[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:
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    tags/1.4.38-rc1/ChangeLog   (with props)

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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
+

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