[svn-commits] bebuild: tag 10.12.0-rc2 r378737 - /tags/10.12.0-rc2/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Wed Jan 9 13:50:15 CST 2013


Author: bebuild
Date: Wed Jan  9 13:50:11 2013
New Revision: 378737

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=378737
Log:
Importing files for 10.12.0-rc2 release.

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+2013-01-09  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.12.0-rc2 Released.
+
+2013-01-08 23:55 +0000 [r378683-378689]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_queue.c: app_queue: Fix incorrect assertion. (issue
+	  ASTERISK-16115)
+
+	* /, configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+	  apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+	  that is in only one queue. When ringinuse=no queue members can
+	  receive more than one call if these calls happen at nearly the
+	  same time. * Fix so a queue member does not receive more than one
+	  call from a queue. NOTE: This fix does not prevent multiple calls
+	  to a member if the member is in more than one queue. * Did some
+	  refactoring to eliminate some code redundancy. (issue
+	  ASTERISK-16115) Reported by: nik600 Patches:
+	  jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+	  uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+	  changes adding the queues.conf check_state_unknown option as it
+	  was added in an attempt to fix this problem. The fix did not need
+	  to be optional. The fix should not have tried to explicitly set
+	  the device state. Setting the device state by something other
+	  than the device introduces a race condition. I also could not see
+	  how the change would be effective other than delaying the
+	  app_queue code long enough for the device state to propagate to
+	  app_queue. ........ Merged revisions 378663 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-02 17:40 +0000 [r378286-378320]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_calendar.c, include/asterisk/devicestate.h,
+	  channels/chan_local.c, /, main/ccss.c, channels/chan_sip.c,
+	  apps/app_meetme.c, channels/chan_agent.c, main/devicestate.c,
+	  include/asterisk/channel.h, res/res_jabber.c, apps/app_queue.c,
+	  channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+	  channels/chan_skinny.c, include/asterisk/event_defs.h,
+	  main/features.c, main/event.c, apps/app_confbridge.c,
+	  apps/confbridge/conf_state_empty.c, funcs/func_devstate.c:
+	  Prevent exhaustion of system resources through exploitation of
+	  event cache Asterisk maintains an internal cache for devices in
+	  the event subsystem. The device state cache holds the state of
+	  each device known to Asterisk, such that consumers of device
+	  state information can query for the last known state for a
+	  particular device, even if it is not part of an active call. The
+	  concept of a device in Asterisk can include entities that do not
+	  have a physical representation. One way that this occurred was
+	  when anonymous calls are allowed in Asterisk. A device was
+	  automatically created and stored in the cache for each anonymous
+	  call that occurred; this was possible in the SIP and IAX2 channel
+	  drivers and through channel drivers that utilized the
+	  res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
+	  These devices are never removed from the system, allowing
+	  anonymous calls to potentially exhaust a system's resources. This
+	  patch changes the event cache subsystem and device state
+	  management to no longer cache devices that are not associated
+	  with a physical entity. (issue ASTERISK-20175) Reported by:
+	  Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
+	  patches: event-cachability-3.diff uploaded by jcolp (license
+	  5000) ........ Merged revisions 378303 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c, main/http.c, res/res_jabber.c,
+	  channels/sip/include/sip.h: Resolve crashes due to large stack
+	  allocations when using TCP Asterisk had several places where
+	  messages received over various network transports may be copied
+	  in a single stack allocation. In the case of TCP, since multiple
+	  packets in a stream may be concatenated together, this can lead
+	  to large allocations that overflow the stack. This patch modifies
+	  those portions of Asterisk using TCP to either favor heap
+	  allocations or use an upper bound to ensure that the stack will
+	  not overflow: * For SIP, the allocation now has an upper limit *
+	  For HTTP, the allocation is now a heap allocation instead of a
+	  stack allocation * For XMPP (in res_jabber), the allocation has
+	  been eliminated since it was unnecesary. Note that the HTTP
+	  portion of this issue was independently found by Brandon Edwards
+	  of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
+	  wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
+	  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
+	  5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
+	  wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
+	  uploaded by wdoekes (license 5674) ........ Merged revisions
+	  378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-31 14:43 +0000 [r378218]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
+	  without crypto info This ensures that Asterisk rejects encrypted
+	  media streams (RTP/SAVP audio and video) that are missing
+	  cryptographic keys and ensures that the incoming SDP is
+	  consistent with RFC4568 as far as having a crypto attribute
+	  present for any SAVP streams. Review:
+	  https://reviewboard.asterisk.org/r/2204/ ........ Merged
+	  revisions 378217 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-18 17:38 +0000 [r378120]  Kinsey Moore <kmoore at digium.com>
+
+	* main/channel.c, /: Add test events for time limit-related hangups
+	  This patch adds hangup-related test events in order to support
+	  testing of time-limited bridges. This aids in testing the S() and
+	  L() bridge options. (issue SWP-4713) ........ Merged revisions
+	  378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-17 23:08 +0000 [r378089-378093]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/loader.c, /: Fix potential double free when unloading a
+	  module. ........ Merged revisions 378092 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_local.c, /: Make chan_local module references tied
+	  to local_pvt lifetime. The chan_local module references were
+	  manually tied to the existence of the ;1 and ;2 channel links. *
+	  Made chan_local module references tied to the existence of the
+	  local_pvt structure as well as automatically take care of the
+	  module references. * Tweaked the wording of the local_fixup()
+	  failure warning message to make sense. Review:
+	  https://reviewboard.asterisk.org/r/2181/ ........ Merged
+	  revisions 378088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-14 21:30 +0000 [r378037]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, apps/app_queue.c: app_queue: Revert bad ringinuse=no patch.
+	  With the option ringinuse=no set, the patch committed for
+	  ASTERISK-16115 causes non-SIP queue members to never be called
+	  because the device state is checked after a channel is created to
+	  determine if the member is busy. These queue members always get
+	  the "Member %s is busy, cannot dial" message. Most channel
+	  drivers other than chan_sip use the default device state
+	  handling. The default device-state state is considered in use or
+	  unknown if the channel exists or not respectively. (closes issue
+	  ASTERISK-20801) Reported by: rmudgett Patches:
+	  jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
+	  patch uploaded by rmudgett ........ Merged revisions 378036 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-13 20:52 +0000 [r377992]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_state.c,
+	  apps/confbridge/include/confbridge.h,
+	  include/asterisk/bridging.h, apps/app_confbridge.c,
+	  apps/confbridge/conf_state_multi_marked.c: confbridge: Fix MOH on
+	  simultaneous user entry to a new conference. When two users
+	  entered a new conference simultaneously, one of the callers hears
+	  MOH. This happened if two unmarked users entered simultaneously
+	  and also if a waitmarked and a marked user entered
+	  simultaneously. * Created a confbridge internal MOH API to
+	  eliminate the inlined MOH handling code. Note that the conference
+	  mixing bridge needs to be locked when actually starting/stopping
+	  MOH because there is a small window between the conference join
+	  unsuspend MOH and actually joining the mixing bridge. * Created
+	  the concept of suspended MOH so it can be interrupted while
+	  conference join announcements to the user and DTMF features can
+	  operate. * Suspend any MOH until the user is about to actually
+	  join the mixing bridge of the conference. This way any pre-join
+	  file playback does not need to worry about MOH. * Made post-join
+	  actions only play deferred entry announcement files. Changing the
+	  user/conference state during that time is not protected or
+	  controlled by the state machine. (closes issue ASTERISK-20606)
+	  Reported by: Eugenia Belova Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/2232/
+
+2012-12-13 13:48 +0000 [r377947]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: Ensure Min-SE is included in outbound
+	  INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+	  value is not 90 (the default) and session timers are not
+	  disabled. This has the effect of Asterisk following RFC4028 more
+	  closely with regard to 422 responses and preventing situations in
+	  which Asterisk would be forced to temporarily accept a call to
+	  tear it down based on a Session-Expires below the locally
+	  configured Min-SE. (issue SWP-5051) Review:
+	  https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+	  Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-12 22:41 +0000 [r377923]  Rusty Newton <rnewton at digium.com>
+
+	* /, sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
+	  sounds/Makefile to 1.4.12 for new Extra Sounds releases See
+	  CHANGES-* files in English extra 1.4.12 tarballs for new sound
+	  prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
+	  (closes AST-755) Reported by: John Bigelow ........ Merged
+	  revisions 377922 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 21:57 +0000 [r377848-377882]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/timing.c, main/channel.c, main/data.c, main/stun.c, /,
+	  main/file.c, main/http.c, main/aoc.c, main/image.c, main/cel.c:
+	  Cleanup CLI commands on exit for several files. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  unregister-cli-multiple-all.patch (license #5909) patch uploaded
+	  by Corey Farrell ........ Merged revisions 377881 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/udptl.c, /: Cleanup udptl on exit. * Cleanup CLI commands on
+	  exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+	  Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell Modified ........ Merged revisions
+	  377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 20:48 +0000 [r377842]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_clialiases.c, /: Fix crash that can occur if CLI
+	  registration fails for an aliased command. A recent memory leak
+	  fix in main/cli.c causes an ast_cli_entry's command field to be
+	  freed and NULLed if ast_cli_register() fails. res_clialiases was
+	  ignoring the return value of ast_cli_register() and was then
+	  passing the NULL command off to a a hash function. This resulted
+	  in a crash. The fix is not to ignore the erroneous return value.
+	  If ast_cli_register() fails, then we do not continue trying to
+	  process the current alias. ........ Merged revisions 377840 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 20:42 +0000 [r377705-377838]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
+	  CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to
+	  taskprocessor.c missed in v10 -r374178. (issue ASTERISK-20649)
+	  Reported by: Corey Farrell Patches:
+	  taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell taskprocessor-cleanup-10-only.patch
+	  (license #5909) patch uploaded by Corey Farrell Modified ........
+	  Merged revisions 377837 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
+	  exit. * Unreference hints and statecbs containers on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
+	  Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
+	  Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell Modified ........ Merged revisions
+	  377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
+	  destroy verbosers and logchannels lists on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  logger-cleanup-all.patch (license #5909) patch uploaded by Corey
+	  Farrell Modified ........ Merged revisions 377771 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/indications.c: Cleanup indications on exit. * Made
+	  ast_unregister_indication_country() unlink the found tone zone
+	  before selecting a new default_tone_zone to make it impossible to
+	  select the tone zone being unregistered again. * Ringcadence is
+	  no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
+	  commands and destroy default_tone_zone on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  indications-cleanup-all.patch (license #5909) patch uploaded by
+	  Corey Farrell Modified ........ Merged revisions 377740 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/event.c: Cleanup event on exit. * Cleanup CLI commands on
+	  exit. * v10 only: Merged v1.8 -r374177 change to event.c missed
+	  in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey
+	  Farrell Patches: event_shutdown-10-only.patch (license #5909)
+	  patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch
+	  (license #5909) patch uploaded by Corey Farrell ........ Merged
+	  revisions 377708 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/dnsmgr.c, /: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
+	  and CLI commands on exit. (issue ASTERISK-20649) Reported by:
+	  Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
+	  patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
+	  (license #5909) patch uploaded by Corey Farrell Modified ........
+	  Merged revisions 377704 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 16:53 +0000 [r377624-377656]  Kinsey Moore <kmoore at digium.com>
+
+	* /, res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
+	  When using res_fax_digium, the T.38 CED tone was not being
+	  provided properly which would cause some incoming faxes to fail.
+	  This was not an issue with res_fax_spandsp since it does not
+	  strictly honor the send_ced flag and sends the CED tone whenever
+	  receiving a T.38 fax. (closes issue FAX-343) Reported-by:
+	  Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions
+	  377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Handle Session-Expires less than local
+	  Min-SE in 200 OK Ensure that a call is immediately torn down if a
+	  Session-Expires value received in a 200 OK is less than the local
+	  Min-SE. This also prevents Asterisk from allowing calls with
+	  Session-Expires below the RFC4028-mandated minimum (90s). (closes
+	  issue ASTERISK-20653) Review:
+	  https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
+	  ........ Merged revisions 377623 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 06:41 +0000 [r377558-377592]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* channels/chan_unistim.c, /: Fix codec mismatch Fix code to send
+	  in both rx and tx open stream messages correct codecs. Found that
+	  on phase 0/1 phones wrong codecs cause to no audio in some
+	  situations. (issue ASTERISK-20183) ........ Merged revisions
+	  377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_unistim.c, /: Fix crash on transfer initiated from
+	  insreeen menu on Unistim phones. Removed CDR-related code that
+	  moved to do_masquarade before. (closes issue ASTERISK-20417)
+	  Reported by: Rudolf Migalin ........ Merged revisions 377557 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.12.0-rc1 Released.
+
+2012-12-10 01:39 +0000 [r377504-377510]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* main/xmldoc.c, /: Improve documentation by making all of the
+	  colors used readable, no matter what the background color is.
+	  Dark blue on a black background is unreadable, as is yellow on a
+	  light background. This patch turns on the bright attribute for
+	  colors when on a dark background and turns *off* the bright
+	  attribute when the -W command line option is used (indicating a
+	  _light_ background). This ensures that text is readable in both
+	  cases. Patch by: tilghman Review:
+	  https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
+	  377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, addons/cdr_mysql.c: Remove some dead code and additionally
+	  handle a case that wasn't handled. ........ Merged revisions
+	  377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-08 00:29 +0000 [r377399-377432]  Richard Mudgett <rmudgett at digium.com>
+
+	* contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
+	  allow/disallow in MySQL contrib script. Using the contrib
+	  sippeers.sql script to create the sippeers MySQL table would
+	  result in being unable to place calls if you set the disallow
+	  value to all. (closes issue ASTERISK-20756) Reported by: Andre
+	  Luis Patches: sippeers.patch patch uploaded by Andre Luis
+	  ........ Merged revisions 377431 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
+	  allocation dumps. ........ Merged revisions 377398 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-07 21:58 +0000 [r377382]  Kinsey Moore <kmoore at digium.com>
+
+	* codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
+	  show" CLI command. In r306010 "Asterisk media architecture
+	  conversion - no more format bitfields", the logic for
+	  incrementing encoders and decoders when opening transcoder
+	  channels was changed without making the corresponding change when
+	  decrementing encoder / decoder channels. The result being that
+	  when a channel was destroyed, codec_dahdi couldn't properly tell
+	  if it was an encoder or decoder, and the default case is to
+	  assume it was a decoder. This could result in negative numbers
+	  for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
+	  encoders/decoders of 92 channels are in use. (closes issue
+	  ASTERISK-19921) Patch-by: Shaun Ruffell
+
+2012-12-06 23:56 +0000 [r377354]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_config_parser.c, apps/app_confbridge.c:
+	  confbridge: Fix some resource leaks on conference teardown. *
+	  Made destroy_conference_bridge() destroy a missed ast_mutex_t and
+	  ast_cond_t. * Made join_conference_bridge() init the
+	  ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
+	  destroy them unconditionally. * Made join_conference_bridge()
+	  abort if the new conference could not be added to the conferences
+	  container. * Made leave_conference() discard any post-join
+	  actions if join_conference_bridge() had to abort early. * Made
+	  the join_conference_bridge() diagnostic messages better describe
+	  what happened. * Renamed leave_conference_bridge() to
+	  leave_conference() and made it only take a conference user
+	  pointer. The conference pointer was redundant. * Made
+	  conf_bridge_profile_copy() use struct copy instead of memcpy(). *
+	  No need to lock the conference in start_conf_record_thread()
+	  since all of the callers already have it locked.
+
+2012-12-05 16:57 +0000 [r377261]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc
+	  on an already dealloced session When srtp_create fails, the
+	  session may be dealloced or just not alloced. At the same time
+	  though, the session pointer might not be set to NULL in this
+	  process and attempting to srtp_dealloc it again will cause a
+	  segfault. This patch checks for failure of srtp_create and sets
+	  the session pointer to NULL if it fails. (closes issue
+	  ASTERISK-20499) Reported by: tootai Review:
+	  https://reviewboard.asterisk.org/r/2228/ ........ Merged
+	  revisions 377256 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 16:49 +0000 [r377258]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
+	  connections. During the TLS re-work in chan_sip some TLS specific
+	  code was moved into a separate function. This function operates
+	  on a copy of the incoming SIP request. This copy was never
+	  deinitialized causing a memory leak for each request processed.
+	  This function is now given a SIP request structure which it can
+	  use to copy the incoming request into. This reduces the amount of
+	  memory allocations done since the internal allocated components
+	  are reused between packets and also ensures the SIP request
+	  structure is deinitialized when the TLS connection is torn down.
+	  (closes issue ASTERISK-20763) Reported by: deti ........ Merged
+	  revisions 377257 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 02:09 +0000 [r377038-377241]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/format.c: * Fix registering core show codecs/codec CLI
+	  commands twice. * Fix registering atexit format_attr_shutdown()
+	  more than once.
+
+	* apps/confbridge/conf_config_parser.c: confbridge: Fix several
+	  small issues. * Made func_confbridge_helper() allow an empty
+	  value when setting options. You previously could not
+	  Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
+	  dialplan. * Made func_confbridge_helper() handle its datastore
+	  better if multiple threads attempt to set the first CONFBRIDGE
+	  option value on the channel. * Made the func_confbridge_helper()
+	  only output one diagnostic message concerning the option. * Made
+	  the bridge video_mode able to repeatedly change in the config
+	  file and CONFBRIDGE dialplan function. The video_mode option
+	  values are an enum and not independent of each other. * Made
+	  handle_cli_confbridge_show_bridge_profile() better handle the
+	  video_mode option. * Simplified datastore handling code in
+	  conf_find_user_profile() and conf_find_bridge_profile(). * Made
+	  parse_bridge(), parse_user(), and parse_menu() use var->file
+	  instead of CONFBRIDGE_CONFIG because the var could have been from
+	  an include file. (closes issue ASTERISK-20655) Reported by:
+	  Birger "WIMPy" Harzenetter
+
+	* apps/app_confbridge.c: confbridge: Update online XML
+	  documentation.
+
+	* /, main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
+	  Convert atexits list to a mutex instead of a rd/wr lock. The lock
+	  is only write locked. * Move CLI verbose Asterisk ending message
+	  to where AMI message is output in really_quit() to avoid further
+	  surprises about using stuff already shutdown. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell ........ Merged
+	  revisions 377165 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, include/asterisk/_private.h, main/stdtime/localtime.c,
+	  main/asterisk.c: Cleanup core main on exit. * Cleanup time zones
+	  on exit. * Make exit clean/unclean report consistent for AMI and
+	  CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey
+	  Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch
+	  uploaded by Corey Farrell core-cleanup-11-trunk.patch (license
+	  #5909) patch uploaded by Corey Farrell Modified ........ Merged
+	  revisions 377135 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/config.c: Cleanup config cache on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  config-cleanup-all.patch (license #5909) patch uploaded by Corey
+	  Farrell ........ Merged revisions 377104 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/cli.c, /: Cleanup CLI resources on exit and CLI command
+	  registration errors. (issue ASTERISK-20649) Reported by: Corey
+	  Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
+	  uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+	  #5909) patch uploaded by Corey Farrell Modified ........ Merged
+	  revisions 377073 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
+	  do_reload() return handling since it never returned anything
+	  other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
+	  Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
+	  Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell Modified ........ Merged revisions
+	  377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/ccss.c: Fix CCSS CLI commands and logger level not
+	  unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
+	  Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
+	  Corey Farrell ........ Merged revisions 377037 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 21:33 +0000 [r376951]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
+	  RELEASE_COMPLETE in response to SETUP. Fix sending a
+	  RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+	  have a B channel available to assign to the call. (closes issue
+	  ABE-2869) Reported by: Guenther Kelleter Patches:
+	  setup-reject_2.diff (license #6372) patch uploaded by Guenther
+	  Kelleter Modified ........ Merged revision 376949 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ........ Merged revisions 376950 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 17:06 +0000 [r376920]  Sean Bright <sean at malleable.com>
+
+	* /, funcs/func_volume.c: Minor spelling fix to the VOLUME
+	  documentation. ........ Merged revisions 376919 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 16:23 +0000 [r376916]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Fix potential crashes during SIP attended
+	  transfers. The principal behind this patch is simple. During a
+	  transfer, we manipulate channels that are owned by a separate
+	  thread than the one we currently are running in, so it makes
+	  sense that we need to grab a reference to the channels so that
+	  they cannot disappear out from under us. In the wild, crashes
+	  were sometimes seen when the transferring party would hang up the
+	  call before the transfer target answered the call. The most
+	  common place to see the crash occur was when attempting to send a
+	  connected line update to the transferer channel. (closes issue
+	  ASTERISK-20226) Reported by Jared Smith Patches:
+	  ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+	  Tested by: Jared Smith ........ Merged revisions 376901 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 22:58 +0000 [r376865-376869]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
+	  local_devicestate(). Regression introduced by ASTERISK-20390 fix.
+	  (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
+	  rmudgett ........ Merged revisions 376868 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+	  ........ Merged revisions 376864 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 21:51 +0000 [r376835]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* /, channels/chan_sip.c: Improve Code Readability And Fix Setting
+	  natdetected Flag For 1.8, 10, 11 and trunk we are are improving
+	  the code readability. For 11 and trunk, auto nat detection was
+	  added. The natdetected flag was being set to 1 when the host
+	  address in the VIA header did not specifiy a port. This patch
+	  fixes this by setting the port on the temporary sock address used
+	  to SIP_STANDARD_PORT in order for the sock address comparison to
+	  work properly. (closes issue ASTERISK-20724) Reported by: Michael
+	  L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2206/ ........ Merged
+	  revisions 376834 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 00:45 +0000 [r376759-376789]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG atexit
+	  unreleased malloc memory summary. * Adds the following CLI
+	  commands to control MALLOC_DEBUG reporting of unreleased malloc
+	  memory when Asterisk is shut down. memory atexit list on memory
+	  atexit list off memory atexit summary byline memory atexit
+	  summary byfunc memory atexit summary byfile memory atexit summary
+	  off * Made check all remaining allocated region blocks atexit for
+	  fence violations. * Increased the allocated region hash table
+	  size by about three times. It still isn't large enough
+	  considering the number of malloced blocks Asterisk uses. * Made
+	  CLI "memory show allocations anomalies" use
+	  regions_check_all_fences(). Review:
+	  https://reviewboard.asterisk.org/r/2196/ ........ Merged
+	  revisions 376788 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
+	  "memory show allocations" misspelling of anomalies option. The
+	  command will still accept the original misspelling. *
+	  Miscellaneous tweaks to CLI "memory show allocations" command
+	  output format. * Made CLI "memory show summary" summarize by line
+	  number instead of by function if a filename is given. * Made CLI
+	  "memory show summary" sort its output by filename or
+	  function-name/line-number depending upon request. * Miscellaneous
+	  tweaks to CLI "memory show summary" command output format.
+	  ........ Merged revisions 376758 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-28 16:30 +0000 [r376726]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c, /: manager: Make challenge work with
+	  allowmultiplelogin=no Prior to this patch, challenge would yield
+	  a multiple logins error if used without providing the username
+	  (which isn't really supposed to be an argument to challenge) if
+	  allowmultiplelogin was set to no because allowmultiplelogin finds
+	  a user with a zero length login name. This check is simply
+	  disabled for the challenge action when the username is empty by
+	  this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+	  Patches: challenge_action_nomultiplelogin.diff uploaded by
+	  Jonathan Rose (license 6182) ........ Merged revisions 376725
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-27 23:58 +0000 [r376628-376689]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
+	  char. The '-' char is supposed to be ignored by the dialplan
+	  extension matching. Unfortunately, it's treatment is not handled
+	  consistently throughout the extension matching code. * Made the
+	  old exten matching code consistently ignore '-' chars. * Made the
+	  old exten matching code consistently handle case in the matching.
+	  * Made ignore empty character sets. * Fixed ast_extension_cmp()
+	  to return -1, 0, or 1 as documented. The only user of it in
+	  pbx_lua.c was testing for -1. It was originally returning the
+	  strcmp() value for less than which is not usually going to be -1.
+	  * Fix character set sorting if the sets have the same number of
+	  characters and start with the same character. Character set [0-9]
+	  now sorts before [02-9a] as originally intended. * Updated some
+	  extension label and priority already in use warnings to also
+	  indicate if the extension is aliased. (closes issue
+	  ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+	  Harzenetter Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/2201/ ........ Merged
+	  revisions 376688 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
+	  pbx/pbx_dundi.c: Remove unnecessary channel module references. *
+	  Removed call to ast_module_user_hangup_all() in
+	  res_config_mysql.c since it is effectively a noop. No channels
+	  can attach a reference to that module. * Removed call to
+	  ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+	  of unload_module() has already called it. * Removed redundant
+	  channel module references in pbx_dundi.c. The registered dialplan
+	  function callback dispatchers for the read/read2/write callbacks
+	  already reference the module before calling. * pbx_dundi: Moved
+	  unregistering CLI commands, DUNDi switch, and dialplan functions
+	  to the first thing the unload_module() does. This will reduce the
+	  chance of new channels using DUNDi services while the module is
+	  being torn down. ........ Merged revisions 376657 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
+	  and use better names. * Update doxygen of AST_LIST_REMOVE().
+	  ........ Merged revisions 376627 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-22 23:56 +0000 [r376587]  Matthew Jordan <mjordan at digium.com>
+
+	* main/lock.c, /, main/logger.c, include/asterisk/lock.h:
+	  Re-initialize logmsgs mutex upon logger initialization to prevent
+	  lock errors Similar to the patch that moved the fork earlier in
+	  the startup sequence to prevent mutex errors in the recursive
+	  mutex surrounding the read/write thread registration lock, this
+	  patch re-initializes the logmsgs mutex. Part of the start up
+	  sequence before forking the process into the background includes
+	  reading asterisk.conf; this has to occur prior to the call to
+	  daemon in order to read startup parameters. When reading in a
+	  conf file, log statements can be generated. Since this can't be
+	  avoided, the mutex instead is re-initialized to ensure a reset of
+	  any thread tracking information. This patch also includes some
+	  additional debugging to catch errors when locking or unlocking
+	  the recursive mutex that surrounds locks when the DEBUG_THREADS
+	  build option is enabled. DO_CRASH or THREAD_CRASH will cause an
+	  abort() if a mutex error is detected. (issue ASTERISK-19463)
+	  Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
+	  376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-20 17:01 +0000 [r376522]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c, channels/sip/include/sip.h: Add "Require:
+	  timer" to 200 OK responses when appropriate. The method by which
+	  the Require header is added to 200 responses is inspired by the
+	  method that Olle Johansson uses in his darjeeling-prack branch.
+	  (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
+	  behest of Olle Johansson Review:
+	  https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
+	  376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-19 19:44 +0000 [r376470]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* /, channels/chan_sip.c, main/security_events.c,
+	  main/indications.c: Fix most leftover non-opaque ast_str uses.
+	  Instead of calling str->str, one should use ast_str_buffer(str).
+	  Same goes for str->used as ast_str_strlen(str) and str->len as
+	  ast_str_size(str). Review:
+	  https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+	  376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-18 20:18 +0000 [r376414-376431]  Matthew Jordan <mjordan at digium.com>
+
+	* /, main/utils.c, main/stdtime/localtime.c, main/asterisk.c:
+	  Reorder startup sequence to prevent lockups when process is sent
+	  to background Although it is very rare and timing dependent, the
+	  potential exists for the call to 'daemon' to cause what appears
+	  to be a deadlock in Asterisk during startup. This can occur when
+	  a recursive mutex is obtained prior to the daemon call executing.
+	  Since daemon uses fork to send the process into the background,
+	  any threading primitives are unsafe to re-use after the call.

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