[asterisk-commits] bebuild: tag 11.8.0-rc1 r405532 - /tags/11.8.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jan 14 13:37:51 CST 2014


Author: bebuild
Date: Tue Jan 14 13:37:48 2014
New Revision: 405532

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=405532
Log:
Importing files for 11.8.0-rc1 release.

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    tags/11.8.0-rc1/ChangeLog   (with props)

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+2014-01-14  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.8.0-rc1 Released.
+
+2014-01-14 18:43 +0000 [r405434-405487]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: No BYE message sent after
+	  INVITE with Replaces Setting channel state DOWN is an unnecessary
+	  step that was only being done in handle_invite_replaces(). This
+	  changes that by removing the call and reducing locking. (closes
+	  issue ASTERISK-23010) Reported by: Ryan Tilton Review:
+	  https://reviewboard.asterisk.org/r/3116/ ........ Merged
+	  revisions 405486 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound
+	  register regression In ASTERISK-12117, an improvement to insure
+	  consistant local from tags on outbound registrations resulted in
+	  an undesirable behavior - caused by leftover unexpired sip_pvt
+	  dialogs (with the previous cseq number), resulting in many
+	  uncessary REGISTER requests. Instead of significant rework of
+	  transmit_register(), this change deletes the dialogs after a 200
+	  OK response indiciating a successful registration, keeping the
+	  old dialogs from interfering with normal operation. (closes issue
+	  ASTERISK-22946) Reported by: Stephan Eisvogel Review:
+	  https://reviewboard.asterisk.org/r/3109/ ........ Merged
+	  revisions 405433 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-14 17:26 +0000 [r405431]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/asterisk.c, configs/logger.conf.sample, main/cli.c,
+	  include/asterisk/logger.h, main/pbx.c, main/manager.c,
+	  funcs/func_timeout.c, apps/app_dumpchan.c, main/logger.c,
+	  UPGRADE.txt, apps/app_verbose.c: verbosity: Fix performance of
+	  console verbose messages. The per console verbose level feature
+	  as previously implemented caused a large performance penalty. The
+	  fix required some minor incompatibilities if the new rasterisk is
+	  used to connect to an earlier version. If the new rasterisk
+	  connects to an older Asterisk version then the root console
+	  verbose level is always affected by the "core set verbose"
+	  command of the remote console even though it may appear to only
+	  affect the current console. If an older version of rasterisk
+	  connects to the new version then the "core set verbose" command
+	  will have no effect. * Fixed the verbose performance by not
+	  generating a verbose message if nothing is going to use it and
+	  then filtered any generated verbose messages before actually
+	  sending them to the remote consoles. * Split the "core set debug"
+	  and "core set verbose" CLI commands to remove the per module
+	  verbose support that cannot work with the per console verbose
+	  level. * Added a silent option to the "core set verbose" command.
+	  * Fixed "core set debug off" tab completion. * Made "core show
+	  settings" list the current console verbosity in addition to the
+	  root console verbosity. * Changed the default verbose level of
+	  the 'verbose' setting in the logger.conf [logfiles] section. The
+	  default is now to once again follow the current root console
+	  level. As a result, using the AMI Command action with "core set
+	  verbose" could again set the root console verbose level and
+	  affect the verbose level logged. (closes issue AST-1252) Reported
+	  by: Guenther Kelleter Review:
+	  https://reviewboard.asterisk.org/r/3114/
+
+2014-01-14 15:32 +0000 [r405362-405380]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Hangup transferer/transferee when
+	  transfer to Parking fails When performing a SIP transfer to a
+	  Park extension, if the Park fails, chan_sip will currently not
+	  hang up either the transferer or the transfer target. This
+	  results in the channels being orphaned with no thread to service
+	  frames, resulting in stuck channels. This patch immediately hangs
+	  up the two channels if a Park fails. (closes issue
+	  ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes
+	  issue ASTERISK-23047) Reported by: Tommy Thompson Tested by:
+	  Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107
+
+	* res/Makefile: res/Makefile: alias dist-clean to distclean A 'make
+	  distclean' is equivalent to 'make dist-clean' in the top most
+	  Makefile. This patch updates the res/Makefile to recognize both
+	  distclean and dist-clean. Note that this is needed for removing
+	  build.mak, which can run into problems if the source file of
+	  Asterisk or its path is changed after build.mak is generated.
+	  (issue ASTERISK-22480) Reported by: Matt Jordan
+
+2014-01-10 17:50 +0000 [r405281]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/logger.c: Logging callid: Fix some sizeof() references per
+	  coding guidelines.
+
+2014-01-09 16:49 +0000 [r405234]  Kevin Harwell <kharwell at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Fails to resume WebRTC
+	  call from hold In ast_rtp_ice_start if the ice session create
+	  check list failed, start check was never initiated and
+	  ice_started was never set to true. Upon re-entering the function
+	  (for instance, [un]hold) it would try to create the check list
+	  again with duplicate remote candidates. Fixed so that if the
+	  create check list fails the necessary data structures are
+	  properly re-initialized for any subsequent retries. Note, it was
+	  decided to not stop ice support (by calling ast_rtp_ice_stop) on
+	  a check list failure because it possible things might still work.
+	  However, a debug message was added to help with any future
+	  troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis
+	  Valentinavičius Patches: works_on_my_machine.patch uploaded by
+	  xytis (license 6558)
+
+2014-01-09 15:41 +0000 [r405215]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/confbridge/conf_state_multi_marked.c, apps/app_confbridge.c:
+	  app_confbridge: Fix crash caused when waitmarked/marked users
+	  leave together When waitmarked users join a ConfBridge, the
+	  conference state is transitioned from EMPTY -> INACTIVE. In this
+	  state, the users are maintined in a waiting users list. When a
+	  marked user joins, the ConfBridge conference transitions from
+	  INACTIVE -> MULTI_MARKED, and all users are put onto the active
+	  list of users. This process works correctly. When the marked user
+	  leaves, if they are the last marked user, the MULTI_MARKED state
+	  does the following: (1) It plays back a message to the bridge
+	  stating that the leader has left the conference. This requires an
+	  unlocking of the bridge. (2) It moves waitmarked users back to
+	  the waiting list (3) It transitions to the appropriate state: in
+	  this case, INACTIVE However, because it plays the prompt back to
+	  the bridge before moving the users and before finishing the state
+	  transition, this creates a race condition: with the bridge
+	  unlocked, waitmarked users who leave the conference (or are
+	  kicked from it) can cause a state transition of the bridge to
+	  another state before the conference is transitioned to the
+	  INACTIVE state. This causes the state machine to get a bit wonky,
+	  often leading to a crash when the MULTI_MARKED state attempts to
+	  conclude its processing. This patch fixes this problem: (1) It
+	  prevents kicked users from being kicked again. That's just a
+	  nicety. (2) More importantly, it fixes the race condition by only
+	  playing the prompt once the state has transitioned correctly to
+	  INACTIVE. If waitmarked users sneak out during the prompt being
+	  played, no harm no foul. Review:
+	  https://reviewboard.asterisk.org/r/3108/ (closes issue AST-1258)
+	  Reported by: Steve Pitts
+
+2014-01-09 14:12 +0000 [r405161]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* apps/app_dumpchan.c, /: "Minimun" typo. ........ Merged revisions
+	  405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-08 16:17 +0000 [r405081-405091]  Kinsey Moore <kmoore at digium.com>
+
+	* /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support
+	  for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
+	  available on newer operating systems. (closes issue
+	  ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
+	  Reported by: George Joseph Patch by: George Joseph ........
+	  Merged revisions 405090 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* UPGRADE.txt, /: UPGRADE: Add a note about non-functionality Add a
+	  note that the "retry on 403 response to REGISTER" for chan_sip is
+	  non-functional in the versions in which it was first introduced.
+	  ........ Merged revisions 405088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Add the missing part of r400140 When the
+	  patch to add retry-on-forbidden-response was committed, part of
+	  the patch for chan_sip was not committed which caused the feature
+	  to be entirely nonfunctional. This corrects the code in question.
+	  (closes issue ASTERISK-17138) Review:
+	  https://reviewboard.asterisk.org/r/2874 ........ Merged revisions
+	  405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-03 22:24 +0000 [r404888]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
+	  on rasterisk Even since the fixes of AST-2013-007, Asterisk
+	  prints the following warning on startup if the user decided to
+	  live dangerously: Privilege escalation protection disabled! See
+	  https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
+	  message is intended for the logs and interactive startup. No need
+	  for it to appear on a remote console. This commit removes it from
+	  there. (closes issue ASTERISK-23084) Review:
+	  https://reviewboard.asterisk.org/r/3101/ ........ Merged
+	  revisions 404861 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-03 21:58 +0000 [r404773-404858]  Kevin Harwell <kharwell at digium.com>
+
+	* cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading
+	  Upon reload the module unconditionally "unloaded" the module
+	  (freeing memory and setting pointers to NULL) and then when
+	  attempting a "load" if the config file had not changed then
+	  nothing would be reinitialized. By moving the "unload" to occur
+	  conditionally (reload only) after an attempted configuration
+	  load, but before module "loading" alleviates the issue. The
+	  module now loads/unloads/reloads correctly. (closes issue
+	  ASTERISK-22871) Reported by: Matteo ........ Merged revisions
+	  404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices
+	  PRI channel dnid on output dahdi show channels output slices the
+	  callerid (which is dnid copied over on PRI channels). If the
+	  channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
+	  then the output slices 1408409XXXX down to 1408409XXX. This patch
+	  just opens it up to 15 chars so you can see the whole thing.
+	  (closes issue ASTERISK-22918) Reported by: outtolunc Patches:
+	  svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
+	  (license 5198) ........ Merged revisions 404784 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_meetme.c: app_meetme: compiler warning Fixed a
+	  compiler warning (errors in 'dev-mode') given by gcc version
+	  4.8.1. The one in app_meetme involved the
+	  'sizeof-pointer-memaccess' (see:
+	  http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it
+	  would no longer issue a warning and can compile again in
+	  'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
+	  ........ Merged revisions 404742 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-02 19:35 +0000 [r404675]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* /, funcs/func_strings.c: func_strings: use memmove to prevent
+	  overlapping memory on strcpy When calling REPLACE() with an empty
+	  replace-char argument, strcpy is used to overwrite the the
+	  matching <find-char>. However as the src and dest arguments to
+	  strcpy must not overlap, it causes other parts of the string to
+	  be overwritten with adjacent characters and the result is
+	  mangled. Patch replaces call to strcpy with memmove and adds a
+	  test suite case for REPLACE. (closes issue ASTERISK-22910)
+	  Reported by: Gareth Palmer Review:
+	  https://reviewboard.asterisk.org/r/3083/ Patches:
+	  func_strings.patch uploaded by Gareth Palmer (license 5169)
+	  ........ Merged revisions 404674 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-31 21:26 +0000 [r404579-404604]  Kevin Harwell <kharwell at digium.com>
+
+	* /, cel/cel_pgsql.c: cel_pgsql: deadlock on unload and
+	  core_event_dispatcher A deadlock can happen between a thread
+	  unloading or reloading the cel_pgsql module and the
+	  core_event_dispatcher taskprocessor thread. Description of what
+	  is happening: Thread 1 (for example, a netconsole thread): a
+	  "module reload cel_pgsql" is launched the thread enter the
+	  "my_unload_module" function (cel_pgsql.c) the thread acquire the
+	  write lock on psql_columns the thread enter the
+	  "ast_event_unsubscribe" function (event.c) the thread try to
+	  acquire the write lock on ast_event_subs[sub->type] Thread 2
+	  (core_event_dispatcher taskprocessor thread): the taskprocessor
+	  pop a CEL event the thread enter the "handle_event" function
+	  (event.c) the thread acquire the read lock on
+	  ast_event_subs[sub->type] the thread callback the "pgsql_log"
+	  function (cel_pgsql.c), since it's a subscriber of CEL events the
+	  thread try to acquire a read lock on psql_columns (closes issue
+	  ASTERISK-22854) Reported by: Etienne Lessard Patches:
+	  cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
+	  6394) ........ Merged revisions 404603 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/channel.c: channels.c: core show channeltypes slicing 'core
+	  show channeltypes' type column is being sliced, resulting in
+	  incomplete type names. (closes issue ASTERISK-22919) Reported by:
+	  outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded by
+	  outtolunc (license 5198)
+
+2013-12-20 21:15 +0000 [r404457]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* /, main/say.c: say.c: correct time for polish In
+	  ast_say_date_with_format_pl(), change ast_say_number() to use
+	  tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
+	  by: Robert Mordec Review:
+	  https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
+	  uploaded by veilen (license 6555) ........ Merged revisions
+	  404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-19 16:57 +0000 [r404344-404351]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* res/res_fax.c: res_fax.c: crash on framehook with no dsp in fax
+	  detect In fax_detect_framehook() a null pointer reference can
+	  occur where a voice frame is processed but no dsp is attached to
+	  the fax detection structure. The code block that rejects frames
+	  that detection cannot be processed on is checking for dsp but
+	  falls through when it should instead return, as this change
+	  implements. (closes issue ASTERISK-22942) Reported by: adomjan
+	  Review: https://reviewboard.asterisk.org/r/3076/
+
+	* main/db.c: astdb: crash in sqlite3 during shutdown When Asterisk
+	  is shut down, the astdb_atexit() function releases (finalize) the
+	  previously initiated (prepared) SQL statements in sqlite3.
+	  Another thread making a subsequent request can cause a crash in
+	  sqlite3. This patch eliminates that issue by resetting the
+	  statement pointer after it is released/cleared. The sqlite3 code
+	  detects the null pointer, and aborts the operation cleanly.
+	  (closes issue AST-1265) Reported by: Alexander Hömig (closes
+	  issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
+	  Review: https://reviewboard.asterisk.org/r/3078/
+
+2013-12-19 08:15 +0000 [r404318]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c,
+	  addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/oochannels.c:
+	  Handle temporary failures on gk registration Introduce new
+	  'stopped' state for gk client and restart gk client on failures
+	  Remove ooh323 stack command lock as it is not need now. (closes
+	  issue ASTERISK-21960) Reported by: Dmitry Melekhov Patches:
+	  ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested
+	  by: Dmitry Melekhov
+
+2013-12-18 22:34 +0000 [r404275]  Jason Parker <jparker at digium.com>
+
+	* main/manager.c: Add AMI event for presence state. Review:
+	  https://reviewboard.asterisk.org/r/3039/
+
+2013-12-18 20:19 +0000 [r404219]  Richard Mudgett <rmudgett at digium.com>
+
+	* addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler
+	  warnings. ........ Merged revisions 404212 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-18 11:59 +0000 [r404136]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_calendar.c: res_calendar: Protect channel when adding
+	  datastore. This change adds a missing channel lock when adding a
+	  datastore to a channel. ........ Merged revisions 404135 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-18 00:28 +0000 [r404045-404087]  Rusty Newton <rnewton at digium.com>
+
+	* /, funcs/func_strings.c: func_strings: Documentation fix for
+	  QUOTE() Example output was inaccurate. (issue ASTERISK-22970)
+	  (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
+	  func_strings.patch uploaded by Gareth Palmer (license 5169)
+	  ........ Merged revisions 404081 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/rtp_engine.c, channels/chan_iax2.c, apps/app_chanspy.c,
+	  apps/app_mixmonitor.c, include/asterisk/test.h, main/channel.c:
+	  Several components: fixing Typos in comments and code,
+	  "avaliable" instead of "available" (issue ASTERISK-23021) (closes
+	  issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty
+	  Newton Patches: available.patch uploaded by Jeremy Lainé (license
+	  6561)
+
+2013-12-16 17:14 +0000 [r403917]  David M. Lee <dlee at digium.com>
+
+	* funcs/func_realtime.c, main/pbx.c, main/tcptls.c,
+	  funcs/func_db.c, /, README-SERIOUSLY.bestpractices.txt,
+	  configs/asterisk.conf.sample, funcs/func_shell.c,
+	  funcs/func_env.c, funcs/func_lock.c, UPGRADE.txt,
+	  include/asterisk/pbx.h, main/asterisk.c: security: Inhibit
+	  execution of privilege escalating functions This patch allows
+	  individual dialplan functions to be marked as 'dangerous', to
+	  inhibit their execution from external sources. A 'dangerous'
+	  function is one which results in a privilege escalation. For
+	  example, if one were to read the channel variable SHELL(rm -rf /)
+	  Bad Things(TM) could happen; even if the external source has only
+	  read permissions. Execution from external sources may be enabled
+	  by setting 'live_dangerously' to 'yes' in the [options] section
+	  of asterisk.conf. Although doing so is not recommended. (closes
+	  issue ASTERISK-22905) Review:
+	  http://reviewboard.digium.internal/r/432/ ........ Merged
+	  revisions 403913 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-16 15:55 +0000 [r403855-403863]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
+	  prevent memory corruption During dialplan execution in
+	  pbx_extension_helper(), the contexts global read lock prevents
+	  link list corruption, but was released with a pointer to the
+	  ast_exten and data later used in variable substitution. Instead,
+	  this patch removes pbx_substitute_variables() and locates a copy
+	  of the ast_exten data on the stack before releasing the lock,
+	  where ast_exten could get free'd by another thread performing a
+	  module reload. (issue AST-1179) Reported by: Thomas Arimont
+	  (issue AST-1246) Reported by: Alexander Hömig Review:
+	  https://reviewboard.asterisk.org/r/3055/ ........ Merged
+	  revisions 403862 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd
+	  length 16 bit message This patch prevents an infinite loop
+	  overwriting memory when a message is received into the
+	  unpacksms16() function, where the length of the message is an odd
+	  number of bytes. (closes issue ASTERISK-22590) Reported by: Jan
+	  Juergens Tested by: Jan Juergens ........ Merged revisions 403853
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-11 19:14 +0000 [r403635]  Russell Bryant <russell at russellbryant.com>
+
+	* /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf
+	  reload If you set a peer's outboundproxy and then removed it from
+	  the config, this would not get picked up in a config reload. This
+	  patch fixes that by resetting it in set_peer_defaults(). Closes
+	  ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
+	  ........ Merged revisions 403634 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-09 03:11 +0000 [r403450]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_fax_spandsp.c, /: res_fax_spandsp: Always init T.38
+	  session to avoid crashes during state change Prior to this patch,
+	  res_fax_spandsp was conservative with how it initialized the
+	  spandsp T.38 context. It would only initialize it if the driver
+	  thought the current state was a T.38 fax. While this works fine
+	  in nominal situations, in certain off nominal situations,
+	  res_fax_spandsp can believe that a T.38 fax will not occur when
+	  in fact one has started. In particular, this was discovered when
+	  res_fax would fall back to audio after timing out on a T.38
+	  upgrade. The SIP channel driver would continue to retry the
+	  re-INVITE and - if the remote end responded after res_fax timed
+	  out with a 200 OK - a T.38 frame would be delivered to the
+	  res_fax stack when it no longer expected it. As it turns out,
+	  there does not appear to be any downside to always initializing
+	  the T.38 context, other than the actual memory allocation. Since
+	  that avoids this off nominal situation (and others which are
+	  equally likely hard to predict), this is the safest way to avoid
+	  this problem. Much thanks to Torrey as well for providing a
+	  scenario that reproduces this issue. (closes issue
+	  ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
+	  Searle patches: always-init-t38.patch uploaded by awinters
+	  (License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
+	  ........ Merged revisions 403449 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-02 17:55 +0000 [r403288]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: Check and reject non-digits e164 values on
+	  peers and general sections in ooh323.conf Regenerate e164
+	  endpoint list on reload ooh323 (issue ASTERISK-22901) Reported
+	  by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch
+
+2013-11-22 17:11 +0000 [r403015]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/translate.c: translate: Move freeing of frame to after it
+	  is used. When translating from one format to another it is
+	  possible to inform the translation function that the source frame
+	  should be freed. This was previously done immediately but shortly
+	  afterwards the frame that was freed was accessed and used again.
+	  This change moves code around a bit so that the frame is now
+	  freed after it has been completely used. (closes issue
+	  ASTERISK-22788) Reported by: Corey Farrell Patches:
+	  translate-access-after-free-11up.patch uploaded by coreyfarrell
+	  (license 5909) translate-access-after-free-1.8.patch uploaded by
+	  coreyfarrell (license 5909) ........ Merged revisions 403014 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-12 15:00 +0000 [r402709]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID
+	  read Asterisk will sometimes core dump during caller id read on
+	  analog channels due to a negative return value from the read() in
+	  my_get_callerid that slips through as a negative length argument
+	  to callerid_feed() if the errno returned by DAHDI is ELAST. This
+	  change ensures that the negative return is treated properly even
+	  when it is ELAST. (closes issue ASTERISK-22746) Reported by:
+	  Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
+	  uploaded by Michael Walton (License 6502) ........ Merged
+	  revisions 402708 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-11 19:26 +0000 [r402686]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_confbridge.c: Get rid of some inaccurate comments. I'm
+	  doing some unrelated work in app_confbridge and finding these
+	  "invalid pin" comments to be annoying. Get out!
+
+2013-11-11 15:35 +0000 [r402646]  Kinsey Moore <kmoore at digium.com>
+
+	* /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
+	  current app_queue code from 1.8 up to trunk the upper and lower
+	  penalties can be set to 0 but the value is interpreted to be
+	  disabled instead of actually setting limits. This is especially
+	  evident if min and max limits are set to 0 and members with
+	  penalties of 0 and 1 are in the queue since the member with
+	  penalty 1 will still receive calls. This patch adjusts the
+	  special disabled value to be INT_MAX instead of 0. (closes issue
+	  ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
+	  Reported by: Schmooze Com ........ Merged revisions 402645 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-08 22:48 +0000 [r402605]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+	  keep same local (from) tag for outgoing register requests For
+	  outbound register requests the tag on the From line was updated
+	  every 20 seconds prior to a successful registration and also once
+	  for each registration renewal. That behavior can possibly cause
+	  the registration to be denied because of the different tag, and
+	  is not aligned with the intention of RFC 3261 8.1.3.5 "...
+	  request constitutes a new transaction and SHOULD have the same
+	  value of the Call-ID, To, and From of the previous request...".
+	  This updates chan_sip to have a field to keep the local tag in
+	  the registration structure and use that tag for registration
+	  requests where the callid is also unchanged. (closes issue
+	  ASTERISK-12117) Reported by: Pawel Pierscionek Review:
+	  https://reviewboard.asterisk.org/r/2988/ ........ Merged
+	  revisions 402604 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-05 15:11 +0000 [r402450-402469]  Kevin Harwell <kharwell at digium.com>
+
+	* /: Recorded merge of revisions 402468 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  chan_sip: notify dialog info ignores presentation indicator in
+	  callerid The presentation indicator in a callerid (e.g. set by
+	  dialplan function Set(CALLERID(name-pres)= ...)) is not checked
+	  when SIP Dialog Info Notifies are generated during extension
+	  monitoring. Added a check to make sure the name and/or number
+	  presentations on the callee (remote identity) are set to allow.
+	  If they are restricted then "anonymous" is used instead. (closes
+	  issue AST-1175) Reported by: Thomas Arimont Review:
+	  https://reviewboard.asterisk.org/r/2976/
+
+	* channels/chan_sip.c: chan_sip: notify dialog info ignores
+	  presentation indicator in callerid The presentation indicator in
+	  a callerid (e.g. set by dialplan function
+	  Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+	  Info Notifies are generated during extension monitoring. Added a
+	  check to make sure the name and/or number presentations on the
+	  callee (remote identity) are set to allow. If they are restricted
+	  then "anonymous" is used instead. (closes issue AST-1175)
+	  Reported by: Thomas Arimont Review:
+	  https://reviewboard.asterisk.org/r/2976/
+
+2013-11-02 02:11 +0000 [r402407-402425]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_state_inactive.c,
+	  apps/confbridge/conf_state_single_marked.c,
+	  apps/confbridge/include/confbridge.h,
+	  apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
+	  apps/confbridge/conf_state_multi_marked.c,
+	  apps/confbridge/conf_state.c,
+	  apps/confbridge/conf_state_single.c: confbridge: Separate user
+	  muting from system muting overrides. The system overrides the
+	  user muting requests when MOH is playing or a waitmarked user is
+	  waiting for a marked user to join. System muting overrides
+	  interfere with what the user may wish the muting to be when the
+	  system override ends. * User muting requests are now independent
+	  of the system muting overrides. The effective muting is now the
+	  logical or of the user request and system override. * Added a
+	  Muted column to the CLI "confbridge list <conference>" command. *
+	  Added a Muted header to the AMI ConfbridgeList action
+	  ConfbridgeList event. (closes issue AST-1102) Reported by: John
+	  Bigelow Review: https://reviewboard.asterisk.org/r/2960/
+
+	* main/config.c, configs/confbridge.conf.sample: config: Allow
+	  ConfBridge DTMF menus to have '#' as the first digit. ConfBridge
+	  allows custom DTMF menus to be created in the confbridge.conf
+	  file by assigning a DTMF key sequence to a sequence of actions as
+	  follows: DTMF-sequence = action,action... Unfortunately, the
+	  normal config file processing code interprets an initial '#'
+	  character as starting a directive such as #include. * Add the
+	  ability to escape the first non-blank character in a config line
+	  so the '#' character can be used without triggering the directive
+	  processing code. (closes issue AFS-2) (closes issue
+	  ASTERISK-22478) Reported by: Nicolas Tanski Patches:
+	  jira_asterisk_22478_v11.patch (license #5621) patch uploaded by
+	  rmudgett (modified) Review:
+	  https://reviewboard.asterisk.org/r/2969/
+
+2013-11-01 12:31 +0000 [r402345]  Kinsey Moore <kmoore at digium.com>
+
+	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+	  channels/chan_sip.c: chan_sip: Fix RTCP port for SRFLX ICE
+	  candidates This corrects one-way audio between Asterisk and
+	  Chrome/jssip as a result of Asterisk inserting the incorrect RTCP
+	  port into RTCP SRFLX ICE candidates. This also exposes an ICE
+	  component enumeration to extract further details from candidates.
+	  (closes issue ASTERISK-21383) Reported by: Shaun Clark Review:
+	  https://reviewboard.asterisk.org/r/2967/
+
+2013-10-31 15:59 +0000 [r402288]  Matthew Jordan <mjordan at digium.com>
+
+	* main/loader.c, /: core/loader: Don't call dlclose in a while loop
+	  For awhile now, we've noticed continuous integration builds
+	  hanging on CentOS 6 64-bit build agents. After resolving a number
+	  of problems with symbols, strange locks, and other shenanigans,
+	  the problem has persisted. In all cases, gdb shows the Asterisk
+	  process stuck in loader.c on one of the infinite while loops that
+	  calls dlclose repeatedly until success. The documentation of
+	  dlclose states that it returns 0 on success; any other value on
+	  error. It does not state that repeatedly calling it will
+	  eventually clear those errors. Most likely, the repeated calls to
+	  dlclose was to force a close by exhausting the references on the
+	  library; however, that will never succeed if: (a) There is some
+	  fundamental error at work in the loaded library that precludes
+	  unloading it (b) Some other loaded module is referencing a symbol
+	  in the currently loaded module This results in Asterisk sitting
+	  forever. Since we have matching pairs of dlopen/dlclose, this
+	  patch opts to only call dlclose once, and log out as an ERROR if
+	  dlclose fails to return success. If nothing else, this might help
+	  to determine why on the CentOS 6 64-bit build agent things are
+	  not closing successfully. Review:
+	  https://reviewboard.asterisk.org/r/2970 ........ Merged revisions
+	  402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-29 23:42 +0000 [r402225]  Rusty Newton <rnewton at digium.com>
+
+	* sounds/Makefile, /: Updates for 1.4.25 core sounds and 1.4.14
+	  extra sounds, plus new en_GB language set The new sound packages
+	  relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
+	  ASTERISK-20782 Modified sounds/Makefile for the new sound
+	  versions and to account for the new en_GB language set. (issue
+	  ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
+	  ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged
+	  revisions 402224 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-29 12:49 +0000 [r402151]  Matthew Jordan <mjordan at digium.com>
+
+	* main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c:
+	  Remove some spammy debug messages; improve clarity of others
+	  Debug messages aren't free. Even when the debug level is
+	  sufficiently low such that the messages are never evaluated,
+	  there is a cost to having to parse Asterisk logs that contain
+	  debug messages that (a) fail to convey sufficient information or
+	  (b) occur so frequently as to be next to meaningless. Based on
+	  having to stare at lots of DEBUG messages, this patch makes the
+	  following changes: * channel.c: When copying variables from a
+	  parent channel to a child channel, specify the channels involved.
+	  Do not log anything for a variable that is not inherited; the
+	  fact that it doesn't have an _ or __ already signifies that it
+	  won't be inherited. * pbx.c: Specify what function evaluation has
+	  occurred that created the result. * translate.c: Bump up the
+	  translator path messages to 10. I've never once had to use these
+	  debug messages, and for each format that is registered (on
+	  startup) and unregistered (on shutdown) the entire f^2 matrix is
+	  logged out. For short tests in the Asterisk Test Suite, this
+	  should make finding the actual test much easier. * xmldoc.c: The
+	  debug message that 'blah' is not found in the tree is expected.
+	  Often, description elements - which are not required - are not
+	  provided. This debug message adds no additional value, as it is
+	  not indicative of an error or helpful in debugging which element
+	  did not contain a 'blah' element as a child. If an element is
+	  supposed to contain a child element, then that XML tree should
+	  have failed validation in the first place. Review:
+	  https://reviewboard.asterisk.org/r/2966/ ........ Merged
+	  revisions 402150 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-28 14:50 +0000 [r402111]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_sip.c, UPGRADE.txt: chan_sip: Clarify 'Forcerport'
+	  Setting Displayed When Running "sip show peers" While looking at
+	  ASTERISK-22236, Walter Doekes pointed out that when running "sip
+	  show peers", the setting being displayed can be confusing. The
+	  display of "N" used to mean NAT (i.e. yes). The NAT setting has
+	  gone through many different changes resulting in the display of
+	  different characters to try and convey what the current setting
+	  is for 'Forcerport' (A for Auto and Forcerport is currently on, a
+	  for Auto but Forcerport is off, Y for yes, and N for no). During
+	  the initial code review to try and clarify these settings
+	  (especially since "N" no longer meant what it used to mean in
+	  prior versions of Asterisk), Mark Michelson suggested using the
+	  full space available to display the settings which helped to make
+	  the settings very clear. That was a great suggestion. Therefore,
+	  this patch does the following: * The column for 'Forcerport' now
+	  will show: Auto (Yes), Auto (No), Yes, or No. * A column for the
+	  'Comedia' setting has been added. It too will display the setting
+	  in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No. *
+	  UPGRADE.txt has been updated to document this change. (closes
+	  issue ASTERISK-22728) Reported by: Walter Doekes Tested by:
+	  Michael L. Young Patches:
+	  asterisk-forcerport-display-clarification_v3.diff uploaded by
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2941
+
+2013-12-17  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.7.0 Released.
+
+2013-12-16  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.7.0-rc2 Released.
+
+	* AST-2013-006 - app_sms: BufferOverflow when receiving odd length 16
+	  bit message
+
+	  This patch prevents an infinite loop overwriting memory when a
+	  message is received into the unpacksms16() function, where the length
+	  of the message is an odd number of bytes.
+	  (closes issue ASTERISK-22590)
+
+	* AST-2013-007 - security: Inhibit execution of privilege escalating
+	  functions
+
+	  This patch allows individual dialplan functions to be marked as
+	  'dangerous', to inhibit their execution from external sources.
+
+	  A 'dangerous' function is one which results in a privilege
+	  escalation. For example, if one were to read the channel variable
+	  SHELL(rm -rf /) Bad Things(TM) could happen; even if the external
+	  source has only read permissions.
+
+	  Execution from external sources may be enabled by setting
+	  'live_dangerously' to 'yes' in the [options] section of
+	  asterisk.conf. Although doing so is not recommended.
+

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