[asterisk-commits] lmadsen: tag 1.4.37-rc1 r287631 - /tags/1.4.37-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 20 13:29:08 CDT 2010
Author: lmadsen
Date: Mon Sep 20 13:29:04 2010
New Revision: 287631
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=287631
Log:
Importing files for 1.4.37-rc1 release.
Added:
tags/1.4.37-rc1/.lastclean (with props)
tags/1.4.37-rc1/.version (with props)
tags/1.4.37-rc1/ChangeLog (with props)
Added: tags/1.4.37-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.37-rc1/.lastclean?view=auto&rev=287631
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--- tags/1.4.37-rc1/ChangeLog (added)
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+2010-09-20 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.37-rc1 Released.
+
+2010-09-20 15:48 +0000 [r287555] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113
+
+2010-09-19 15:56 +0000 [r287469] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Make sure we always free variables properly in
+ manager originate. (closes issue #17891) reported, solved and
+ tested by oej Review: https://reviewboard.asterisk.org/r/869/
+
+2010-09-17 21:06 +0000 [r287386] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: Blank columns should get set on reload, not
+ ignored. (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+
+2010-09-17 13:34 +0000 [r287307] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113
+
+2010-09-16 22:12 +0000 [r287197] Jason Parker <jparker at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Add LSB headers for Debian
+ init script, since Debian will complain if it isn't there.
+ Headers were taken from trunk. (closes issue #17958) Reported by:
+ javyer
+
+2010-09-16 20:04 +0000 [r287114-287118] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113
+
+ * main/cdr.c: Don't stop printing cdr variables if we encounter one
+ with a blank name or value. (closes issue #17900) Reported by:
+ under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2010-09-15 20:20 +0000 [r286941-286956] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: whitespace fix
+
+ * apps/app_voicemail.c: Ensure mailbox is not filled to capacity
+ before doing message forwarding. Specifically, before prompting
+ to record a prepended message the capacity is checked first. If
+ the mailbox is full the extension will be reprompted. ABE-2517
+
+2010-09-14 19:26 +0000 [r286679-286756] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson
+
+ * main/channel.c: Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342)
+
+2010-09-13 15:12 +0000 [r286381] Jason Parker <jparker at digium.com>
+
+ * tests: Add stuff to svn:ignore for tests/ directory. (closes
+ issue #17983) Reported by: oej
+
+2010-09-11 16:59 +0000 [r286267] Olle Johansson <oej at edvina.net>
+
+ * main/file.c: Handle error response when we can't make file
+ compatible Review: https://reviewboard.asterisk.org/r/911/
+
+2010-09-10 22:54 +0000 [r286222] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Return -1 if chan_local doesn't support an
+ option
+
+2010-09-10 20:35 +0000 [r286114] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c: Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/
+
+2010-09-10 20:33 +0000 [r286113] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: An outgoing call may not get hung up if a
+ pre-connect incoming ISDN call is disconnected. If the ISDN link
+ a pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances.
+
+2010-09-10 20:03 +0000 [r286070] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes sip extension state update DEADLOCK
+ PROBLEM: In chan_sip, and all the other channel drivers, it is
+ common for us to hold the tech_pvt lock while we ask the Asterisk
+ core about an extension and context. Every time we do this the
+ locking order becomes, (1. tech_pvt lock ---> 2. global context
+ lock). In chan_sip when a dialog subscribes to a hint, that
+ locking order is reversed in the extensionstate callback which
+ will occur outside of the channel_driver's monitor loop. So, on
+ an extension state update we have (1. global context lock ---->
+ 2. tech_pvt lock). Typically when we have to do a reversed
+ locking order like this we'd just do some sort of deadlock
+ avoidance to fix the problem... That will not work here. There
+ are more locks involved here than just the context and tech_pvt.
+ Those are the two that are colliding, but it is impossible to
+ give up the context lock because the global hints list lock MUST
+ be held as well and we can not give that lock up during the
+ extensionstate callback traversal... The locking order for the
+ context and hints are (1. global context lock ----> 2. hints list
+ lock). Deadlock avoidance is not an option here. SOLUTION: The
+ solution this patch implements is to queue the extension state
+ updates into a list and send the NOTIFY messages out during the
+ do_monitor pvt traversal. This clears out the problem of having
+ to hold the context lock before the tech_pvt lock entirely.
+ (closes issue #17888) Reported by: zerohalo
+
+2010-09-10 19:25 +0000 [r286059] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, funcs/func_channel.c,
+ include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h: Inherit CHANNEL() writes to both sides
+ of a Local channel Having Local (/n) channels as queue members
+ and setting the language in the extension with
+ Set(CHANNEL(language)=fr) sets the language on the Local/...,2
+ channel. Hold time report playbacks happen on the Local/...,1
+ channel and therefor do not play in the specified language. This
+ patch modifies func_channel_write to call the setoption callback
+ and pass the CHANNEL() write info to the callback. chan_local
+ uses this information to look up the other side of the channel
+ and apply the same changes to it. (closes issue #17673) Reported
+ by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
+
+2010-09-10 18:22 +0000 [r285889-286023] Tilghman Lesher <tlesher at digium.com>
+
+ * main/test.c: Missing newline
+
+ * include/asterisk/select.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ tests/test_poll.c: Fix Mac OS X build. This also fixes a rather
+ grievous calculation error for the offset of ast_fdset, which was
+ masked on Linux and FreeBSD, because these platforms check the
+ first 256 FDs regardless of the bitmask setting (due to backwards
+ compatibility).
+
+2010-09-09 22:34 +0000 [r285817] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * codecs/gsm/Makefile: GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+
+2010-09-09 20:06 +0000 [r285742] Jason Parker <jparker at digium.com>
+
+ * main/channel.c: Transmit silence when reading DTMF in
+ ast_readstring. Otherwise, you could get issues with DTMF
+ timeouts causing hangups. (closes issue #17370) Reported by:
+ makoto Patches: channel-readstring-silence-generator.patch
+ uploaded by makoto (license 38)
+
+2010-09-09 17:20 +0000 [r285638] Brett Bryant <bbryant at digium.com>
+
+ * res/res_musiconhold.c: Fixes an issue with MOH where it doesn't
+ recover cleanly when it can't play a file and would just stop,
+ instead of continuing to find the next playable file in the MOH
+ class. (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/
+
+2010-09-08 22:07 +0000 [r285566] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: In retrans_pkt, do not unlock pvt until the
+ end of the function on a transmit failure.
+
+2010-09-07 20:30 +0000 [r285266-285365] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_config.c: Catch invalid extensions at the parser, instead
+ of making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL
+
+ * main/poll.c: Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+
+2010-09-07 17:45 +0000 [r285194] Brett Bryant <bbryant at digium.com>
+
+ * apps/app_voicemail.c: Fixes voicemail.conf issues where mailboxes
+ with passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash]
+
+2010-09-06 06:54 +0000 [r285088] Tilghman Lesher <tlesher at digium.com>
+
+ * BSDmakefile (added), makeopts.in: Silly convenience script for
+ BSD platforms.
+
+2010-09-03 16:10 +0000 [r284881] Terry Wilson <twilson at digium.com>
+
+ * apps/app_chanspy.c: Properly detect when a sound file doesn't
+ exist ast_fileexists returns -1 for error and 0 for a
+ non-existant file. The existing code treated missing files as
+ though they were existed.
+
+2010-09-02 20:25 +0000 [r284777] Brett Bryant <bbryant at digium.com>
+
+ * main/manager.c: Fixes a bug in manager.c where the default
+ configuration values weren't reset when the manager configuration
+ was reloaded. (closes issue #17917) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/883/
+
+2010-09-02 16:47 +0000 [r284703] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Removed relatedpeer code from
+ sip_autodestruct Handling of the relatedpeer structure associated
+ with a sip_pvt should be done during the final sip_destruction
+ function, not in sip_autodestruct.
+
+2010-09-01 18:49 +0000 [r284393-284478] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, main/asterisk.c, main/poll.c,
+ include/asterisk/select.h (added), channels/chan_phone.c,
+ channels/chan_misdn.c, configure,
+ include/asterisk/autoconfig.h.in, res/res_features.c,
+ configure.ac, channels/chan_alsa.c,
+ include/asterisk/poll-compat.h, include/asterisk/channel.h,
+ tests/test_poll.c (added): Ensure that all areas that previously
+ used select(2) now use poll(2), with implementations that need
+ poll(2) implemented with select(2) safe against 1024-bit
+ overflows. This is a followup to the fix for the pthread timer in
+ 1.6.2 and beyond, fixing a potential crash bug in all supported
+ releases. (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/
+
+ * channels/chan_sip.c: Don't send a devstate change on
+ poke_noanswer if the state did not change. (closes issue #17741)
+ Reported by: schmidts Patches: chan_sip.c.patch uploaded by
+ schmidts (license 1077)
+
+2010-08-31 18:57 +0000 [r284316] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample: Update say.conf.sample to match the
+ rules in say.c (closes issue #17835) Reported by: RoadKill
+ Patches: say.conf.sample.patch.rules uploaded by RoadKill
+ (license 933) Tested by: RoadKill
+
+2010-08-27 22:17 +0000 [r283960] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Parse all "Accept" headers for SIP SUBSCRIBE
+ requests. (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671)
+
+2010-08-27 20:29 +0000 [r283880] Jason Parker <jparker at digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c: Fix issue with
+ decoding ^-escaped characters in realtime. (closes issue #17790)
+ Reported by: denzs Patches: 17790-chunky.diff uploaded by qwell
+ (license 4) Tested by: qwell, denzs
+
+2010-08-27 15:11 +0000 [r283834] Terry Wilson <twilson at digium.com>
+
+ * main/config.c: Use ast_free since ast_variable_new uses
+ ast_calloc
+
+2010-08-26 15:22 +0000 [r283690] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixed how Asterisk destroys a dialog on
+ channel hangup before invite receives a response. If an
+ ast_channel with a SIP tech pvt hangs up before the sip dialog
+ gets a response to its outgoing INVITE, Asterisk used to
+ pretend_ack the INVITE. This is not rfc compliant and results in
+ confusion at the other endpoint. sip_pretend_ack will ack and
+ remove all the packets in the retransmit queue. This means that
+ the INVITE will stop retransmitting, and that any response to
+ that INVITE that comes after the pretend_ack occurs will be
+ ignored. Instead of faking any sort of acknowledgement for an
+ outgoing INVITE during an internal hangup, we should let the
+ protocol stack process the INVITE transaction and terminate the
+ dialog properly. This is achieved by setting the PENDING_BYE
+ flag. When this flag is used, once the dialog proceeds to an
+ escapable state the transaction will either be canceled with a
+ SIP_CANCEL or completed followed immediately by a BYE. Attempting
+ to do this any other way is incorrect. If the endpoint is not
+ responding to the INVITE request, the INVITE must continue to be
+ retransmitted until it times out which will result in the dialog
+ being destroyed.
+
+2010-08-24 16:01 +0000 [r283380] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: This fix makes sure the ast_channel hangs up
+ correctly when the dialog's PENDING_BYE flag is set. When the
+ pending bye flag is used, it is possible that the dialog will
+ terminate and leave the sip_pvt->owner channel up. This is
+ because we never hangup the ast_channel after sending the SIP_BYE
+ request. When we receive the response for the SIP_BYE we set
+ need_destroy which we would expect to destroy the dialog on the
+ next do_monitor loop, but this is not the case. The dialog will
+ only be destroyed once the owner is hungup even with the
+ need_destroy flag set. This patch sets the softhangup flag on the
+ ast_channel when a SIP_BYE request is sent as a result of the
+ pending bye flag.
+
+2010-09-13 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.36 Released.
+
+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>
+
[... 29223 lines stripped ...]
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