[asterisk-commits] bebuild: tag 11.2.0-rc1 r377526 - /tags/11.2.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Dec 9 19:58:53 CST 2012


Author: bebuild
Date: Sun Dec  9 19:58:49 2012
New Revision: 377526

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377526
Log:
Importing files for 11.2.0-rc1 release.

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    tags/11.2.0-rc1/ChangeLog   (with props)

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--- tags/11.2.0-rc1/ChangeLog (added)
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+2012-12-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.2.0-rc1 Released.
+
+2012-12-10 01:41 +0000 [r377505-377511]  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
+	  ........ Merged revisions 377510 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, 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
+	  ........ Merged revisions 377504 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-09 01:22 +0000 [r377462]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_motif.c: Add missing support for "who hung up" to
+	  chan_motif. (closes issue ASTERISK-20671) Reported by: Matt
+	  Jordan Review: https://reviewboard.asterisk.org/r/2208/
+
+2012-12-08 00:29 +0000 [r377401-377433]  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 ........ Merged
+	  revisions 377432 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, 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 ........ Merged
+	  revisions 377399 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-07 22:02 +0000 [r377383]  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 ........ Merged revisions
+	  377382 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-06 23:58 +0000 [r377355]  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. ........ Merged
+	  revisions 377354 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-06 17:28 +0000 [r377340]  Russell Bryant <russell at russellbryant.com>
+
+	* main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl
+	  show' CLI command allows you to show the details about a specific
+	  named ACL in acl.conf. This patch adds tab completion to the
+	  command. Review: https://reviewboard.asterisk.org/r/2230/
+
+2012-12-06 14:11 +0000 [r377319]  Matthew Jordan <mjordan at digium.com>
+
+	* main/manager.c: Fix memory leak in 'manager show event' when
+	  command entered incorrectly When the CLI command 'manager show
+	  event' was run incorrectly and its usage instructions returned, a
+	  reference to the event container was leaked. This would prevent
+	  the container from being reclaimed when Asterisk exits. We now
+	  properly decrement the count on the ao2 object using the nifty
+	  RAII_VAR macro. Thanks to Russell for helping me stumble on this,
+	  and Terry for writing that ridiculously helpful macro.
+
+2012-12-05 17:08 +0000 [r377262]  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 ........ Merged
+	  revisions 377261 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-05 16:50 +0000 [r377259]  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 ........ Merged
+	  revisions 377258 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-05 02:19 +0000 [r377213-377244]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/format.c, /: Fix registering core show codecs/codec CLI
+	  commands twice. ........ Merged revisions 377241 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* 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(). (closes
+	  issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter
+	  ........ Merged revisions 377227 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, apps/app_confbridge.c: confbridge: Update online XML
+	  documentation. ........ Merged revisions 377212 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-04 12:59 +0000 [r377195]  Russell Bryant <russell at russellbryant.com>
+
+	* contrib/scripts/install_prereq: Add libuuid to install_prereq for
+	  Fedora. I ran this script and my build failed. pjproject requires
+	  this.
+
+2012-12-03 22:58 +0000 [r377039-377167]  Richard Mudgett <rmudgett at digium.com>
+
+	* 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 ........ Merged
+	  revisions 377166 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* main/asterisk.c, /, include/asterisk/_private.h,
+	  main/stdtime/localtime.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 ........ Merged
+	  revisions 377136 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* 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 ........ Merged
+	  revisions 377105 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* 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 ........ Merged
+	  revisions 377074 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* 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
+	  ........ Merged revisions 377070 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, 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 ........ Merged
+	  revisions 377038 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-03 14:54 +0000 [r377021]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_motif.c: Fix an RTP instance reference count leak
+	  in chan_motif. When setting up an RTP instance the RTCP portion
+	  of the instance keeps a reference to the instance itself. In
+	  order to release this reference and stop RTCP the stop API call
+	  must be called before destroying the instance. (closes issue
+	  ASTERISK-20751) Reported by: joshoa
+
+2012-12-01 00:46 +0000 [r376983]  Joshua Colp <jcolp at digium.com>
+
+	* configs/motif.conf.sample, channels/chan_motif.c: Tweak extension
+	  used for incoming calls received on Motif. Based on feedback from
+	  numerous individuals this patch tweaks incoming calls to first
+	  look for an extension with the name of the endpoint. If no such
+	  extension exists the call will silently fall back to the "s"
+	  extension as it previously did.
+
+2012-11-30 21:35 +0000 [r376952]  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 ........ Merged
+	  revisions 376951 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-30 17:07 +0000 [r376921]  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 ........ Merged
+	  revisions 376920 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-30 16:36 +0000 [r376917]  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 ........ Merged
+	  revisions 376916 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 22:59 +0000 [r376866-376870]  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 ........ Merged
+	  revisions 376869 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+	  ........ Merged revisions 376864 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376865 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 21:57 +0000 [r376836]  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 ........ Merged
+	  revisions 376835 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 17:17 +0000 [r376822]  Pedro Kiefer <pedro at kiefer.com.br>
+
+	* channels/chan_sip.c: Fix chan_sip websocket payload handling
+	  Websocket by default doesn't return an ast_str for the payload
+	  received. When converting it to an ast_str on chan_sip the last
+	  character was being omitted, because ast_str functions expects
+	  that the given length includes the trailing 0x00. payload_len
+	  only has the actual string length without counting the trailing
+	  zero. For most cases this passed unnoticed as most of SIP
+	  messages ends with \r\n. (closes issue ASTERISK-20745) Reported
+	  by: Iñaki Baz Castillo Review:
+	  https://reviewboard.asterisk.org/r/2219/
+
+2012-11-29 00:46 +0000 [r376760-376790]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/asterisk.c, /, main/astmm.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 ........ Merged
+	  revisions 376789 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, 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 ........ Merged
+	  revisions 376759 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-28 16:37 +0000 [r376727]  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 ........
+	  Merged revisions 376726 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-28 00:08 +0000 [r376629-376690]  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 ........ Merged
+	  revisions 376689 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* 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 ........ Merged
+	  revisions 376658 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* /, 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 ........ Merged
+	  revisions 376628 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-22 23:58 +0000 [r376588]  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
+	  ........ Merged revisions 376587 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-20 21:58 +0000 [r376561]  David M. Lee <dlee at digium.com>
+
+	* res/res_http_websocket.c: Added missing newlines to websocket
+	  ast_logs. Without these newlines, log messages just continue
+	  tacking onto the same line, and do not flush immediately.
+
+2012-11-20 18:57 +0000 [r376550]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/sip/include/sip.h, /, channels/chan_sip.c: 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
+	  ........ Merged revisions 376522 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-20 17:37 +0000 [r376540]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_sip.c: Reduce CLI spam of "Extension Changed"
+	  device state messages. Asterisk 11 follows RFC3265 that states
+	  that after every subscribe or resubscribe a notify should be
+	  sent. Thus the console if filled continuously with the following
+	  after every subscribe; == Extension Changed 8512[phones] new
+	  state IDLE for Notify User cisco1 In Asterisk 1.8 only changes
+	  would be sent. Thus only when a device state changed was anything
+	  emitted to the console. fix: Only print to console when device
+	  state isn't forced. (closes issue ASTERISK-20706) Reported by:
+	  alecdavis Tested by: alecdavis alecdavis (license 585)
+
+2012-11-19 19:54 +0000 [r376471]  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
+	  ........ Merged revisions 376470 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-18 20:22 +0000 [r376415-376441]  Matthew Jordan <mjordan at digium.com>
+
+	* main/asterisk.c, /, main/utils.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. Implementations of pthread
+	  recursive mutexes are highly likely to store the thread
+	  identifier of the thread that previously obtained the mutex. If
+	  the mutex was locked prior to the fork, a subsequent unlock
+	  operation will potentially fail as the thread identifier is no
+	  longer valid. Since the mutex is still locked, all subsequent
+	  attempts to grab the mutex by other threads will block. This
+	  behavior exhibited itself most often when DEBUG_THREADS was
+	  enabled, as this compile time option surrounds the mutexes in
+	  Asterisk with another recursive mutex that protects the storage
+	  of thread related information. This made it much more likely that
+	  a recursive mutex would be obtained prior to daemon and unlocked
+	  after the call. This patch does the following: a) It backports a
+	  patch from Asterisk 11 that prevents the spawning of the
+	  localtime monitoring thread. This thread is now spawned after
+	  Asterisk has fully booted. b) It re-orders the startup sequence
+	  to call daemon earlier during Asterisk startup. This limits the
+	  potential of threading primitives being accessed by
+	  initialization calls before daemon is called. c) It removes calls
+	  to ast_verbose/ast_log/etc. prior to daemon being called.
+	  Developers should send error messages directly to stderr prior to
+	  daemon, as calls to ast_log may access recursive mutexes that
+	  store thread related information. d) It reorganizes when thread
+	  local storage is created for storing lock information during the
+	  creation of threads. Prior to this patch, the read/write lock
+	  protecting the list of threads in ast_register_thread would
+	  utilize the lock in the thread local storage prior to it being
+	  initialized; this patch prevents that. On a very related note,
+	  this patch will *greatly* improve the stability of the Asterisk
+	  Test Suite. Review: https://reviewboard.asterisk.org/r/2197
+	  (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
+	  mjordan ........ Merged revisions 376428 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376431 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* apps/confbridge/conf_state.c, /: Add a test event that reports
+	  changes in ConfBridge state This patch adds a test event to
+	  ConfBridge that reports transitions between states in ConfBridge.
+	  This is used by tests in the Asterisk Test Suite that verify
+	  state changes based on the entering/leaving of conference
+	  participants. ........ Merged revisions 376414 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-16 19:59 +0000 [r376391]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_monitor.c, /: monitor: prevent attempts to move/remove
+	  recordings skipped with 'i' and 'o'. The i and o options for
+	  monitor skip the input and output sides of a recording
+	  respectively. This patch addresses a problem in those options
+	  when monitor is called without specifying a specific filename
+	  where monitor will try to move the recording that was skipped.
+	  Since this usually doesn't exist when these options are used, it
+	  would produce a warning when it does this in most cases, but it
+	  is conceivable that there are use cases where this could result
+	  in moving/removing a file unintentionally. (closes issue
+	  ASTERISK-20641) Reported by: Jonathan Rose Review:
+	  https://reviewboard.asterisk.org/r/2190/ ........ Merged
+	  revisions 376389 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376390 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-16 00:09 +0000 [r376339-376343]  David M. Lee <dlee at digium.com>
+
+	* /, utils/extconf.c: Fixed extconf.c breakage introduced in
+	  r376306. To quote wdoekes: > Note that I'm not confirming
+	  legitimacy of having that file in tree at > all. Is anyone using
+	  aelparse/conf2ael? ........ Merged revisions 376340 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376342 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* utils/Makefile, tests/test_astobj2_thrash.c (added),
+	  utils/utils.xml, /, utils/hashtest.c (removed),
+	  tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
+	  include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit
+	  tests. Both hashtest and hashtest2 are manual testing apps that
+	  thrash hash tables (hashtab and ao2 containers, respectively), by
+	  spinning up several threads that randomly insert, delete, lookup
+	  and iterate over the hash table. If the app doesn't crash, the
+	  hash table probably passes the test. Those utils are not a part
+	  of the typical Asterisk build, so they do not usually get
+	  compiled. This all makes them less that useful. This patch
+	  removes those manual test programs and replaces them with
+	  Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+	  also attempts to make the tests more deterministic. * Rather than
+	  spinning up some number of threads that operate on the hash table
+	  randomly, spin up four threads that concurrenly add, remove,
+	  lookup and iterate over the hash table. * Each thread checks the
+	  state of the hash table both during and after execution, and
+	  indicates a test failure if things are not as expected. * Each
+	  thread times out after 60 seconds to prevent deadlocking the unit
+	  test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+	  Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+	  revisions 376306 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376315 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-15 23:03 +0000 [r376310]  Jonathan Rose <jrose at digium.com>
+
+	* /, apps/app_meetme.c: app_meetme: Fix channels lingering when
+	  hung up under certain conditions Channels would get stuck and
+	  MeetMe would repeatedly display an Unable to write frame to
+	  channel error in the conf_run function if hung up during certain
+	  sound prompts such as during user count announcements. This patch
+	  fixes that by reintroducing a hangup check in the meetme's main
+	  loop (also in conf_run). (closes issue ASTERISK-20486) Reported
+	  by: Michael Cargile Review:
+	  https://reviewboard.asterisk.org/r/2187/ Patches:
+	  meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
+	  Rose (license 6182) ........ Merged revisions 376307 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376308 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-15 02:08 +0000 [r376264]  Rusty Newton <rnewton at digium.com>
+
+	* apps/app_voicemail.c, /: Patch to play correct sound file when a
+	  voicemail's urgent status is removed We were attempting to play
+	  "vm-urgent-removed", which didn't exist. Now we play
+	  "vm-marked-nonurgent" which exists and is the correct sound file.
+	  Previous behavior was silence and a warning on the CLI. (issue
+	  ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
+	  Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
+	  uploaded by Rusty Newton (license 5829) ........ Merged revisions
+	  376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 376263 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-14 19:53 +0000 [r376234]  Richard Mudgett <rmudgett at digium.com>
+
+	* pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
+	  Future dated call files are ignored when astspooldir is relative
+	  to the current directory. The queue_file() assumed that the qdir
+	  needed to be prepended if the given filename did not start with a
+	  '/'. If astspooldir is relative it is not going to start from the
+	  root directory obviously so it will not start with a '/'. The
+	  filename used in queue_file() ultimately results in qdir
+	  prepended multiple times. * Made queue_file() not prepend qdir if
+	  the filename contains a '/'. (closes issue ASTERISK-20593)
+	  Reported by: James Le Cuirot Patches:
+	  0004-Fix-future-call-files-from-relative-directories.patch
+	  (license #6439) patch uploaded by James Le Cuirot ........ Merged
+	  revisions 376232 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376233 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-13 18:48 +0000 [r376217]  Brent Eagles <beagles at digium.com>
+
+	* main/channel.c, /: Patch to prevent stopping the active generator
+	  when it is not the silence generator. This patch introduces an
+	  internal helper function to safely check whether the current
+	  generator is the one that is expected before deactivating it. The
+	  current externally accessible ast_channel_stop_generator()
+	  function has been modified to be implemented in terms of the new
+	  function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
+	  ........ Merged revisions 376199 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376208 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-12 20:45 +0000 [r376168]  Joshua Colp <jcolp at digium.com>
+
+	* main/pbx.c, /: Properly check if the "Context" and "Extension"
+	  headers are empty in a ShowDialPlan action. The code which
+	  handles the ShowDialPlan action wrongly assumed that a non-NULL
+	  return value from the function which retrieves headers from an
+	  action indicates that the header has a value. This is incorrect
+	  and the contents must be checked to see if they are blank.
+	  (closes issue ASTERISK-20628) Reported by: jkroon Patches:
+	  asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
+	  ........ Merged revisions 376166 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 376167 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-12 20:16 +0000 [r376144]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
+	  Problem When adding a dynamic hint, if an extension contains an
+	  underscore no variable subsitution is being performed. This patch

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