[asterisk-commits] lmadsen: tag 1.6.2.15-rc1 r295034 - /tags/1.6.2.15-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Nov 15 12:17:37 CST 2010


Author: lmadsen
Date: Mon Nov 15 12:17:34 2010
New Revision: 295034

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295034
Log:
Importing files for 1.6.2.15-rc1 release.

Added:
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    tags/1.6.2.15-rc1/.version   (with props)
    tags/1.6.2.15-rc1/ChangeLog   (with props)

Added: tags/1.6.2.15-rc1/.lastclean
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Added: tags/1.6.2.15-rc1/ChangeLog
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--- tags/1.6.2.15-rc1/ChangeLog (added)
+++ tags/1.6.2.15-rc1/ChangeLog Mon Nov 15 12:17:34 2010
@@ -1,0 +1,27611 @@
+2010-11-15  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.15-rc1 Released.
+
+2010-11-15 07:42 +0000 [r294988]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_curl.c: It is possible to crash Asterisk by feeding
+	  the curl engine invalid data. (closes issue #18161) Reported by:
+	  wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
+	  tilghman (license 14) Tested by: tilghman
+
+2010-11-12 21:14 +0000 [r294904-294910]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c: Return correct error code if lock path
+	  fails. The recent changes to open_mailbox actually caused it to
+	  be fixed, but let's be consistent. Reported by alecdavis in
+	  asterisk-dev.
+
+	* apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
+	  Nov 2010) | 16 lines 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 02:44 +0000 [r294822]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
+	  Nov 2010) | 11 lines 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 21:57 +0000 [r294639-294733]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+	  | 18 lines 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) (closes issue #17779) Reported by: lftsy Patches:
+	  bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+	  zerohalo ........
+
+	* main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+	  revisions 294384 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+	  | 47 lines 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
+	  ........
+
+2010-11-10 23:16 +0000 [r294571]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/features.c: Actually pay attention to documented settings in
+	  features.conf. (closes issue #16757) Reported by: voxter Patches:
+	  20101012__issue16757.diff.txt uploaded by tilghman (license 14)
+	  Review: https://reviewboard.asterisk.org/r/994/
+
+2010-11-10 12:41 +0000 [r294500]  Russell Bryant <russell at digium.com>
+
+	* main/devicestate.c: Improve a debug message to be more readable
+	  and consistent. (closes issue #18282) Reported by: klaus3000
+	  Patches: ast_devstate2str-patch.txt uploaded by klaus3000
+	  (license 65)
+
+2010-11-09 20:27 +0000 [r294429]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Detect GMime properly on systems where
+	  gmime flags and libs are configured with pkg-config. (closes
+	  issue #16155) Reported by: jcollie Patches:
+	  20100917__issue16155.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman
+
+2010-11-08 22:30 +0000 [r294277-294312]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_timing_timerfd.c: add missing unlock not present in
+	  294277
+
+	* main/timing.c, main/channel.c, res/res_timing_timerfd.c,
+	  include/asterisk/timing.h: Fix playback failure when using IAX
+	  with the timerfd module. To fix this issue the alert pipe will
+	  now be used when the timerfd module is in use. There appeared to
+	  be a race that was not solved by adding locking in the timerfd
+	  module, but needed to be there anyway. The race was between the
+	  timer being put in non-continuous mode in ast_read on the channel
+	  thread and the IAX frame scheduler queuing a frame which would
+	  enable continuous mode before the non-continuous mode event was
+	  read. This race for now is simply avoided. (closes issue #18110)
+	  Reported by: tpanton Tested by: tpanton I put tested by tpanton
+	  because it was tested on his hardware. Thanks for the remote
+	  access to debug this issue!
+
+2010-11-08 20:50 +0000 [r294242]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Go off hold when we get an empty reinvite
+	  telling us to. (closes issue 0014448) Reported by: frawd (closes
+	  issue #17878) Reported by: frawd
+
+2010-11-05 00:06 +0000 [r293969]  Shaun Ruffell <sruffell at digium.com>
+
+	* codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
+	  Nov 2010) | 17 lines 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. ........
+
+2010-11-03 18:31 +0000 [r293806]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
+	  Nov 2010) | 20 lines 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 23:07 +0000 [r293723]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+	  | 8 lines 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 21:26 +0000 [r293647]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
+	  Nov 2010) | 6 lines 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 1.6.2.14 Released.
+
+2010-09-20  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.14-rc1 Released.
+
+2010-09-20 15:56 +0000 [r287556-287558]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Use ast_str when processing hint state changes
+	  Merged revisions 287555 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+	  2010) | 5 lines Use ast_dynamic_str when processing hint state
+	  changes (related to issue #17928) Reported by: mdu113 ........
+
+	* /: Revert r287556.
+
+	* /: Use ast_str when processing hint state changes Merged
+	  revisions 287555 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+	  2010) | 5 lines Use ast_dynamic_str when processing hint state
+	  changes (related to issue #17928) Reported by: mdu113 ........
+
+2010-09-19 16:06 +0000 [r287470]  Olle Johansson <oej at edvina.net>
+
+	* main/manager.c, /: Merged revisions 287469 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+	  lines 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 21:08 +0000 [r287387]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 287386 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+	  | 7 lines 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 13:36 +0000 [r287308]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 287307 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+	  2010) | 5 lines 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 [r287198]  Jason Parker <jparker at digium.com>
+
+	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
+	  7 lines 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 20:06 +0000 [r287115-287119]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 287118 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+	  2010) | 8 lines 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, /: Merged revisions 287114 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+	  2010) | 8 lines 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 20:28 +0000 [r286998]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15
+	  Sep 2010) | 7 lines 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 19:27 +0000 [r286681-286757]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+	  2010) | 13 lines 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, /: Merged revisions 286679 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+	  2010) | 7 lines 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) ........
+
+2010-09-14 05:06 +0000 [r286527-286587]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/realtime/mysql/voicemail_messages.sql (added),
+	  contrib/realtime/mysql/voicemail_data.sql (added): Add
+	  documentation on missing backend tables for Voicemail
+
+	* main/features.c: C precedence got me
+
+	* main/features.c: Refactor conversion to ast_poll() to fix
+	  callparking regression.
+
+2010-09-13 19:38 +0000 [r286456]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c: Remove "Internal IP" from sip show settings,
+	  as it's not at all useful to display. (closes issue #17840)
+	  Reported by: oej
+
+2010-09-11 17:05 +0000 [r286268]  Olle Johansson <oej at edvina.net>
+
+	* /, main/file.c: Merged revisions 286267 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+	  lines Handle error response when we can't make file compatible
+	  Review: https://reviewboard.asterisk.org/r/911/ ........
+
+2010-09-10 22:56 +0000 [r286223]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 286222 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
+	  Sep 2010) | 1 line Return -1 if chan_local doesn't support an
+	  option ........
+
+2010-09-10 20:55 +0000 [r286117]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri,
+	  10 Sep 2010) | 4 lines Load iax.conf before registering any
+	  functions/applications/actions. Review:
+	  https://reviewboard.asterisk.org/r/914/ ........
+
+2010-09-10 20:42 +0000 [r286116]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10
+	  Sep 2010) | 11 lines 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 20:35 +0000 [r286115]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/pbx.h, include/asterisk/frame.h,
+	  channels/chan_local.c, /, funcs/func_channel.c,
+	  include/asterisk/channel.h: Merged revisions 286059 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10
+	  Sep 2010) | 16 lines 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 18:30 +0000 [r285930-286024]  Tilghman Lesher <tlesher at digium.com>
+
+	* tests/test_heap.c, /, main/test.c: Merged revisions 286023 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010)
+	  | 2 lines Missing newline ........
+
+	* include/asterisk/select.h: Another fix for Mac OS X. While trying
+	  to fix this the "right" way, I wandered into dependency hell. Two
+	  hours later, I backed out, and just removed the offending code.
+	  ast_inline_api only goes one level deep and then it breaks. Ouch.
+
+	* tests/test_poll.c, include/asterisk/select.h, /, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+	  285889 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+	  | 7 lines 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 22:49 +0000 [r285818]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+	  2010) | 8 lines 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 20:09 +0000 [r285744]  Jason Parker <jparker at digium.com>
+
+	* main/channel.c, /: Merged revisions 285742 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+	  9 lines 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 18:50 +0000 [r285639-285710]  Brett Bryant <bbryant at digium.com>
+
+	* main/pbx.c: Fixes an issue with dialplan pattern matching where
+	  the specificity for pattern ranges and pattern special characters
+	  was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+	  Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+	  657) Tested by: Nick_Lewis
+
+	* res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
+	  Sep 2010) | 7 lines 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 22:11 +0000 [r285563-285567]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
+	  | 2 lines In retrans_pkt, do not unlock pvt until the end of the
+	  function on a transmit failure. ........
+
+	* channels/chan_sip.c: Fixes interoperability problems with session
+	  timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
+	  "Require" header. This is not to our benefit and RFC 4028 section
+	  7.1 even warns against it. It is possible for one endpoint to
+	  perform session-timer refreshes while the other endpoint does not
+	  support them. If in this case the end point performing the
+	  refreshing puts "timer" in the Require field during a refresh,
+	  the dialog will likely get terminated by the other end. 2. Change
+	  the behavior of 'session-timer=accept' in sip.conf (which is the
+	  default behavior of Asterisk with no session timer configuration
+	  specified) to only run session-timers as result of an incoming
+	  INVITE request if the INVITE contains an "Session-Expires"
+	  header... Asterisk is currently treating having the "timer"
+	  option in the "Supported" header as a request for session timers
+	  by the UAC. I do not agree with this. Session timers should only
+	  be negotiated in "accept" mode when the incoming INVITE supplies
+	  a "Session-Expires" header, otherwise RFC 4028 says we should
+	  treat a request containing no "Session-Expires" header as a
+	  session with no expiration. Below I have outlined some situations
+	  and what Asterisk's behavior is. The table reflects the behavior
+	  changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+	  1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+	  "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+	  "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+	  4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+	  "Session-Expires" header 5. Outgoing INVITE: HAS
+	  "Session-Expires". Active - Asterisk will have an active refresh
+	  timer regardless if the other endpoint does. Inactive - Asterisk
+	  does not have an active refresh timer regardless if the other
+	  endpoint does. XXXXXXX - Not possible for mode.
+	  ______________________________________ |SITUATIONS |
+	  'session-timer' MODES | |___________|________________________| |
+	  | originate | accept | |-----------|------------|-----------| |1.
+	  | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+	  Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+	  -------------------------------------- (closes issue #17005)
+	  Reported by: alexrecarey
+
+2010-09-08 20:56 +0000 [r285532]  Brett Bryant <bbryant at digium.com>
+
+	* apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
+	  the amount of time left in a conference, if music on hold was
+	  playing, it doesn't restart. (closes issue #17408) Reported by:
+	  sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+	  sysreq (license 1009) Tested by: sysreq
+
+2010-09-08 20:42 +0000 [r285526-285529]  Jason Parker <jparker at digium.com>
+
+	* res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
+	  guidelines in moh rescan fix. Also fix the documentation that got
+	  me in trouble.
+
+	* res/res_musiconhold.c: Fixes issue where moh files were no longer
+	  rescanned during a reload. (closes issue #16744) Reported by: pj
+	  Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
+	  by: qwell
+
+2010-09-07 20:31 +0000 [r285267-285366]  Tilghman Lesher <tlesher at digium.com>
+
+	* pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+	  | 9 lines 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, /: Merged revisions 285266 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+	  | 4 lines Use poll, if indicated to do so, in the ast_poll2
+	  implementation. This fixes the unit tests on FreeBSD 8.0.
+	  ........
+
+2010-09-07 17:49 +0000 [r285196]  Brett Bryant <bbryant at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
+	  Sep 2010) | 10 lines 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 06:55 +0000 [r285089]  Tilghman Lesher <tlesher at digium.com>
+
+	* makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
+	  | 2 lines Silly convenience script for BSD platforms. ........
+
+2010-09-03 18:15 +0000 [r284958]  Brett Bryant <bbryant at digium.com>
+
+	* channels/chan_iax2.c: This is a patch provided for issue #17935
+	  to add the ActionID to the IAXregistry AMI response. (closes
+	  issue #17935) Reported by: alexkuklin Patches: iaxshowreg
+	  uploaded by alexkuklin (license 1115) Tested by: alexkuklin
+
+2010-09-03 16:20 +0000 [r284897]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+	  | 5 lines 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
+	  existed. ........
+
+2010-09-02 20:54 +0000 [r284778]  Brett Bryant <bbryant at digium.com>
+
+	* main/manager.c, /: Merged revisions 284777 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
+	  | 7 lines 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 16:48 +0000 [r284704]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+	  | 7 lines 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. ........
+
+2010-09-02 16:07 +0000 [r284399-284665]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_usbradio.c: Fixing build.
+
+	* apps/app_queue.c: Don't reset queue stats on a module reload.
+	  (closes issue #17535) Reported by: raarts Patches:
+	  20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+
+	* configure, include/asterisk/autoconfig.h.in: Failed to rerun
+	  bootstrap.sh after last commit
+
+	* res/res_jabber.c, main/rtp.c, main/poll.c,
+	  include/asterisk/select.h (added), channels/chan_usbradio.c,
+	  channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
+	  include/asterisk/poll-compat.h, tests/test_poll.c (added),
+	  main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
+	  configure.ac, channels/console_video.c,
+	  include/asterisk/channel.h: Merged revisions 284478 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
+	  Sep 2010) | 11 lines 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/ ........
+
+	* res/res_config_pgsql.c: Don't warn on floats and timestamps
+	  (closes issue #17082) Reported by: coolmig
+
+	* /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+	  | 7 lines 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 18:59 +0000 [r284317]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
+	  Aug 2010) | 7 lines 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 ........
+
+2010-08-30 22:27 +0000 [r284280]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
+	  we should not be trying to write to it. 2) Call _exit(0), not
+	  exit(0), to avoid running shutdown routines in a child. 3) Use
+	  endian, not processor, detection to ensure bytes are written in
+	  the correct order. (closes issue #15706) Reported by: modelnine
+	  Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
+	  modelnine (license 865) Tested by: gmartinez
+
+2010-08-27 22:27 +0000 [r284002]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+	  | 8 lines 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 20:30 +0000 [r283881]  Jason Parker <jparker at digium.com>
+
+	* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+	  revisions 283880 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+	  8 lines 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 ........
+
+2010-08-26 15:24 +0000 [r283381-283691]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+	  | 19 lines 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.
+	  ........
+
+	* channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
+	  xml body so pickup can occur. When pedantic mode is used, the
+	  dialog-info xml generated during a ringing event must contain the

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