[asterisk-commits] bebuild: tag 1.8.26.0-rc1 r405528 - /tags/1.8.26.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jan 14 13:33:21 CST 2014


Author: bebuild
Date: Tue Jan 14 13:33:18 2014
New Revision: 405528

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=405528
Log:
Importing files for 1.8.26.0-rc1 release.

Added:
    tags/1.8.26.0-rc1/.lastclean   (with props)
    tags/1.8.26.0-rc1/.version   (with props)
    tags/1.8.26.0-rc1/ChangeLog   (with props)

Added: tags/1.8.26.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/.lastclean?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/.lastclean (added)
+++ tags/1.8.26.0-rc1/.lastclean Tue Jan 14 13:33:18 2014
@@ -1,0 +1,1 @@
+40

Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.8.26.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/.version?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/.version (added)
+++ tags/1.8.26.0-rc1/.version Tue Jan 14 13:33:18 2014
@@ -1,0 +1,1 @@
+1.8.26.0-rc1

Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.8.26.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/ChangeLog?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/ChangeLog (added)
+++ tags/1.8.26.0-rc1/ChangeLog Tue Jan 14 13:33:18 2014
@@ -1,0 +1,47186 @@
+2014-01-14  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.26.0-rc1 Released.
+
+2014-01-14 18:35 +0000 [r405433-405486]  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/
+
+	* 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/
+
+2014-01-14 15:31 +0000 [r405379]  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 (closes issue ASTERISK-23047)
+	  Reported by: Tommy Thompson Review:
+	  https://reviewboard.asterisk.org/r/3107
+
+2014-01-09 14:11 +0000 [r405160]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* apps/app_dumpchan.c: "Minimun" typo.
+
+2014-01-08 16:00 +0000 [r405033-405090]  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
+
+	* 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.
+
+	* 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
+
+2014-01-06 17:31 +0000 [r404951]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* funcs/func_strings.c: func_strings: fix for memmove patch test In
+	  r404674 the AST_TEST_DEFINE(test_REPLACE) test was added that
+	  made use of a function that doesn't exist in 1.8. This fixes that
+	  by reverting to directly accessing chan varshead. Reported by:
+	  Tzafrir Cohen (issue ASTERISK-22910)
+
+2014-01-03 22:06 +0000 [r404861]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* main/asterisk.c: asterisk.c: supress 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/
+
+2014-01-03 21:57 +0000 [r404742-404857]  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
+
+	* 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)
+
+	* apps/app_meetme.c, channels/chan_unistim.c: chan_unistim.c,
+	  app_meetme: compiler warnings Fixed a couple of compiler warnings
+	  (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. The one in
+	  chan_unistim was issuing an array out of bounds message. Fixed
+	  both so they would no longer issue warnings and can compile again
+	  in 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
+
+2014-01-02 19:32 +0000 [r404674]  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)
+
+2013-12-31 21:25 +0000 [r404603]  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)
+
+2013-12-20 21:12 +0000 [r404456]  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)
+
+2013-12-18 19:47 +0000 [r404212]  Richard Mudgett <rmudgett at digium.com>
+
+	* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/memheap.c,
+	  addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
+	  addons/ooh323c/src/perutil.c, addons/ooh323cDriver.c,
+	  addons/ooh323c/src/ooSocket.c: ooh323c: Fix gcc 4.6.3 compiler
+	  warnings.
+
+2013-12-18 11:58 +0000 [r404135]  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.
+
+2013-12-18 00:27 +0000 [r404044-404081]  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)
+
+	* 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 16:36 +0000 [r403913]  David M. Lee <dlee at digium.com>
+
+	* include/asterisk/pbx.h, main/asterisk.c, 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: 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/
+
+2013-12-16 15:53 +0000 [r403853-403862]  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/
+
+	* 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
+
+2013-12-11 19:11 +0000 [r403634]  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/
+
+2013-12-09 03:10 +0000 [r403449]  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)
+
+2013-11-22 17:10 +0000 [r403014]  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)
+
+2013-11-12 14:55 +0000 [r402645-402708]  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)
+
+	* 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
+
+2013-11-08 22:46 +0000 [r402604]  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/
+
+2013-11-05 15:08 +0000 [r402468]  Kevin Harwell <kharwell at digium.com>
+
+	* 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/
+
+2013-10-31 15:57 +0000 [r402287]  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
+
+2013-10-29 23:41 +0000 [r402224]  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)
+
+2013-10-29 14:52 +0000 [r402192]  David M. Lee <dlee at digium.com>
+
+	* configure, configure.ac, makeopts.in, Makefile: Backport r373119
+	  from 11 to go along with RAII_VAR support. In order to use nested
+	  functions on some versions of GCC (e.g. GCC on OS X), the
+	  -fnested-functions flag must be passed to the compiler. This
+	  patch adds detection logic to ./configure to add the flag if
+	  necessary.
+
+2013-10-29 12:40 +0000 [r402150]  Matthew Jordan <mjordan at digium.com>
+
+	* main/translate.c, main/xmldoc.c, main/channel.c, main/pbx.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/
+
+2013-12-17  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.25.0 Released.
+
+2013-12-16  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.25.0-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.
+
+	  (closes issue ASTERISK-22905)
+
+2013-10-28  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.25.0-rc1 Released.
+
+2013-10-25 21:51 +0000 [r401959-402000]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
+	  rtp payloads copy and improve argument names In function
+	  ast_rtp_instance_early _bridge_make_compatible the use of
+	  instance 0/1 as arguments doesn't clearly communicate a direction
+	  that the copying of payloads from the source channel to the
+	  destination channel will occur, making it more probable to have
+	  the arguments to ast_rtp_codecs_payloads_copy() put in the
+	  reverse order. This patch renames the arguments with _dst and
+	  _src suffixes and corrects the copy direction.
+
+	* include/asterisk/pbx.h, main/pbx.c: pbx.c: fix confused match
+	  caller id that deleted exten still in hash This fixes a bug where
+	  a zero length callerid match adjacent to a no match callerid
+	  extension entry would be deleted together, which then resulted in
+	  hashtable references to free'd memory. A third state of the
+	  matchcid value has been added to indicate match to any extension
+	  which allows enforcing comparison of matchcid on/off without
+	  errors. (closes issue AST-1235) Reported by: Guenther Kelleter
+	  Review: https://reviewboard.asterisk.org/r/2930/
+
+2013-10-25 17:21 +0000 [r401619-401914]  Jonathan Rose <jrose at digium.com>
+
+	* utils/clicompat.c: Put clicompat-r2.patch back in We've figured
+	  out how to resolve the problems this was causing in 12/trunk, so
+	  this can go back in now. (issue ASTERISK-22467) Reported by:
+	  Corey Farrell Patches: clicompat-r2.patch uploaded by
+	  coreyfarrell (license 5909)
+
+	* utils/clicompat.c: revert clicompat-r2.patch from r401704 Patch
+	  caused the following build errors against testsuite
+	  https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
+	  (issue ASTERISK-22467) Reported by: Corey Farrell
+
+	* main/utils.c: utils: Fix memory leaks and missed unregistration
+	  of CLI commands on shutdown Final set of patches in a series of
+	  memory leak/cleanup patches by Corey Farrell (closes issue
+	  ASTERISK-22467) Reported by: Corey Farrell Patches:
+	  main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
+	  main-utils-11.patch uploaded by coreyfarrell (license 5909)
+	  main-utils-12up.patch uploaded by coreyfarrell (license 5909)
+
+	* tests/test_linkedlists.c: test_linkedlists: Fix memory leak
+	  (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+	  test_linkedlists-1.8.patch uploaded by coreyfarrell (license
+	  5909) test_linkedlists-11up.patch uploaded by coreyfarrell
+	  (license 5909)
+
+	* main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
+	  reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+	  jitterbuf-jb_reset-leak-1.8.patch
+	  jitterbuf-jb_reset-leak-11up.patch
+
+	* main/astobj2.c: astobj2: Unregister debug CLI commands at exit
+	  (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+	  astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
+	  (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
+	  coreyfarrell (license 5909)
+
+	* apps/app_voicemail.c: app_voicemail: Memory Leaks against tests
+	  (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+	  app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
+	  app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
+
+	* main/asterisk.c, utils/clicompat.c, channels/chan_dahdi.c,
+	  codecs/ilbc/doCPLC.c, main/data.c, main/app.c: memory leaks:
+	  Memory leak cleanup patch by Corey Farrell (second set) Also
+	  covers ast_app_parse_timelen-fail-zero-length.patch, but the
+	  patch was replaced with one of my own. (issue ASTERISK-22467)
+	  Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch
+	  uploaded by coreyfarrell (license 5909) clicompat-r2.patch
+	  uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch
+	  uploaded by coreyfarrell (license 5909)
+	  data-cleanup-test-registration.patch uploaded by coreyfarrell
+	  (license 5909) main-asterisk-kill-listener.patch uploaded by
+	  coreyfarrell (license 5909)
+
+	* tests/test_dlinklists.c, funcs/func_math.c,
+	  channels/sip/reqresp_parser.c, main/test.c,
+	  main/editline/readline.c: memory leaks: Memory leak cleanup patch
+	  by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
+	  Corey Farrell Patches:
+	  chan_sip-parse_contact_header_test-free-contacts.patch uploaded
+	  by coreyfarrell (license 5909) cli-filename-completion-leak.patch
+	  uploaded by coreyfarrell (license 5909) func_math.patch uploaded
+	  by corefarrell (license 5909) main-test-cleanup.patch uploaded by
+	  coreyfarrell (license 5909) test_dlinklists.patch uploaded by
+	  coreyfarrell (license 5909)
+
+	* main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
+	  Address jittery DTMF events in RTP streams (closes issue
+	  ASTERISK-21170) Reported by: NITESH BANSAL Patches:
+	  dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
+	  Review: https://reviewboard.asterisk.org/r/2938/
+
+2013-10-23 16:34 +0000 [r401577]  Richard Mudgett <rmudgett at digium.com>
+
+	* cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a filter
+	  when the CDR value is empty. Extra CDR records are written if a
+	  filtered CDR value is empty because the filter is not checked.
+	  (closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria
+
+2013-10-23 15:19 +0000 [r401537]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_mgcp.c: chan_mgcp: Properly handle malformed media
+	  lines This corrects a situation in which a media line was not
+	  parsed properly and resulted in a crash. (closes issue
+	  ASTERISK-21190) Reported by: adomjan Patches:
+	  chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
+
+2013-10-23 11:10 +0000 [r401497]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Fix an issue where an incompatible audio
+	  format may be added to SDP. If preferred codecs included any
+	  non-audio format the code would mistakenly add the audio format,
+	  even if it was not a joint capability with the remote side.
+	  (closes issue ASTERISK-21131) Reported by: nbougues Patches:
+	  patch_unsupported_codec_1.8.patch uploaded by nbougues (license
+	  6470)
+
+2013-10-22 22:36 +0000 [r401445]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP is
+	  not available during SSRC change In r400089, a patch was put in
+	  to correct erroneous RTCP statistic resets. Unfortunately,
+	  ast_rtp_read can be called on an RTP instance that does not have
+	  RTCP information. This patch prevents that crash by only
+	  resetting the statistics if we do actually have an RTCP instance.
+	  (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
+	  Bigelow
+
+2013-10-22 00:13 +0000 [r401378]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_analog.c: chan_dahdi: Fix unable to get index
+	  warning when transferring an analog call. Transferring an analog
+	  call using flashhooks generated an unable to get index WARNING
+	  message when the transfer is completed. * Removed unnecessary
+	  analog subchannel shell games when transferring a call using
+	  flashhooks. Thanks to Tzafrir Cohen for mentioning this in a
+	  comment on issue ASTERISK-22720.
+
+2013-10-21 19:45 +0000 [r401325]  Kevin Harwell <kharwell at digium.com>
+
+	* main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
+	  tgetstr(), when libncurses5-dev isn't installed Include the
+	  appropriate declarations when not using termcap, but term+curses
+	  and [n]curses do not exist. (closes issue ASTERISK-22351)
+	  Reported by: A. Iglesias Patches:
+	  issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
+	  by wdoekes (license 5674)
+
+2013-10-18 14:40 +0000 [r401178]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* main/channel.c: Properly copy/remove the device state cache flag
+	  over a masquerade. In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE
+	  flag was added that tells the devstate system to not cache states
+	  for non-real devices. However, when optimizing away channels
+	  (ast_do_masquerade), that flag wasn't copied. In my case, using
+	  Local devices as queue members created a situation where the
+	  endpoint was considered in use, but the state change of the
+	  device being available again was ignored (not cached). The
+	  endpoint channel was optimized into the (previously) Local
+	  channel, but kept the do-not-cache flag. The end result being
+	  that the queue member apparently stayed in use forever. (closes
+	  issue ASTERISK-22718) Reported by: Walter Doekes Review:
+	  https://reviewboard.asterisk.org/r/2925/
+
+2013-10-17 15:22 +0000 [r401119]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_jabber.c: Reduce log level of a non-pubsub error message
+	  Drop an error log message to debug level 1 since distributed
+	  device state functions correctly when receiving this message and
+	  it spams the logs. (closes issue ASTERISK-22410) Reported by:
+	  abelbeck Patches:
+	  asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
+	  uploaded by abelbeck (License 5903)
+
+2013-10-16 11:04 +0000 [r401049]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* apps/app_queue.c: Don't check all realtime queues when doing
+	  "queue show some_queue". When using realtime queues, queues have
+	  to be fetched from the database every now and then to see if any
+	  info has been changed or to see if the queue has been removed.
+	  When fetching info for an individual queue, the pruning of other
+	  queues is unnecessarily costly. Review:
+	  https://reviewboard.asterisk.org/r/2907/
+
+2013-10-15 14:52 +0000 [r400970]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Prevent chan_sip from sending duplicate
+	  BYEs. When a 200 OK for an initial INVITE is received, we were
+	  doing the right thing by ACKing and sending an immediate BYE.
+	  However, we also were doing the wrong thing and queuing an answer
+	  frame, thus causing the call to be answered. This would cause the
+	  call to be hung up by the channel thread, thus resulting in a
+	  second BYE being sent out. In this fix, I also have set the
+	  hangupcause to be correct since the initial BYE being sent by
+	  Asterisk had an unknown hangup cause. I have changed to using
+	  "Bearer capabilty not available" since the call was hung up due
+	  to an SDP offer/answer error. (closes issue ASTERISK-22621)
+	  reported by Kinsey Moore
+
+2013-10-14 21:40 +0000 [r400907]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: chan_dahdi: Reflect the set software gain
+	  in the CLI "dahdi show channel" output. * Remember the swgain
+	  setting from CLI "dahdi set swgain" command so the CLI "dahdi
+	  show channel" output will reflect the current setting. * Updated
+	  CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
+	  (issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
+	  jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded
+	  by rmudgett
+
+2013-10-14 21:32 +0000 [r400906]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Do not increment the SDP version between 183
+	  and 200 responses. Bumping the SDP version number can cause
+	  interoperability problems since receivers of the responses will
+	  expect that a 200 SDP will be identical to a previous 183 SDP.
+	  (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches:
+	  dont-increment-session-version-in-2xx-after-183.patch uploaded by
+	  NITESH BANSAL (License #6418)
+
+2013-10-08 22:26 +0000 [r400694-400767]  Kinsey Moore <kmoore at digium.com>
+
+	* configure, configure.ac: Add warning when compiling with iODBC
+	  support When running configure, libiodbc2 development headers
+	  will fulfill the requirement for ODBC development headers, but
+	  will not function properly. This adds a warning when libiodbc2
+	  development headers are detected instead of unixodbc development
+	  headers. (closes issue ASTERISK-22459) Reported by: Patrick
+	  Maille Tested by: Walter Doekes Patches:
+	  issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
+	  (License 5674)
+
+	* funcs/func_config.c: Fix func_config list entry allocation The
+	  AST_CONFIG dialplan function defined in func_config.c allocates
+	  its config file list entries using ast_malloc. List entry
+	  allocations destined for use with Asterisk's linked list API must
+	  be ast_calloc()d or otherwise initialized so that list pointers
+	  are set to NULL. These uses of ast_malloc have been replaced by
+	  ast_calloc to prevent dereferencing of uninitialized pointer
+	  values when traversing the list. (closes issue ASTERISK-22483)
+	  Reported by: Brian Scott
+
+2013-10-06 17:07 +0000 [r400622]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* apps/app_queue.c: Fix Regression With Queuelog EXITWITHKEY Only
+	  Logging Two Out Of Four Fields Commit r62462 added two extra
+	  fields for logging "the original position the caller entered the
+	  queue at, and the amount of time the caller was waiting in the
+	  queue." But when r75969 was merged from 1.4 into trunk (r75977),
+	  these two fields disappeared. Those two extra fields were not
+	  logged in 1.4 and when the patch was merged, those fields went
+	  away. Therefore, this is a regression and was caught by the
+	  reporter because he was reading the awesome "Asterisk: The
+	  Definitive Guide" book. (closes issue ASTERISK-22197) Reported
+	  by: Dalius M. Tested by: Dalius M. Patches:
+	  asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
+	  Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2901/
+
+2013-10-03 22:51 +0000 [r400469]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Don't ignore expires value in
+	  contact header if it lacks semicolon (closes issue
+	  ASTERISK-22574) Reported by: Filip Jenicek Patches:
+	  chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
+
+2013-10-03 18:25 +0000 [r400393]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_rtp_multicast.c: Ensure res_rtp_mutlicast sets SSRC
+	  properly This fixes a bug where the SSRC field on multicast RTP
+	  can be stuck at 0 which can cause problems for endpoints trying
+	  to make sense of incoming streams. (closes issue ASTERISK-22567)
+	  Reported by: Simone Camporeale Patches:
+	  22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
+	  (License 6536)
+
+2013-10-02 21:30 +0000 [r400314]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_iax2.c: Cast Integer Argument To Unsigned Char The
+	  member reg in the peercnt structure is an unsigned char and
+	  peercnt_modify() is expecting an unsigned char argument which
+	  gets assigned to peercnt->reg. This patch fixes that by casting
+	  the integer argument being passed to peercnt_modify to unsigned
+	  char.

[... 46476 lines stripped ...]



More information about the asterisk-commits mailing list