[asterisk-commits] russell: tag 1.4.1 r57561 - in /tags/1.4.1:
.lastclean .version ChangeLog
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Mar 2 16:16:11 MST 2007
Author: russell
Date: Fri Mar 2 17:16:10 2007
New Revision: 57561
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57561
Log:
importing files for 1.4.1 release
Added:
tags/1.4.1/.lastclean (with props)
tags/1.4.1/.version (with props)
tags/1.4.1/ChangeLog (with props)
Added: tags/1.4.1/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.1/.lastclean?view=auto&rev=57561
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--- tags/1.4.1/ChangeLog (added)
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+2006-03-02 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.1 released.
+
+2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at transnexus.com>
+
+ * doc/osp.txt: Update OSP documentation for v1.4.
+
+2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Move message from verbose to debug
+
+2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf at digium.com>
+
+ * sounds/Makefile: updated the sound tarball versions in Makefile
+
+2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * /: Block patch only needed in 1.2
+
+2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin at 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 at 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 at 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 at edvina.net>
+
+ * /: Blocking patch for 1.2 only
+
+2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * main/acl.c: Don't output debug unless we asked for it
+
+2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Formatting
+
+2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp at 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 at edvina.net>
+
+ * channels/chan_sip.c: Add some comments on queue system behaviour
+ and how it affects the SIP channel
+
[... 4349 lines stripped ...]
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