[asterisk-commits] bebuild: tag 10.12.0-rc1 r377518 - /tags/10.12.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Dec 9 19:52:00 CST 2012


Author: bebuild
Date: Sun Dec  9 19:51:58 2012
New Revision: 377518

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377518
Log:
Importing files for 10.12.0-rc1 release.

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    tags/10.12.0-rc1/ChangeLog   (with props)

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+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.
+	  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
+
+	* 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.
+
+2012-11-16 19:41 +0000 [r376390]  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
+
+2012-11-16 00:08 +0000 [r376315-376342]  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
+
+	* 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
+
+2012-11-15 22:55 +0000 [r376308]  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
+
+2012-11-15 01:50 +0000 [r376263]  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
+
+2012-11-14 19:50 +0000 [r376233]  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
+
+2012-11-13 18:20 +0000 [r376208]  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
+
+2012-11-12 20:44 +0000 [r376167]  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
+
+2012-11-12 20:15 +0000 [r376143]  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
+	  changes from checking if the extension contains an underscore to
+	  checking if the extension begins with an underscore. (closes
+	  issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
+	  Steven T. Wheeler, Michael L. Young Patches:
+	  asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
+	  L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2188/ ........ Merged
+	  revisions 376142 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-08 21:59 +0000 [r376088]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
+	  Turns out the "helpful" setting of ms and res in this macro is
+	  completely useless after the timeout antipattern fix. If you're a
+	  new guy looking to write code, don't write a macro like this one.
+	  ........ Merged revisions 376087 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-08 21:07 +0000 [r376030-376059]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/sig_ss7.c: chan_dahdi/SS7: Made reject incoming call
+	  for an in-alarm or blocked channel. If a SS7 call comes in
+	  requesting a CIC that is in-alarm, the call is accepted and
+	  connects if the extension exists in the dialplan. The call does
+	  not have any audio. * Made release the call immediately with
+	  circuit congestion cause. (closes issue ASTERISK-20204) Reported
+	  by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
+	  #5621) patch uploaded by rmudgett ........ Merged revisions
+	  376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* include/asterisk/utils.h, include/asterisk/astmm.h, /,
+	  main/utils.c, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG
+	  enhancements. * Makes malloc() behave like calloc(). It will
+	  return a memory block filled with 0x55. A nonzero value. * Makes
+	  free() fill the released memory block and boundary fence's with
+	  0xdeaddead. Any pointer use after free is going to have a pointer
+	  pointing to 0xdeaddead. The 0xdeaddead pointer is usually an
+	  invalid memory address so a crash is expected. * Puts the freed
+	  memory block into a circular array so it is not reused
+	  immediately. * When the circular array rotates out a memory block
+	  to the heap it checks that the memory has not been altered from
+	  0xdeaddead. * Made the astmm_log message wording better. * Made
+	  crash if the DO_CRASH menuselect option is enabled and something
+	  is found. * Fixed a potential alignment issue on 64 bit systems.
+	  struct ast_region.data[] should now be aligned correctly for all
+	  platforms. * Extracted region_check_fences() from
+	  __ast_free_region() and handle_memory_show(). * Updated
+	  handle_memory_show() CLI usage help. Review:
+	  https://reviewboard.asterisk.org/r/2182/ ........ Merged
+	  revisions 376029 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-07 17:16 +0000 [r375995]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c,
+	  main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c,
+	  res/res_fax.c, apps/app_record.c, channels/chan_agent.c,
+	  main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
+	  channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
+	  channels/chan_dahdi.c, apps/app_waitforring.c,
+	  channels/sig_analog.c: Multiple revisions 375993-375994 ........
+	  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov
+	  2012) | 30 lines Fix misuses of timeouts throughout the code.
+	  Prior to this change, a common method for determining if a
+	  timeout was reached was to call a function such as
+	  ast_waitfor_n() and inspect the out parameter that told how many
+	  milliseconds were left, then use that as the input to
+	  ast_waitfor_n() on the next go-around. The problem with this is
+	  that in some cases, submillisecond timeouts can occur, resulting
+	  in the out parameter not decreasing any. When this happens
+	  thousands of times, the result is that the timeout takes much
+	  longer than intended to be reached. As an example, I had a
+	  situation where a 3 second timeout took multiple days to finally
+	  end since most wakeups from ast_waitfor_n() were under a
+	  millisecond. This patch seeks to fix this pattern throughout the
+	  code. Now we log the time when an operation began and find the
+	  difference in wall clock time between now and when the event
+	  started. This means that sub-millisecond timeouts now cannot play
+	  havoc when trying to determine if something has timed out. Part
+	  of this fix also includes changing the function ast_waitfor() so
+	  that it is possible for it to return less than zero when a
+	  negative timeout is given to it. This makes it actually possible
+	  to detect errors in ast_waitfor() when there is no timeout.
+	  (closes issue ASTERISK-20414) reported by David M. Lee Review:
+	  https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
+	  mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
+	  lines Remove some debugging that accidentally made it in the last
+	  commit. ........ Merged revisions 375993-375994 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-06 18:27 +0000 [r375965]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/features.h, main/channel.c, /, main/features.c,
+	  include/asterisk/channel.h, .cleancount: Fix stuck DTMF when
+	  bridge is broken. When a bridge is broken by an AMI Redirect
+	  action or the ChannelRedirect application, an in progress DTMF
+	  digit could be stuck sending forever. * Made simulate a DTMF end
+	  event when a bridge is broken and a DTMF digit was in progress.
+	  (closes issue ASTERISK-20492) Reported by: Jeremiah Gowdy
+	  Patches: bridge_end_dtmf-v3.patch.txt (license #6358) patch
+	  uploaded by Jeremiah Gowdy Modified to
+	  jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch
+	  (license #5621) patch uploaded by rmudgett Tested by: rmudgett
+	  Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged
+	  revisions 375964 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.11.0 Released.
+
+2012-12-06  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.11.0-rc3 Released.
+
+	* chan_local: Fix local_pvt ref leak in local_devicestate().
+
+	Regression introduced by ASTERISK-20390 fix.
+
+	(closes issue ASTERISK-20769)
+	Reported by: rmudgett
+
+2012-12-05  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.11.0-rc2 Released.
+
+	* 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
+
+2012-11-06  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.11.0-rc1 Released.
+
+2012-11-05 23:00 +0000 [r375894]  Matthew Jordan <mjordan at digium.com>
+
+	* main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+	  res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+	  bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+	  include/asterisk/timing.h, res/res_musiconhold.c,
+	  channels/chan_iax2.c, res/res_fax_spandsp.c,

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