbacks. This non-zero return value indicates to the
- Asterisk core that it would like an inband DTMF generator put on
- the channel. However, this is the wrong thing to do. It should
- *always* return 0, instead. When the digit begin and end
- functions are called on the proxied channel, the underlying
- channel will indicate whether inband DTMF is needed or not, and
- the generator will be put on that one, and not the Agent channel.
- (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
- by me)
-
-2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
- lines This transcoder message needn't be a NOTICE. I've seen it
- cause confusion more than a few times. ........
-
-2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: If no variables were passed into
- pbx_substitute_variables_helper_full(), then don't even bother
- creating a temporary bogus channel, since that is only for
- allowing certain functions to operate on the variables as if they
- were on a channel. Most importantly, this fixes a crash. (issue
- #9613, reported by callguy, fixed by me)
-
-2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
- lines Issue #7351 - SIP Cancel fails due to the wrong contact
- uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
- - THANKS!!!! THis was a hard one to catch. ........
-
- * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
- annoying DEBUG messages not controlled by option_debug (DEA).
- Thanks!
-
-2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
- lines Revert previous fix for when the IAX2 channel goes funky
- (that's the technical term). This is causing legit calls to be
- prematurely hung up. (issue #9600 reported by justdave) ........
-
- * main/channel.c: Missed an ast_app_group_discard during merge.
- Thanks blitzrage!
-
- * res/res_monitor.c: Don't always say that the channel is being
- paused if it is actually being unpaused in the Manager ack
- message. (reported by jsmith in #asterisk-bugs)
-
- * main/config.c, /: Merged revisions 61958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
- lines Don't count failed include attempts against the
- configuration include level. (issue #9593 reported by mostyn)
- ........
-
-2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
- | 2 lines handle a very bizarre race condition with channels
- being redirected before a simple switch can be started on them
- (issue #9286) ........
-
-2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
- 2 lines If the callerid= option is specified, but empty, clear
- any previous data. ........
-
- * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
- 2 lines Ensure that callerid settings are reset on a reload.
- ........
-
-2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c, main/channel.c, include/asterisk/app.h,
- funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
- lines Merge rewritten group counting support. No more storing
- data on the variable list of the channels. That was bad, mmmk?
- (issue #7497 reported by sabbathbh) ........
-
-2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
- 3 lines Fix a typo where cid_num got copied instead of cid_ani.
- (issue #9587, reported and patched by xrg) ........
-
-2007-04-24 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.3 released.
-
-2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 61786 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
- 4 lines Don't crash if a manager connection provides a username
- that exists in manager.conf but does not have a password, and
- also requests MD5 authentication. (ASA-2007-012) ........
-
- * main/channel.c, include/asterisk/channel.h: Improve DTMF handling
- in ast_read() even more in response to a discussion on the
- asterisk-dev mailing list. I changed the enforced minimum length
- of a digit from 100ms to 80ms. Furthermore, I made it now enforce
- a gap of 45ms in between digits. These values are not
- configurable in a configuration file right now, but they can be
- easily changed near the top of main/channel.c.
-
-2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
- | 1 line removed #if 0 block from chan_phone, chan_zap, and
- chan_modem restart_monitor() ........
-
-2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com>
-
- * main/dial.c: Add a few more state changes in
- handle_frame_ownerless() so that the SLA code will get notified
- of these changes even when an owner channel is not provided. This
- isn't from a specific bug report, it's just something I noticed
- while poking around.
-
-2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
- lines Allow RFC2833 to be sent in the response SDP when an INVITE
- comes in without SDP. (issue #9546 reported by mcrawford)
- ........
-
-2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Some dialplan functions, such as CUT(), expect to
- operate on variables on a channel. So, this little hack lets them
- work in places where a channel doesn't exist, such as within
- DUNDi configuration. (issue #9465, reported and patched by
- Corydon76, testing by blitzrage)
-
- * main/channel.c: Ensure that digits passing through Asterisk have
- a reasonable minimum length. It is currently 100 ms. If someone
- thinks this should be different, feel free to speak up. (related
- to issues #8944, #9250, and #9348)
-
-2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Avoid invalid seqno cycling detection. Per comment
- from Dave Troy: This adds back in some simple typecasting I had
- in an earlier version which I realize now may be breaking things.
- Issue #9554.
-
- * main/loader.c, /: Merged revisions 61704 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
- lines Fix an issue that I noticed while looking over issue 9571.
- The reload timestamp was getting set after reloading the built-in
- stuff, and before the modules. ........
-
-2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Remove a stray debug message introduced by a recent
- commit.
-
-2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
- lines If the '* to hangup' option is not enabled, we don't need
- to disable * as a valid exit key. If it was enabled, this
- statement would've never been checked in the first place. Issue
- #9552 ........
-
-2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com>
-
- * main/config.c, apps/app_voicemail.c, main/manager.c,
- include/asterisk/config.h: Fix the UpdateConfig manager action to
- properly treat "variables" and "objects" differently (a=b versus
- a=>b). (issue #9568, reported by pari, patch by me)
-
-2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
- lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
- Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........
-
-2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/manager.c: Bug 9557 - simple reason why reading a function
- always returned NULL
-
- * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
- funcs/func_groupcount.c, /, funcs/func_timeout.c,
- funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
- | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
- Channel parameter can cause Asterisk to crash. The reason this
- needs to be fixed in the functions instead of in AMI is because
- Channel can legitimately be NULL, such as when retrieving global
- variables. ........
-
-2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: allow external build systems to extract the
- required sound file versions
-
-2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net>
-
- * main/rtp.c: Clean upp formatting, add some doxygen stuff while
- we're in cleaning mode... Thanks Kevin!
-
- * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)
-
-2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: #9483, half of patch by twilson to solve 302
- redirect issues
-
- * /: Blocking AstHoloPatch from 1.2
-
-2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: This is a fix to the way CDR merge handles the data
- that results from ForkCDR.
-
-2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
- lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
- the same as OUTBOUND_GROUP except it will get unset after use so
- it won't get accidentally inherited. (issue #BE-140) ........
-
- * apps/app_speech_utils.c: Do not bother looking for a result if
- none are present.
-
- * channels/chan_sip.c: For those very verbose SIP implementations
- that attach tons of info to the Contact header... let's increase
- our variable sizes. (issue #9535 reported by jeffg)
-
-2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Eliminate a compiler warning with
- ODBC_STORAGE enabled so that it will build under dev-mode.
-
-2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com>
-
- * channels/chan_oss.c: A fix for chan_oss that resulted from the
- CDR changes; it helps to use the right info.
-
-2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't assume the callid of a dialog will be
- set, as in some circumstances it may not. (issue #9534 reported
- by tecnoxarxa)
-
-2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
- 5 lines If someone sets the "useragent" option in sip.conf to be
- empty, then don't add the User-Agent header at all. It is an
- optional header, anyway. Also, the bug report says that some of
- Japan's SIP providers don't allow it for some weird reason.
- (issue #9488, reported by makoto, fixed by me) ........
-
-2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com>
-
- * channels/chan_misdn.c: Don't export AOCD variables on
- misdn_hangup anymore, this was mainly a fix for trunk..
-
-2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
- 6 lines Fix a bug with switching between host=dynamic and using
- specific hosts for peers. The code would only reset the peer's
- address when it is dynamic if it was a new peer structure. Now,
- it will also reset the address if it was already in the peer
- list, but before the reload, it was not dynamic. (issue #9515,
- reported by caio1982, fixed by me) ........
-
- * main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
- bkruse) In passing, constify the elements of the mimetypes table.
-
- * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
- 5 lines Remove the attempt at reporting configuration errors in
- sip.conf. This can cause a bunch of improper messages when using
- realtime. I give up. As oej tried to convince me when I put this
- in, there is just no easy way to do it. (inspired by a message on
- the -dev list) ........
-
-2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com>
-
- * channels/chan_misdn.c: Export AOCD variables on misdn_hangup.
-
- * channels/chan_misdn.c: Ignore facility messages in case we don't
- have a corresponding channel object.
-
- * channels/chan_misdn.c: AOCD's are now exported to asterisk
- channel variables.
-
-2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com>
-
- * main/Makefile, main/http.c, main/minimime (removed): File upload
- support was added to solve some needs for the Asterisk GUI.
- However, after much discussion, it has been decided that adding
- this to 1.4 is not in the best interests of the project. It has
- been removed here, but will remain in trunk.
-
-2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
- 2007) | 2 lines msns config parameter defaults to '*' ........
-
-2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com>
-
- * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
- previous fix to overcome a compiler warning; the app NoCDR() has
- been updated to mark the channel CDR as POST_DISABLED instead of
- destroying the CDR; this way its flags are propagated thru a
- bridge and the CDR is actually dropped. The cases where only one
- channel in a bridge has a CDR was cleaned up.
-
-2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
- lines - Don't send ActionID before Response: header. - Don't use
- a blank in an AMI header ........
-
-2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/minimime/mm_envelope.c, res/res_features.c: fix up some
- warnings found using --enable-dev-mode
-
- * main/minimime/Doxyfile (removed),
- main/minimime/tests/messages/CVS (removed),
- main/minimime/tests/CVS (removed): remove some more stuff we
- don't need
-
-2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com>
-
- * main/minimime/test (removed): Remove another directory that
- should no longer be there
-
- * main/minimime/Make.conf (removed), main/minimime/mytest_files
- (removed), main/minimime/.cvsignore (removed), main/minimime/sys
- (removed), main/minimime/mm-docs (removed): Remove various files
- that I thought I already removed.
-
-2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Use the appropriate interface name with
- COMPLETECALLER. Issue 9395.
-
-2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com>
-
- * channels/chan_oss.c, main/channel.c, main/cdr.c,
- channels/chan_phone.c, channels/chan_misdn.c,
- channels/chan_skinny.c, channels/chan_features.c,
- channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
- channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
- channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
- channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
- include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_iax2.c: This is a big improvement over the current
- CDR fixes. It may still need refinement, but this won't have as
- many folks bothered.
-
-2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: Add final new line after JabberEvent
-
-2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
- lines Allow matching on names shorter than 3 chars. This also
- fixes the case where somebody wants to match on less then 3
- chars. Issue 9071 ........
-
-2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
- | 2 lines Don't check for error when lowering priority (according
- to the manpage, it should never happen anyway). It might could
- happen, though, if another thread messed with the priority, so
- safeguard against that (reported via -dev list). ........
-
- * channels/chan_local.c, /: Merged revisions 60846 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
- Apr 2007) | 2 lines Bug 9505 - If the return value for
- local_queue_frame is set, then p->lock is no longer valid.
- ........
-
-2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
- lines When calling a device that then forwards us elsewhere... we
- have to make our channels compatible if it is the only channel
- being dialed. (issue #9445 reported by marcelbarbulescu) ........
-
- * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
- MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)
-
-2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
- | 2 lines Gosub called within a Macro resets the arguments
- improperly and causes general weirdness. (Issue 8329) ........
-
- * main/http.c: Fix --enable-dev-mode
-
- * channels/chan_oss.c: Off by one error, resulting in a crash
- (Issue 9500)
-
- * /, main/file.c: Merged revisions 60660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
- | 2 lines Bug 9486 - memory leak when opening a filestream
- ........
-
-2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com>
-
- * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
- main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
- main/manager.c, main/minimime/mm_mimepart.c,
- main/minimime/test.sh, configure, include/asterisk/compat.h,
- main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
- main/minimime/tests/parse.c, main/minimime/mm_base64.c,
- main/minimime/mm_mimeutil.c, main/minimime/mm.h,
- main/minimime/tests, main/minimime/mm_header.c,
- main/minimime/mm_error.c, main/Makefile,
- main/minimime/mm_codecs.c, main/minimime/mm_param.c,
- configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
- include/asterisk/manager.h, main/minimime/strlcpy.c,
- configs/http.conf.sample, main/minimime/mm_parse.c,
- main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
- main/minimime/mm_util.c, main/minimime/mm_envelope.c,
- main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
- main/minimime/tests/messages/test2.txt,
- main/minimime/tests/messages/test3.txt,
- main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
- main/minimime/tests/messages/test4.txt,
- main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
- main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
- main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
- main/minimime/mimeparser.l, main/minimime/mm_context.c,
- main/minimime/mimeparser.tab.h, main/minimime (added),
- main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
- main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
- main/minimime/mimeparser.y, Makefile.moddir_rules,
- main/minimime/sys, main/minimime/tests/Makefile: To be able to
- achieve the things that we would like to achieve with the
- Asterisk GUI project, we need a fully functional HTTP interface
- with access to the Asterisk manager interface. One of the things
- that was intended to be a part of this system, but was never
- actually implemented, was the ability for the GUI to be able to
- upload files to Asterisk. So, this commit adds this in the most
- minimally invasive way that we could come up with. A lot of work
- on minimime was done by Steve Murphy. He fixed a lot of bugs in
- the parser, and updated it to be thread-safe. The ability to
- check permissions of active manager sessions was added by Dwayne
- Hubbard. Then, hacking this all together and do doing the
- modifications necessary to the HTTP interface was done by me.
-
-2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * UPGRADE.txt: clarified a sentence in the format_wav section
-
- * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
- plan to remove GAIN code from trunk
-
-2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: When a station picks up a trunk that was on
- hold, make the hints reflect that nobody has the trunk on hold
- anymore.
-
- * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
- reported by francesco_r, fixed by me) * The original behavior was
- that if one station put a call on hold, another one picked it up,
- and then hung up, the code would still consider the call on hold
- by the first station, so the trunk would not be hung up. However,
- to better comply with what most people seem to expect it to
- behave, it will now hang up the trunk. * Fix a problem with
- "barge=no". This was only intended to prevent people from joining
- calls that are in progress. However, it also prevented other
- people from picking up a call that was on hold. This has been
- fixed. * When there are no active stations on a trunk and it is
- on hold, the code now indicates the HOLD and UNHOLD conditions to
- the trunk channel. This allows music on hold to be played to the
- trunk when it is on hold.
-
-2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we check the faxdetect option
- before doing fax processing
-
- * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
- lines There should only be one code path for doing DTMF
- conditionals on channels. This fixes it. ........
-
-2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
- | 2 lines remove undocumented 'cardsmode' parameter and stop
- searching for transcoders during reload() ........
-
-2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Add support for returning different
- types of results (ie: NBest).
-
-2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * formats/format_wav.c: modified default GAIN for issue 5823,
- thanks jrwalliker
-
-2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com>
-
- * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
- some clarification to the example configs for CDRs, on how to
- select a backend. Also, made cdr-csv the default if you 'make
- samples', and no other changes.
-
-2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
- lines Just because we can't find the voicemail configuration
- file, doesn't mean that the module failed to load. The user could
- be using realtime. Issue #9473 ........
-
-2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com>
-
- * main/http.c: Add the MIME type for gif by request from Pari
-
-2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
- lines Only unlock our pvt and net locks if we are actually going
- to try to lock the owner again. (issue #9472 reported by zoa)
- ........
-
-2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 60134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
- 6 lines It is valid to redirect channels via the manager
- interface that are not in the UP state. Instead of checking for
- that to prevent to ensure a dead channel doesn't get redirected,
- just use the ast_check_hangup() API call. (issue #9457, reported
- by Callmewind, patch by me) (related to issue #8977) ........
-
- * channels/chan_sip.c: Add a Content-Length of 0 to the response
- built by transmit_response_with_unsupported(). (issue #9454,
- reported by makoto, fixed by me)
-
- * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
- 4 lines Fix the return value of handle_common_options() so that
- it always properly indicates whether it handled the option or
- not. (issue #9455, reported by Netview, fixed by me) ........
-
- * apps/app_meetme.c: Fix a problem where if a trunk was hung up
- while it was on hold, all of the hints would reflect the line
- still on hold, even though it should reflect that it is back to
- not in use. (issue #9459, reported by francesco_r, fixed by me)
-
- * /: Blocked revisions 60016 via svnmerge ........ r60016 | russell
- | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a
- missing "\r\n" in the body of the NOTIFY that is sent to indicate
- the status of a transfer. (issue #9388, reported by rarritt)
- ........
-
- * /: Blocked revisions 60014 via svnmerge ........ r60014 | russell
- | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the
- more generic check for "sed -r" support that was already present
- in 1.4. (related to issue #9399) ........
-
- * /: Blocked revisions 60012 via svnmerge ........ r60012 | russell
- | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On
- Darwin, the -r argument to sed is not valid. It has to be -E.
- (issue #9399, reported by jcovert) ........
-
-2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Don't clash when a person both speaks
- and uses DTMF.
-
-2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
- 4 lines Don't attempt to report configuration errors in
- build_user(). oej pointed out that for a "friend" entry, this
- won't work, because all user options are valid for peers, but not
- the other way around. ........
-
- * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
- 3 lines Make chan_sip report when it encounters an unknown
- option. (issue #9440, reported by nightcrawler) ........
-
- * /, main/app.c: Merged revisions 59886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
- 5 lines When doing a built-in blind or attended transfer, restore
- the ability to use '#' to terminate the number and immediately do
- the transfer instead of having to dial the number and just wait
- for the feature digit timeout. (issue #8366, xueliangliang)
- ........
-
- * Makefile: Ensure that menuselect gets executed in dependency
- check mode every time you run make.
-
-2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
- Merged revisions 59788,59803 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
- lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
- or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
- 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........
-
-2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
- Merged revisions 59623-59624,59639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
- 1 line we can now make 30 channels on a PRI (before we forgot
- chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
- (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
- r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
- 1 line added option which allows us to accept incoming SETUP
- Messages without automatically sending Proceeding or Setup
- Acknowledge, this is useful with some broken switches and if you
- want to Release incoming calls without previously having
- acknowledged them. The new option is
- noautorespond_on_setup=yes|no default is no, so we don't break
- the existing behaviour ........
-
-2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
- lines Increase the maximum size for a string of mailboxes to
- 1024. (issue #9270 reported by rtucker) ........
-
-2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
- not the test. As per 9435, thanks to marcelbarbulescu
-
-2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com>
-
- * main/netsock.c, /: Merged revisions 59608 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
- 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
- This is needed by the patch that went in for issue 7874.
- chan_iax2 needs to be able to create socket that is lisetning on
- INADDR_ANY, but also be able to bind sockets to specific
- addresses. (Thanks to Stevenson on the asterisk-dev mailing list
- for explaining why this flag was needed.) ........
-
-2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com>
-
- * configure, main/Makefile, acinclude.m4: Add linux-uclibc host
- arch..."thingy". Sorry, I don't know what it's called...
-
-2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
- include/asterisk/cdr.h: several changes via kpflemings review
-
- * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
- include/asterisk/cdr.h: These mods fix CDR issues from 8221,
- 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
- from transfer situations.
-
- * configs/extensions.conf.sample: A small clarification to keep
- bugs from being filed, and confusion from rising, if
- clearglobalvars is set, and globals are set in the AEL file.
- (9419)
-
-2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com>
-
- * res/res_jabber.c: When building a response to a subscription, the
- "from" must be the full Jabber ID. This fixes some problems where
- jabber users are not able to add their Asterisk account to their
- user list, since they are unable to get Asterisk to approve their
- subscription. (issue #8210, reported by caspy, and verified by
- bradtem)
-
-2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
- lines Keep a global array of variables indicating whether certain
- conference rooms are in use. This ensures that two people going
- into a new dynamic conference when the 'e' option is set don't go
- into the same conference room. (issue #8835 reported by eliel)
- ........
-
-2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, /: Merged revisions 59357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
- 5 lines If an error occurs when reading from an RTP socket, and
- the error code does not indicate that we should try again, then
- return NULL instead of a "null frame". This will prevent Asterisk
- from trying over and over again, and eventually causing the
- system to crash. (issue #8285, john) ........
-
- * /: Blocked revisions 59355 via svnmerge ........ r59355 | russell
- | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport
- the change to chan_iax2 to return NULL instead of a "null frame"
- from its read callback. See revision 59341 to the 1.4 branch for
- more info. ........
-
- * channels/chan_iax2.c: When the IAX2 read callback gets called,
- return NULL instead of a "null frame". This will cause Asterisk
- to hangup the call instead of keep trying whatever it was doing.
- Under normal conditions, this function would *never* be called.
- However, the author of this patch says an error will occur that
- will cause it to get called every 100 thousand calls or so. When
- this does happen, it puts the channel in a loop that eventually
- brings down the system. So, hangup up the call is certainly a
- better alternative. (issue #8286, john)
-
- * Makefile: Export the GTK2 library and include information to sub
- Makefiles.
-
-2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
- | 3 lines Issue 9415 - No point to getting a diagnostic field if
- we aren't doing anything with the information. (Plus, it tends to
- crash the Postgres ODBC driver.) ........
-
- * /: Blocked revisions 59299 via svnmerge ........ r59299 |
- tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines
- Change ENV section to use setenv, instead of putenv (Alexandru
- Pirvulescu <sigxcpu@gmail.com>, reported via -dev list) ........
-
-2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Another crash that I thought we had fixed already
- - Issue 9396
-
- * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
- | 2 lines Oops ........
-
- * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
- | 2 lines Fix a few remaining bad mmap(2) return values ........
-
-2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
- 3 lines Fix the check of the return value from mmap(). Thanks to
- Corydon for catching this one. ........
-
- * apps/app_directory.c: Fix app_directory to actually compile with
- ODBC_STORAGE, and update the code to the latest res_odbc API.
-
- * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
- The Makefile did not properly ensure that this information got
- copied from what was selected for app_voicemail. (issue #9224)
-
- * channels/chan_sip.c: Fix the check that ensures that the CHANNEL
- function's first argument is "rtpqos". Thanks, Corydon. :)
-
-2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
- asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
- this case the proper thing to do. This change now has it doing
- the proper thing.
-
-2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
- 4 lines Fix the use of the "sourceaddress" option when "bindaddr"
- is set to 0.0.0.0 instead of having each interface explicitly
- listed. (issue #7874, patch by stevens) ........
-
- * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
- function to just be additional parameter of the CHANNEL function.
- This way, it will be possible for other RTP based channel drivers
- to expose this information in the future.
-
-2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
- Mär 2007) | 1 line fixed #9355 ........
-
-2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Oops, this should be case insensitive
-
-2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
- asterisk). I turned a duplicate context from a WARNING to an
- ERROR. Now you get a module load failure, and asterisk just
- exits. That's better than a crash, right\?
-
-2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Change this to a single dp function to make
- oej happy.
-
-2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com>
-
- * main/config.c: Fix for 9257; by eliminating the globals in
- main/config.c, we make it thread-safe, which is a minimum
- requirement.
-
-2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Add ability to specify no timeout. This
- means as soon as the prompt is done playing it moves on to the
- next priority.
-
-2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Somehow the code for building the email for
- voicemail got out of sync. This change makes a few tweaks to get
- 1.4 in sync with trunk. (issue #9301)
-
- * apps/app_meetme.c: Fix some codec negotiation problems when
- CallerID support is not enabled in SLA. (issue #9308, reported by
- twilson)
-
-2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Make SpeechBackground obey the digit
- timeout value.
-
-2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Rename the new dialplan functions to match
- the variable name
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
- AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
- some because they get set in sip_hangup. So, there are common
- situations where the variables will not be available in the
- dialplan at all. So, this patch provides an alternate method for
- getting to this information by introducing AUDIORTPQOS and
- VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
- with some testing by blitzrage)
-
-2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
- pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
- and STANDALONE_AEL
-
-2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
- include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
- channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
- provides a dsp pipeline for i.e. echo cancellation modules, make
- chan_misdn use it. * add a check for linux/mISDNdsp.h to
- configure.ac and update the autogenerated files: 'configure',
- 'autoconfig.h.in' (the 'configure' script was not in sync with
- the latest configure.ac, so the diff is a bit bigger than
- expected).
-
-2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com>
-
- * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
- aelparse binary! DONT_OPTIMIZE should now work once again.
-
-2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
- lines Only try to handle a response if it has a response code.
- (ASA-2007-011) ........
-
-2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com>
-
- * /: blocking out the fix in 59187... already incorporated here
-
- * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
- line Added a few words in the Macro doc strings about the
- behavior of macros with hangups (et al.), as per 9337 ........
-
-2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: don't allow string input to overrun the
- buffer to hold it (ASA-2007-010)
-
- * channels/chan_misdn.c: remove variables that are no longer used
- (--enable-dev-mode is good, developers should be using it)
-
-2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com>
-
- * utils/Makefile: The stuff in utils was compiling with -O6 even if
- DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
-
-2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com>
-
- * main/http.c: Add svg mimetype for pari.
-
- * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
- lines Indicate the filename changed when it is changed. (issue
- #9311 reported by jsmith) ........
-
- * channels/chan_sip.c: Until we can do media level parsing for
- sendrecv/etc just use the first value found. This crept up when a
- phone was offered audio+video and returned an inactive video
- stream. chan_sip thought the phone said to put the person on hold
- but that was totally wrong. (issue #9319 reported by benbrown)
-
-2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/logger.c: Fix defines for inline stack backtraces (only used
- by developers anyway)
-
-2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com>
-
- * channels/iax2-parser.c: Copy len variable as well, should fix
- remaining IAX2 DTMF issues.
-
-2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com>
-
- * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
- return it to its previous, untouched, state.
-
- * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
- The fix for the AEL <<security hole>> (bug 9316) is here...
-
-2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h: Merged revisions
- 58849-58850,59062-59063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
- 1 line added method standard_dec for dialing out on groups, to
- avoid conflicts, which caused issues with some ISDN providers
- ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
- Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
- crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
- avoid sending a disconnect when we already received one. ........
- r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
- 1 line modified a loglevel ........
-
-2007-03-19 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.4.2 released.
-
-2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Oops, this should have been a %d all along
-
-2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
- reported by ajohnson)
-
-2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
- via -dev list)
-
-2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
- code 0 (reported by qwerty1979)
-
-2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Don't return a non-zero return code if the
- profile doesn't exist, to match what the documentation says it
- already does. (#9307 Reported by kkiely)
-
-2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com>
-
- * apps/app_page.c: Wait for the async thread to exit when hanging
- up all of the paged phones under all circumstances. (issue #9181
- reported by PhilSmith)
-
-2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: fix a couple SLA documentation
- references
-
- * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
- (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
- doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
- doc/channelvariables.txt (added), doc/ael.txt (added),
- doc/billing.tex (removed), build_tools/prep_tarball,
- doc/callingpres.txt (added), doc/enum.txt (added),
- doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
- doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
- doc/security.txt (added), doc/imapstorage.txt (added),
- doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
- doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
- doc/iax.txt (added), doc/ael.tex (removed),
- doc/channelvariables.tex (removed), doc/enum.tex (removed),
- doc/security.tex (removed), doc/math.txt (added), Makefile,
- doc/imapstorage.tex (removed), doc/privacy.tex (removed),
- doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
- (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
- doc/chaniax.txt (added), doc/app-sms.txt (added),
- doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
- doc/ices.txt (added), doc/dundi.tex (removed),
- doc/linkedlists.txt (added), doc/queuelog.txt (added),
- doc/extconfig.txt (added), doc/radius.txt (added),
- doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
- doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
- (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
- doc/queuelog.tex (removed), doc/configuration.txt (added),
- doc/asterisk-conf.txt (added), doc/sla.pdf (added),
- doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
- (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
- doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
- doc/channels.txt (added), doc/ip-tos.tex (removed),
- doc/extensions.txt (added), doc/queues-with-callback-members.txt
- (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
- doc/misdn.txt (added), doc/manager.txt (added),
- doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
- doc/billing.txt (added), doc/localchannel.txt (added),
- doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
- (added), doc/00README.1st (added): Making these documentation
- changes in the 1.4 branch upset various people, so these chanes
- will only be done in the trunk.
-
- * build_tools/prep_tarball: Add the --pdf option to the usage of
- rubber in prep_tarball
-
- * Makefile, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- configure script checking for GTK2 and some additional Makefile
- targets to support gmenuselect
-
-2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
- common syntax and update the resulting appdocs TeX file
-
-2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com>
-
- * doc/asterisk.tex: add a link to the rubber homepage
-
-2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_setcdruserfield.c, main/pbx.c,
- apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
- Expand deprecation warnings from simply warning on use to the
- builtin documentation.
-
-2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com>
-
- * doc/asterisk.tex, Makefile: Add Asterisk version information to
- the generated PDF
-
- * build_tools/prep_tarball: have prep_tarball attempt to build
- asterisk.pdf
-
-2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_realtime.c: Function works fine, but the documentation
- is backwards.
-
-2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com>
-
- * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
- (added), doc/freetds.txt (removed), doc/odbcstorage.txt
- (removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
- doc/model.txt (removed), doc/channelvariables.txt (removed),
- doc/ael.txt (removed), doc/billing.tex (added),
- doc/callingpres.txt (removed), doc/enum.txt (removed),
- doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
- doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
- doc/security.txt (removed), doc/imapstorage.txt (removed),
- doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
- doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
- doc/iax.txt (removed), doc/ael.tex (added),
- doc/channelvariables.tex (added), doc/enum.tex (added),
- doc/security.tex (added), doc/math.txt (removed), Makefile,
- doc/imapstorage.tex (added), doc/privacy.tex (added),
- doc/realtime.txt (removed), doc/dundi.txt (removed),
- doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
- (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
- doc/ast_appdocs.tex (added), doc/realtime.tex (added),
- doc/ices.txt (removed), doc/dundi.tex (added),
- doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
- doc/extconfig.txt (removed), doc/radius.txt (removed),
- doc/cliprompt.tex (added), doc/chaniax.tex (added),
- doc/hardware.txt (removed), doc/mp3.txt (removed),
- doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
- (added), doc/queuelog.tex (added), doc/configuration.txt
- (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
- (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
- doc/h323.txt (removed), doc/mp3.tex (added),
- doc/configuration.tex (added), doc/asterisk-conf.tex (added),
- doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
- doc/ip-tos.tex (added), doc/extensions.txt (removed),
- doc/queues-with-callback-members.txt (removed), doc/apps.txt
- (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
- (removed), doc/manager.txt (removed), doc/jitterbuffer.tex
- (added), doc/extensions.tex (added), doc/billing.txt (removed),
- doc/localchannel.txt (removed),
- doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
- (removed), doc/00README.1st (removed): Merge changes from
- svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
- directory into a single LaTeX formatted document so that we can
- generate a PDF, HTML, or other formats from this information. *
- Add a CLI command to dump the application documentation into
- LaTeX format which will only be include if the configure script
- is run with --enable-dev-mode. * The PDF turned out to be close
- to 1 MB, so it is not included. However, you can simply run "make
- asterisk.pdf" to generate it yourself. We may include it in
- release tarballs or have automatically generated ones on the web
- site, but that has yet to be decided.
-
-2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Don't assume that the pvt structure will
- still exist after calling schedule_delivery as it may not. (issue
- #9278 reported by fmachado)
-
-2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Some people like to put "limitonpeer"
- instead of "limitonpeers" in their configuration. While we're at
- it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
- issue #9172)
-
- * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
- examples section
-
- * doc/security.txt, /: Merged revisions 58896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
- 3 lines Add a note to the security file that the Asterisk CLI and
- log files may contain sensitive information, and that people
- should keep this in mind. ........
-
- * configs/sla.conf.sample, apps/app_meetme.c: By default, don't
- attempt to do any CallerID handling at all with SLA because it is
- known to not work properly in some situations. However, add an
- option to enable it for those that would like to use it anyway.
- The short story behind this is that to properly handle CallerID
- with SLA, we need the ability to change the CallerID on an
- existing call, and we are not ready to handle that.
-
-2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Issue 9162 -
- pbx_substitute_variables_helper assumes the buffer is initialized
- to all zeroes. This fixes a case where it wasn't.
-
-2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Ensure that the blinky lights show that the
- trunk stopped ringing when the trunk hangs up before a station
- has answered it. (issue #9234, reported by francesco_r)
-
- * configs/sla.conf.sample: fix the reference to the SLA
- documentation
-
-2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
- lines Issue #9229 - No port in request URI on register to non
- default SIP ports (neelakantan) ........
-
- * channels/chan_sip.c: Don't hangup the call on OK or errors on
- MESSAGE and INFO inside of a dialog (like video update requests).
-
- * channels/chan_sip.c: Issue #9251 - Clear From URI from user
- attributes (tgrman)
-
-2007-03-12 16:52 +0000 [r58833] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 58832 via svnmerge ........ r58832 | file |
- 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't
- use the assembler version of fetchadd_int under Intel Macs.
- (issue #9254 reported by darrell budic) ........
-
-2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 57034,57523,57753,58558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
- 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
- bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
- 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
- r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
- 1 line fixed another place where the out_cause was hardcoded to
- 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
- Mar 2007) | 1 line we can free channel 31 as well, since we can
- occupy it ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: added UU transceiving and
- corect handling for rdnis
-
-2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Allow RFC2833 compensation to compensate for even
- stupider implementations by queueing up the end frame at the
- start, not the actual end. (issue #8963 reported by AndrewZ)
-
- * channels/chan_sip.c, configs/sip.conf.sample: Add
- matchexterniplocally setting which only substitutes your
- externip/externhost setting if it matches the localnet setting. I
- know of at least two people who need opposite settings, so I made
- it an option! (issue #8821 reported by kokoskarokoska)
-
-2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a few more places in chan_iax2 where
- the ast_frame used for receiving a frame was not properly
- initialized. - Interpolating a frame when the jitterbuffer is in
- use - decrypting a frame when IAX2 encryption is on - frames in
- an IAX2 trunk
-
- * apps/app_meetme.c: Make the compiler happy and initialize a
- variable.
-
- * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
- Merge some updates to the SLA documentation. I plan to keep
- working on this to explain all of the expected behavior with call
- handling, configuration details for specific phones, and other
- things. However, I got tired of doing it in plain text, so I
- switched to using LaTeX. I have included the PDF version. I
- haven't been able to get a nice looking plain text version out of
- it yet, but I'm not terribly concerned since this is supposed to
- be more of the manual, while the plain text sample configuration
- file is the reference.
-
-2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix spelling of unavailable in voicemail
- documentation. (issue #9248 reported by tensai)
-
- * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
- lines If we are unable to lookup the host in a c line we have to
- abort, otherwise the previous data is gone and we will
- (potentially) have no data when all is said and done. ........
-
-2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Hang up the channel that put the call on hold
- in the event processing thread to avoid a race condition. Also,
- if the station originated the call that it is putting on hold,
- don't hang up the trunk if it was the only station on the call
- and it is hanging up due to hold and not a normal hangup.
-
- * channels/chan_zap.c: Add a missing break statement so that
- handling the above event does not incorrectly destroy the
- channel. (issue #9242, andrew)
-
-2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Fix segfault (Issue 9236)
-
-2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Refactor hold handling a bit so that it does
- not require keeping the call up when a call is put on hold.
-
-2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Make early SDP seeding even smarter! We have to check
- codecs in the make_compatible function too. (issue #9221 reported
- by marcelbarbulescu)
-
- * main/dsp.c, /: Merged revisions 58388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
- lines Only print out debug message if the definition that makes
- the variables shows up was actually defined. (issue #9233
- reported by serginuez) ........
-
-2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/http.c: this change was not needed; fclose() handles closing
- the file descriptor already
-
- * apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
- value
-
- * main/http.c: fix two cases where HTTP session file descriptors
- would not be closed
-
-2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, configure, configure.ac: If we receive
- ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
- tzafrir) Also, update the configure script to make sure that we
- don't try to build chan_zap if the installed version of zaptel
- does not include ZT_EVENT_REMOVED.
-
- * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
- Moore) Merged revisions 58242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
- 7 lines Fix a problem where the Asterisk channel name could be
- that of the wrong IAX2 user for a call. This is because the first
- step of choosing this name is to look for an IAX2 peer that
- happens to have the same IP/port number that this call is coming
- from and assuming that is it. However, this is not always
- correct. So, I have made it change this name after authentication
- happens since at that point, we have an exact match. ........
-
-2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
- at least one matching codec before attempting early bridge SDP
- seeding. (issue #9221 reported by marcelbarbulescu)
-
-2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com>
-
- * /: Blocked revisions 58167 via svnmerge ........ r58167 | russell
- | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a
- misplaced block of code in the 1.2 version of the patch to fix
- issue #8977 ........
-
- * main/manager.c, /: Merged revisions 58164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
- 4 lines If the channels acquired using the manager Redirect
- action are not up, then don't attempt to do anything with them.
- It could lead to weird behavior, including crashes. (issue #8977)
- ........
-
-2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
- line Fix for 9220: Eyebeam cannot renew subscriptions for
- presence info. Reason: re-SUBSCRIBE requests don't include Accept
- headers, which the rfc says are optional (to put it tersely), (it
- uses MAY), and luckily, the sip_pvt struct has the format info
- stored, so we simply leave it if the format is set, and the
- accept header null. ........
-
-2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com>
-
- * configs/voicemail.conf.sample: Clarify the documentation of the
- dialout and sendvoicemail options. (issue #9000, caio1982 and
- serge-v)
-
-2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
- lines Change error message to proper message ........
-
-2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com>
-
- * channels/chan_skinny.c: Return an error of transmit_response is
- called without a session. (issue #9002)
-
-2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Since chan_iax2 does not support reception
- of DTMF with duration ensure that it is set to 0 on the frame.
- (issue #8521 reported by gdhgdh)
-
- * apps/app_meetme.c: Don't create a listen channel and record the
- conference unless the option is turned on. (issue #9204 reported
- by francesco_r)
-
- * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
- lines Make create_dirpath use our standard for return values. -1
- is failure, 0 is success. (issue #9205 reported by ballares)
- ........
-
-2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57825 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
- line Fixed a typo introduced via 9156 (either the gotos or their
- doc strings are wrong) ........
-
-2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Don't allow a NULL pointer to reach
- ast_frdup. (issue #9155 reported by cmaj)
-
- * res/res_jabber.c: Don't reference a potentially NULL pointer.
- (issue #9199 reported by klolik)
-
- * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
- reported by edgreenberg)
-
-2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
- pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
- Updated the regression tests
-
-2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
- | 2 lines Memory leak of a list, if call recording was abandoned
- ........
-
-2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/say.c: submitted patch for Georgian language, issue 9010,
- submitted by Alexander Shaduri
-
-2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: add missing configuration template.
- Thanks to Lacy Moore on asterisk-users for pointing this out\!
-
-2007-03-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.1 released.
-
-2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the check that is used to
- determine whether zaptel transcoder support is present. The
- interface has changed.
-
-2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
- lines If a SIP message comes in and goes to a method handler that
- requires additional values that may not be present then send back
- an error. ........
-
-2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57458 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
- line further refinement in wording of goto documentation, as per
- 9156, goto not proceeding to next instruction ........
-
- * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
- right, but 9184 points out the problem-- the escape is removed by
- pbx_config, and pbx_ael should also, before sending it down into
- the pbx engine. Also, you have to insert it back in, if you are
- generating extensions.conf code from the AEL.
-
-2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com>
-
- * main/file.c: Return the correct digit that interrupted the
- stream. This fixes exiting the Background application when using
- the m option. (issue #9176, mjagdis)
-
- * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
- include/asterisk/channel.h: Merge changes from
- svn/asterisk/team/russell/sla_updates * Originally, I put in the
- documentation that only Zap interfaces would be supported on the
- trunk side. However, after a discussion with Qwell, we came up
- with a way to make IP trunks work as well, using some things
- already in Asterisk. So, here it is, this now officially supports
- IP trunks. * Update the SLA documentation to reflect how to setup
- IP trunks. * Add a section in sla.txt that describes how to set
- up an SLA system with voicemail. * Simplify the way DTMF
- passthrough is handled in MeetMe. * Fix a bug that exposed itself
- when using a Local channel on the trunk side in SLA. The
- station's channel needs to be passed to the dial API when dialing
- the trunk. * Change a WARNING message to DEBUG in channel.h. This
- message is of no use to users.
-
-2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 57317 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
- 2007) | 2 lines Don't even attempt to optimize things when a
- proxy channel is involved. It will just explode in weird and
- unexplaineable ways. (issue #9175 reported by
- clegall_proformatique) ........
-
-2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
-
-2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
- docs
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
- from svn/asterisk/team/russell/sla_updates * Add support for
- private hold. By setting "hold=private" for a trunk, only the
- station that put the call on hold will be able to retrieve it
- from hold. Also, by setting "hold=private" for a station, any
- call that station puts on hold can only be retrieved by that
- station.
-
- * apps/app_meetme.c: Minor formatting change
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
- svn/asterisk/team/russell/sla_updates * Add support for the
- "barge=no" option for trunks. If this option is set, then
- stations will not be able to join in on a call that is on
- progress on this trunk.
-
-2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
- line a small documentation update, to reflect reality in the goto
- doc strings, as per 9156, Goto does not proceed to next prio if
- jump fails ........
-
-2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
- 2007) | 2 lines Fix a few more issues with the agent logoff CLI
- command. (issue #9123 reported by arbrandes) ........
-
-2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
- changes from svn/asterisk/team/russell/sla_updates * Add support
- for station ring delays. Ring delays can be set globally for a
- station or for specific trunks on the station. * Fix a few bugs
- in existing code. * Restructure and Reorganize code to improve
- readability and maintainability. * Improve formatting of the "sla
- show (trunks|stations)" CLI commands.
-
-2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Picky compiler...
-
- * apps/app_speech_utils.c: Better handle timeouts when the
- individual speaks after everything has been played but before the
- timeout ends.
-
-2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: I was surprised that I had not yet downgraded
- missing goto targets and macro call defs to a warning, in case
- they are in extensions.conf; I rectified this problem. Also, A
- goto in a macro to a target in a catch block was not being found;
- I fixed this too; the cause was that I needed to treat catch
- statements like an extension in the find_match code.
-
-2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Fix voicemail email attachments. I missed
- the conversion of one of the line endings and there was an extra
- one where it should not have been. (issue #9128)
-
-2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
- picky... show deprecation warning in application help, too
- (reported via list)
-
-2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com>
-
- * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
- if a device was not specified in alsa.conf, then we just use the
- system default, instead of creating our own default of hw:0,0.
- (issue #9139)
-
-2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
- lines Obey the clearglobalvars option in extensions reload (or
- dialplan reload depending on your version). (issue #9146 reported
- by ramonpeek) ........
-
-2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a crash in my last change to
- iax2_indicate(). (issue #9150)
-
-2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com>
-
- * apps/app_record.c: Update app_record documentation to use new CLI
- command, core show file formats. (issue #9151 reported by junky)
-
- * main/pbx.c: Use ast_strlen_zero to see if the language and/or
- context argument is not present for Background instead of just
- checking if it is NULL. (issue #9141 reported by mjagdis)
-
-2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Do more complete locking of the
- chan_iax2_pvt struct in the indicate callback. (Problem brought
- up by Ben Smithurst on the asterisk-dev list)
-
-2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Allow both of the show version files and core
- show file versions CLI commands to work. (issue #9135 reported by
- mvanbaak)
-
-2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Move a comment to be in the correct struct.
-
- * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell
- | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure
- that lock.h is included in utils.c with AST_API_MODULE defined so
- that the implementations will be properly included when the
- AST_INLINE_API functions are not going to be inlined. (issue
- #9124, festr) ........
-
-2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/channel.c, /: Merged revisions 56684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
- | 3 lines Issue 9130 - If prev is the last item on the channel
- list, then evaluating additional conditions (e.g. name prefix)
- will cause a NULL dereference. ........
-
-2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure to set a speeddials parent on
- creation. Don't crash if hold is pressed when no call is active.
- Don't return in places that we shouldn't..
-
-2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: update to match zaptel 1.4 API change that
- was committed a few minutes ago
-
-2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /: Merged revisions 56504 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
- 8 lines Fix up a couple more signal handlers to not do bad things
- that could cause various undesirable results. The other day, I
- made Asterisk deadlock by hitting Control-C because of a bad
- signal handler. Now, signal handlers just set a flag and write to
- an alert pipe for the flag to be handled. Then, there is another
- thread that is monitoring for these flags. If being run in
- console mode, it is just the main thread. If Asterisk is in the
- background, a thread is created to do it. ........
-
-2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com>
-
- * main/sched.c: Change log notice to debug. It is possible for a
- scheduled item to execute and be deleted at close to the same
- time and unavoidable. If this happens this message creeps up.
-
-2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
- 4 lines Don't destroy mutexes before unregistering all of the
- entry points from the core. Also, fix a potential memory leak
- from not destroying the locks for all of the possible call
- numbers (about 32k of them). ........
-
-2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_version_h: build special version strings for
- AADK/S800i builds
-
-2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: The IMAP storage code uses the same code to
- build the email that is used when voicemail is sent via email
- using something like sendmail. In the patch from bug 8033 to fix
- various IMAP storage problems, the line endings in the email file
- were changed in the code from "\n" to "\r\n". However, this
- breaks sending regular voicemail to email. So, this change
- conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
- enabled. (issue #9128, patch by jarjarbinks, modified by me to
- not break IMAP storage)
-
-2007-02-22 23:25 +0000 [r56280] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 56279 via svnmerge ........ r56279 | file |
- 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always
- defer Agent logoff if any channels are up until they hang up.
- (issue #9123 reported by arbrandes) ........
-
-2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
- doc/sla.txt: Merge changes from team/russell/sla_updates. This
- batch of changes to the SLA code does a few different things. * I
- made the SLA code event driven instead of having to act in a lot
- of busy loops while dialing things to wait for state changes.
- This makes the code more efficient and readable at the same time.
- * I have implemented a couple of new features. The first is
- inbound trunk ringing timeouts. This is an option that defines
- how long to let an incoming call on a trunk to ring. * I have
- also implemented ring timeouts for stations. They may be
- specified for the entire station, meaning it is how long to let
- the station ring before giving up. You can also specify a ring
- timeout for a specific trunk on a station. So, you can say that
- you only want a specific station to ring 5 seconds if it is line1
- ringing, but otherwise, there is no timeout.
-
-2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
- lines Only change the original or clone channel if it's the
- channel behind the proxy channel, not if it's just a regular
- bridged channel. ........
-
-2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: Update OSP documentation for v1.4.
-
-2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move message from verbose to debug
-
-2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com>
-
- * sounds/Makefile: updated the sound tarball versions in Makefile
-
-2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Restructure a little bit of code to reduce
- nesting. There is no functionality change here.
-
- * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
- 3 lines If we receive a frame that is not in any of the
- negotiated formats, then drop it. (potentially issue #8781 and
- SPD-12) ........
-
-2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: Print out deprecation notice on usage output of CLI
- commands. (issue #8925 reported by blitzrage)
-
-2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c: disable unloading of embedded modules... there is
- a fundamental problem with doing so that will not be fixed in
- this version of Asterisk due to its invasiveness
-
-2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
- lines Change naughty warning message to provide useful
- information. If a write now fails on a channel in meetme it will
- tell you the channel name instead of spitting out the wrong error
- message. ........
-
-2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Fix locking issue, and accept
- "transport-accept" as a valid accept message. This should solve
- issues 8970 and 8503.
-
-2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Simplify the last change to app_meetme, and
- move the call to dispose_conf() up into the block where we know a
- conf exists.
-
-2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Only dispose of the conference if one was
- created.
-
- * apps/app_speech_utils.c: Only start playing the next file if we
- have not been quieted.
-
- * channels/chan_sip.c: Add a flag that indicates whether a SIP
- dialog is an outgoing call or not. SIP_OUTGOING originally did it
- but it was repurposed to the direction of the last transaction,
- which can cause update_call_counter to falsely decrease the wrong
- counters. (please don't hurt me oej) (issue #8943 reported by
- mdu113)
-
-2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/make_version: Merged revisions 55868 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
- Feb 2007) | 2 lines use new tag version script ........
-
-2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
- after transfer (decrement inuse early on transferer's call leg)
-
-2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
- Issue 7764, patch by sailer
-
-2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Improve the reference counting to fix bugs
- where people report seeing conferences listed that have no
- members. (issue #9073)
-
- * /: Blocked revisions 55750 via svnmerge ........ r55750 | russell
- | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix
- random crashes when using the MeetMe application. This patch
- converts list handling to use the linked list macros and most
- importantly, implements reference counting on the ast_conference
- objects. The reference counting was first backported from 1.4.
- However, that code has some problems that caused the reference
- count to never hit zero. Those problems are fixed in this patch
- and will be resolved in 1.4 and trunk next, with a different
- patch. (issues #7647, #9073, #9106, BE-115). ........
-
-2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Better handle dropped IMAP connections.
- (issue #9054 reported by bsmithurst)
-
- * channels/chan_sip.c: Return behavior I removed. I did not
- remember that you could just add a localnet entry to make it
- work.
-
- * channels/chan_sip.c: Don't test our own address against the
- localnet settings. At least one person has had issues as a result
- of this from #7051 so I'm reversing it. (issue #8821 reported by
- kokoskarokoska)
-
- * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
- 2007) | 2 lines Defer clearing callback information if channels
- are up until they are hung up. This ensures the hangup process
- goes smoothly and no channels get hung in limbo. (issue #8088
- reported by kebl0155) ........
-
-2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com>
-
- * main/http.c: Add the Asterisk version information to the Server
- header in HTTP responses. (requested by Pari)
-
- * include/asterisk/manager.h: Increase the maximum number of
- manager headers to 128, at the request of Pari.
-
- * /: Blocked revisions 55588 via svnmerge ........ r55588 | russell
- | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert
- a tab to spaces so that the documentation is printed out properly
- aligned. ........
-
-2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
- with strdupa (thanks file) 55555!
-
-2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: Change the formatting of sla.conf.sample
- to make it more readable. (issue #9112, blitzrage)
-
-2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: - Not sending arguments to an application is
- not "out of memory" - Making error messages a bit more clear
-
-2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
- | 2 lines forcename and forcegreetings options should check to
- see if the recording already exists ........
-
-2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_iax2.c: Changed iax2 process thread to detached to
- correct memory leak due to left over thread context on thread
- exit. Modified module unload process to avoid deadlocks on
- pthread cancels
-
-2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net>
-
- * /, apps/app_record.c: Merged revisions 55277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
- lines Documentation update (#9053, jsmith) ........
-
- * /: Block patch that was made only for 1.2 (already implemented in
- 1.4 and trunk)
-
-2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Add missing membername option to AddQueueMember
- documentation. (issue #9088 reported by seanbright)
-
-2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue where callerid would not be
- displayed on some phones. Issue 8995, initial patch and research
- done by wedhorn
-
-2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
- lines Answer the channel before recording privacy information.
- (issue #8926 reported by lmamane) ........
-
- * apps/app_queue.c: Make the 'i' option of Queue actually work.
- (issue #8986 reported by utis)
-
- * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
- lines Allow chan_sip to handle attended transfers from a SIP
- phone that is sitting behind chan_agent. Yes folks, all it took
- was one line of code. (issue #8784 reported by pzieba) ........
-
-2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: If the
- pg_config application is found, but there is probably executing
- it, then consider postgres unavailable. (issue #8637)
-
- * codecs/gsm/Makefile: Filter out yet another architecture that
- does not work with the optimizations in the built-in libgsm.
- (issue 8637, ovi)
-
- * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
- revisions 55005 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
- 9 lines Revert the change I did in revisions 54955, 54969, and
- 54970, in 1.2, 1.4, and trunk. I decided that once a conference
- is created from meetme.conf, it is acceptable behavior that the
- pin can not be changed until the conference goes away. I also
- added a note in meetme.conf to describe this behavior. We still
- have another issue in 1.4 and trunk where some conferences with
- no users don't go away. That is the real bug that needs to be
- addressed here. ........
-
-2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
- 2007) | 2 lines Do not send indications through ast_indicate in
- chan_agent but instead go directly to the technology. This way
- when indications are emulated they happen on the Agent channel
- and do not screw up formats on the channels. (issue #8439
- reported by punkgode) ........
-
-2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
- 5 lines For conferences that are configured in meetme.conf, check
- the configuration file every time someone joins the conference
- instead of only when the conference is first created. This is to
- ensure that changes to the pin numbers in the config file are
- always honored. (issue #9073) ........
-
-2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Need to check macro extension as well as macro
- context for directed pickup.
-
-2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
- default. It was set to enabled in pbx.c. However, if the option
- was not present in extensions.conf, then pbx_config.c would set
- it back to disabled.
-
- * res/res_features.c: Clean up a few coding guidelines issues -
- spaces to tabs, use sizeof() to pass the size of a static buffer,
- add spaces ...
-
-2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: Clarify a restart message. It's silly, but the
- reporter had a very valid point. Issue 9079
-
-2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Allow directed pickup to pick up the real
- context instead of the macro context if a Macro is used. (issue
- #8984 reported by jamesb63)
-
-2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #7541 - Handle multipart attachments
- to SIP messages - even if boundary is quoted.
-
- * /, res/res_agi.c: Merged revisions 54771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
- lines Issue #9069 - If we open with TH we should not close with
- /TD. (seanbright) ........
-
-2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Don't let dtmf leak over into the engine
- and let it skew the results... also give DTMF results priority.
- (issue #9014 reported by surftek)
-
- * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
- lines Use a separate variable to indicate execution should
- continue instead of the return value. (issue #8842 reported by
- pluto70) ........
-
- * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
- #9068 reported by mhardeman)
-
-2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net>
-
- * /: Block patch only needed in 1.2
-
-2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
- lines When handling glare on a PRI, move the requested channel
- rather than hang up the old one. Fix for 8957 and 9011. ........
-
-2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
- with it's placement, feel free to change it. (issue #9045
- reported by gork)
-
-2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Remove a couple of leftover debug messages
-
- * include/asterisk/devicestate.h: Fix the documentation on the
- return values from device state provider registration and
- deletion.
-
- * channels/chan_sip.c: If we fail to create the SIP socket, then
- return -1 from reload_config() so that load_module() will return
- AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
- spammed with error messages every time chan_sip tries to send a
- message.
-
-2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net>
-
- * /: Blocking patch for 1.2 only
-
-2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Change
- ast_set_state_callback() to ast_dial_set_state_callback()
-
- * main/dial.c, apps/app_meetme.c, apps/app_page.c,
- include/asterisk/dial.h: - Add the ability to register a callback
- to monitor state changes in an asynchronous dial operation. -
- Rename the various references to "status" to "state" in the dial
- API
-
-2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Make the --without-oss argument work.
- (issue #9026 reported by puzzled)
-
-2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com>
-
- * configs/users.conf.sample: Fix a typo where "vmpassword" should
- be "vmsecret"
-
-2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Fix VLDTMF reception
-
- * apps/app_echo.c: Much simpler than previous one ;-)
-
- * main/channel.c: Provide correct DTMF duration
-
- * main/cli.c: Bring deprecated 'debug channel <x|all>' command back
-
-2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac, acinclude.m4: don't display the
- --with-imap message unless --with-imap was specified without a
- path use '-n' instead of '! -z' for tests
-
-2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Add some output for "show application
- SLAStation/SLATrunk"
-
- * channels/chan_sip.c: Change some text to properly state "On
- Hold", which was already done in trunk.
-
- * configs/sla.conf.sample, include/asterisk/app.h,
- include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
- channels/chan_sip.c, doc/sla.txt (added),
- include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
- team/russell/sla_rewrite This is a completely new implementation
- of the SLA functionality introduced in Asterisk 1.4. It is now
- functional and ready for testing. However, I will be adding some
- additional features over the next week, as well. For information
- on how to set this up, see configs/sla.conf.sample and
- doc/sla.txt. In addition to the changes in app_meetme.c for the
- SLA implementation itself, this merge brings in various other
- changes: chan_sip: - Add the ability to indicate HOLD state in
- NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
- the channel is not bridged to another channel. linkedlists.h: -
- Add support for rwlock based linked lists. dial.c: - Add the
- ability to run ast_dial_start() without a reference channel to
- inherit information from.
-
- * apps/app_echo.c: When the Echo() application receives the digit
- '#', echo that back as well. Since we already sent the BEGIN
- frame for that digit, it makes sense to send the END as well.
-
-2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_gtalk.c: another dependency
-
- * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
- funcs/func_odbc.c, res/res_adsi.c: add some inter-module
- dependencies
-
- * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
- scripts to work when both MODULEINFO and MAKEOPTS are present in
- a source file
-
-2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Temporarily change musicclass on channel to one
- specified in Dial so that the 'm' option functions properly.
- (issue #8969 reported by christianbee)
-
-2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/imapstorage.txt, configure, configure.ac: clarify the fact
- that voicemail IMAP storage cannot be built against a distro's
- binary c-client library package (at least not at this time)
-
-2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net>
-
- * main/acl.c: Don't output debug unless we asked for it
-
-2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Fix timeout issue when utterance is
- longer then timeout itself.
-
-2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/loader.c: Issue 9007 - Mutex not released on early return
-
- * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
- | 2 lines Issue 9003 - If fullname is empty, quote() passes back
- "\"" ........
-
-2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com>
-
- * main/db1-ast/Makefile: When building libdb1.a, put the additional
- flags needed at the beginning of ASTCFLAGS, instead of at the
- end. This way, we ensure that we find the local headers first
- before accidentally trying to use headers that exist in locations
- specified in the ASTCFLAGS passed from the main Makefile. (issue
- #8637, ovi)
-
- * main/Makefile: The clean target actually needs to run "distclean"
- on editline. This is because we need to make sure that its
- configure script gets executed again, because the CFLAGS we want
- to pass to editline may have changed.
-
-2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: We can not reliably do P2P bridging with DTMF passing
- back with compensation if we need to listen for DTMF frames.
- (issue #8962 reported by caio1982)
-
-2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: When parsing the NTP timestamp in a sender report
- message, you are supposed to take the low 16 bits of the integer
- part, and the high 16 bits of the fractional part. However, the
- code here was erroneously taking the low 16 bits of the
- fractional part. It then shifted the result 16 bits down, so the
- result was always zero. This fix makes it grab the appropriate
- high 16 bits, instead. (issue #8991, pointed out by
- andre_abrantes)
-
-2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Directly load say.conf in load_module
- instead of calling the reload function. (issue #8946 reported by
- junky)
-
- * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
- lines Fix a few potential memory leaks with realtime users and
- peers. (issue #8999 reported by bsmithurst) ........
-
-2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
- | 2 lines Issue 7440 - Macro called from Macro from the h
- extension exits prematurely ........
-
-2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 52843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
- 1 line fixed some possible segfaults. also fixed an very
- important bug which occurs on high load (when calls are very fast
- generated) ........
-
-2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_jabber.c: Text fix for jabber reload command (reported by
- bkruse via IRC)
-
- * main/manager.c, /: Merged revisions 53245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
- | 2 lines Issue 8987 - Status could return two responses
- (mnicholson) ........
-
-2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting
-
-2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Ensure say_cfg is NULL when the module is
- loaded. (issue #8946 reported by junky)
-
- * apps/app_playback.c: Unregister Playback CLI commands as well as
- dialplan application. (issue #8946 reported by junky)
-
-2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add some comments on queue system behaviour
- and how it affects the SIP channel
-
-2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make SIPDtmfMode application work with
- recent capability changes, and also fix an RTP stack issue when
- the auto option was used. (issue #8972 reported by mdu113)
-
-2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
- 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when
- the dial application exits early because of invalid arguments
- instead of just leaving it empty. (issue #8975) ........
-
- * /: Blocked revisions 53134 via svnmerge ........ r53134 | russell
- | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert
- some changes that accidentally got committed as a part of another
- fix. ........
-
-2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
- because due to compatibilities with CS1000 reported at
- www.voip-info.org
-
-2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com>
-
- * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
- warning to the console that things might possibly be
- misconfigured when queue member's states are still 'Not in Use'
- when we're about to bridge them with a caller from queue. Also,
- put some documentation quoted from oej's queues.txt efforts
- started in /trunk today. This commit puts #7433 into feedback
- state for 1.4, and pending no further negative feedback, it will
- finally be closed.
-
-2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Correct a copy/pasted error message line for RTCP.
-
- * main/config.c, /: Merged revisions 53117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
- lines Pass the glob expanded filename to process_text_line so
- that error messages contain the actual filename, not the original
- include one. (issue #8959 reported by tzafrir) ........
-
- * Makefile: Add systemname to asterisk.conf generation per recent
- discussions about it. (issue #8968 reported by blitzrage)
-
-2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
- p2p RTP call setup in SIP. You can enable it in sip.conf, but it
- is now considered experimental until we solve the
- AST_CONTROL_ANSWER with payload and videocaps stuff.
-
-2007-02-01 23:16 +0000 [r53108] Jason Parker <jparker@digium.com>
-
- * /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell |
- 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a
- small typo. Synopsis lines shouldn't have a newline ........
-
-2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
- lines Copy noncodeccapability over to the joint variable so that
- telephone-event will get transmitted in the sent INVITE. ........
-
- * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
- here as well, but it apparently required both dev mode and no
- optimizations to creep up.
-
- * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
- lines Don't negotiate RFC2833 when not configured to do so.
- (issue #8799 reported by mdu113) ........
-
-2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com>
-
- * funcs/func_strings.c: Fix the FIELDQTY function to not crash.
- (reported by blitzrage and Corydon on IRC)
-
-2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net>
-
- * /: Going backwards, blame file.
-
-2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
- 2007) | 2 lines Return previous behavior of having MOH pick up
- where it was left off. (issue #8672 reported by sinistermidget)
- ........
-
- * funcs/func_strings.c: Make func_strings build under dev mode.
- Didn't I do this today already in the berkeley DB?
-
-2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
- call counter - If it's still set at time of dialog destruction,
- make sure we decrement the device call counter properly before we
- destroy the dialog
-
- * apps/app_queue.c: Change debug level for state change message
- that is not really informative when debugging app_queue
-
- * channels/chan_sip.c: Cleaning up the devicestate callback
- function
-
-2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Oops.
-
- * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
- | 2 lines Bug 8965 ........
-
-2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Add missing 'F' letter to getopt so it magically
- becomes a valid option. (issue #8960 reported by tzafrir)
-
-2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
- | 2 lines No wonder FIELDQTY doesn't work with functions... the
- documentation in pbx.c was wrong ........
-
-2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix silly logic. We really want to write
- UDPTL frames out when the call is up.
-
-2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Add explanation of port= in combination
- with defaultip= (thanks jsmith)
-
-2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: we update the name on any first reply of
- our setup
-
-2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: chan_h323 is very stable, so let it built
- by default
-
-2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: When going on hold have the side that was put on hold
- reinvite back to Asterisk. When going off hold have the side that
- was taken off hold reinvited back to the other party.
-
- * main/rtp.c: Add more frame types to forward in the RTP bridge
- loops.
-
-2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com>
-
- * main/cdr.c, main/manager.c, pbx/pbx_spool.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
- pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
- main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
- channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
- Merged revisions 53045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
- 3 lines Fix a bunch of places where pthread_attr_init() was
- called, but pthread_attr_destroy() was not. ........
-
- * apps/app_userevent.c: Remove an extra \r\n from manager user
- events. (issue #8955, mnicholson)
-
- * main/rtp.c, /: Merged revisions 53039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
- 3 lines Use the proper format string to print unsigned values in
- the rtp debug output. (issue #8954, wmis) ........
-
- * apps/app_queue.c: Only changed the paused status in an existing
- queue member if the paused column exists.
-
- * apps/app_queue.c: Instead of always creating a realtime queue
- member as unpaused, read the "paused" column and use that value
- for the paused status of the member. (issue #8949, jmls)
-
- * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
- (issue #8363, johnlange)
-
- * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
- #8942, lters)
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- codecs/codec_gsm.c: When we are checking for a system installed
- version of libgsm, we need to check for gsm.h as well.
- Furthermore, when checking for this header, it may be located in
- a gsm/ sub directory, so check for that, as well. (issue #8773)
-
- * /: Blocked revisions 52954 via svnmerge ........ r52954 | russell
- | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't
- print a message indicating that we don't know what to do with a
- proceeding control frame in ast_request_and_dial(). We just need
- to ignore it. (reported by JerJer on #asterisk-dev) ........
-
- * channels/chan_sip.c: Only set the DTMF flag on the rtp structure
- if the DTMF mode is actually RFC2833, not just that it is not
- INFO. This makes it get set for inband DTMF as well, which is not
- valid. (issue #8936)
-
- * main/asterisk.c, /: Merged revisions 52903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
- 9 lines The SIGHUP handler was implemented to allow admins to
- send SIGHUP to a running Asterisk process to reload the
- configuration. However, doing the actual reload in the signal
- handler itself is a very bad thing to do, because the reload
- process includes calling non-reentrant functions such as
- malloc/calloc/etc. If Asterisk is running in the background, then
- the reload will happen immediately. However, if running in
- console mode, the reload doesn't work until something is typed at
- the console. That sort of defeats the purpose, but I don't see an
- easy way to get around it at this point. ........
-
- * /: Blocked revisions 52857 via svnmerge ........ r52857 | russell
- | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment
- out the parts in the Makefile that make codec_zap get built. It
- will not yet build against zaptel 1.2, so I am disabling it to
- prevent further bug reports until it gets merged. (issue #8940)
- ........
-
-2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Drop the deprecated show commands since the
- original ones were changed back. (issue #8937 reported by
- PCadach)
-
-2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Revert reprecation of h.323 gk cycle
- command from pre-1.4 version instead of duplicated h323 cycle gk
-
- * res/res_odbc.c: Don't play with free()'d pointers
-
- * configure, acinclude.m4: Handle non-standard OpenH323/PWLib
- library names
-
-2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
- 5 lines Fix the extraction of the timestamp from video frames. It
- was using the mapping for a mini-frame instead of a video-frame,
- which caused it to get invalid data. (issue #8795, mihai)
- ........
-
-2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
- 2007) | 2 lines Now that filename is part of the structure and
- since it comes before postprocess... we have to add it to our
- postprocess line. (reported on asterisk-dev by Boris Bakchiev)
- ........
-
-2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Add a missing quotation mark. This was pointed out
- by jcmoore on #asterisk-dev.
-
- * main/manager.c: Remove a recursive lock of the manager session.
- This was pointed out by zandbelt in issue #8711.
-
-2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * pbx/pbx_config.c: Argument number correction
-
-2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com>
-
- * main/Makefile: ASTLDFLAGS needs to be passed to the editline
- configure script as LDFLAGS. (issue #8928, zandbelt)
-
- * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
- mode translation. P2P bridging can only be used when the DTMF
- modes don't match if the core is monitoring DTMF in both
- directions. Then, the core will handle the translation.
- Otherwise, this bridging method can not be used. (issue #8936)
-
- * main/manager.c: The session lock can not be held while calling
- action callbacks. If so, then when the WaitEvent callback gets
- called, then no event can happen because the session can't be
- locked by another thread. Also, the session needs to be locked in
- the HTTP callback when it reads out the output string. This fixes
- the deadlock reported in both 8711 and 8934. Regarding issue
- 8711, there still may be an issue. If there is a second action
- requested before the processing of the first action is finished,
- there could still be some corruption of the output string buffer
- used to build the result. (issue #8711, #8934)
-
-2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Use ast_calloc instead of malloc.
-
-2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com>
-
- * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
- backport to 1.4). It was committed to trunk via 7663. But it
- wasn't so much an enhancement as a fix for the bad language
- output for portuguese in Brazil, so, after a lot of prodding from
- patient Brazilians, here is the same fix for 1.4
-
-2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Set quota information to 0 when creating a
- vm_state. (issue #8924 reported by neutrino88)
-
-2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com>
-
- * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
- the last commit to the adaptive jitterbuffer code. - Specifically
- indicate to the compiler that the "dropem" variable only needs
- one but. - Change formatting to conform to coding guidelines.
-
-2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com>
-
- * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
- jitterbuf, whereas it would not complain about, and would allow
- itself to be overfilled (per the max_jitterbuf parameter). Now it
- rejects any data over and above that size, and complains about
- it.
-
-2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * configure, configure.ac: Suggested change to fix normal usage of
- --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
- list)
-
-2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
- lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
- follow documentation. (issue #7677 reported by amilcar) ........
-
- * main/manager.c: Have the manager interface send back an "Already
- logged in" message instead of "Invalid/Unknown Command" when the
- client authenticates for a second time. (issue #8509 reported by
- pari)
-
- * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
- lines Make the last context entry read in the dominant one.
- (issue #8918 reported by pj) ........
-
- * main/file.c: Fix core show file formats CLI command.
-
-2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com>
-
- * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
- lines Allow dequeueing of frames with negative timestamp by
- moving jitterbuffer frames check to jb_next. (issue #8546
- reported by harmen) ........
-
- * channels/chan_sip.c: Drop out variables I accidentally put in.
-
- * channels/chan_sip.c: Decrement onHold count if we are hung up on
- and still on hold. (issue #8909 reported by alexh42)
-
- * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
- 2007) | 2 lines Add another note about audio files being played
- back to each bridged party. (issue #8718 reported by ppyy)
- ........
-
-2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, configs/users.conf.sample: By suggestion
- from kpfleming last week, change "vmpassword" to "vmsecret".
-
- * configure, configure.ac: Remove libnsl as a required lib for
- libiksemel to work. This change was already made in the trunk.
- (issue #8762)
-
- * /: Blocked revisions 52137 via svnmerge ........ r52137 | russell
- | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a
- seg fault when running this application with no arguments from
- AGI. (issue #8905, junky) ........
-
- * include/asterisk/dial.h: Fix the formatting of doxygen comments
- to properly indicate that the comment documents the previous
- entity, as opposed to the next one.
-
-2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com>
-
- * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
- line updated check_expr via 8322 (refactoring of expression
- checking impl); elfring contributed a nice code reorg, I
- contributed some time to get it working again, better messages
- ........
-
-2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c (added), apps/app_page.c, main/Makefile,
- include/asterisk/dial.h (added): Merge in dialing API and the
- app_page that uses it. (issue #BE-118)
-
- * channels/chan_sip.c: Fix changing channel formats when joint
- capability changes and there are no audio formats... I didn't
- break it originally! (issue #8535 reported by ivoc)
-
-2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com>
-
- * configure: rebuild configure script to reflect last chan_h323
- related changes.
-
-2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: added fix from #8899
-
- * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
- Jan 2007) | 1 line fixed the busy problem (dialstatus was not
- busy when we called a busy extension) ........
-
-2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Show capabilities *and* preference in
- general settings in "sip show settings" (reported by Clona/Telio
- - Thanks!)
-
-2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz>
-
- * acinclude.m4: Allow x64 builds of H.323 (please, rebuild
- configure)
-
-2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 51843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
- 6 lines Fix an issue related to synchronization of recordings
- when using Monitor(). The bug is a miscalculation of the amount
- to seek the stream for writing to disk when the number of samples
- coming in and out of a channel do not match up. (issue #8298,
- #8887, report and patch by guillecabeza, patch files created and
- testing done by whoiswes) ........
-
- * apps/app_while.c, /: Merged revisions 51828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
- 4 lines Don't set a new value for the END_ variable on the
- channel before using the old value. If you do, it will lead to
- accessing a memory address that has been free()'d. (issue #8895,
- arkadia) ........
-
-2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
- channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_alsa.c,
- channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
- drivers to use module referencing so that unloading them while in
- use will not result in crashes. (issue #8897 reported by junky)
-
-2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix some bugs in process_message(). The manager
- session lock needs to be held when sending some sort of response,
- or calling one of the manager action callbacks. This resolves an
- issue where people using the GUI would get random crashes when
- they start clicking around a lot. (issue #8711, reported and
- debugged by zandbelt)
-
- * main/http.c: Fix setting the default port of 8088 on 64-bit or
- big-endian machines.
-
- * main/manager.c: When traversing the list of manager actions, the
- iterator needs to be initialized to the list head *after* locking
- the list. Also, lock the actions list in one place it is being
- accessed where it was not being done.
-
-2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: this mod from 8593 (dstchannel in cdr is
- empty when transfer call).
-
- * main/callerid.c: via 8748 (callerid.c loses name when returning
- PRIVATE_NUMBER flag), the user suggested this mod, saying it
- would allow 'WITHHELD' to appear in the name field, which would
- be useful
-
-2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
- 6 lines * more additions to make the RESTART message work * added
- fix for misdn_call to allow SETUPs with empty extensions,
- replaced the strtok_r functions with strsep for that (inspired by
- Sandro Cappellazzo, thanks) ........ r50506 | crichter |
- 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
- L2 UP, the L1 is UP definitely too, so we set the L1 state up as
- well. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c: manually merged r49922 and r50335, because
- of conflicts. this commint includes addition of the ISDN RESTART
- Message
-
-2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
- startup if h323 configuration file not found (reported by
- mithraen)
-
-2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only change audio formats on the channel if
- we have an audio format to change to. (issue #8535 reported by
- ivoc)
-
- * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
- 2007) | 2 lines Yield before reading from zaptel timing source
- under Solaris so that other threads get a chance to do things.
- (issue #7875 reported by bob) ........
-
-2007-01-22 19:41 +0000 [r51411] Russell Bryant <russell@digium.com>
-
- * /: Blocked revisions 51410 via svnmerge ........ r51410 | russell
- | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge
- codec_zap support for the transcoder card. This is a standalone
- codec module so it will not affect anything else. ........
-
-2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This fixes 8836, according to dnatural
-
-2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
- 2007) | 2 lines Move filestream creation to Mixmonitor loop. This
- will prevent a blank file from being created if no frames ever
- pass through to be recorded. (issue #7589 reported by
- steve_mcneil) ........
-
- * /: Blocked revisions 51359 via svnmerge ........ r51359 | file |
- 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly
- declare what codecs are supported by default globally since using
- a bitmask for all may include ones we don't need. (issue #8357
- reported by gknispel_proformatique) ........
-
-2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com>
-
- * configs/say.conf.sample: Fix Italian numeral support in say.conf
- for "_[2-9]00" case. "2131" would've translated to something
- along the lines of (pardon my..Italian {or lack thereof})
- "duecentocentotrentuno", which makes no sense at all.
-
- * configs/say.conf.sample: Fix German language support in say.conf
- Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
- einundzwanzig has the same format as zweiundzwanzig (as do all
- other "_ZX" spoken numerals) Fix support for numbers in the
- 10,000,000 to 99,999,999 range. Add support for numbers in the
- 100,000,000 to 999,999,999 range.
-
-2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Remove an unused instance of an unnamed enum.
-
- * apps/app_meetme.c: Remove another duplicated definition
-
- * apps/app_meetme.c: Remove a variable that was declared twice.
-
- * codecs/gsm/Makefile: Add a couple more processors that need
- optimizations excluded. (issue #8637)
-
- * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
- AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
- thing. So, a digit would have been interpreted incorrectly here.
- Since the channel driver will always have the begin and end
- callbacks called for a digit, only support the button-down and
- button-up messages.
-
- * .cleancount: Bump the cleancount since my last commit changed the
- channel structure.
-
- * channels/chan_oss.c, main/rtp.c, main/channel.c,
- channels/chan_phone.c, channels/chan_misdn.c,
- channels/chan_skinny.c, channels/chan_features.c,
- channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
- channels/chan_zap.c, channels/chan_local.c, main/frame.c,
- channels/chan_sip.c, channels/chan_agent.c,
- include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_iax2.c: Merge the changes from the
- /team/group/vldtmf_fixup branch. The main bug being addressed
- here is a problem introduced when two SIP channels using SIP INFO
- dtmf have their media directly bridged. So, when a DTMF END frame
- comes into Asterisk from an incoming INFO message, Asterisk would
- try to emulate a digit of some length by first sending a DTMF
- BEGIN frame and sending a DTMF END later timed off of incoming
- audio. However, since there was no audio coming in, the DTMF_END
- was never generated. This caused DTMF based features to no longer
- work. To fix this, the core now knows when a channel doesn't care
- about DTMF BEGIN frames (such as a SIP channel sending INFO
- dtmf). If this is the case, then Asterisk will not emulate a
- digit of some length, and will instead just pass through the
- single DTMF END event. Channel drivers also now get passed the
- length of the digit to their digit_end callback. This improves
- SIP INFO support even further by enabling us to put the real
- digit duration in the INFO message instead of a hard coded 250ms.
- Also, for an incoming INFO message, the duration is read from the
- frame and passed into the core instead of just getting ignored.
- (issue #8597, maybe others...)
-
- * main/asterisk.c: Merged revisions 51300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
- 4 lines Fix a memory leak on command line tab completion. The
- container for the matches was freed, but the individual matches
- themselves were not. (issue #8851, arkadia) ........
-
-2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c: chan_zap compiles without libpri after
- committing 7877 patch
-
- * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
- | 3 lines issue 7877: chan_zap module reload does not use
- default/initialized values on subsequent loads. Reset
- configuration variables to default values prior to parsing
- configuration file. ........
-
-2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block this patch since it is already here
-
-2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, main/channel.c, main/pbx.c,
- funcs/func_strings.c, main/app.c: Add some more checks for
- option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
- patch(es) by tgrman
-
-2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com>
-
- * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
- Ensure that the locations given to the Asterisk configure script
- for ncurses, curses, termcap, or tinfo are further passed along
- to the editline configure script. This fixes some
- cross-compilation environments. (issue #8637, reported by ovi,
- patch by me)
-
-2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
- Jan 2007) | 2 lines If a timezone is not specified, assume
- localtime (instead of gmtime) (Issue #7748) ........
-
-2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Only start timeout once we reach the end
- of the files to play back.
-
-2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Fix an issue with file name completion in "module
- load" and "load". Issue 8846
-
-2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Copy MOH settings when calling a peer so
- that if they put someone on hold or get put on hold themselves
- they get the right music class. (issue #8840 reported by mdu113)
-
-2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Fix an issue with deprecated commands
-
-2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
- Jan 2007) | 2 lines Document all the fields, including the
- indication that "uniqueid" should not be renamed. ........
-
-2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Make the "hasmanager" option in users.conf
- actually have an effect. (issue #8740, LnxPrgr3)
-
-2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Build the IMAP remote directory string
- better and properly. Fix an issue with encoding the GSM voicemail
- when attaching to the voicemail. (issue #8808 reported by
- akohlsmith)
-
- * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
- (issue #8840 reported by mdu113)
-
-2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com>
-
- * funcs/func_odbc.c: Fix some instances where when loading
- func_odbc, a double-free could occur. Also, remove an unneeded
- error message. If the failure condition is actually a memory
- allocation failure, a log message will already be generated
- automatically.
-
- * channels/chan_zap.c: Instead of dividing the offset by 2
- directly, make it more clear that the offset is being scaled by
- the size of the elements in the buffer. (Inspired by a discussing
- on the asterisk-dev list about this code)
-
- * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
- 3 lines Move the check for a failure of ast_channel_alloc() to
- before locking the pvt structure again. Otherwise, on a failure,
- this will cause a deadlock. ........
-
-2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/utils.c: Merged revisions 51194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
- | 4 lines When ast_strip_quoted was called with a zero-length
- string, it would treat a NULL as if it were the quoting character
- (and would thus return the string in memory immediately following
- the passed-in string). ........
-
-2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: re-add "password" for realtime voicemail
-
-2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Return the correct result when directly writing out a
- packet so that the core doesn't then decide to handle it the
- regular way again. (issue #8833 reported by rcourtna)
-
-2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_voicemail.c: a few more coding style cleanups and one
- bug fix (from AnthonyL)
-
-2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
- scheduler callback.
-
-2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Fix issue with dtmf continuation packets when the
- dtmf digit is 0... Issue 8831
-
- * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
- IMAP storage and realtime voicemail. Also update the vmdb sql
- script for IMAP specific options. Issue 8819, initial patches by
- bsmithurst (slightly modified by me)
-
- * doc/voicemail_odbc_postgresql.txt: change documentation to
- reflect new procedure in 1.4/trunk
-
-2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
- 51161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
- | 2 lines Add documentation walkthrough on getting Postgres to
- work with voicemail (from Issue 8513) ........
-
- * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
- | 2 lines Postgres driver doesn't like a NULL pointer when
- retrieving the length (Bug 8513) ........
-
-2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c: minor things i missed before i get jumped
- on
-
-2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 51145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
- lines Return previous behavior. ParkedCalls will be able to do
- DTMF based transfers again. trunk however will get an option to
- allow this to be set on/off. (issue #8804 reported by nortex)
- ........
-
-2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com>
-
- * main/file.c: Display more useful output when streaming files.
- Include the channel name to which the file is being played. Issue
- 8828, patch by junky.
-
-2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
- lines Add none as a valid callgroup/pickupgroup option. I
- consider it a bug that it would inherit it all the way down and
- not have any way to reset it to nothing - so that's why it is in
- 1.2. (issue #8296 reported by gkloepfer) ........
-
-2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com>
-
- * main/config.c: It is possible for the config pointer to be NULL
- here, so it needs to be checked before dereferencing it.
-
-2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
- changing voicemail password in users.conf from voicemail main,
- written by AnthonyL bug #8436
-
-2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com>
-
- * Makefile.rules: Filter out a few CFLAGS that are not valid
- CXXFLAGS.
-
-2007-01-15 23:10 +0000 [r50988] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /: Blocked revisions 50987 via svnmerge ........ r50987 |
- tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines
- Check return value before dereferencing (Bug 8822) ........
-
-2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
- | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
- lines Solves issue with forwarding voicemails from folders other
- than inbox. patch by anthonyl. ........
-
-2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: re-add deprecated "show version" CLI command.
-
-2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Move event processing into do_message so that it
- gets executed again when events are tripped.
-
-2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, main/Makefile,
- configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
- ACX_PTHREAD macro from the Autoconf macro archive for setting up
- compiler pthreads support... should improve portability to
- platforms with unusual pthreads requirements
-
-2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com>
-
- * main/astmm.c: Add missing newlines for two memory CLI commands.
-
-2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
- main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
- main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
- main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
- main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
- main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
- main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
- main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
- main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
- main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
- main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
- main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
- main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
- main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
- main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
- main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
- Jan 2007) | 2 lines Bug 8814 - db should look for its header
- using a relative path, instead of the system path (Fixes FreeWRT)
- ........
-
-2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, build_tools/make_sample_voicemail (added): when
- building the sample greetings for maibox 1234@default during
- 'make samples', build a greeting for each language and file
- format the user selected to install with menuselect (reported by
- Brian Capouch on asterisk-dev)
-
-2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Only write a frame out to the channel if one
- exists. There are cases where one may not and would therefore
- cause the channel driver to segfault. (issue #8434 reported by
- slimey)
-
- * res/res_snmp.c: Only join the snmp thread on an unload if the
- thread is actually running. (issue #8810 reported by junky)
-
-2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample: Update documentation to state that
- you shouldn't use realtime static with voicemail.conf
-
-2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: We need to check for res being 0 in do_message
- itself, otherwise our headers will get lost.
-
-2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/pbx.c, /: Merged revisions 50561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
- | 2 lines minor documentation clarification ........
-
-2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Remove check for channel state as it can
- definitely be something other then ring, and also clean up the
- code a bit. This should solve the parking issues and maybe some
- attended transfer issues people have been seeing.
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
- support to see whether NAT was detected (yay symmetric RTP) and
- also add a check in chan_sip so that if NAT has been detected and
- the reinvite behind nat option has been turned off, then just do
- partial bridge. (issue #8655 reported by mnicholson)
-
- * apps/app_speech_utils.c: Merge speech-multi branch which adds
- support for joining multiple sound files together to be played
- one after another in SpeechBackground.
-
- * main/config.c: Fix parsing when using something like ldap
- settings. (done by anthonyl)
-
- * channels/chan_sip.c: Fix chan_sip not working issue. Let's not
- prematurely return 0. (issue #8783 reported by st41ker)
-
-2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
- it fail to load if the config file existed. Issue 8777
-
-2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
- lines Add another return value to dial_exec_full that indicates
- execution is going to continuing at a new
- extension/context/priority and to just let it slide. (issue #8598
- reported by jon) ........
-
- * main/pbx.c: Ensure data's existence before trying to access it.
- (issue #8774 reported by rcourtna)
-
-2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 50227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
- 6 lines Make the number that represents the major version number
- a single digit instead of 2. Using two digits makes it an octal
- number when put into version.h, which breaks the compilation of
- any out of tree module that checks the version for any version
- after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev
- mailing list, who gave credit to vihai for pointing it out)
- ........
-
-2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Re-add CLI command that should have only been
- deprecated in 1.4. Thanks kshumard! (reported in person, so no
- associated issue #)
-
-2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
- | 4 lines The advent of realtime has enabled people to use commas
- in the fullname field. This could cause an issue with sending
- voicemails, when the field is unquoted. (Issue 8595) ........
-
-2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - handle re-invites properly in sip_hangup()
- - Add some invitestate status changes just to be sure
-
-2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
- where it wouldn't ever parse a password, since it was using
- "secret" instead of "password" Issue 8761, reported by and patch
- suggestion from ssokol.
-
-2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_senddtmf.c: we can't unlock a channel if we cant find
- it. - AnthonyL bug #8741
-
-2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Disable the more intense packet2packet bridging until
- the bugs can be worked out.
-
-2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8677 - Handle failure of T.38
- re-invite This is not a fix, but adding an error message to tell
- the admin that we have a bad configuration. We should not send
- T.38 re-invites to devices that can't handle it (with the current
- architecture where you have to hard-code t.38 support per
- device). To really fix this, we need to figure out a way to tell
- the incoming call that the re-invite failed, so we can signal
- failure on that end and go back to the original call.
-
- * channels/chan_sip.c: Issue #8524, support multiple via header
- values (tardieu) Thanks!
-
- * channels/chan_sip.c: We only need one forward declaration
-
- * channels/chan_sip.c: Issue 8735: Terminate state when extension
- is unavailable for subscription
-
-2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
- lines Ensure we use the default refresh value of 60 if the remote
- server does not send one. (issue #8746 reported by maethor)
- ........
-
-2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: since we use AC_PATH_TOOL to find tools,
- we should use the results it provides for us (reported by Brian
- Capouch on the asterisk-dev list)
-
-2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
- | 2 lines If openstream fails, then we crash (Issue 8564)
- ........
-
- * channels/chan_sip.c: Second condition was a subset of the first,
- so hold was never decremented, thus hint stayed stuck (Issue
- 8747)
-
-2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
- byte of allocated memory! This looks like it may have been a
- chicken/egg scenario.. You had to call a cleanup func, because
- everything was allocated. Then since you had to call a cleanup
- func, you were forced to allocate - ie; strdup("").
-
-2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, acinclude.m4: one more time...
-
- * configure, acinclude.m4: proper fix for r49712
-
- * configure, acinclude.m4: if --with-foo=<path> is specific for a
- configure option, ensure that it is used for header file checking
- as well
-
- * main/manager.c: ast_func_read() needs a writable copy of the
- function name to be passed
-
-2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
- chan_zap also depend on zaptel. This fixes an issue (8727) with
- zaptel being in a different directory, using --with-zaptel.
-
-2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: don't 'consume' the params list before we try to
- use it again
-
- * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
- main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
- main/db.c, channels/chan_zap.c, channels/chan_sip.c,
- apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
- utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
- apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
- AMI and AMI/HTTP requests by nearly 20K in most cases
-
-2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Don't keep repeating the warning over and over
- when the end of the call is reached. (issue #8724 reported by
- xrg)
-
-2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
- | 2 lines ensure that threads which are supposed to be detached
- (because we aren't going to wait on them) are created properly
- ........
-
- * channels/chan_iax2.c: revert the dynamic_list insertion change...
- that was not the right thing to do
-
- * channels/chan_iax2.c: create the IAX2 processing threads as
- background threads so they will use smaller stacks when we create
- a dynamic thread, put it on the dynamic_list right away so we
- don't lose track of it
-
-2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: It's possible for the iax2 pvt to
- disappear, so if it has... don't bother looking for dpentries.
-
-2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/threadstorage.h, main/asterisk.c,
- build_tools/cflags.xml, include/asterisk.h, main/Makefile,
- main/threadstorage.c (added), main/utils.c: add support for
- tracking thread-local-storage objects that exist via
- 'threadstorage' CLI commands
-
-2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com>
-
- * main/config.c: Only free comments and line buffer once we reach
- the first level. (issue #8678 reported by ssokol, fixed by
- anthonyl)
-
-2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2-parser.c, main/frame.c: don't mark these
- allocations as 'cache' allocations when caching has been disabled
-
- * channels/iax2-parser.c: if we're going to decrement the frame
- count when we free a frame, we should inrement it when we create
- one :-)
-
- * channels/iax2-parser.c, channels/iax2-parser.h,
- channels/chan_iax2.c: only do IAX2 frame caching for voice and
- video frames
-
- * main/frame.c: don't do frame header caching in the core if
- LOW_MEMORY is defined
-
- * channels/iax2-parser.c: don't define this type either if
- LOW_MEMORY is enabled
-
-2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
- | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
- lines converted a lot of 256 to PATH_MAX and some white space
- fixes. ........
-
-2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode
-
- * codecs/Makefile: make building of codec_gsm against the system
- GSM library actually work
-
-2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
- | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
- lines good catch russell sorry i missed that. fix magic number
- with proper sizeof ........
-
-2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com>
-
- * funcs/func_realtime.c: Fix the REALTIME() dialplan function.
- ast_build_string() advances the string pointer to the position to
- begin the next write into the buffer. So, this pointer can not be
- used to copy the contents of the string later. The beginning of
- the buffer must be saved. Interestingly enough, this code could
- not have ever worked. (Pointed out by Sebb on IRC, thanks!)
-
-2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
- | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
- lines When using ODBC_STORAGE VoicemailMain doesn't create the
- subdirectories for a mailbox such as the INBOX directory. this
- patch solves that problem, was written by anthony be-125 ........
-
-2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
- configs/misdn.conf.sample: Merged revisions
- 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
- 1 line changed a few debugs to higher debug levels ........
- r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
- 1 line added the export and import of the MISDN_ADDRESS_COMPLETE
- Variable to inidcate wether the extension is already completely
- dialed or if there might come additional digits by information
- elements. also added some docs for that. ........ r48467 |
- crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
- removed FIXUP state. added check for channel allocation conflict
- when we create a setup while the other site creates a setup on
- the same channel, besides the check we resolve this conflict.
- ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
- Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
- preselected channel we just accept it, even when we're NT. added
- some checks for segfaults. ........ r48576 | crichter |
- 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
- reject a channel, because it's in use already, we shouldn't
- process the setup anymore. made the channel allocation a bit
- easier and more understandable, removed a few unused lines
- ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
- Jan 2007) | 1 line added check for channel ranges in the
- set/empty channel functions. set pmp_l1_check default to no.
- added misdn restart pid cli command. added cleaning of channel
- when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
- 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
- check for bridging in misdn_call to avoid setting
- echocancellation when 2 mISDN channels are involved and when
- bridging is set. That lead to a kernel panic before under
- different situations, because we switched about 2 times between
- hardware bridging and echocancelation * readded MISDN_URATE
- variable which got lost before, this should make app_v110 work
- again * fixed typo ........
-
-2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, Makefile.rules: various Makefile improvements to get
- chan_vpb (and any other C++ modules) to build properly
-
-2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Check pvt structure presence before passing
- to send_command. This gets rid of the irritating message about a
- packet without pvt structure. This happens because the scheduled
- item is getting cancelled at almost the exact moment it is
- getting executed.
-
-2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
- pbx/ael/ael.flex: This is a slight modification to Josh's edits
- for #8579; both files edited were the produced by flex; so the
- source files need to be changed instead, and the generated files
- regenerated.
-
-2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
- enabled at that point in the code
-
-2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Allow fractions of a second in the Wait()
- application, like it says it allows.
-
-2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: remove comment that is unrelated to this
- function
-
-2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net>
-
- * configs/features.conf.sample: Adding note on effect of
- applicationmap features on re-invites
-
-2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
- configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
- and transcoding support in the system's Zaptel installation, and
- make only the modules that need those features dependent on them
- (this will allow building the other Zaptel-using parts of
- Asterisk against older versions of Zaptel or those on other
- platforms that haven't caught up yet to the Linux version)
-
- * Makefile: use a simpler (and portable) method to ensure that
- menuselect is built as a host binary
-
- * Makefile: revert this change until a better solution can be
- found... 'env -i' was not being used properly, but even when
- changed to do so, this process fails during cross-compilation
- because the menuselect build still sees 'CC' as set to the
- cross-compiler
-
-2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: remove incomplete implementation of dnsmgr.
- Let's fix this in trunk.
-
-2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_config.c: IAX has been deprecated for quite some time so
- we had better use IAX2 when creating the dial string for users.
- (issue #8697 reported by ssokol)
-
- * channels/chan_zap.c: Use asprintf to build the channel names
- instead of custom function. I believe the custom function is
- doing some things that are not portable across all
- implementations. (issue #8570 reported by hterag & issue #8692
- reported by nicolasg)
-
- * main/rtp.c: If the Packet2Packet bridge is being broken because
- of a masquerade then attempt to read a frame in so the masquerade
- actually happens. Otherwise weirdness will occur. (issue #8696
- reported by kjotte)
-
- * channels/chan_iax2.c: Initialize the packet queue in load_module
- instead of just declaring the list with the default value. (issue
- #8695 reported by ssokol)
-
-2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
- comma args converted to vertical bars. I hope this change does
- little harm.
-
-2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: put this value into the correct property
-
- * /, BUGS: Merged revisions 49045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
- | 2 lines location of the bug posting guidelines has changed
- ........
-
- * sample.call: simple commit to test CIA integration
-
-2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me
-
- * main/http.c: saw this in passing... fix a small typo
-
-2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: new versions of sounds
-
-2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com>
-
- * main/http.c: make the uris_lock a rwlock instead of a mutex lock
- - needs to be forward ported to trunk
-
-2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/lock.h: Backport support for read/write locks.
-
-2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
- pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
- pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
- pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
- from the ael stuff. Also, threw in a minor fix to frame.c to
- avoid build-killing compiler warnings.
-
-2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com>
-
- * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
- available when LOW_MEMORY is used and things are being built in
- the utils directory, so we need to resort to the old method of
- strncpy. (issue #8579 reported by mottano)
-
-2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
- main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
- main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
- main/http.c, main/logger.c: since these variables all have static
- duration, none of them need initializers (they default to zero
- anyway)
-
- * include/asterisk/options.h, main/asterisk.c, main/file.c: move
- extern declaration for this option to a header file where it
- belongs provide an initial value for 'languageprefix' option,
- instead of relying on randomness to provide a useful value
-
-2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Only include acl.h and lock.h once
-
- * channels/chan_sip.c: Only set rfc2833compensate flag once
- (handle_common_options)
-
- * channels/chan_sip.c: - Remove checking for T38 options twice.
- Keeping them in handle_common_options
-
-2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: make the option actually match the
- documentation
-
- * channels/iax2-parser.c, include/asterisk/utils.h,
- include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
- memory' and 'show memory summary' to distinguish memory
- allocations that were done for caching purposes, so they don't
- look like memory leaks
-
-2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
- politically correct
-
- * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
- cisco MWI support. Normally we try not to change our software for
- bugs in other devices. But in this case, the Cisco phones are so
- widespread so we try to implement a fix while waiting for a
- bugfix from Cisco.
-
- * channels/chan_sip.c: - Make sure handle_common_options return 1
- when we found a common option - Move uncommon (only global)
- option away from handle_common_options Reported by rizzo. Thanks!
-
- * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
- re-sending invite with auth.
-
- * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
- (rizzo, #8600)
-
-2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Get rid of a needless memory allocation and
- only create a conference structure in find_conf_realtime if data
- was read from realtime. (issue #8669 reported by robl)
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
- API call that initializes an RTP structure. We need this because
- chan_sip is cheeky and uses a temporary RTP structure for codec
- purposes, and the API calls that are used rely on the lock.
- (Pointed out on asterisk-dev by Andy Wang)
-
- * configure, configure.ac: Clean up autoconf file (gets rid of
- warnings seen when rebuilding configure) and rebuild configure.
-
-2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com>
-
- * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
- 6 lines Fix an error introduced by copying and pasting the
- handling of the >= operator for the MATH function. If a single
- equal sign was used as an operator, the function would treat it
- is as if it were the >= operator. Now, it properly handles it as
- an invalid operator. (issue #8665, patch by tempest1) ........
-
- * channels/chan_oss.c: Fix a typo in an error message that
- indicated that the MGCP channel type could not be registered,
- instead of the correct type, OSS.
-
- * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
- 3 lines Check for the proper return value on an error in a call
- to mmap(). This was reported by Andy Wang on the asterisk-dev
- list. Thanks! ........
-
- * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
- 3 lines Remove a couple of misplaced dots in log messages. This
- was reported by Andrea Spadaccini on the asterisk-dev mailing
- list. ........
-
- * main/http.c: Implement locking for the list of URI handlers to
- make it thread-safe.
-
-2006-12-23 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0 released.
-
-2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com>
-
- * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.
-
- * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia
-
-2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com>
-
- * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
- lines Add new silence sound files to the spec for Redhat. (issue
- #8652 reported by alvaro_palma_aste) ........
-
-2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
- builds.
-
- * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
- it is then passed to the IMAP store file function. (issue #8614
- reported by punknow)
-
- * doc/snmp.txt: find is not the same as bind when it comes to
- documentation. (issue #8626 reported by johann8384)
-
-2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: suppress compiler warnings in this module
- until it can be improved
-
-2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
- lines Free localuser structure when we fail to dial (issue #8612
- reported by rizzo) ........
-
-2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_sms.c: fix a bogus datalen in the frames generated by
- app_sms (causing noisy output if you listen to the output!) This
- affects trunk as well, whereas 1.2 is ok.
-
-2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com>
-
- * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
- type for these unixODBC API calls, eliminating warnings on 64-bit
- platforms that use the 'new' 64-bit types for ODBC API calls
-
-2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com>
-
- * Makefile: Use env -i to start a fresh environment when going to
- build menuselect. This is more portable then using unset. (issue
- #8543 reported by jtodd)
-
-2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/channel.h: unbreak the macro used for
- incrementing the frame counters. I don't know when the bug was
- introduced, but with the typical usage c->fin =
- FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
- trunk as well (fix coming).
-
-2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Put thread into proper list if we abort
- handling due to an error, and also hold the lock while putting it
- back into the proper idle list so we don't prematurely get a
- signal. (issue #8604 reported by arkadia)
-
-2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
- utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
- utils/ael_main.c: remove some now-unnecessary explicit includes
- of autoconfig.h clean up per-file dependencies during 'make
- clean'
-
- * build_tools/prep_tarball: need an additional argument here to
- make the downloads actually occur
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
- these calls from thinking they have multiple arguments
-
- * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
- funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
- main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
- main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
- simplify dependency tracking system, using the compiler's
- built-in method for generating them, and only doing dependency
- tracking if developer mode is enabled via the configure script
-
- * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
- really, really have to have autoconfig.h included before all
- other headers (especially system headers), the Makefile will now
- force it to happen (this will fix build problems with files like
- ast_expr2f.c, where we can't control the inclusion order in the
- file itself)
-
- * funcs/func_curl.c: instead of initializing the curl library every
- time the CURL() function is invoked, do it only once per thread
- (this allows multiple calls to CURL() in the dialplan for a
- channel to run much more quickly, and also to re-use connections
- to the server) (thanks to JerJer for frequently complaining about
- this performance problem)
-
-2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Turn payload_lock into bridge_lock and make it
- encompass all RTP structure contents that may relate to bridge
- information, including who we are bridged to.
-
- * channels/chan_iax2.c: Hold call structure lock in places where a
- qualify or peer action can destroy it.
-
- * channels/chan_iax2.c: Lock network retransmission queue in all
- places that it is used.
-
-2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
- from 1.2)
-
- * channels/chan_sip.c: Update to latest IANA spec
-
-2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Use a wakeup variable so that we don't wait
- on IO indefinitely if packets need to be retransmitted.
-
- * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
- structure can change AFTER a bridge has started. This comes from
- the packet handling of the SIP response when indication that it
- was answered has been sent. Therefore we need to protect this
- data with a lock when we read/write. (issue #8232 reported by
- tgrman)
-
- * main/rtp.c: Remove direct RTCP bridging. I've come to the
- conclusion that we should handle this through the core and not
- just forward it on. Should solve a few bugs.
-
-2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0-beta4 released.
-
-2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
- is the way it should have been done.
-
-2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
-
- * sounds/Makefile: new sounds package with 100% more silence
-
- * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.2 ........
- r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
- | 4 lines app_externalivr needs a real silence file, and
- additional changes to add silence files into core instead of
- extra patch provided by bug 8177 with minor additions. ........
-
-2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Return non-existant callerid handling to
- that which it was before. In 1.4 and trunk callerid can be
- allocated but not have any contents so we have to use
- ast_strlen_zero before passing it to the relevant functions.
- (issue #8567 reported by pabelanger)
-
-2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: STRFTIME() does not actually require an
- argument (issue 8540)
-
-2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Merge in my latest RTP changes. Break out RTP and
- RTCP callback functions so they no longer share a common one.
-
- * apps/app_meetme.c: Use the correct API call to say a device state
- changed. (Yes, I'm a nub.)
-
- * apps/app_meetme.c: Don't access the conference structure after it
- has been freed.
-
-2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
- res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
- apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
- | 5 lines When doing a fork() and exec(), two problems existed
- (Issue 8086): 1) Ignored signals stayed ignored after the exec().
- 2) Signals could possibly fire between the fork() and exec(),
- causing Asterisk signal handlers within the child to execute,
- which caused nasty race conditions. ........
-
-2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
- line This version applies the patch suggested by stevens in bug
- 7836 (make inbound channel RINGING state consistent with other
- channels). ........
-
-2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use locking when accessing the
- registrations list. This list is not actually used very often, so
- the likelihood of there being a problem is pretty small, but
- still possible. For example, if the CLI command to list the
- registrations was called at the same time that a reload was
- occurring and the registrations list was getting destroyed and
- rebuilt, a crash could occur. In passing, go ahead and convert
- this list to use the linked list macros.
-
- * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
- | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
- locking when accessing the registrations list. This list is not
- actually used very often, so the likelihood of there being a
- problem is pretty small, but still possible. For example, if the
- CLI command to list the registrations was called at the same time
- that a reload was occurring and the registrations list was
- getting destroyed and rebuilt, a crash could occur. ........
-
-2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
- Dec 2006) | 3 lines Ensure that the file position is not
- incremented beyond the total number of files available for
- playback. (issue #8539, ulogic) ........
-
-2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
-
- * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
- killed bug 8423 -- OriginateSuccess and OriginateError incomplete
- channel name. May it rest in peace.
-
-2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
- retransmitted to Asterisk
-
-2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
-
- * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
- Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
- in the sample configuration file. (issue #8526, arkadia) ........
-
-2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Don't send Contact on MESSAGE
-
-2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
-
- * configure.ac: Fix curl version number testing to be much more
- friendly to non-bash shells. Issue 8508, patch by me. This
- *SHOULD* be POSIX compliant now..
-
-2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Merging the invitestate-1.4 branch after
- successful testing. Will check if I can solve this with less
- changes in 1.2.
-
- * configs/sip.conf.sample: Add missing s from another repository.
- (thanks jcmoore!)
-
- * configs/sip.conf.sample: Updating sip.conf.sample with
- information about T38 not working when chan_local or chan_agent
- is involved in the call. I don't know how big a fix that would be
- to solve, but this is the current state of affairs. (Chan_sip
- currently checks if the other side of the bridge has a SIP tech.
- We could/should implement another check, possibly for udptl_write
- or some flag in the ast_channel structure).
-
-2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Oops, forgot to release the odbc handle
-
- * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
- | 6 lines If the recording in the database is too large, it will
- fail to retrieve with an mmap error. Not too sure why this
- doesn't happen when we put it in the database, also, but since
- that doesn't seem to be broken, I'm not going to fix it (at least
- until someone reports it). Solution is to ask for the file in
- smaller chunks. (Bug 8385) ........
-
-2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix an issue which didn't allow
- unavail/greet/busy/etc messages from being saved into ODBC (and
- probably IMAP).
-
- * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
- 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
- change from 8016 - this breaks other stuff... Needs further
- review. Tip: When you've reported a bug about something and
- somebody has put up a patch for it.. It's not a good idea to open
- a completely new bug and say that something is broken because of
- the patch in the other bug - PLEASE mention something in the bug
- where the patch was actually created. ........
-
- * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
- 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
- issue where a message isn't saved correctly when using ODBC
- storage and reviewing a message. Issue 8016 - patch by sokhapkin.
- ........
-
-2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
- 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
- generic bridge tells us not to retry, and we have a frame to spit
- out then break the bridge. Props to markit in #asterisk-bugs for
- bringing this up. ........
-
-2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample: Add documentation to
- voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
- blitzrage.
-
- * doc/snmp.txt: Attempt to document some of the dependencies that
- are needed for net-snmp Issue 8499 - initial patch by blitzrage.
-
-2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
-
- * sounds/Makefile: When "fetch" is in use, instead of "wget",
- --continue is not a valid option. (issue #8451)
-
-2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Removing one of two pieces of code to
- handle 481 response on INVITE - Move handling of REFER response
- to handle_response_refer()
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
- configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
- transmission happens - Encapsulate RTP timers in the rtp
- structure so we have one for video and one for audio The video
- one is not used in 1.4, really. Will be used for RTP keepalives
- when we can send something that video phones support in the RTP
- stream. I now this is a big architectual change at this stage for
- 1.4, but decided it was needed to avoid future bug reports. -
- Document the RTP NAT keepalive option in sip.conf.sample Issue
- 7679 in the bug tracker. Please test.
-
-2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h: Backport the comment containing the
- warning regarding the limitations on the usage of this function.
- It is thread safe, but not technically reentrant.
-
-2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
- | 2 lines if Dial() is going to send music-on-hold to the calling
- party, it has to send PROGRESS first to ensure that the reverse
- audio path has been setup first (BE-106) ........
-
-2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
-
- * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
- FreeBSD 6.1 does not include wget by default. However, it has
- fetch which will work just fine for our purposes of downloading
- the sounds packages. So, check for both wget and fetch and the
- configure script and use what was found to download them. If
- neither one was found, and sound packages are selected that must
- be downloaded, the install process will print out an informative
- error message indicating the situation. Also, fix a couple places
- where "make" was hard coded into some output messages by
- replacing them with the $(MAKE) variable. (issue #8451, initial
- patch by pabelanger, with additional modifications by me)
-
-2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 48183 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
- lines Fix a small typo - issue 8848, reported by pabelanger
- ........
-
-2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/cli.c: Double-unlock error (reported by blitzrage on IRC)
-
-2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
- "limitonpeers" patch from trunk, to fix a lot of issues with
- queues and SIP device states - Remove support for T.38 early
- media, since it's impossible. (Two patches in one - extra friday
- evening offer due to being off line from svn today... :-)
-
-2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
- do a partial bridge for Google Talk since we need to handle STUN.
- (issue #8448 reported by phsultan)
-
-2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue 8319 - change noncecount before
- using it.
-
-2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
-
- * /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
- 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
- write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
- driver. (issue #8390 reported by hselasky) ........
-
- * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
- lines Only print out debug message if bridged channel is not
- NULL. (issue #8412 reported by jubilex) ........
-
- * /, res/res_features.c: Merged revisions 48154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
- lines Do not listen for DTMF on the bridge that comes into
- existence when ParkedCall is executed. This means native bridging
- can now occur for this. (issue #8406 reported by kebl0155)
- ........
-
- * main/cdr.c, /: Merged revisions 48151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
- lines Print certain CDR messages out at the NOTICE level versus
- WARNING since they can occur when used with the CDR applications
- and are perfectly fine. (issue #8367 reported by dartvader)
- ........
-
- * /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
- 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
- the pointer to the allocated block of memory so that we can free
- it and not cause a memory leak. (issue #8449 reported by arkadia)
- ........
-
- * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
- 2006) | 2 lines Document 'port' for SIP peers, came up because of
- the current mailing list thread. (issue #8450 reported by
- blitzrage) ........
-
-2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
-
- * doc/manager.txt: Explain status reports and make codefreeze more
- happy :-)
-
- * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
- GS 487 adapter without CSEQ on separate line in the REGISTER
- request. Imported from 1.2.
-
-2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
- mm_login. (issue #8420 reported by slimey)
-
-2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Explain the use device status system
- implemented in SIP for subscriptions, queues and manager a bit
- better. Like in 1.2, you will get more detailed information if
- you set a call limit for a device. When the call limit is
- reached, the status system will report a device as busy. For
- queues, setting a call limit per SIP device is propably a
- requirement. In most cases, it will work much better if you only
- use type=peer and not type=friend. We might decide to backport
- the new setting from trunk to apply all call limits to the peer
- part of a friend only.
-
-2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 48106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
- lines If the frame was duplicated before writing out then we need
- to free it. (issue #8429 reported by edguy3) ........
-
-2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
-
-2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Don't crash if the mailstream was not
- created.
-
-2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
-
- * Makefile: Export several more variables in top level Makefile.
- Inspired by issue 8438.
-
-2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
- 2006) | 2 lines According to the research I have done we never
- needed to include compiler.h in the first place so let's not!
- (issue #8430 reported by edguy3) ........
-
- * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
- lines Use the proper function to get the new message count
- instead of always using the filesystem. (issue #8421 reported by
- slimey) ........
-
-2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
- Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
- ........
-
-2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Remove a couple of unused variables (issue #8380,
- casper)
-
-2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
- lines Do not reference the freed outgoing structure in the debug
- message. (issue #8425 reported by arkadia) ........
-
-2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Change logging message
-
-2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com>
-
- * funcs/func_cdr.c: might as well also document the raw values of
- the flag vars
-
- * /, funcs/func_cdr.c: A little bit of func_cdr documentation
- upgrade-- no bug# involved, although 8221 may have inspired it.
-
-2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
- and future releases, you can disable subscription support totally
- or per peer in sip.conf with allowsubscribe = yes | no
-
-2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com>
-
- * main/translate.c: bug 8189 posted this fix for main/translate.c
- for PLC
-
-2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
- Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
- beatufied some logs, changed some loglevels. changed the default
- value of block_on_alarm ........
-
-2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Don't allocate unused variable.
-
-2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Video will never reach Packet2Packet bridging and can
- do more harm then good.
-
-2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If we have the non standard G726-32 setting turned on
- we want to return G726-32 to the SDP, not our AAL2 string. (issue
- #8330 reported by voipgate)
-
-2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
- provisional response. Let's not treat that as early media.
- (discovered at the AVTF meeting in Paris).
-
-2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Oops, merge missed release of odbc object
-
- * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
- | 2 lines Failing to trap -1 error from mmap causes segfault
- (Issue 8385) ........
-
-2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c, /: Merged revisions 47859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
- lines Don't forget to byte swap if we are exiting the smoother
- feed early. (issue #8287 reported by arturs) ........
-
- * /: Blocked revisions 47855 via svnmerge ........ r47855 | file |
- 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free
- history items at the end of use of the temporary SIP pvt
- structure. (issue #8383 reported by benh) ........
-
- * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists.
-
- * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c,
- include/asterisk/channel.h: Use a separate variable in the
- channel structure to store the context that the channel was
- dialed from. (issue #8382 reported by jiddings)
-
-2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Explain properly how videosupport works.
- Committ from Asterisk Video Task Force meeting in Paris!
-
- * /, channels/chan_sip.c: Make sure we destroy scheduled items and
- not use them ever again after destruction (rizzo)
-
-2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header
- contains angle brackets (the bug was only in a corner case where
- the < was right after the opening quote, and the fix is trivial).
-
-2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com>
-
- * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially
- pointed out by mrobinson.
-
- * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell |
- 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a
- couple of typos in applications.. Initially spotted by mrobinson.
- ........
-
-2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, doc/billing.txt: update documentation regarding IAX2 transfers
- and CDRs Merged revisions 47776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
- | 2 lines update clearly wrong documentation regarding cdr_custom
- ........
-
-2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Compare technology using the pointers
- instead of a straight comparison based on name. (issue #8228
- reported by dean bath)
-
- * /: Blocked revisions 47761 via svnmerge ........ r47761 | file |
- 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for
- the header file specifically in all cases, not just the existence
- of the directory. (issue #8358 reported by mrness) ........
-
-2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: check for pre-1.4 versions of Zaptel and
- abort the configure script if found with an appropriate error
- message
-
-2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
- notification optional, in order to avoid a lot of extra database
- lookups for all those realtime users out there.
-
-2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 47750 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
- 2006) | 2 lines Because of the way chan_local is written we
- should be extra careful and make sure our callback functions have
- a tech_pvt. (issue #8275 reported by mflorell) ........
-
- * apps/app_meetme.c: Don't unreference the SLA object if there is
- no SLA object in the devicestate callback. (issue #8354 reported
- by loloski)
-
-2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup
-
- * UPGRADE.txt: Warn users about change in canreinvite
-
- * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
- authenticated (according to the RFC) - Update docs on
- canreinvite. "nonat" is the recommended setting for most users
- with phones behind a NAT.
-
-2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 47711 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
- 2006) | 2 lines Make sure that the pvt structure exists before
- trying to do fixup on Local channels. (issue #7937 reported by
- mada123, fix by alamantia with mods by me) ........
-
-2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL
-
-2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: We need to ensure timelimit stuff is included as
- well so warnings get played. (issue #8050 reported by KNK)
-
-2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/file.c: don't try to call fclose() if fopen() failed
-
-2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Improve SIP history - Never send reply to
- ACK (again...)
-
-2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
- | 4 lines ensure that message duration is included in email
- notifications for forwarded messages (BE-96, fix by me after
- corydon used his clue-bat on me) ensure that duration in the
- message metadata is updated if prepending is done during
- forwarding (related to BE-96) remove prototype for API call that
- does not exist ........
-
- * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
- Nov 2006) | 2 lines clear the category's variable tail pointer as
- well when variables are detached from it ........ r47688 |
- kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
- lines when appending a list of variable to a category, ensure the
- tail pointer points to the last variable in the list ........
- r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
- | 2 lines when re-writing the config file, don't repeat the path
- if it hasn't changed ........
-
- * main/config.c, /: Merged revisions 47682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
- | 2 lines ouch... don't use printf, use ast_log/ast_verbose
- ........
-
-2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: fix longest match search in find_cli. Trunk already
- fixed. 1.2 not affected (well, i have no idea, the code is
- totally different there).
-
-2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Send error message when we can't allocate
- SIP dialog, possibly due to limitation of file descriptors.
- (imported from 1.2)
-
-2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If NAT detection is turned on or already detected
- then say NAT is active when setting the remote RTP peer when
- doing early bridging. (issue #8365 reported by marcelbarbulescu)
-
-2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/term.c: more formatting cleanup, and avoid running off the
- end of the string
-
-2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Turn notice about unknown RTCP packet type into a
- debug message instead.
-
-2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
- platforms (this variable is an 'int' anyway, comparing it to
- 'signed long' is not useful)
-
-2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
- lines Update copyright information in the ADSI logo blob.
- ........
-
- * channels/chan_sip.c: Only keep the video RTP structure around if
- 1. Video support is enabled and 2. A video codec is enabled on
- the dialog
-
- * funcs/func_uri.c: Small documentation clarification for
- URIENCODE. (issue #8294 reported by salaud)
-
-2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Conversion of res_odbc API to include ast_
- prefix did not completely transition app_voicemail when
- ODBC_STORAGE is used (reported on IRC by caio1982, not in
- bugtracker)
-
-2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com>
-
- * apps/app_amd.c: Use LOG_DEBUG to print out the indication that
- app_amd is using default settings instead of using LOG_NOTICE.
- This stops needless logging of this information under normal
- circumstances. (issue #8361 reported by Seb7)
-
-2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Update documentation to fit the
- implementation...
-
- * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
- retransmission system if it's an OPTION packet from peerpoke
-
-2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
- lines Initialize global pointers for connection and result to
- NULL. (issue #8356 reported by james) ........
-
-2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
- | 2 lines Having more than 255 old messages caused corruption in
- the new/old count ........
-
-2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com>
-
- * main/config.c: This solves bug 8342, whereby a crash occurs under
- certain circumstances while reading a config file with comments--
- a call to CB_ADD shouldn't happen if withcomments is zero
-
-2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/cli.c, channels/chan_sip.c: Re-enable old deprecated
- commands
-
-2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: - Don't reply to INVITE already replied
- to when we get BYE - Declare errmsg as int. Oops.
-
-2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
- the messed if, but we all forgot to update the regressions. Until
- now.
-
-2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
- found... just confuses users
-
-2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
- lines When sending an SMS with a user data header properly set
- the UDH flag in the first byte. (issue #8347 reported by
- hoffmeis) ........
-
- * main/cli.c: Free full command string upon unregistering of CLI
- command. Backported from revision 47536 from rizzo.
-
-2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Only produce error message about sip history
- once
-
-2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com>
-
- * configure, acinclude.m4: AC_PROG_SED is included in autoconf
- 2.60, but apparently it is not included in 2.59. So, to maintain
- compatability with 2.59 since it is a small change, copy this
- macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
- #8345)
-
-2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
- | 2 lines If the execute fails a second time, make sure that we
- don't pass back a stale handle ........
-
- * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
- | 2 lines Don't play dialtone if the seizing the channel fails
- (Bug 7754) ........
-
-2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
- DEA!!!)
-
- * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
- UDPTL in sdp...
-
- * channels/chan_sip.c: - Don't destroy SIP dialog because of a
- failed T.38 re-invite. Wait for a bye. Final response to a
- re-invite does not mean that the session dies, only that the
- re-invite fails. - Keep RTP active during processing of T.38
- re-invite. If the re-invite fails, RTP needs to remain as before
- the re-invite. Issue 8338 - darren1713. Please test.
-
- * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
- -Add some comments to t.38 code
-
-2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
- 4 lines Only do the check to determine whether the channel
- calling this function is an IAX2 channel when getting the IP
- address using the special argument, CURRENTCHANNEL. (issue #8341,
- jcovert) ........
-
- * Makefile: Add the target "menuconfig" as an alias for the
- "menuselect" target. This is just a favor to users so that if you
- accidentally type "make menuconfig" instead of "make menuselect",
- it still works. (inspired by a comment on IRC from wangster
- calling me an "especially devious asterisk developer" for having
- it be menuselect instead of menuconfig. :) )
-
- * main/term.c: Tweak the formatting of this new function to better
- conform to coding guidelines.
-
-2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com>
-
- * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
- safe output!
-
-2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we don't use 32 bits when we only
- need one bit.
-
-2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: ...and make sure that the dialog is
- destroyed, even if we don't get any answer on the bye... This is
- the channel that remains dead after the SIP transfer
-
- * channels/chan_sip.c: Add debug output while trying to trace bug
- in bug report
-
- * channels/chan_sip.c: Make sure we destroy dialog...
-
- * /, channels/chan_sip.c: Small cleanup of handle_request_invite()
- - imported from 1.2 with changes
-
-2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
- callerid name for switches that bork on it.
-
-2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
- SDP (alphaque)
-
-2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/prep_moduledeps: grep -m is not available on BSD, so
- use head -1 instead
-
-2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c: Only split up extension and context if a
- value exists. (issue #8332 reported by loloski)
-
-2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c,
- channels/chan_iax2.c: Discussion of these CLI changes resulted in
- more consistency (Bug 8236)
-
-2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
- removing them should be LOG_NOTICE, not LOG_DEBUG
-
- * apps/app_queue.c: reflect addition/removal of dynamic queue
- members in queue_log, so that people using dialplan replacement
- for AgentCallbackLogin can still track login/logout (issue #7736,
- reported/patched by whoiswes but this commit was written by me
- and covers all three paths for AQM/RQM)
-
-2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Rip out half implementation of 491 response
- support, since it wasn't implemented properly and caused memory
- leaks in the case of us getting 491's, which Asterisk actually
- sends... Since it is a bit too complicated to fix this, I'll rip
- it out of 1.4 and put it on the to-do-list for future releases.
- Now, we handle this as congestion, which it really is. Issue
- #8331
-
- * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
- Thanks fenlander!
-
-2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Fix building of chan_h323 by completeing
- some structure definitions. (issue #8327 reported by Mithraen)
-
- * apps/app_voicemail.c: Do conversion in a more easier to read and
- working way for \r, \n, and \t. (issue #8324 reported by
- johnlange)
-
-2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, channels/chan_zap.c,
- build_tools/prep_moduledeps: Work around an issue that caused
- menuselect to display a bogus description for app_voicemail and
- chan_zap. These modules use some preprocessor directives to
- determine what it will report to Asterisk as its description.
- However, the way we extract this information from the source
- files for menuselect is not smart enough to figure this out.
- (issue #8326, #8328)
-
-2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
- 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
- higher as, well, it's apparently going to be removed. This should
- make all you FC6 fans happy as your Asterisk will now build
- without any mods. ........
-
-2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com>
-
- * main/cli.c: fix tab completion for "core debug channel" and "core
- no debug channel"
-
- * main/cli.c: Fix "core show channel". Also, fix tab completion for
- both "core show channel" and "core show channels".
-
- * main/cli.c: Fix "core debug channel <whatever>". I guess someone
- needs to go through and audit every CLI command that changed
- number of arguments ...
-
- * main/asterisk.c: revert the previous change, which actually
- modified the deprecated command, "show profile". Now, actually
- apply the change to "core show profile".
-
- * main/asterisk.c: Fix argument parsing for the "core show profile"
- CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)
-
- * main/cli.c: Fix another CLI command, "core show uptime" ...
- (issue #8323, reported by johnlange, fixed by myself)
-
- * main/asterisk.c: fix "core show version" to reflect the new
- number of arguments for this CLI command (issue #8316, kshumard)
-
-2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com>
-
- * main/channel.c: This update fixes 7531
-
- * channels/chan_skinny.c: Committed in behalf of 8190.
-
-2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/frame.c: the battle over CLI command formats has broken
- stuff...
-
- * channels/chan_sip.c: add simple fix for SDP to report proper
- sample rate for G.722 media sessions
-
-2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com>
-
- * utils/streamplayer.c: I occasionally get email from users that
- are trying to figure out what this does, or due to some
- misunderstanding as to what it is supposed to do, can't get it to
- work. So, I have added some text here to hopefully explain what
- this application does and does not do.
-
- * channels/chan_gtalk.c: Make this module build again
-
- * configure, configure.ac, acinclude.m4: Copy the macros from
- libtool.m4 to our own acinclude.m4 such that libtool is no longer
- required to be installed to be able to generated the configure
- script.
-
-2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
-
-2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com>
-
- * channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
- channels/chan_misdn.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
- include/asterisk/stringfields.h, apps/app_voicemail.c,
- main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
- channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
- channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
- channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
- solve the problem in bug 7506. It's a lot of rework to solve a
- fairly small problem... such is life.
-
-2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c: Make MOH work as it did before in
- chan_local, without this then it can go funky when transfers and
- MOH are involved. (issue #7671 reported by jmls)
-
-2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/musiconhold.conf.sample: clean up sample config, and make
- native file playback the more obvious default choice
-
-2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c: large overhaul to voicemail imap support.
- Allows support for more imap servers, also a better
- implementation of several parts of the original work. patch
- provided by 8033 with major upgrades.
-
-2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
- continue.
-
-2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Fixing the attack shield so it doesn't
- produce attacks... Issue 8265 - never reply to an ACK
-
-2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
- Nov 2006) | 5 lines If random order is enabled for files mode
- music on hold, set a random initial position, instead of always
- starting at the first file, and doing the random operation only
- when switching to the next file. (bug reported by John Lange on
- the asterisk-dev mailing list) ........
-
-2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
- transfer from "john" Thank you!
-
-2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com>
-
- * main/cli.c: Fix another bug in "core set debug" ...
-
- * main/asterisk.c, main/cli.c: Really fix the "core set debug" and
- "core set verbose" CLI commands.
-
- * main/cli.c: fix the "atleast" option to the "core set verbose"
- and "core set debug" CLI commands
-
-2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This fix introduced via bug 8233
-
-2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org>
-
- * bootstrap.sh: align bootstrap.sh with the version in trunk (needs
- to be blocked as it is already in trunk)
-
- * configure.ac: add proper environment vars to detect modules on
- freebsd. (already applied to trunk so it needs to be blocked
- there)
-
-2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
- channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
- changes making the CLI more consistent with "category verb
- arguments" (continuation of issue 8236)
-
- * main/config.c, main/cli.c, main/channel.c, main/manager.c,
- channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
- main/http.c, main/file.c, main/logger.c, main/image.c,
- res/res_indications.c, main/asterisk.c, res/res_odbc.c,
- channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
- channels/chan_local.c, main/frame.c, channels/chan_sip.c,
- res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
- res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
- Reverse change of "show" to "list" and make several other
- commands more consistent with "category verb arguments"
-
-2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move check for codec translation to
- sip_call() instead of in add_sdp. No one bothers with the result
- of add_sdp anyway... Yet...
-
- * channels/chan_sip.c: Disable code for T38 over TCP and RTP since
- there's no trace of actual functionality for it :-)
-
-2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
- Nov 2006) | 3 lines ignore files in a music on hold directory
- that begin with '.' (issue #8249, cboie) ........
-
-2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix
-
-2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: don't send INVITE when we have determined
- that we can't offer any audio formats due to lack of transcoding
- support (or incorrect configuration)
-
-2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
- lines Repeat after me oej: I will at least make sure my code
- compiles before I commit it. ........
-
-2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)
-
-2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com>
-
- * /, main/callerid.c: Add the missing call to free described in
- issue #8268. Also, add a bunch of missing calls to free in
- callerid_feed_jp().
-
- * main/say.c: fix saying one hundred and two hundred in hebrew
- (issue #7810, eldadran)
-
- * Makefile, configure, codecs/gsm/Makefile, configure.ac,
- build_tools/strip_nonapi, makeopts.in: Fixes for
- cross-compilation on mips (issue #8058, ywalther, with some
- modifications)
-
- * aclocal.m4, build_tools/menuselect-deps.in, configure,
- build_tools/embed_modules.xml, configure.ac: Add a check in the
- configure script to determine whether ld is GNU ld or not. This
- is needed because module embedding only works for gnu ld. GNU ld
- is now listed as a dependency for all of the module embedding
- options in menuselect. (issue #8143)
-
-2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com>
-
- * channels/chan_gtalk.c: bind address support from bug 8164
-
-2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com>
-
- * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
- accept longer strings or mass confusion and a lot of lost time is
- the result
-
-2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com>
-
- * main/Makefile: Force poll() emulation for Darwin to always be on.
- It's too broken to consider being used. This resolves the console
- issue OSX users have been seeing. I would have liked to autoconf
- this but I haven't been able to come up with a test case that
- works. Que sera.
-
-2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
- 9 lines soxmix and Asterisk expect different file extensions for
- certain formats. This was already handled for the wav49 format.
- However, it was not handled for ulaw and alaw. I fixed this in
- such a way that using the alternate extensions for ulaw and alaw
- will only happen if we know we're calling soxmix, and not a
- custom script defined using the MONITOR_EXEC variable. The wav49
- processing was left alone so that external scripts will see no
- behavior change. (issue #7550, reported by mnicholson, proposed
- patch by junky, committed fix is a bit different) ........
-
-2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: It's another round of chan_iax2 fixes!
- Should hopefully fix the deadlock issues people have been
- reporting. IAXtel now has qualify turned on for 800 peers and it
- is handling it fine.
-
-2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com>
-
- * main/config.c: Cleanups suggested by Russell.
-
-2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Prevent an infinite loop when config
- processing gets to a jitterbuffer option
-
-2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com>
-
- * main/translate.c: Fix "core show translation" output. Issue
- #8243, patch by Damin.
-
-2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/translate.h, main/translate.c: add an API so
- that translators can activate/deactivate themselves when needed
-
- * include/asterisk/translate.h, main/translate.c: revert changes
- that were the wrong way to address this... proper fix coming
-
- * main/translate.c: let's set the seen flag early enough to
- actually make a difference...
-
- * include/asterisk/translate.h, main/translate.c: don't re-do setup
- operations for translators that can dynamically register
- themselves
-
-2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /: Blocked revisions 46662 via svnmerge ........ r46662 |
- tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines
- Move thread-unsafe initializer to the module loading code; add
- the corresponding function to the module unload to fix a memory
- leak. ........
-
-2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net>
-
- * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
- #8089 - Fix the ENUM support (picking one record by number).
- Thanks otmar!
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
- when we're supposed to support ;rport. Issue #7473.
-
- * /, channels/chan_sip.c: If peer fails ACL check, fail peer at
- REGISTER
-
- * channels/chan_sip.c: Fix T38 too. Thanks, tgrman !
-
-2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com>
-
- * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
- boot process to ensure it starts after stuff like MySQL (issue
- #8253, Alric)
-
- * /, main/utils.c: Merged revisions 46560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
- 3 lines When handling the case where the hostname is just an IPV4
- numeric address, be sure to set the address type. (issue #8247,
- alexr) ........
-
- * /, res/res_agi.c: Merged revisions 46557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
- 3 lines fix some copy/paste bugs in the checking of arguments for
- the "control stream file" AGI command (issue #8255, mnicholson)
- ........
-
- * main/translate.c: Add a small tweak to the code that checks to
- see whether destination formats are translatable based on the
- source format. If we have already determined that there is no
- translation path in one direction, don't bother checking the
- other direction.
-
-2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/translate.c: when unregistering a translator, don't rebuild
- the translation matrix unless needed when filtering formats out
- of an offer, ensure we check for translation ability in both
- directions
-
- * include/asterisk/linkedlists.h: ensure that items removed from a
- list are always unlinked from the list (next pointer set to NULL)
-
-2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Don't explicitly link in crypt as it is
- not used on some platforms.
-
- * channels/chan_iax2.c: We need to lock the pvt structure during
- retransmission as another worker thread may be doing something as
- well.
-
-2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net>
-
- * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
- include/asterisk/doxyref.h, channels/chan_sip.c,
- main/ast_expr2f.c, include/asterisk/module.h,
- formats/format_ogg_vorbis.c, main/app.c,
- include/asterisk/channel.h, include/asterisk/lock.h,
- include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
- kshumard. An extra big thankyou is given to everyone that
- contributes to doxygen! THANK YOU!
-
- * main/rtp.c, /: Bind RTCP to the same IP as RTP
-
- * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
- redirects (imported from 1.2)
-
- * /, channels/chan_sip.c: Issue #7608 - Notifications sent with
- wrong content-type (imported from 1.2, modified)
-
- * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
- was reported for trunk, but obviously exists in 1.4 too.
-
- * channels/chan_sip.c: Restoring the old logic, since working
- around it and fixing it seemed too complicated. - The
- SIP_OUTGOING flag indicates the direction of the last transaction
- in the dialog. - The initreq stores the last request in the
- dialog, the request that opened the latest transaction. Please
- now retry all the 1.4 bug reports with mixed to/from headers,
- tags etc in ACK, BYE, CANCEL. Thanks!
-
- * channels/chan_sip.c: Accepting a message twice may be
- misinterpreted...
-
- * channels/chan_sip.c: - 183 is not reliable message... - Error
- should not have SDP
-
-2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com>
-
- * utils/Makefile: Don't build muted on OpenBSD, it is not
- supported.
-
-2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: move the copy of the default settings to the
- global settings back out of process_zap, so that they aren't
- overwritten when process_zap is called multiple times
-
-2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net>
-
- * contrib/asterisk-ng-doxygen: Put some doxygen pressure on
- Christian :-)
-
-2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
- res/res_musiconhold.c: We should always be using _exit() after a
- fork() or vfork() instead of exit(). This is because exit() does
- some extra cleanup which in some implementations of vfork(), for
- example, can actually modify the state of the parent process,
- causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
-
- * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell
- | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We
- should always be using _exit() after a fork() or vfork() instead
- of exit(). This is because exit() does some extra cleanup which
- in some implementations of vfork(), for example, can actually
- modify the state of the parent process, causing very weird bugs
- or crashes. (issue #7971, Nick Gavrikov) ........
-
- * channels/chan_zap.c: Instead of iterating all of the options once
- to look for jitterbuffer options, and then again for everything
- else, move the processing of jitterbuffer options into the main
- loop so that there are no erroneous messages about ignoring
- unknown options. (issue #8226)
-
-2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 46350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
- 1 line fixed a bug which caused chan_misdn to try to allocate 2
- times the same channel on high load, which then caused
- instability of mISDN. removed a useless function from isdn_lib.c
- ........
-
- * channels/misdn_config.c: fixed not compile issue, which was just
- introduced
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
- Merged revisions 46176 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
- 1 line added nttimeout option to configure wether we disconnect
- calls on NT timeouts or not during an overlapdial session
- ........
-
-2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
- lines oops - somebody forgot to change this - long ago, probably.
- ........
-
- * CHANGES: grammar check
-
-2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Corrections to changes (Multiparking is not included)
-
-2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com>
-
- * main/translate.c: - If the source has no audio or no video
- portion, do not call powerof() to get the format index. - Don't
- run through the audio and video loops if there is no audio or
- video portion of the source If 0 is passed to powerof, it will
- return -1. This value of -1 was then being used as an array index
- in these loops, which caused a crash on some systems. Other than
- this issue, this code works as we expected it to. If a format is
- not in the source, and we have to translation path to it, it is
- not offered in the list of acceptable destination formats. (fixes
- issue #8231)
-
-2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com>
-
- * CHANGES: update to reflect G.722 addition
-
-2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com>
-
- * doc/backtrace.txt: update backtrace documentation to reflect
- changes in 1.4 (issue #8230, kshumard)
-
-2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com>
-
- * main/config.c, main/manager.c: Fix config comment code
- preservation code (thanks murf!)
-
-2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Old todo note - Don't add Contact header on
- BYE and Cancel
-
-2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com>
-
- * configure.ac: fix error output when checking for openh323 to
- refer to openh323 instead of pwlib (issue #8222, misaksen)
-
-2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Somewhat ugly code to try to fix issue
- #7608. Since the problem was not very well defined, the fix is a
- bit fuzzy too... Thanks to Luigi for accidentally spotting the
- possible problem!
-
-2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: update warning message to include "agi" option
- (issue #8225, jmls)
-
-2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: use 1.4.3 extra sounds with corrected silence
- files
-
- * sounds/sounds.xml, sounds/Makefile: add support for prebuilt
- G.722 prompts and music on hold files
-
-2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: show settings doesn't produce a list of
- similar objects, it should stay a "show"
-
-2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
- channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
- pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
- main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
- cdr/cdr_custom.c, channels/chan_mgcp.c,
- apps/app_parkandannounce.c, apps/app_voicemail.c,
- channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
- res/res_adsi.c, main/utils.c, apps/app_ices.c,
- pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
- apps/app_getcpeid.c: apparently developers are still not aware
- that they should be use ast_copy_string instead of strncpy... fix
- up many more users, and fix some bugs in the process
-
-2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/pbx.c: WaitExten truncates decimals of times to wait,
- instead of accepting them (Bug 8208)
-
-2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
- channels/chan_h323.c, channels/chan_iax2.c,
- include/asterisk/frame.h: add passthrough and file format support
- for G.722 16KHz audio (issue #5084, original patch by andrew,
- updated by mithraen)
-
- * channels/chan_sip.c, main/translate.c: code zone experiment:
- don't offer formats in the outbound INVITE that aren't either
- passthrough or translatable
-
- * main/translate.c: if multiple translators are registered for the
- same source/dest combination, ensure that the lowest-cost one is
- always inserted earlier in the list
-
-2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com>
-
- * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
- #8147)
-
-2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: We need to initialize our scheduler pthread
- condition... yes.
-
-2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org>
-
- * main/http.c: merge 45152 don't leak descriptors in http.c
-
- * channels/chan_sip.c: merge 45966 refer_to_domain potentially
- containing options
-
- * channels/chan_sip.c: merge 46026 improper checks on get_header()
- return values
-
- * channels/chan_sip.c: merge 46045 prevent NULL args to
- ast_strdupa() in chan_sip.c
-
-2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com>
-
- * Makefile: Restore the ability to remove the firmware directory
- without causing the installation to fail (issue #8111)
-
-2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/translate.c: ensure that the translation matrix is properly
- lock-protected every place it is used
-
- * include/asterisk/translate.h, main/translate.c: add an API call
- to allow channel drivers to determine which media formats are
- compatible (passthrough or transcode) with the format an existing
- channel is already using
-
- * doc/imapstorage.txt: simplify and correct voicemail IMAP storage
- build instructions
-
-2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/channel.c: Pass through a frame if we don't know what it is,
- rather than trying to pass a NULL, which will segfault a channel
- driver (Bug 8149)
-
-2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com>
-
- * utils/muted.c, utils/ael_main.c: In muted.c, check the return
- value of strdup. In ael_main.c, check the return value of calloc.
- (issue #8157) In passing fix a few minor bugs in ael_main.c. The
- last argument to strncpy() was a hard-coded 100, where it should
- have been 99. I changed this to use sizeof() - 1.
-
- * apps/app_meetme.c: Fix the descriptions of some of the
- MeetMeAdmin options (issue #8098, mflorell)
-
- * res/res_jabber.c: don't crash when an incoming message has no
- "from" (issue #8205, jmls)
-
-2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com>
-
- * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
- lines Don't leak memory mmmk? ........
-
-2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
- Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
- couldn't be initialized it would cause a segfault after 'reload'.
- Reported by Drew/Matt thx. ........
-
-2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com>
-
- * res/res_monitor.c: Add a couple missing unregistrations of
- manager actions and remove duplicate unregistrations of
- applications. (issue #8194, jmls)
-
-2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Don't use promotion on Darwin because it doesn't
- seem to work quite right in all cases, this should solve the
- unresolved symbol issue people have been seeing.
-
- * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
- installed in the proper location (reported on asterisk-dev
- mailing list)
-
-2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Let's understand SIP: - REFER can create
- dialog, Asterisk does not support it yet - NOTIFY can create
- dialog in Asterisk's implementation (voicemail) even though we
- don't support the server side of it. In this case, the standard
- is a side issue ;-) - Added extened functionality for unsupported
- methods (PING, PUBLISH) so we don't create PVT's for those
- either. Russellb needs to judge what to do with this in 1.2, but
- I think the current implementation n 1.2 is a bug since we're
- sending bad replies to NOTIFY and REFER outside of dialogs
-
-2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com>
-
- * res/res_jabber.c: Let's remember to unregister JabberStatus too
- (issue #8184 reported by jmls)
-
- * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
- 2006) | 2 lines Respect language selection when seeing if the
- file exists (issue #8178 reported by mnicholson) ........
-
- * channels/chan_sip.c: If the jitterbuffer is forced on then we
- can't partially bridge (reported by wangster on #asterisk-dev)
-
-2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Don't leak the actual thread-specific
- sip_pvt struct
-
-2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: don't leak memory when a chan_sip thread is
- destroyed that has a thread-local temp_pvt allocated
-
-2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Don't modify things if we are using vfork as
- this is very bad and may cause unexpected behavior (issue #7970
- reported by Nick Gavrikov)
-
-2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: remove duplicate declarations
-
-2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org>
-
- * main/http.c: merge from trunk: move ast_variables_destroy() to a
- better place in handle_uri() to avoid leaking memory on non
- existing files.
-
-2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Don't segfault if you're using a channel driver that
- doesn't turn RTCP on
-
-2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't attempt to access private data members of
- the pthread_mutex_t object, because this does not work on all
- linux systems. Instead, just access the reentrancy field in the
- ast_mutex_info struct when DEBUG_THREADS is enabled. If
- DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
- DEBUG_THREADS on as well. (issue #8139, me)
-
- * configs/sip_notify.conf.sample: update entry to reboot a snom
- phone (issue #7850, pnlarsson)
-
-2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0-beta3 released.
-
-2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/stringfields.h, main/ast_expr2.c,
- main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
- optimize the 'quick response' code a bit more... no more malloc()
- or memset() for each response expand stringfields API a bit to
- allow reusing the stringfield pool on a structure when needed,
- and remove some unnecessary code when the structure was being
- freed
-
-2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't create a "real" pvt structure for
- requests that shouldn't be able to create one. Instead use a
- temporary pvt and fill it with enough information so we can send
- a reply.
-
-2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Adding information about Marks
- direct-RTP hack to the docs...
-
-2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com>
-
- * LICENSE: provide licensing language for IAXy firmware file
-
-2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
- directed pickup (BE-85).
-
-2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net>
-
- * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
- your support!
-
- * channels/chan_sip.c: Don't destroy dialog for unexpected REFER
- response...
-
-2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com>
-
- * funcs/func_rand.c: update the doc string for both AEL and
- extensions.conf users.
-
-2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/acl.c don't drop the entire permit/deny list when an attempt
- is made to add an invalid entry (BE-92)
-
-2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c: Clear the quiet flag too since we are
- restarting a recognition again (reported on -dev by Stephan
- Edelman)
-
- * res/res_speech.c: Check return value from engine in case of
- failure (ie: out of licenses) (reported on -dev mailing list)
-
-2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest17 (added),
- pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
- pbx/ael/ael-test/ael-vtest17 (added),
- pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
- this release via these changes
-
-2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: avoiding warning, fixing potential bug
-
-2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com>
-
- * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
- codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
- codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
- codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
- codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
- codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
- codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
- codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
- codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
- codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
- codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
- codecs/lpc10/analys.c, codecs/lpc10/onset.c,
- codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
- codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
- codecs/lpc10/median.c, codecs/lpc10/encode.c,
- codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
- codecs/lpc10/invert.c: And file said... let the compiler warnings
- STOP!
-
- * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
- reported by mnicholson)
-
- * apps/app_playback.c: Move say.conf existence check to do_say
- function since it is called from multiple places (issue #8144
- reported by kshumard)
-
-2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
- we have multiple bindings (reported on asterisk-dev)
-
-2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Complete merging in RPID screen changes
- (issue #8101 reported by hristo, patch by oej in revision 44757)
-
- * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
- the background refresh item back into the scheduler if enabled
- since it is deleted during reload. (issue #8142 reported by
- p_lindheimer)
-
-2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/utils.c: use a configure script test for PMTU discovery
- control instead of just assuming it's available on Linux
-
-2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
- echocandisable issues when bridged. this caused a kernel panic
- sometimes.. also some minor formatting fixes
-
- * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
- got a wrong isdn cause at RELEASE_COMPLETE
-
-2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: merge formatting and minor code
- simplifications from trunk
-
-2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com>
-
- * channels/chan_gtalk.c: fix for bug 7764.
-
-2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: we can only send one 'a=ptime' attribute per
- media session, not one for each format
-
- * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
- main/utils.c: ensure that IAX2 and SIP sockets allow UDP
- fragmentation when running on Linux (thanks to Brian Candler on
- the asterisk-dev list for the tip)
-
-2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com>
-
- * main/manager.c: fix a silly typo in a comment that I saw while
- reading the commit list
-
-2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com>
-
- * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
- #8135 reported by ssokol)
-
-2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com>
-
- * main/manager.c: append_event must be called while holding the
- session lock
-
-2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com>
-
- * res/res_jabber.c: change some debug output to use LOG_DEBUG
- instead of verbose output
-
-2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com>
-
- * main/db1-ast/Makefile: These are already set by the parent
- Makefile.. There is no need to have this here (it doesn't
- actually work anyways).
-
-2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c: removed warning because of missing
- prototype declaration
-
-2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Do not set default/global values in the
- variable declaration, set it in reload_config()
-
-2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Move some stuff around so that a NOTIFY
- dialog won't hang around until the end of the world under certain
- circumstances
-
-2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz>
-
- * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
- CHANNEL() function sometime mix parameter and value
-
-2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_logic.c: Lost of a bit of logic when this was
- simplified between 1.2 and 1.4 (Bug 8117)
-
-2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Bail out if we have no refer structure and
- we get a refer response
-
-2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: more merge from trunk (comments and change a
- static function name)
-
-2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only set DTMF information if an RTP
- structure exists
-
-2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
- support of dynamically enabling hdlc on bchannels
-
-2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: whitespace changes related to previous
- commit
-
- * channels/chan_sip.c: merge a few code simplifications that have
- gone into trunk during last week, to reduce differences between
- the two branches and make porting fixes easier.
-
-2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix a problem where phones that go
- "missing" never got unregistered. Issue #8067, reported by pj,
- patch by Anthony LaMantia (with minor whitespace modifications)
-
-2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
- the deadlock
-
- * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
- (issue #8115 reported by vazir)
-
-2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: do not dereference p if we
- know it is NULL
-
-2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx, channels/chan_h323.c,
- channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
- caller's transfer capability too
-
-2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: put common code in a
- function to avoid repetitions.
-
- * channels/chan_sip.c: remove hardwired usage of 5060, use
- DEFAULT_SIP_PORT instead
-
- * channels/chan_sip.c: option_debug checking
- before printing to debug channel.
-
- * channels/chan_sip.c: backport simplifications on sip_register,
- usage of ast_set2_flag(), and fixes to the handling of failed
- module loading.
-
- * channels/chan_sip.c: improve and document function
- get_in_brackets(), introducing a helper function
- find_closing_quote() of more general use.
-
-2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/linkedlists.h: ensure that mutex locks inside
- list heads are initialized properly on platforms that require
- constructor initialization (issue #8029, patch from timrobbins)
-
- * CHANGES: remove Jingle as per mog
-
-2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Remove the seqno check for RFC2833, the handler is
- smart enough to not need it.
-
-2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com>
-
- * CHANGES: various cleanups
-
-2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: When the sequence number rolls over then reset the
- recorded sequence number for DTMF (issue #8106 reported by
- bungalow)
-
- * main/file.c: Even more frames to treat as though the remote side
- disappeared (issue #8097 reported by eldadran)
-
-2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c, main/http.c: make sure sockets are blocking when
- they should be blocking.
-
-2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed segfault which happens during
- hold/transfer action
-
- * channels/chan_misdn.c: if INFORMATION Message come with keypad
- instead of called party number, we just use the keypad as called
- party number.
-
- * 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:
- added the option 'reject_cause' to make it possible to set
- the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
- which is automatically rejected because chan_misdn does not
- support that kind of callwaiting. Therefore chan_misdn supports
- now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
- now gets the info if the requested channel is incoming or
- outgoing to make the 3. channel possible
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
- removed a useless bc field, added setting of frame.delivery fields,
- some minor code cleanups
-
-2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com>
-
- * main/file.c: Treat busy control frames as hangup in the file streaming
- core (issue #8097 reported by eldadran)
-
-2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
- Many thanks to Doug!
-
-2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
- hanging by a thread if the other side is already setup with T.38
-
-2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/app.c: don't segfault when an argument without a close
- parenthesis is found stop parsing as soon as that situation
- occurs
-
-2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com>
-
- * CHANGES: I put the accumulated changes from the commit logs and
- inspection, into CHANGES. Hope everyone approves!
-
- * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
- install process sticks muted.conf in /etc/asterisk, so that's
- where muted should look for it, right?
-
-2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't totally bail out if T.38 was
- negotiated
-
-2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: fix Polycom presence notification again
-
-2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org>
-
- * utils/Makefile: as far as i can tell astman only uses newt...
-
- * Makefile: put linker flags in ASTLDFLAGS where they belong
-
-2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
- requests add workaround for new Polycom firmware SUBSCRIBE
- requests (bug is known to exist in 2.0.1 firmware)
-
- * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
- work
-
-2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
- pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
- pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
- pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
- pbx/ael/ael-test/ael-test16/extensions.ael (added),
- pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
- pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
- pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
- problems reported in bug 8090
-
-2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
- main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
- channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
- main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
- include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
- channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
- main/devicestate.c, main/utils.c, res/res_musiconhold.c,
- channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
- thread creation code a bit reduce standard thread stack size
- slightly to allow the pthreads library to allocate the stack+data
- and not overflow a power-of-2 allocation in the kernel and waste
- memory/address space add a new stack size for 'background'
- threads (those that don't handle PBX calls) when LOW_MEMORY is
- defined
-
-2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com>
-
- * configs/muted.conf.sample: I've been meaning to add some
- explanation about muted... here it is
-
- * configs/manager.conf.sample: CLI reverbification update to this
- config file
-
- * apps/app_macro.c: In response to bug 7776, a Warning has been
- added to the doc string for Macro().
-
-2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/asterisk.c, main/loader.c, main/term.c, Makefile,
- include/asterisk.h: ensure that local include files are always
- used avoid a duplicate function name (term_init())
-
-2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com>
-
- * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
- client without resource.
-
-2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: fix a logic error in my previous fix to the queue
- reload code
-
-2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Change default presentation indicator
- to "user provided not screened" if octet 3a missed in
- CallingPartyNumber IE
-
-2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Use VideoSupport instead so it is considered
- a valid XML attribute name. (issue #8075 reported by renemendoza)
-
-2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Fix preparation of type and
- presentation of calling number
-
-2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com>
-
- * doc/jingle.txt, channels/chan_jingle.c (removed),
- include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
- res/res_jabber.c: updated res_jabber for even better component
- support, soon will be jep-0100 compliant. also removed
- chan_jingle and infromed info from jingle.txt, chan_gtalk still
- works and should be used in this version.
-
-2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Change the fd on the I/O context in case it
- changed during the reload, which is indeed possible. (issue #7943
- reported by eclubb)
-
- * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
- instead of hardcoding the path for the error message (issue #7942
- reported by eclubb)
-
-2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz>
-
- * configs/users.conf.sample, pbx/pbx_config.c: Missed part of
- userconf functionality for chan_h323
-
-2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com>
-
- * main/io.c: Shrink when current_ioc is unused. It is set to -1 when
- unused, not 0. (issue #7941 reported by eclubb)
-
-2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz>
-
- * doc/realtime.txt: Typo fix
-
- * channels/chan_h323.c: Optimization of oh323_indicate(): less
- locks - less problems, plus single exit point
-
-2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
- you're not talking about a channel :)
-
-2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Do not simulate any audio tones if we got
- PROGRESS message
-
-2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com>
-
- * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
- be empty. The cause is that since ASTDATADIR is explicitly
- exported using "export ASTDATADIR" at the top of the Makefile,
- make no longer considers the variable "undefined", so the
- Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
- #8063, reported by akohlsmith, fixed by me)
-
- * configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
- option in the sample queues.conf (issue #8065, adamg)
-
-2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: sync with trunk - move variable declarations
- to the beginning of a block.
-
-2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz>
-
- * main/rtp.c: Allow one-way RTP streams (device->Asterisk)
-
-2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
- build problems: - with AST_DEVMODE, building codecs/lpc10 fails
- because of lots of warnings, and the configure step in editline
- fails as well. Fix this by removing the -Werror in these steps. -
- on FreeBSD (but probably on other platforms as well), the final
- link of asterisk fails because AST_LIBS was not exported to the
- subdirs Makefiles. Add a proper fix in the top-level Makefile (a
- possible alternative way is to add "export AST_LIBS" near the
- beginning of the file). With this fix, i believe that some of the
- platform-specific conditionals in main/Makefile are redundant
- (because they should be already dealt with in the top level
- Makefile) but i don't have a platform to check. Merging to head
- will happen in a moment.
-
-2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
- of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
- by phsultan with a small fix by me, myself or I. Thanks,
- Philippe! (This was caused by my changes to the transaction
- handling)
-
- * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
- VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
- sends ACK not on OK message only (when remote party answers) but
- on RINGING message too, so when we send 200 OK message, we get
- unidentified ACK message (because INVITE acknowledged on RINGING
- message already), so 200 OK retransmits within its retransmission
- interval then call gets dropped. If someone else knows how to
- provide workaround for such cases, please, fix it in correct way.
- Thanks to ssh from #asteriskru for provide access to his box to
- study and fix this case.
-
-2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com>
-
- * agi, utils: ignore temporary files made by the Makefiles during a
- build
-
- * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
- codecs/Makefile, utils/Makefile, configure,
- build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
- Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
- pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
- system bugs, and convert Makefiles to be compatible with GNU make
- 3.80
-
-2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c, main/cli.c: Fix a bug with the removal of
- 'atleast' argument to 'core verbose' and 'core debug'. Add that
- argument back in.
-
-2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
- carefully when no CallingNumber IE available
-
- * channels/h323/ast_h323.cxx: Fake display name by called number on
- incoming calls (until passing connected number/connected name is
- not implemented)
-
- * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
- includes
-
- * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
- pass TON/PRESENTATION information - original
- H323Connection::SendSignalSetup() destroys Q.931 fields.
-
-2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/Makefile: yet another place where we were not using the
- correct CFLAGS by default
-
- * main/Makefile: missed one conversion to ASTCFLAGS
-
-2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx, channels/chan_h323.c,
- channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
- TON/PRESENTATION information too
-
-2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
- main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
- Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
- CFLAGS and LDFLAGS for build of Asterisk components, because they
- are also then used for non-Asterisk components (like menuselect);
- use our own variables instead
-
- * configure, configure.ac: support --without-curl in configure
- script
-
- * Makefile.rules: another cross-compile fix
-
- * Makefile: a couple more environment settings that can't leak into
- the menuselect build
-
- * main/cli.c: proper fix for ast_group_t change
-
- * include/asterisk/lock.h: eliminate compiler warning when
- DEBUG_CHANNEL_LOCKS is enabled and users of this header file
- don't also include channel.h
-
-2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Fix incorrect argument order for member names,
- on persisted members. Issue 8047, patch by jmls.
-
-2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c, res/res_monitor.c,
- include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
- channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
- main/udptl.c, main/frame.c, funcs/func_timeout.c,
- channels/chan_sip.c, apps/app_festival.c,
- channels/iax2-provision.c, apps/app_alarmreceiver.c,
- res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
- Put in missing \ns on the end of ast_logs (issue #7936 reported
- by wojtekka)
-
-2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: fix buggy (and overly complex) loop used during reload
- of app_queue for static member list updating
-
-2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Extend call establishment timeout
-
-2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Make sure the pvt exists before accessing
- it again as it may have gone away (issue #7562 reported by Seb7
- and issue #7939 reported by sorg)
-
- * main/cli.c: Warning be gone!
-
-2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: app_queue is comparing the device names incorrectly
- while checking their statuses. It's internal list of interfaces
- includes the dial string, while the argument passed to this
- function does not have the dial string (/n for a local channel).
- This causes it to ignore the device state changes because it
- thinks it belongs to none of its members. (#8040 reported and
- patch by tim_ringenbach)
-
-2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Stop the stream after waitstream returns so that our
- formats get restored. (issue #7370 reported by kryptolus)
-
-2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Fix compiler warning
-
-2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
- tim_ringenbach reported and patched)
-
- * apps/app_queue.c: Autopause not working for queue members. (#8042
- - jmls reported and patch)
-
-2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
- remote side to start media on outgoing PROGRESS message
-
- * include/asterisk/compiler.h: Put attribute tag at correct place
-
-2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
- when the call could not be properly established in misdn_call.
- also removed the ACK_HDLC stuff which is not really needed.
-
-2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Do not open transmit channel until
- TCS is received
-
- * main/file.c: Don't warn on HOLD/UNHOLD control frames
-
- * main/file.c: Don't treat unknown control frames as voice
-
-2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Avoid inability to lock directory log message by
- creating the directory ahead of time. (Issue 7631)
-
-2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com>
-
- * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
- not being set under certain circumstances. Fix a minor issue, to
- make it use the filenames that were parsed, instead of the entire
- argument string. Fix Background() to return -1 like Playback(),
- if no args are specified.
-
-2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Compensate for out of order packets better if RFC2833
- compensation is turned on.
-
- * channels/chan_iax2.c: Get rid of two functions from a time now
- past (we THINK these are from pre-recursive lock time) that may
- be contributing to two open issues on the bug tracker (7562/7939)
- and that has the potential to just make bad things happen if the
- timing is right.
-
-2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com>
-
- * main/channel.c,res/res_features.c: Fix a problem that occurred if
- a user entered a digit
- that matched a bridge feature that was configured using multiple
- digits, and the digit that was pressed timed out in the feature
- digit timeout period. For example, if blind transfer is
- configured as '##', and a user presses just '#'. In this
- situation, the call would lock up and no longer pass any frames.
- (issue #7977 reported by festr, and issue #7982 reported by
- michaels and valuable input provided by mneuhauser and kuj. Fixed
- by me, with testing help and peer review from Joshua Colp). There
- are a couple of issues involved in this fix: 1) When
- ast_generic_bridge determines that there has been a timeout, it
- returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
- this result, it calls ast_generic_bridge over again with the same
- timestamp for the next event. This results in an endless loop of
- nothing until the call is terminated. This is resolved by simply
- changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
- sees a timeout. 2) I also changed ast_channel_bridge such that if
- in the process of calculating the time until the next event, it
- knows a timeout has already occured, to immediately return
- AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
- anyway. 3) In the process of testing the previous two changes, I
- ran into a problem in res_features where ast_channel_bridge would
- return because it determined that there was a timeout. However,
- ast_bridge_call in res_features would then determine by its own
- calculation that there was still 1 ms before the timeout really
- occurs. It would then proceed, and since the bridge broke out and
- did *not* return a frame, it interpreted this as the call was
- over and hung up the channels. The reason for this was because
- ast_bridge_call in res_features and ast_channel_bridge in
- channel.c were using different times for their calculations.
- channel.c uses the start_time on the bridge config, which is the
- time that the feature digit was recieved. However, res_features
- had another time, 'start', which was set right before calling
- ast_channel_bridge. 'start' will always be slightly after
- start_time in the bridge config, and sometimes enough to round up
- to one ms. This is fixed by making ast_bridge_call use the same
- time as ast_channel_bridge for the timeout calculation. ........
-
-2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
- versioning, since Asterisk has it's own
-
-2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make rfc2833compensate a global option.
-
-2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Backport revision 43754 from the trunk,
- which removes an unused buffer from mm_login to close bug 8038,
- as well as addresses some formatting and coding guidelines issues
- in passing. Originally, I did not commit this to 1.4 since it is
- not necessarily fixing a bug. However, since the IMAP storage
- code is brand new, I decided it would be better to make the
- change here as well, in case someone has to work on this code to
- address issues in the very near future. I don't want to make
- unnecessary merge problems going to the trunk.
-
-2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com>
-
- * configs/extensions.ael.sample: This change to extensions.ael was
- to fix bug 8031; the install scripts are causing it to be copied
- to /etc/asterisk/extensions.ael, and because it is a fairly
- direct conversion of the original extensions.conf, the macro and
- context names clash with the existing extensions.conf. So, I put
- an ael- in front of all macros and contexts, and checked every
- goto and macro call. Also, this file compiles under aelparse.
-
-2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Back in revision 4798, this message was changed from
- using ast_cli() to directly calling write(). During this change,
- checking if this was a remote console was removed. This caused
- this message about using "exit" or "quit" to exit an Asterisk
- console to come up in times where it did not make sense. This
- change restores the check to see if this is a remote console
- before printing the message. (fixes BE-65)
-
-2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com>
-
- * .cleancount, main/cli.c, channels/chan_sip.c,
- include/asterisk/channel.h: Use proper type to represent the group variable
- (issue #8025 reported by makoto)
-
-2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Add missing newline character in the warning
- message about deprecated TOS values in configuration.
-
- * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
- mailbox definitions, don't introduce a length limit on the
- definition by using a 256 byte temporary storage buffer. Instead,
- make the temporary buffer just as big as it needs to be to hold
- the entire mailbox definition. (fixes BE-68)
-
-2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c: Strip options off the argument passed for
- devicestate in chan_local. (issue #8034 reported by pcardozo)
-
- * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
- overhaul of the whisper support. 1. We need to duplicate the
- frame from ast_translate 2. We need to ensure we always have
- signed linear coming in for signed linear combining. 3. We need
- to ensure we are always feeding signed linear out. 4. Properly
- store and restore write format when beeping on the channel we are
- whispering on. 5. Properly discontinue the stream on the channel
- for the beep. (issue #8019 reported by timkelly1980)
-
-2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: update to use 1.4.3 core sounds, with corrected
- beep/beeperr/tt-monkeys files
-
-2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com>
-
- * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
- Dan Austin. Maximum values were incorrect, which is why this is
- being put in 1.4
-
- * channels/chan_skinny.c: Add proper codec support to chan_skinny.
- Works with at least ulaw, alaw, and g729a. This is technically a
- "new feature", but there are justifications for it. I found a bug
- with the recent rtp packetization changes, which caused the media
- setup to fail under certain circumstances, particularly when
- using allow=all, or having no allow= statements (globally or on
- the device). I could have either removed the rtp packetization
- features, or I could add proper codec support (which, without, I
- think most people would consider to be a bug anyways).
-
-2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Should have moved these lines up in the
- merge, instead of removing them
-
- * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
- delete=yes was ignored 2) maxmessages was ignored
-
-2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
- channels/h323/cisco-h225.asn: Fix ASN1 description of
- non-standard Cisco extensions
-
- * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
- changes of trunk: 1) r43540: Avoid possible deadlock on channel
- destruction 2) r43590: Disable fastStart if requested by remote
- side
-
-2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com>
-
- * sounds/Makefile: One more fix for sounds installation - this time
- for portability. Reported to asterisk-dev mailing list.
-
-2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com>
-
- * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
- crashing if trying to play an OGG moh file.
-
-2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
- channels/chan_h323.c: Merged revisions 43472,43495 from trunk
-
-2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com>
-
- * channels/iax2-provision.c: Fix a CLI command registration issue
- where an erroneous message claiming that "iax2 show provisioning"
- was already registered. This was because this command was
- registering itself as both the command, as well as the command it
- is deprecating. (issue #8022, reported by bjweeks, fixed by
- myself)
-
- * channels/chan_iax2.c:Check to see if the channel that is activating the
- IAXPEER function is actually an IAX2 channel before proceeding to
- process it to avoid crashing. (issue #8017, reported by admott,
- fixed by myself)
-
-2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: don't output the 'build complete' message when the
- target being run is already going to do an installation
-
-2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
- properly. Remove reload support, since it doesn't
- actually...work.
-
-2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This commits a change to return
- MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
- goes well for bug 8004
-
- * pbx/pbx_ael.c: If the extensions.ael file not found, or
- unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.
-
-2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Make sure we explicitly set the CLI command to not be
- deprecated, if it isn't.
-
-2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: use rebuilt extra sounds
-
- * main/channel.c: all the Linux systems I have don't use
- '__m_count' for this field, so I don't know where this came
- from...
-
-2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com>
-
- * include/asterisk/threadstorage.h: backport the compatability fix
- to use attribute_malloc instaed of __attribute__ ((malloc))
-
- * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
- could not be configured (issue #8006, Mithraen)
-
- * main/frame.c: Suppress a compiler warning about the use of a
- potentially uninitialized variable. It couldn't actually happen,
- though.
-
-2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: First shot at unload_module in
- chan_skinny.. More to come.
-
-2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com>
-
- * include/asterisk/jabber.h, channels/chan_gtalk.c,
- res/res_jabber.c: updates for better compontent support
-
-2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
- actually documented how the new features in res_odbc actually
- work. (Oops)
-
-2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_oss.c: Some more clean up in the load function for
- chan_oss (issue #8002 reported by Mithraen with minor mods by
- moi)
-
- * channels/chan_mgcp.c: Clean up chan_mgcp's module load function
- (issue #8001 reported by Mithraen with mods by moi)
-
-2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/Makefile, build_tools/strip_nonapi (added): add another
- attempt to strip non-API symbols from the final binary... script
- will need to be extended to work on non-Linux systems
-
-2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_url.c: Fix documentation to reflect how Url() really
- works
-
- * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates
-
-2006-09-21 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0-beta2 released.
-
-2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/Makefile: remove this change... it requires binutils 2.17
-
-2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com>
-
- * build_tools/make_version: fix minor typo in the way version is
- handled
-
-2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0-beta1 released.