[asterisk-commits] lmadsen: tag 1.6.2.16-rc1 r298185 - /tags/1.6.2.16-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Dec 13 10:19:11 CST 2010


Author: lmadsen
Date: Mon Dec 13 10:19:05 2010
New Revision: 298185

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=298185
Log:
Importing files for 1.6.2.16-rc1 release.

Added:
    tags/1.6.2.16-rc1/.lastclean   (with props)
    tags/1.6.2.16-rc1/.version   (with props)
    tags/1.6.2.16-rc1/ChangeLog   (with props)

Added: tags/1.6.2.16-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.16-rc1/.lastclean?view=auto&rev=298185
==============================================================================
--- tags/1.6.2.16-rc1/.lastclean (added)
+++ tags/1.6.2.16-rc1/.lastclean Mon Dec 13 10:19:05 2010
@@ -1,0 +1,1 @@
+36

Propchange: tags/1.6.2.16-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.6.2.16-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.6.2.16-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.6.2.16-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.16-rc1/.version?view=auto&rev=298185
==============================================================================
--- tags/1.6.2.16-rc1/.version (added)
+++ tags/1.6.2.16-rc1/.version Mon Dec 13 10:19:05 2010
@@ -1,0 +1,1 @@
+1.6.2.16-rc1

Propchange: tags/1.6.2.16-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.6.2.16-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.6.2.16-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.6.2.16-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.16-rc1/ChangeLog?view=auto&rev=298185
==============================================================================
--- tags/1.6.2.16-rc1/ChangeLog (added)
+++ tags/1.6.2.16-rc1/ChangeLog Mon Dec 13 10:19:05 2010
@@ -1,0 +1,28720 @@
+2010-12-13  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.16-rc1 Released.
+
+2010-12-10 16:24 +0000 [r298050]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/netsock.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Portability issue on OpenSolaris. Also detect the
+	  required structure element, because OpenSolaris defines
+	  SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+	  #18442) Reported by: ranjtech Patches:
+	  20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+	  Tested by: ranjtech
+
+2010-12-09 22:10 +0000 [r297960]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+	  | 14 lines Ignore spurious REGISTER requests If a REGISTER
+	  request with a Call-ID matching an existing transaction is
+	  received it was possible that the REGISTER request would
+	  overwrite the initreq of the private structure. This info is used
+	  to generate messages for other responses in the transaction. This
+	  patch ignores REGISTER requests that match non-REGISTER
+	  transactions. (closes issue #18051) Reported by: eeman Tested by:
+	  twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+
+2010-12-08 18:04 +0000 [r297908]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/extensions.conf.sample: Use inheritance to get correct
+	  results for SIPFROMDOMAIN. (from an internal Digium discussion)
+
+2010-12-07 22:58 +0000 [r297824]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 297823 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+	  | 12 lines Revert code that changed SSRC for DTMF. Some previous
+	  behavior was attempted to be restored, but mistakingly I did not
+	  realize that the previous behavior was incorrect. This fixes DTMF
+	  not being detected since DTMF shouldn't cause the SSRC to change.
+	  (related to issue #17404) (closes issue #18189) (closes issue
+	  #18352) Reported by: marcbou Tested by: cmbaker82 ........
+
+2010-12-07 22:40 +0000 [r297713-297819]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+	  utils/muted.c, /: Merged revisions 297818 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+	  | 4 lines Use non-deprecated APIs for CoreAudio Review:
+	  https://reviewboard.asterisk.org/r/1040/ ........
+
+	* apps/app_followme.c, /: Merged revisions 297689 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+	  | 8 lines Don't create a Local channel if the target extension
+	  does not exist. (closes issue #18126) Reported by: junky Patches:
+	  followme.diff uploaded by junky (license 177) (partially
+	  restructured by me to avoid a possible memory leak) ........
+
+2010-12-06 22:03 +0000 [r297605]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+	  | 12 lines Improve handling of REGISTER requests with multiple
+	  contact headers. The changes here attempt to more strictly follow
+	  RFC 3261 section 10.3. Basically the following will now cause a
+	  400 Bad Response to be returned, if: - multiple Contact headers
+	  are present with one set to expire all bindings ("*") - wildcard
+	  parameter is specified for Contact without Expires header or
+	  Expires header is not set to zero. ABE-2442 ABE-2443 ........
+
+2010-12-03 17:40 +0000 [r297534]  Sean Bright <sean at malleable.com>
+
+	* channels/chan_console.c: The CLI command should not contain
+	  <placeholder>s, these are for descriptions.
+
+2010-12-02 20:06 +0000 [r297405]  Paul Belanger <pabelanger at digium.com>
+
+	* Makefile, /: Merged revisions 297404 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+	  2010) | 7 lines Resolve compile error under FreeBSD We now set
+	  _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+	  ASTCFLAGS to override the setting. Review:
+	  https://reviewboard.asterisk.org/r/1043/ ........
+
+2010-12-02 18:07 +0000 [r297311]  Terry Wilson <twilson at digium.com>
+
+	* /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+	  | 12 lines Initialize offset for adaptive jitter buffer When the
+	  adaptive jitter buffer is enabled in sip.conf, the first frame
+	  placed in the jitter buffer fails with something like:
+	  jb_warning_output: Resyncing the jb. last_delay 0, this delay
+	  -215886466, threshold 1000, new offset 215886466 This happens
+	  because the offset is not initialized before calling jb_put().
+	  This patch modifies jb_put_first_adaptive() to set the offset to
+	  the frame's timestamp. Review:
+	  https://reviewboard.asterisk.org/r/1041/ ........
+
+2010-12-02 13:16 +0000 [r297229]  Russell Bryant <russell at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+	  | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+	  This is in response to some questions on IRC. To the user, there
+	  was nothing that made it obvious that this error had anything to
+	  do with DAHDI not being loaded. ........
+
+2010-12-02 08:55 +0000 [r297186]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5
+	  lines If we get a NOTIFY from a non-existing subscription we
+	  should answer with 481, not bad event. If we answer 481 the
+	  subscription that we don't want will be cancelled. ........
+
+2010-12-01 17:52 +0000 [r297073]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+	  | 23 lines Fix not stopping MOH when transfered local channel
+	  queue member is answered. The problem here is only present when
+	  local channels are used with the MOH passthru option as well as
+	  no optimization (/nm). I will describe the slightly bizarre
+	  scenario that was used to test, where phones B and C are queue
+	  members: Phone A dials into a queue with two members using local
+	  channels and the above options. Phone B answers. Phone A blind
+	  transfers phone B into the same queue. Phone A hangs up. Phone C
+	  answers, but phone B didn't stop playing MOH. In this scenario,
+	  the unhold frame that should have gotten to phone B never arrived
+	  due to the masquerade from the blind transfer. This is usually
+	  fine since app_queue manages the starting and stopping of MOH.
+	  However, with the passthrough option enabled when app_queue
+	  attempts to stop MOH it tries to do so on the local channel
+	  rather than the real channel. The easiest solution was to just
+	  make sure to send an unhold frame during the transfer since it
+	  wouldn't make sense to have MOH playing after a transfer anyway.
+	  This only modifies SIP transfers, but the other transfers did not
+	  seem to be a problem. If DTMF based transfers were a problem it
+	  might be okay to add ast_moh_stop to finishup, but I didn't want
+	  to have to add that unless required. ABE-2624 ........
+
+2010-12-01 17:01 +0000 [r296950-296991]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01
+	  Dec 2010) | 5 lines Clarify documentation on how we store codec
+	  preference lists. (closes issue #18397) Reported by: birgita
+	  ........
+
+	* channels/chan_iax2.c: Missed initializations caused startup
+	  errors on Mac OS X (and possibly others, too).
+
+2010-12-01 00:24 +0000 [r296869]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30
+	  Nov 2010) | 4 lines Properly restore backup information file when
+	  hanging up during message prepending. ABE-2654 ........
+
+2010-11-29 22:54 +0000 [r296671]  Paul Belanger <pabelanger at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon,
+	  29 Nov 2010) | 5 lines Make sure nothing else is needed before
+	  destroying the scheduler. (closes issue #18398) Reported by:
+	  pabelanger ........
+
+2010-11-29 07:27 +0000 [r296533]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: I love standards. There are so many to choose from.
+	  Except when there isn't one. Linux and *BSD disagree on the
+	  elements within the ucred structure. Detect which one is in use
+	  on the system. (closes issue #18384) Reported by: bjm Patches:
+	  cred-diffs uploaded by bjm (license 473)
+	  20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+	  (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+	  tilghman (license 14) Tested by: tilghman, bjm
+
+2010-11-27 10:39 +0000 [r296466]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_meetme.c: 18 characters is too short for most date/times
+	  (20 is the usual, but we add more in case of greater precision).
+	  (closes issue #18369) Reported by: tnakonz
+
+2010-11-26 12:23 +0000 [r296351]  Olle Johansson <oej at edvina.net>
+
+	* /, main/say.c: Merged revisions 296309 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+	  lines Fix bugs in saying numbers using the Swedish language
+	  syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+	  help from Peter Lindahl. Testing by the ClearIT team during a
+	  coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+	  ........
+
+2010-11-24 23:28 +0000 [r296221]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 296213 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+	  | 6 lines Make Asterisk less crashy. Since we might not put a new
+	  translation path on the channel, go ahead and set it to NULL
+	  right after destroying the old one to ensure we don't try to free
+	  an invalid translation path later on. ........
+
+2010-11-24 22:42 +0000 [r296166]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24
+	  Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port
+	  after FXS port gets a CallWaiting pip. The FXS connected phone
+	  has to have CW/CID support to fail, as it will send back a DTMF
+	  'A' or 'D' when it's ready to receive CallerID. A normal phone
+	  with no CID never fails. Also the SIP phone does not hear MOH
+	  when the CW call is answered. The DTMF end frame is suppressed
+	  when the phone acknowledges the CW signal for CID. The problem is
+	  the DTMF begin frame needs to be suppressed as well. The DTMF
+	  begin frame is causing SIP to start sending the DTMF RTP frames.
+	  Since the DTMF end frame is suppressed, SIP will not stop sending
+	  those DTMF RTP packets. * Suppress the DTMF begin and end frames
+	  when the channel driver is looking for DTMF digits. * Fixed a
+	  couple issues caused by not cleaning up the CID spill if you
+	  answer the CW call while it is sending the CID spill. * Fixed not
+	  sending CW/CID spill to the phone when the call is natively
+	  bridged. (Fixed by not using native bridge if CW/CID is
+	  possible.) * Suppress received audio when sending CW/CID spills.
+	  The other parties involved do not need to hear the CW/CID spills
+	  and may be confused if the CW call is for them. (closes issue
+	  #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+	  uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+	  NOTE: * v1.4 does not have the main problem fixed by suppressing
+	  the DTMF start frames. The other three items fixed are relevant.
+	  * If you really must restore native bridging between analog
+	  ports, you need to disable CW/CID either by configuring
+	  chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+	  dialing the number to temporarily disable CW. ........
+
+2010-11-24 20:23 +0000 [r296001-296083]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 296082 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+	  | 12 lines Fix false reporting of an error by set_format(). In
+	  the case that the native format was able to be changed to match
+	  the new requested format, the code proceeded to attempt to build
+	  a translation path, anyway. The result would be NULL, since no
+	  translation path is necessary and resulted in this function
+	  thinking an error has occurred. This case is now specifically
+	  caught and no attempt to build a translation path is attempted.
+	  Thanks to our automated tests and bamboo.asterisk.org for
+	  catching this problem and making a whole lot of noise when things
+	  started failing. :-) ........
+
+	* apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+	  | 38 lines Handle failures building translation paths more
+	  effectively. The problem scenario occurred on a heavily loaded
+	  system that was using the codec_dahdi module and exceeded the
+	  hardware transcoding capacity. The failure mode at that point was
+	  not good. The report came in to us as an Asterisk lock-up. The
+	  "core show locks" shows a ton of threads locked up (but no
+	  obvious deadlock). Upon deeper investigation, when the system is
+	  in this state, the CPU was maxed out. The CPU was being consumed
+	  by the Asterisk logger spewing messages on every audio frame for
+	  calls set up after transcoder capacity was reached. The purpose
+	  of this patch is to make Asterisk handle failures to create a
+	  translation path in a more graceful manner. If we can't
+	  translate, then the call just needs to be dropped, as it's not
+	  going to work. These are the changes: 1) In set_format() of
+	  channel.c (which is called by set_read_format() and
+	  set_write_format()), it was ignoring if
+	  ast_translator_build_path() failed and returned NULL. It now pays
+	  attention to that case and returns a result reflecting failure.
+	  With this change in place, the bridging code will immediately
+	  detect a failure and end the bridge instead of proceeding to try
+	  to bridge frames that can't be translated and making channel
+	  drivers freak out by sending them frames in a format they weren't
+	  expecting. 2) In ast_indicate_data() of channel.c, failure of
+	  ast_playtones_start() was ignored. It is now reflected in the
+	  return value of the function. This didn't turn out to have any
+	  affect on the bug, but seemed like a good change to leave in. 3)
+	  In app_dial(), when only sending a call to a single endpoint, it
+	  will attempt to do some bridging of its own of early audio. It
+	  uses make_compatible() when it's going to do this. However, it
+	  ignored failure from make compatible. So, even with the fix from
+	  #1, if there was early audio going through app_dial, there would
+	  still be a period of invalid frames passing through. After
+	  detecting failure here, Dial() exits. ABE-2658 ........
+
+2010-11-23 09:36 +0000 [r295907]  Olle Johansson <oej at edvina.net>
+
+	* /, main/say.c: Merged revisions 295906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+	  lines Fix support of saynumber(1,n) in the Swedish language
+	  (closes issue #18353) Reported by: oej Review:
+	  https://reviewboard.asterisk.org/r/1031/ ........
+
+2010-11-22 20:02 +0000 [r295868]  Sean Bright <sean at malleable.com>
+
+	* configs/chan_dahdi.conf.sample: Change some documentation to
+	  suggest dahdi_monitor instead of ztmonitor.
+
+2010-11-22 19:28 +0000 [r295843]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/frame.h, main/channel.c, main/pbx.c, /,
+	  apps/app_macro.c, include/asterisk/channel.h: Merged revisions
+	  295790 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+	  | 46 lines The channel redirect function (CLI or AMI) hangs up
+	  the call instead of redirecting the call. To recreate the
+	  problem: 1) Party A calls Party B 2) Invoke CLI "channel
+	  redirect" command to redirect channel call leg associated with A.
+	  3) All associated channels are hung up. Note that if the CLI
+	  command were done on the channel call leg associated with B it
+	  works. This regression was a result of the fix for issue #16946
+	  (https://reviewboard.asterisk.org/r/740/). The regression affects
+	  all features that use an async goto to execute the dialplan
+	  because of an external event: Channel redirect, AMI redirect, SIP
+	  REFER, and FAX detection. The struct ast_channel._softhangup code
+	  is a mess. The variable is used for several purposes that do not
+	  necessarily result in the call being hung up. I have added
+	  doxygen comments to describe how the various _softhangup bits are
+	  used. I have corrected all the places where the variable was
+	  tested in a non-bit oriented manner. The primary fix is the new
+	  AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+	  the soft hangup requests that do not normally result in a hangup
+	  do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+	  Reported by: SantaFox (closes issue #18185) Reported by:
+	  kwemheuer (closes issue #18211) Reported by: zahir_koradia
+	  (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+	  Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+	  Review: https://reviewboard.asterisk.org/r/1013/ ........
+
+2010-11-20 00:45 +0000 [r295710]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/event.h, main/event.c: Fix cache of device state
+	  changes for multiple servers. This patch addresses a regression
+	  where device states across multiple servers were not being
+	  processing completely correctly. The code works to determine the
+	  overall state by looking at the last known state of a device on
+	  each server. However, there was a regression due to some invasive
+	  rewrites of how the cache works that led to the cache only
+	  storing the last device state change for a device, regardless of
+	  which server it was on. The code is set up to cache device state
+	  change events by ensuring that each event in the cache has a
+	  unique device name + entity ID (server ID). The code that was
+	  responsible for comparing raw information elements (which EID is)
+	  always returned a match due to a memcmp() with a length of 0.
+	  There isn't much code to fix the actual bug. This patch also
+	  introduces a new CLI command that was very useful for debugging
+	  this problem. The command allows you to dump the contents of the
+	  event cache. (closes issue #18284) Reported by: klaus3000
+	  Patches: issue18284.rev1.txt uploaded by russell (license 2)
+	  Tested by: russell, klaus3000 (closes issue #18280) Reported by:
+	  klaus3000 Review: https://reviewboard.asterisk.org/r/1012/
+
+2010-11-19 21:55 +0000 [r295672]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 295628 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
+	  | 8 lines Discard responses with more than one Via This is not a
+	  perfect solution as headers that are joined via commas are not
+	  detected. This is a parsing issue that to fix "correctly" would
+	  necessitate a new SIP parser. Review:
+	  https://reviewboard.asterisk.org/r/1019/ ........
+
+2010-11-18 17:51 +0000 [r295440]  Paul Belanger <pabelanger at digium.com>
+
+	* res/res_jabber.c, include/asterisk/jabber.h: Fix compiler
+	  warnings when using openssl-dev 1.0.0+ Review:
+	  https://reviewboard.asterisk.org/r/1016/
+
+2010-11-16 22:57 +0000 [r295281]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, /: Merged revisions 295280 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010)
+	  | 1 line Dead code elimination in channel.c:ast_channel_bridge()
+	  variable who. ........
+
+2010-12-02  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.15 Released.
+
+2010-11-15  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.15-rc1
+
+2010-11-15 18:24 +0000 [r294988-295062]  Tilghman Lesher <tlesher at digium.com>
+
+	* tests/test_expr.c (added), /: Merged revisions 295026 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010)
+	  | 2 lines Create test verifying results of expression parser
+	  ........
+
+	* 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>)". ........
+
+2010-10-30 01:49 +0000 [r293340-293417]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+	  Oct 2010) | 1 line Remove some more code that serves no purpose.
+	  ........
+
+	* channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+	  Oct 2010) | 1 line Remove some code that serves no purpose.
+	  ........
+
+2010-10-28 19:54 +0000 [r293195-293196]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+	  Reported (though the reporter did not understand he was reporting
+	  a bug) on the asterisk-users list:
+	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+	  ........
+
+	* /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c,
+	  res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+	  Reported (though the reporter did not understand he was reporting
+	  a bug) on the asterisk-users list:
+	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+	  ........
+
+2010-10-28 16:09 +0000 [r293158]  Jeff Peeler <jpeeler at digium.com>
+
+	* funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically
+	  when you're using characters above \x7f or invalid character
+	  escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+	  Patches: issue18060_func_strings_filter_infinite_loop.patch
+	  uploaded by wdoekes (license 717) Tested by: wdoekes
+
+2010-10-26 18:33 +0000 [r293118]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25
+	  Oct 2010) | 29 lines Fix inprocess_container in voicemail to
+	  correctly restrict max messages. The comparison function logic
+	  was off, so the number of sessions for a given mailbox were not
+	  being incremented properly. This problem caused the maximum
+	  number of messages per folder to not be respected when
+	  simultaneously leaving multiple voicemails just below the
+	  threshold. These problems should be fixed by the above, but just
+	  in case: Fixed resequence_mailbox to rely on the actual number of
+	  detected number of files in a directory rather than just assuming
+	  only 10 messages more than the maximum had been left. Also if
+	  more messages than the maximum are deleted they are actually
+	  removed now. The second purpose of this commit should have been
+	  separated out probably, but is related to the above. Again, if
+	  the number of messages in a given voicemail folder exceeds the
+	  maximum set limit make sure to allocate enough space for the
+	  deleted and heard index tracking array. A few random fixes: There
+	  was a forgotten decrement of the inprocess count in
+	  imap_store_file. When using IMAP storage, do not look in the
+	  directory where file based storage messages may still reside and
+	  influence the message count. Ensure to use only the first format
+	  in sendmail. ABE-2516 ........
+
+2010-10-25 19:06 +0000 [r292867]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 292866 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25
+	  Oct 2010) | 27 lines This patch turns chan_local pvts into
+	  astobj2 objects. chan_local does some dangerous things involving
+	  deadlock avoidance. tech_pvt functions like hangup and
+	  queue_frame are provided with a locked channel upon entry. Those

[... 28038 lines stripped ...]



More information about the asterisk-commits mailing list