[svn-commits] lmadsen: tag 1.4.38-rc1 r295028 - /tags/1.4.38-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 15 12:04:15 CST 2010
Author: lmadsen
Date: Mon Nov 15 12:04:13 2010
New Revision: 295028
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295028
Log:
Importing files for 1.4.38-rc1 release.
Added:
tags/1.4.38-rc1/.lastclean (with props)
tags/1.4.38-rc1/.version (with props)
tags/1.4.38-rc1/ChangeLog (with props)
Added: tags/1.4.38-rc1/.lastclean
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--- tags/1.4.38-rc1/ChangeLog (added)
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+2010-11-15 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.38-rc1 Released.
+
+2010-11-15 17:58 +0000 [r295026] Tilghman Lesher <tlesher at digium.com>
+
+ * tests/test_expr.c (added): Create test verifying results of
+ expression parser
+
+2010-11-12 20:49 +0000 [r294903] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix regression causing abort in voicemail
+ after opening a mailbox with no mesgs. In order to be more safe,
+ some error handling code was changed to respect more error
+ conditions including the potential memory allocation failure for
+ deleted and heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger
+
+2010-11-12 02:41 +0000 [r294821] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Asterisk is getting a "No D-channels
+ available!" warning message every 4 seconds. Asterisk is just
+ whining too much with this message: "No D-channels available!
+ Using Primary channel XXX as D-channel anyway!". Filtered the
+ message so it only comes out once if there is no D channel
+ available without an intervening D channel available period.
+ (closes issue #17270) Reported by: jmls
+
+2010-11-11 22:11 +0000 [r294641-294739] Jeff Peeler <jpeeler at digium.com>
+
+ * main/pbx.c: I didn't mean to merge this, sorry
+
+ * channels/chan_sip.c: Fix problem with qualify option packets for
+ realtime peers never stopping. The option packets not only never
+ stopped, but if a realtime peer was not in the peer list multiple
+ options dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) Reported by: lftsy Patches: bug16382-3.patch uploaded by
+ jpeeler (license 325) Tested by: zerohalo
+
+ * main/pbx.c: One small addition to 294384 found while very
+ carefully merging to 1.6.
+
+2010-11-09 17:37 +0000 [r294384] Jeff Peeler <jpeeler at digium.com>
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c: Fix a deadlock
+ in device state change processing. Copied from some notes from
+ the original author (Russell): Deadlock scenario: Thread 1:
+ device state change thread Holds - rdlock on contexts Holds -
+ hints lock Waiting on channels container lock Thread 2: SIP
+ monitor thread Holds the "iflock" Holds a sip_pvt lock Holds
+ channel container lock Waiting for a channel lock Thread 3: A
+ channel thread (chan_local in this case) Holds 2 channel locks
+ acquired within app_dial Holds a 3rd channel lock it got inside
+ of chan_local Holds a local_pvt lock Waiting on a rdlock of the
+ contexts lock A bunch of other threads waiting on a wrlock of the
+ contexts lock To address this deadlock, some locking order rules
+ must be put in place and enforced. Existing relevant rules: 1)
+ channel lock before a pvt lock 2) contexts lock before hints lock
+ 3) channels container before a channel What's missing is some
+ enforcement of the order when you involve more than any two. To
+ fix this problem, I put in some code that ensures that (at least
+ in the code paths involved in this bug) the locks in (3) come
+ before the locks in (2). To change the operation of thread 1 to
+ comply, I converted the storage of hints to an astobj2 container.
+ This allows processing of hints without holding the hints
+ container lock. So, in the code path that led to thread 1's
+ state, it no longer holds either the contexts or hints lock while
+ it attempts to lock the channels container. (closes issue #18165)
+ Reported by: antonio ABE-2583
+
+2010-11-08 18:59 +0000 [r294163] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Modify our handling of 491 responses to drop
+ any pending reinvite retry scheduler entries if we get a new 491.
+ This prevents a scheduler entry from leaking if we receive a 491
+ response when one is pending. If a scheduler entry leaks, the pvt
+ it is associated my get destroyed before the scheduler entry
+ fires, and then memory corruption and crashes can occur when the
+ scheduled reinvite attempts to access and modify the memory of
+ the destroyed pvt. ABE-2543
+
+2010-11-05 00:02 +0000 [r293968] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c: codecs/codec_dahdi: Prevent "choppy" audio
+ when receiving unexpected frame sizes. dahdi-linux 2.4.0
+ (specifically commit 9034) added the capability for the wctc4xxp
+ to return more than a single packet of data in response to a
+ read. However, when decoding packets, codec_dahdi was still
+ assuming that the default number of samples was in each read. In
+ other words, each packet your provider sent you, regardless of
+ size, would result in 20 ms of decoded data (30 ms if decoding
+ G723). If your provider was sending 60 ms packets then
+ codec_dahdi would end up stripping 40 ms of data from each
+ transcoded frame resulting in "choppy" audio. This would only
+ affect systems where G729 packets are arriving in sizes greater
+ than 20ms or G723 packets arriving in sizes greater than 30ms.
+ DAHDI-744.
+
+2010-11-04 21:28 +0000 [r293922] David Vossel <dvossel at digium.com>
+
+ * res/res_features.c: Fixes ringback tone on feature semi-attended
+ transfer ABE-2168
+
+2010-11-03 18:23 +0000 [r293805] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Party A in an analog 3-way call would
+ continue to hear ringback after party C answers. All parties are
+ analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
+ flash hooks to bring C into 3-way call before C answers. (A and B
+ hear ringback) 4) C answers 5) A continues to hear ringback
+ during the 3-way call. (All parties can hear each other.) * Fixed
+ use of wrong variable in dahdi_bridge() that stopped ringback on
+ the wrong subchannel. * Made several debug messages have more
+ information. A similar issue happens if B and C are SIP channels.
+ B continues to hear ringback. For some reason this only affects
+ v1.8 and trunk. * Don't start ringback on the real and 3-way
+ subchannels when creating the 3-way conference. Removing this
+ code is benign on v1.6.2 and earlier.
+
+2010-11-02 23:02 +0000 [r293722] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Add enabled/disabled information for
+ rtautoclear sip show settings output. When setting to zero/"no",
+ the numeric default was shown making it not obvious the disabled
+ setting was respected. (closes issue #18123) Reported by:
+ zerohalo
+
+2010-11-02 21:24 +0000 [r293639] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Make warning message have more useful
+ information in it. Change "Unable to get index, and nullok is not
+ asserted" to "Unable to get index for '<channel-name>' on channel
+ <number> (<function>(), line <number>)".
+
+2010-11-02 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.37 Released.
+
+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
+
[... 29389 lines stripped ...]
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