[svn-commits] lmadsen: tag 1.4.36-rc1 r283274 - /tags/1.4.36-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 23 13:24:08 CDT 2010
Author: lmadsen
Date: Mon Aug 23 13:24:03 2010
New Revision: 283274
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283274
Log:
Importing files for 1.4.36-rc1 release.
Added:
tags/1.4.36-rc1/.lastclean (with props)
tags/1.4.36-rc1/.version (with props)
tags/1.4.36-rc1/ChangeLog (with props)
Added: tags/1.4.36-rc1/.lastclean
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--- tags/1.4.36-rc1/ChangeLog (added)
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+2010-08-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.36-rc1 Released.
+
+2010-08-20 16:46 +0000 [r283048-283123] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. ..........
+
+ * channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending
+ ALERTING is a protocol error The PRI layer in chan_dadhi will
+ check if a PROGRESS message has already been sent, and not allow
+ sending another (although that is technically allowed by the Q931
+ spec), however it does not protect against sending an ALERTING
+ and then sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+
+2010-08-19 21:03 +0000 [r282893] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: tos_sip option was not being set correctly
+ When tos_sip is used, the tos of the sip socket is only set
+ correctly if the socket binding changes on a reload. If the
+ binding stays the same but the TOS changes, the new tos value
+ would not take into effect. This patch fixes that. (closes issue
+ #17712) Reported by: nickb
+
+2010-08-19 02:12 +0000 [r282729] Terry Wilson <twilson at digium.com>
+
+ * configs/sip.conf.sample: Add some documentation about codec
+ negotiation to sip.conf
+
+2010-08-16 17:06 +0000 [r282430] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+
+2010-08-12 22:49 +0000 [r282129] Jason Parker <jparker at digium.com>
+
+ * pbx/pbx_config.c: Register CLI commands before parsing config, in
+ case there is a config error.
+
+2010-08-12 03:00 +0000 [r281911] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: Ensure SSRC is changed when media source is
+ changed to resolve audio delay. This change causes the SSRC to
+ change right before the channels are bridged, which is what used
+ to happen. It seems that fixes were made to attempt limiting SSRC
+ changes, targeted mainly at sending DTMF. DTMF is not affecting
+ the SSRC with this change. There are two other control frames
+ sent in ast_channel_bridge that probably should also be changed
+ to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this
+ change up to the discretion of resolving issue #17007. For
+ reference - old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff
+
+2010-08-11 18:28 +0000 [r281762-281819] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample: Add Danish support to say.conf.sample
+ (closes issue #17836) Reported by: RoadKill Patches:
+ say.conf.sample.patch.dk uploaded by RoadKill (license 933)
+
+ * configs/say.conf.sample: Allow say.conf to handle large numbers
+ ending with multiple zeros. (closes issue #17833) Reported by:
+ RoadKill Patches: say.conf.sample.patch.largenumbers uploaded by
+ RoadKill (license 933)
+
+2010-08-10 17:45 +0000 [r281566] Russell Bryant <russell at digium.com>
+
+ * apps/app_dial.c: Reset visible indication after answer. (closes
+ issue #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts
+
+2010-08-09 20:04 +0000 [r281390] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Prevent loss of Caller ID information set
+ on local channel after masquerade. Caller ID set on the channel
+ before a masquerade occurs when using a local channel would cause
+ the information to be lost. The problem was that the information
+ was set on a channel destined to be hung up. The somewhat
+ confusing fix is to detect if any Caller ID has been set on the
+ channel and if so preswap the Caller ID data so that basically
+ the masquerade puts the data back. (closes issue #17138) Reported
+ by: kobaz Review: https://reviewboard.asterisk.org/r/847/
+
+2010-08-06 21:34 +0000 [r281185] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: chan_sip: fixes provisional keepalive
+ scheduled item crash There is a scheduler item in chan_sip that
+ keeps sending the last provisional message in response to an
+ INVITE Request for a period of time until a final response to
+ that INVITE is sent. Because of the way this scheduler item
+ works, it requires a reference to a sip_pvt pointer to work
+ properly. The problem with this is that it is currently possible
+ (but rare) for the sip_pvt to get destroyed and that scheduler
+ item to still exist. When this occurs, the scheduler event fires
+ and attempts to access a freed sip_pvt which causes a crash.
+ (closes issue #17497) Reported by: anonymouz666 Patches:
+ keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671)
+ Review: https://reviewboard.asterisk.org/r/849/
+
+2010-08-05 07:28 +0000 [r280982] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo
+
+2010-08-04 18:54 +0000 [r280944] Russell Bryant <russell at digium.com>
+
+ * contrib/scripts/astcli (added): Copy astcli back to 1.4 so it's
+ available for automated testing purposes.
+
+2010-08-03 20:49 +0000 [r280811] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_callerid.c, channels/chan_dahdi.c: Prevent DAHDI
+ channels from overriding the callerid, once it's been set by the
+ user. (closes issue #16661) Reported by: jstapleton Patches:
+ 20100414__issue16661.diff.txt uploaded by tilghman (license 14)
+ 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman
+ (license 14) Tested by: jstapleton
+
+2010-07-29 19:04 +0000 [r280448] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: fixes issue with translator frame not getting
+ freed A translator frame even if it local storage so the
+ translation path can be freed. This issue prevented g729 licenses
+ from being freed up. (closes issue #17630) Reported by: manvirr
+ Patches: encoder_fix.diff uploaded by dvossel (license 671)
+ Tested by: manvirr, dvossel
+
+2010-07-29 15:52 +0000 [r280341] Jean Galarneau <jgalarneau at digium.com>
+
+ * apps/app_meetme.c: Fix a dsp structure leak occuring when a local
+ channel is put into a meetme conference, then masquaraded away.
+ ABE-2422
+
+2010-07-28 13:50 +0000 [r280088] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/live_ast: Update help text to be less confusing.
+
+2010-07-27 20:33 +0000 [r279945] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, include/asterisk/audiohook.h, main/audiohook.c:
+ remove empty audiohook write list on channel If a channel has an
+ audiohook write list created on it, that list stays on the
+ channel until the channel is destroyed. There is no reason to
+ keep that list on the channel if it becomes empty. If it is empty
+ that just means we are doing needless translating for every
+ ast_read and ast_write. This patch removes the audiohook list
+ from the channel once it is detected to be empty on either a read
+ or write. If a audiohook is added back to the channel after this
+ list is destroyed, the list just gets recreated as if it never
+ existed to begin with. (closes issue #17630) Reported by: manvirr
+ Review: https://reviewboard.asterisk.org/r/799/
+
+2010-07-24 23:57 +0000 [r279346] Bradley Latus <brad.latus at gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-24 23:27 +0000 [r279344] Jeff Peeler <jpeeler at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Provide a default value for DAHDI_TRANSCODE so when DAHDI is not
+ installed menuselect doesn't get confused: Unknown value '' found
+ in build_tools/menuselect-deps for DAHDI_TRANSCODE
+
+2010-07-23 21:56 +0000 [r279206] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, apps/app_queue.c: SIP promiscuous redirect could
+ fail to dial the redirect. The ast_channel was created with one
+ variable to ast_request() but the call to ast_call() that
+ initiates the outgoing call was using a different variable. The
+ two variables are not equivalent if the call_forward string
+ included a channel technology specifier. e.g., SIP/200
+
+2010-07-23 18:04 +0000 [r279053] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Backport fixes for sip_uri_params_cmp() from
+ trunk.
+
+2010-07-23 17:04 +0000 [r278981-278984] Tilghman Lesher <tlesher at digium.com>
+
+ * autoconf/ast_check_pwlib.m4, configure, configure.ac: Establish a
+ maximum version for openh323 (i.e. not opal), because chan_h323
+ will fail to load, even if it links. (issue #17679) Reported by:
+ am
+
+ * main/asterisk.c: Avoid race with consolethread on shutdown (on
+ parallel processors). (closes issue #17080) Reported by:
+ sybasesql Patches: 20100721__issue17080.diff.txt uploaded by
+ tilghman (license 14) Tested by: sybasesql
+
+2010-07-22 19:31 +0000 [r278701] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DNID does not get cleard on a new call
+ when using immediate=yes with ISDN signaling. When you are using
+ chan_dahdi ISDN signaling with immediate=yes and a call comes in
+ without a DNID then you get the DNID of a previous call.
+ Chan_dahdi does not touch the DNID field on a new call if it does
+ not have a DNID. Made always copy the DNID from the new call. The
+ patches backport the relevant changes from trunk -r210387.
+ (closes issue #17568) Reported by: wuwu Patches:
+ issue17568_v1.4.patch uploaded by rmudgett (license 664)
+ issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
+
+2010-08-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.35 Released.
+
+2010-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.35-rc1 Released.
+
+2010-07-22 14:55 +0000 [r278618] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Allow PLC to function properly when channels use
+ SLIN for audio. If a channel involved in a bridge was using SLIN
+ audio, then translation paths were not guaranteed to be set up
+ properly since in all likelihood the number of translation steps
+ was only 1. This patch enforces the transcode_via_slin behavior
+ if transcode_via_slin or generic_plc is enabled and one of the
+ formats to make compatible is SLIN. AST-352
+
+2010-07-20 22:23 +0000 [r278023-278261] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Delete IMAP messages in reverse order, to
+ ensure reordering after each expunge does not cause deletion of
+ the wrong message. (closes issue #16350) Reported by: noahisaac
+ Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+ (license 14)
+
+ * main/autoservice.c, res/res_features.c,
+ include/asterisk/channel.h: Do not queue up DTMF frames while a
+ call is on hold. (Fixes ABE-2110)
+
+ * main/manager.c: Off-by-one error (closes issue #16506) Reported
+ by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14)
+
+2010-07-19 20:56 +0000 [r277944] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/
+
+2010-07-19 20:16 +0000 [r277906] Jean Galarneau <jgalarneau at digium.com>
+
+ * res/res_features.c: Avoid trying to pickup a parked extension
+ before the park operation is completed. A crash could occur if
+ the extension is picked up while the parking extension is being
+ announced. Testing pu->notquiteyet while searching for a parked
+ extension resolves this crash. (ABE-2418)
+
+2010-07-17 16:59 +0000 [r277738] Tilghman Lesher <tlesher at digium.com>
+
+ * autoconf/ast_func_fork.m4, configure: Remove uclibc cross-compile
+ triplet, as uclibc has a working fork()... it's only uclinux that
+ does not. (closes issue #17616) Reported by: pprindeville
+
+2010-07-16 22:43 +0000 [r277625] Tim Ringenbach <tim.ringenbach at gmail.com>
+
+ * res/res_features.c: Save and restore AST_FLAG_BRIDGE_HANGUP_DONT
+ on attended transfer. ast_bridge_call() clears
+ AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+ ast_bridge_call() is called for a second bridge on the same
+ channel, and it clears that flag, which still needs to get set
+ for when the original ast_bridge_call() gets control back and
+ checks it. Review: https://reviewboard.asterisk.org/r/741
+
+2010-07-16 21:54 +0000 [r277568] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c: Since we split
+ values at the semicolon, we should store values with a semicolon
+ as an encoded value. (closes issue #17369) Reported by: gkservice
+ Patches: 20100625__issue17369.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman
+
+2010-07-16 21:18 +0000 [r277497] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Default to no udptl error correction so that
+ error correction will be disabled in the event that the remote
+ end indicates that they do not support the error correction mode
+ we requested. FAX-128
+
+2010-07-16 20:18 +0000 [r277419] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: priexclusive in chan_dahdi.conf ignored
+ when reloading dahdi module During a reload, the priexclusive and
+ outsignalling parameters are not read in from the config file as
+ intended. Unfortunately, they get set to defaults as a result.
+ This patch makes sure that they do not get set to defaults during
+ a reload. (closes issue #17441) Reported by: mtryfoss Patches:
+ issue17441_v1.4.patch uploaded by rmudgett (license 664)
+ issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+ issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett
+
+2010-07-16 18:30 +0000 [r277327] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Interpret device state AST_DEVICE_UNKNOWN as
+ extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+ Reported by: francesco_r Patches: pbx.c.patch uploaded by
+ viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+
+2010-07-16 18:04 +0000 [r277261] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c: If variable gotten is not set, will segfault on
+ Solaris. (closes issue #17636) Reported by: bklang
+
+2010-07-16 17:29 +0000 [r277247] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: For pass through DTMF tones, measure the actual
+ duration between the begin and end packets on the wire. If it is
+ detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+ emulation. AST-362
+
+2010-07-16 17:10 +0000 [r277182] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_amd.c: Total analysis time error with SIP and silence
+ suppression When using app_amd with SIP providers that have
+ silence suppression on, the iTotalTime count increases
+ exponentially. (closes issue #17656) Reported by: juls
+
+2010-07-15 13:48 +0000 [r276652] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: In a perfect world, the frame source would never
+ be NULL. In the meantime, don't crash when it is.
+
+2010-07-14 11:49 +0000 [r276267] Leif Madsen <lmadsen at digium.com>
+
+ * configs/voicemail.conf.sample: Update documentation for
+ voicemail.conf externpass option.
+
+2010-07-13 19:14 +0000 [r275994-276126] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: Only reset a CDR that exists.
+
+ * res/res_features.c: Use chan->cdr instead of chan_cdr (just like
+ peer->cdr instead of peer_cdr in the last commit).
+
+ * res/res_features.c: Access peer->cdr directly instead of through
+ a saved off reference. At this point in the code, it is possible
+ that peer_cdr may be invalid. Specifically, in the blind transfer
+ code, CDRs are swapped between channels. So, peer_cdr is no
+ longer == peer->cdr. The scenario that exposed a crash in this
+ code was a blind transfer that hit the system call limit, causing
+ the transferee channel to get destroyed after the transfer
+ attempt failed. Even if it succeeds and this code doesn't crash,
+ this code was still trying to reset a CDR on a channel that was
+ now owned by a different thread, which is a BadThing(tm).
+ (ABE-2417)
+
+2010-07-13 14:47 +0000 [r275909] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql,
+ contrib/scripts/realtime_pgsql.sql (removed),
+ contrib/scripts/vmdb.sql (removed),
+ contrib/scripts/iax-friends.sql (removed),
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql, contrib/scripts/meetme.sql
+ (removed), contrib/realtime (added), contrib/realtime/postgresql,
+ contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
+ contrib/realtime/oracle, contrib/realtime/sqlserver,
+ contrib/scripts/sip-friends.sql (removed): Move SQL scripts into
+ their own database-specific directories.
+
+2010-07-12 20:34 +0000 [r275665-275773] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: Make user removals and traversals thread safe
+ in meetme. Race conditions present in meetme involving the user
+ list where a lack of locking has the potential for a user to be
+ removed during a traversal or as in the case of the reporter
+ after checking if the list is empty could cause a crash. Fixing
+ this was done by convering the userlist to an ao2 container.
+ (closes issue #17390) Reported by: Vince Review:
+ https://reviewboard.asterisk.org/r/746/
+
+ * main/channel.c: Change ast_write to not stop generator when
+ called from ast_prod. For SIP channels configured with the
+ progressinband option on, the ringback was being immediately
+ stopped. This problem was due to ast_prod being moved for a
+ deadlock fix in 259858. Prodding the channel after setting up the
+ generator triggered the check in ast_write to stop the generator.
+ The fix here should write the frame the same as was done before
+ the call to ast_prod was moved. (closes issue #17372) Reported
+ by: tech_admin
+
+2010-07-09 19:28 +0000 [r275241-275290] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * main/cli.c: fix tab-completion for unload command. (closes issue
+ #17536) Reported by: junky Patches: unload_vs_mod_unload.diff
+ uploaded by junky (license 177) Tested by: pabelanger
+
+ * channels/chan_sip.c: Fix logging message for stale nonce. (closes
+ issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+ uploaded by kenner (license 1040) Tested by: lmadsen
+
+2010-07-09 18:23 +0000 [r275027-275182] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/loader.c: give a better error message when attempting to
+ unload a module that is not loaded
+
+ * main/loader.c: don't unload modules that returned
+ AST_MODULE_LOAD_DECLINE when they were loaded
+
+ * apps/app_dial.c: Clear the AST_CDR_FLAG_DIALED flag for channels
+ going into the pbx via the G option in app_dial (closes issue
+ #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: jamicque,
+ mnicholson
+
+2010-07-09 15:33 +0000 [r275021] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/test.h, main/test.c: Document that a leading and
+ trailing slash is expected for test categories. Also, emit a
+ warning if a test is registered without one of these.
+
+2010-07-07 18:12 +0000 [r274579] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Close the DAHDI FD on error when
+ processing chan_dahdi toneduration config parameter.
+
+2010-07-07 06:13 +0000 [r274417] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/say.conf.sample: Correct how 100, 200, 300, etc. is said.
+ Also add the crazy British numbers. (closes issue #16102)
+ Reported by: Delvar Patches: say.conf.fix.patch uploaded by
+ Delvar (license 908) (plus a few additional fixes and
+ simplifications by me)
+
+2010-07-06 22:46 +0000 [r274283-274359] Jeff Peeler <jpeeler at digium.com>
+
+ * main/Makefile: Ensure file.o is built correctly. (related to
+ issue #15250)
+
+ * configs/sip.conf.sample: Correct sip.conf.sample comments for
+ prematuremedia option. (closes issue #17513) Reported by: festr
+ Patches: patch uploaded by festr (license 443)
+
+2010-07-06 22:08 +0000 [r274280] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add option to not
+ do a call forward on 482 Loop Detected Asterisk has always set up
+ a forwarded call when receiving a 482 Loop Detected. This
+ prevents handling the call failure by just continuing on in the
+ dialplan. Since this would be a change in behavior, the new
+ option to disable this behavior is forwardloopdetected which
+ defaults to 'yes'. Review:
+ https://reviewboard.asterisk.org/r/764/
+
+2010-07-06 14:29 +0000 [r274157] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c: Fix problem with RFC 2833 DTMF not being accepted. A
+ recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order. The
+ problem was that the check did not account for the fact that the
+ seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+ (closes issue #17571) Reported by: mdeneen Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+2010-07-06 13:52 +0000 [r274093] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Make get_member_status return QUEUE_NO_MEMBERS
+ instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work
+ again. This regression was introduced in 273639. Also fixed
+ whitespace.
+
+2010-07-05 19:48 +0000 [r273981] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, channels/chan_iax2.c: Command 'stop
+ gracefully' doesn't.
+
+2010-07-05 13:51 +0000 [r273884] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * main/config.c: Remove extra line breaks from 'core show config
+ mappings' (closes issue #17583) Reported by: pabelanger Patches:
+ issue17583.patch uploaded by pabelanger (license 224) Tested by:
+ lmadsen
+
+2010-07-02 21:36 +0000 [r273717-273793] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_local.c, configure,
+ include/asterisk/autoconfig.h.in, channels/chan_agent.c,
+ configure.ac, channels/chan_h323.c, include/asterisk/lock.h,
+ include/asterisk/compiler.h: Have the DEADLOCK_AVOIDANCE macro
+ warn when an unlock fails, to help catch potentially large
+ software bugs. (closes issue #17407) Reported by: pdf Patches:
+ 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/751/
+
+ * main/autoservice.c: Autoservice loop optimization causes a busy
+ loop, when channels are serviced while in hangup. (closes issue
+ #17564) Reported by: ramonpeek Patches:
+ 20100630__issue17564.diff.txt uploaded by tilghman (license 14)
+ Tested by: ramonpeek
+
+2010-07-02 15:54 +0000 [r273640] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * apps/app_voicemail.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, channels/chan_sip.c, res/res_agi.c,
+ res/res_jabber.c: Fix various typos, reported by Lintian
+
+2010-07-02 15:46 +0000 [r273639] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: If all members are paused, the wrong status is
+ indicated. (closes issue #17576) Reported by: ramonpeek Patches:
+ diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek
+
+2010-07-01 22:09 +0000 [r273565] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Don't return a partially initialized datastore.
+ If memory allocation fails in ast_strdup(), don't return a
+ partially initialized datastore. Bad things may happen. (related
+ to ABE-2415)
+
+2010-07-01 20:19 +0000 [r273354-273474] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: Allow admin user to join conference without
+ using admin mode and no user pin. Configuring the conference in
+ meetme.conf like the following: conf => 2345,,6666 did not prompt
+ for pin when used without admin mode. This meant that the
+ conference could not be joined as an admin even if the user knew
+ the correct pin. The original bug report was submitted claiming
+ that the blank user pin should deny entry into the conference. I
+ think a better way to handle this would be with a feature
+ enhancement that used the following syntax: conf => 2345,X,6666 -
+ where X denotes no acceptable pin allowed (closes issue #15704)
+ Reported by: modelnine
+
+ * apps/app_meetme.c: Ensure channel placed in meetme in ringing
+ state is properly hung up. An outgoing channel placed in meetme
+ while still ringing which was then hung up would not exit meetme
+ and the channel was not properly destroyed. Specifically checking
+ for this scenario by looking at the appropriate control frames
+ resolves the issue. (closes issue #15871) Reported by: Ivan
+ Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+ (license 229)
+
+2010-06-29 23:15 +0000 [r273057-273060] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Allow the "useragent" value to be restored
+ into memory from the realtime backend. This value is purely
+ informational. It does not alter configuration at all. (closes
+ issue #16029) Reported by: Guggemand Patches:
+ realtime-useragent.patch uploaded by Guggemand (license 897)
+ Tested by: Guggemand
+
+ * main/channel.c: _Really_ skip the channel... don't just retry for
+ another 200 cycles. (Closes issue SWP-1652, ABE-2240)
+
+2010-06-29 21:36 +0000 [r273017] Russell Bryant <russell at digium.com>
+
+ * /: Remove properties that were erroneously merged to 1.4 from one
+ of my branches.
+
+2010-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34 Released.
+
+2010-07-07 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34-rc2 Released.
+
+ * Fix problem with RFC 2833 DTMF not being accepted.
+
+ A recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order.
+ The problem was that the check did not account for the fact that
+ the seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+
+ (closes issue 0017571)
+ Reported by: mdeneen
+ Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
+ via the G option in app_dial
+
+ (closes issue 0017592)
+ Reported by: jamicque
+ Patches:
+ G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: jamicque, mnicholson
+
+2010-06-29 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34-rc1 Released.
+
+2010-06-28 21:50 +0000 [r272921-272925] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth
+
+ * main/config.c: Also trim trailing blanks on #includes
+
+ * main/config.c: Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan
+
+2010-06-28 18:47 +0000 [r272878-272881] Russell Bryant <russell at digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2.
+
+ * main/asterisk.c, Makefile, include/asterisk/test.h (added),
+ build_tools/cflags-devmode.xml, include/asterisk.h,
+ tests/Makefile, tests/test_skel.c, /, main/Makefile, tests
+ (added), include/asterisk/linkedlists.h, main/test.c (added):
+ Backport unit test API to 1.4. Review:
+ https://reviewboard.asterisk.org/r/750/
+
+2010-06-28 17:31 +0000 [r272804] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Decode URI in contact header of 302
+ response. ABE-2352
+
+2010-06-28 17:11 +0000 [r272688-272763] Russell Bryant <russell at digium.com>
+
+ * Makefile: Force SILENTMAKE where it is needed.
+
+ * Makefile: Backport method of setting SUBMAKE from trunk. By
+ setting the PRINT_DIR variable, SUBMAKE will print out the
+ directories it descends into, which is important for editors
+ (like vim) that watch the build output so that they can take you
+ to the file where an error occurred.
+
+2010-06-25 20:17 +0000 [r272562] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Make the structure of the
+ table specified before match the queries and results. (closes
+ issue #17557) Reported by: cmaj
+
+2010-06-24 21:58 +0000 [r272446] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: ss_thread calls pri_grab without lock
+ during overlap dial Recent changes to chan_dahdi with relation to
+ overlap dialing call pri_grab without first obtaining a lock.
+ (closes issue #17414) Reported by: pdf Patches: bug17414.patch
+ uploaded by jpeeler (license 325)
+
+2010-06-23 22:33 +0000 [r272367] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Send AgentComplete manager events in the event
+ of blind and attended transfers. (closes issue #16819) Reported
+ by: elbriga Patches: app_queue.diff uploaded by elbriga (license
+ 482)
+
+2010-06-23 20:57 +0000 [r272255] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_meetme.c: First caller into a dynamic conference now
+ enter pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger
+
+2010-06-23 18:40 +0000 [r272147] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Backport part of revision 136715 to fix
+ callerid in voicemail text files (IMAP only). (closes issue
+ #16945) Reported by: mneuhauser
+
+2010-06-22 17:31 +0000 [r271689-271902] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Decrease the module ref count in sip_hangup
+ when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep
+ the ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain
+
+ * pbx/pbx_dundi.c: Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000
+
+ * configs/sip_notify.conf.sample, channels/chan_sip.c: Modify
+ chan_sip's packet generation api to automatically calculate the
+ Content-Length. This is done by storing packet content in a
+ buffer until it is actually time to send the packet, at which
+ time the size of the packet is calculated. This change was made
+ to ensure that the Content-Length is always correct. (closes
+ issue #17326) Reported by: kenner Tested by: mnicholson, kenner
+ Review: https://reviewboard.asterisk.org/r/693/
+
+2010-06-21 20:37 +0000 [r271552] Jeff Peeler <jpeeler at digium.com>
+
+ * pbx/pbx_ael.c: Do not use sizeof to calculate size of a heap
+ allocated character array. Change left out from 271399. (closes
+ issue #16053) Reported by: diLLec
+
+2010-06-18 20:52 +0000 [r271399-271444] Jeff Peeler <jpeeler at digium.com>
+
+ * pbx/pbx_ael.c: Check for newly added memory allocation failures
+ gracefully during AEL2 parsing.
+
+ * pbx/pbx_ael.c: Fix crash when parsing some heavily nested
+ statements in AEL on reload. Due to the recursion used when
+ compiling AEL in gen_prios, all the stack space was being
+ consumed when parsing some AEL that contained nesting 13 levels
+ deep. Changing a few large buffers to be heap allocated fixed the
+ crash, although I did not test how many more levels can now be
+ safely used. (closes issue #16053) Reported by: diLLec Tested by:
+ jpeeler
+
+2010-06-18 18:54 +0000 [r271339-271340] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: Remove an unnecessary assignment that
+ causes a DEBUG_THREADS build failure on mac os x.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Fix a build problem on Mac OS X with
+ DEBUG_THREADS enabled. This set of changes was already in trunk.
+
+2010-06-18 18:33 +0000 [r271335] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Eliminate deadlock potential in
+ dahdi_fixup(). (This is a backport of 269307, committed to trunk
+ by rmudgett.) Calling dahdi_indicate() when the channel private
+ lock is already held can cause a deadlock if the PRI lock is
+ needed because dahdi_indicate() will also get the channel private
+ lock. The pri_grab() function assumes that the channel private
+ lock is held once to avoid deadlock. (closes issue #17261)
+ Reported by: aragon
+
+2010-06-22 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.33.1 Released.
+
+ * channels/chan_dahdi.c: Merge revision 270404 from the 1.4 branch.
+
+ fixes FXS port still ringing when answered, as reported by Tzafrir
+ on dev-list.
+
+ (issue #17067)
+ Reported by: tzafrir
+ Tested by: alecdavis
+
+2010-06-17 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33 Released.
+
+2010-06-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33-rc2 Released.
+
+2010-06-10 Tilghman Lesher <tlesher at digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+
+2010-06-09 Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33-rc1 Released.
+
+2010-06-01 15:17 +0000 [r266585] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Prevent CLI prompt from distorting output of
+ lines shorter than the prompt. Uses the VT100 method of clearing
+ the line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig
+
+2010-06-01 14:57 +0000 [r266579-266580] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Fix formatting issue with previous patch.
+
+ * channels/chan_sip.c: Missing fallback to audio fax feature when
+ T.38 re-INVITE failed When a T.38 re-INVITE failed with an 488 or
+ 606 answer, we should fallback to audio fax by send a
+ re-re-INVITE without T.38. The function is backported from 1.6
+ asterisk. (closes issue #16795) Reported by: vrban (closes issue
[... 28846 lines stripped ...]
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