[svn-commits] bebuild: tag 1.8.24.0-rc1 r399446 - /tags/1.8.24.0-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Thu Sep 19 09:03:58 CDT 2013


Author: bebuild
Date: Thu Sep 19 09:03:56 2013
New Revision: 399446

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=399446
Log:
Importing files for 1.8.24.0-rc1 release.

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+2013-09-19  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.24.0-rc1 Released.
+
+2013-09-18 19:54 +0000 [r399402]  Kinsey Moore <kmoore at digium.com>
+
+	* main/abstract_jb.c: Fix jitter buffer log file creation This
+	  adjusts '/'-to-'#' replacement to replace all instances of '/'
+	  instead of just the first to ensure that the jitter buffer log
+	  file gets the correct name as per Richard Kenner's suggestion.
+	  (closes issue ASTERISK-21036) Reported by: Richard Kenner
+
+2013-09-18 17:15 +0000 [r399351]  Matthew Jordan <mjordan at digium.com>
+
+	* build_tools/prep_tarball: Update prep_tarball with new
+	  documentation files on the Asterisk wiki This will now pull both
+	  a command reference for the version being prepared, as well as an
+	  Admin Guide that applies to all versions of Asterisk. (issue
+	  ASTERISK-22439) Reported by: Olle Johansson
+
+2013-09-18 01:32 +0000 [r399304]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* main/features.c: Fix Segfault When Syntax Of A Line Under
+	  [applicationmap] Is Invalid When processing the lines under the
+	  [applicationmap] context in features.conf, a segfault occurs from
+	  attempting to process a line with an invalid syntax (basically
+	  missing most of the arguments). Example: [applicationmap]
+	  automon=*6 * This patch moves the checking for empty arguments to
+	  before they are accessed. * Also, checked the "todo" comment and
+	  removed it. Some applications do not require arguments. (closes
+	  issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
+	  Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
+	  Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2803
+
+2013-09-16 16:37 +0000 [r399158]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c: chan_iax2: Fix saving the wrong expiry time
+	  in astdb. When a new IAX2 client registers, the astdb database is
+	  updated with the value of minregexpire defined in iax.conf
+	  instead of using the expiry time that is provided by the client.
+	  The provided expiry time of the client is updated after inserting
+	  the astdb entry. As a consequence, restarting or reloading
+	  asterisk creates clients whose registration may expire before
+	  they reregister. The clients are therefore unavailable after
+	  minregexpire seconds until they reregister. * Move updating of
+	  the expiry time to before inserting into the astdb. (closes issue
+	  ASTERISK-22504) Reported by: Stefan Wachtler Patches:
+	  chan_iax2.c.patch (license #6533) patch uploaded by Stefan
+	  Wachtler
+
+2013-09-13 20:47 +0000 [r399098]  David M. Lee <dlee at digium.com>
+
+	* main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
+	  defined. If MALLOC_DEBUG is enabled, then the debug destructor
+	  for the container is used, which would erroneously write to
+	  /tmp/refs. This patch only uses the debug destructor if ref_debug
+	  is used. (closes issue ASTERISK-22536)
+
+2013-09-13 13:31 +0000 [r399033]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_meetme.c: Fix several crashes in MeetMeAdmin This change
+	  ensures that MeetMeAdmin commands requiring a user actually get a
+	  user and fixes another issue where an extra dereference could
+	  occur for a last-entered user being ejected if a user identifier
+	  was also provided. (closes issue ASTERISK-21907) Reported by:
+	  Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/
+
+2013-09-12 20:09 +0000 [r398937-398977]  Jonathan Rose <jrose at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Revert
+	  r398835 due to failing tests involving originate (issue
+	  ASTERISK-22424) Reported by: Jonathan Rose
+
+	* res/res_musiconhold.c: res_musiconhold: Fix reference leaks
+	  caused when reloading with REF_DEBUG set Due to a faulty function
+	  for debugging reference decrementing, it was possible to reduce
+	  the refcount on the wrong object if two moh classes of the same
+	  name were in the moh class container. (closes issue
+	  ASTERISK-22252) Reported by: Walter Doekes Patches:
+	  18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
+	  6182)
+
+2013-09-12 00:00 +0000 [r398880-398884]  Rusty Newton <rnewton at digium.com>
+
+	* apps/app_queue.c: 'queue add member' help text correction You are
+	  adding dial strings to the queue, not channels. An aribitrary
+	  string could be used, but you are typically referencing a
+	  channel. Correcting the command help text. (issue ASTERISK-22263)
+	  (closes issue ASTERISK-22263) Reported By: Rusty Newton
+
+	* configs/chan_dahdi.conf.sample: Documentation fix -
+	  waitfordialtone is not boolean, it's time in milliseconds
+	  Changing text in chan_dahdi.conf sample to be accurate. (issue
+	  ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
+	  Malcolm Davenport
+
+2013-09-11 19:39 +0000 [r398835]  Jonathan Rose <jrose at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
+	  calls without prior SDP on 200 OK If we receive a 200 OK without
+	  SDP, we will now check to see if the remote address has been
+	  established for that channel's RTP session and if the to tag for
+	  that channel has changed from the most recent to tag in a
+	  response less than 200. If either a change has been made since
+	  the last to-tag was received or the remote address is unset, then
+	  we will drop the call. (closes issue ASTERISK-22424) Reported by:
+	  Jonathan Rose Review:
+	  https://reviewboard.asterisk.org/r/2827/diff/#index_header
+
+2013-09-10 17:53 +0000 [r398757]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/xmldoc.c, main/cli.c, funcs/func_dialgroup.c, main/heap.c,
+	  main/event.c, res/res_musiconhold.c, main/indications.c,
+	  main/asterisk.c: Fix incorrect usages of ast_realloc(). There are
+	  several locations in the code base where this is done: buf =
+	  ast_realloc(buf, new_size); This is going to leak the original
+	  buf contents if the realloc fails. Review:
+	  https://reviewboard.asterisk.org/r/2832/
+
+2013-09-10 17:47 +0000 [r398748-398752]  David M. Lee <dlee at digium.com>
+
+	* utils/check_expr.c: Fixed utils directory breakage from r398748,
+	  this time with extra hate.
+
+	* utils/check_expr.c, utils/ael_main.c, utils/conf2ael.c: Fixed
+	  utils directory breakage from r398648
+
+2013-09-09 23:15 +0000 [r398703]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
+	  completely different from the freed magic number. Race conditions
+	  between freeing a nul terminated string and ast_strdup()'ing it
+	  are more likely to be detected if the fence and freed magic
+	  numbers are completely different.
+
+2013-09-09 19:56 +0000 [r398648]  David M. Lee <dlee at digium.com>
+
+	* main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
+	  DEBUG_THREADS when lock is acquired in __constructor__ This patch
+	  fixes some long-standing bugs in debug threads that were
+	  exacerbated with recent Optional API work in Asterisk 12. With
+	  debug threads enabled, on some systems, there's a lock ordering
+	  problem between our mutex and glibc's mutex protecting its module
+	  list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
+	  thread, the module list will be locked before acquiring our
+	  mutex. In another thread, our mutex will be locked before locking
+	  the module list (which happens in the depths of calling
+	  backtrace()). This patch fixes this issue by moving backtrace()
+	  calls outside of critical sections that have the mutex acquired.
+	  The bigger change was to reentrancy tracking for
+	  ast_cond_{timed,}wait, which wrongly assumed that waiting on the
+	  mutex was equivalent to a single unlock (it actually suspends all
+	  recursive locks on the mutex). (closes issue ASTERISK-22455)
+	  Review: https://reviewboard.asterisk.org/r/2824/
+
+2013-09-06 20:56 +0000 [r398523-398576]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_jabber.c: Commit the remainder of r398523 This is a
+	  missing part of the commit in revision 398523 that corrects the
+	  name of a variable. (issue ASTERISK-22435)
+
+	* res/res_jabber.c: Fix Jabber/XMPP distributed MWI The mailbox and
+	  context are swapped on the receiving end for all users of Jabber
+	  and XMPP distributed MWI in Asterisk 1.8 and all more recent
+	  versions. This swaps those values to be correct when publishing
+	  to the internal event system from Jabber/XMPP distributed MWI
+	  state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested
+	  by: Michael Keuter Patches:
+	  asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
+	  abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
+	  uploaded by abelbeck
+
+2013-09-05 19:00 +0000 [r398235-398456]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c: chan_iax2: Reduce indentation in
+	  __attempt_transmit(). * Reduce indentation in
+	  __attempt_transmit(). * Don't update the static last error time
+	  variable every time in __schedule_action() and socket_read().
+
+	* channels/chan_iax2.c: chan_iax2: Fix stray reference to worker
+	  thread idle_list. * Fix stray reference to idle_list in
+	  cleanup_thread_list(). This may be the reason for the note in
+	  iax2_process_thread() about threads not being removed from the
+	  task lists. * Move cleanup_thread_list(&idle_list) to after the
+	  other lists are cleaned up.
+
+	* channels/chan_iax2.c: chan_iax2: Fix bridgecallno deadlock
+	  avoidance. * Fix bridgecallno deadlock avoidance. When doing
+	  deadlock avoidance, you need to retest the status of values for
+	  each loop to see if you still need the lock for bridgecallno. *
+	  As a safety check, after acquiring the bridgecallno lock you
+	  should check if iaxs[bridgecallno] is NULL just like the current
+	  callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
+	  to after processing any deferred frames to ensure that the
+	  iostate is IDLE when it is placed back into the idle list.
+	  defer_full_frame() tries to ensure iax2_process_thread() wakes up
+	  to process the frame.
+
+	* channels/iax2-parser.c: chan_iax2: Add missing control frame
+	  names to debug frame decode output. (Part 2)
+
+	* channels/iax2-parser.c: chan_iax2: Add missing control frame
+	  names to debug frame decode output.
+
+	* channels/chan_misdn.c: chan_misdn: Fix misdn debug output printed
+	  with arbitrary verbose levels. Fix the misdn debug output to
+	  remote consoles. chan_misdn uses ast_console_puts() which doesn't
+	  know about verbose levels. Better to use ast_verbose() instead.
+	  Without this patch the misdn debug messages are appended to the
+	  verbose level which ever was set by the message sent to the
+	  console before, i.e. any undefined level. (closes issue AST-1218)
+	  Reported by: Guenther Kelleter Patches: misdnlog.patch (license
+	  #6372) patch uploaded by Guenther Kelleter
+
+2013-09-02 07:24 +0000 [r398167]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* cel/cel_custom.c: Be a little more verbose when loading
+	  cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
+
+2013-08-30 18:55 +0000 [r398021-398102]  Kevin Harwell <kharwell at digium.com>
+
+	* channels/chan_sip.c, main/config.c, res/res_security_log.c: Fix
+	  various memory leaks main/config.c - cleanup cache fie includes
+	  res/res_security_log.c - unregister logger level
+	  channesl/chan_sip.c - cleanup io context and notify_types (closes
+	  issues ASTERISK-22378) Reported by: Corey Farrell Patches:
+	  config_shutdown.patch uploaded by coreyfarrell (license 5909)
+	  res_security_log.patch uploaded by coreyfarrell (license 5909)
+	  chan_sip-1.8.patch uploaded by coreyfarrell (license 5909)
+
+	* main/manager.c, res/res_agi.c: Memory leak fix
+	  ast_xmldoc_printable returns an allocated block that must be
+	  freed by the caller. Fixed manager.c and res_agi.c to stop
+	  leaking these results. (closes issue ASTERISK-22395) Reported by:
+	  Corey Farrell Patches: manager-leaks-1.8.patch uploaded by
+	  coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+	  by coreyfarrell (license 5909)
+
+	* main/features.c: Fix memory leak Fixed a features.c test that
+	  leaked a reference to a parked call. This caused chancount to
+	  never reach 0, so graceful shutdown stops. Also added an
+	  unregister test. (closes issue ASTERISK-22413) Reported by: Corey
+	  Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
+	  coreyfarrell (license 5909)
+
+2013-08-30 16:46 +0000 [r398018]  Richard Mudgett <rmudgett at digium.com>
+
+	* tests/test_substitution.c: test_substituition: Fix failed test
+	  reporting to actually report failure. You cannot put the "Testing
+	  <blah> pass/fail" on a single line before actually performing the
+	  test. Now any additional failure information is logged before the
+	  test pass/fail announcement. * Added an additional CDR(answer,u)
+	  test.
+
+2013-08-27 17:55 +0000 [r397710-397756]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
+	  SDP If the SIP channel driver processes an invalid SDP that
+	  defines media descriptions before connection information, it may
+	  attempt to reference the socket address information even though
+	  that information has not yet been set. This will cause a crash.
+	  This patch adds checks when handling the various media
+	  descriptions that ensures the media descriptions are handled only
+	  if we have connection information suitable for that media. Thanks
+	  to Walter Doekes, OSSO B.V., for reporting, testing, and
+	  providing the solution to this problem. (closes issue
+	  ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
+	  issueA22007_sdp_without_c_death.patch uploaded by wdoekes
+	  (License 5674)
+
+	* channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK on
+	  dialog that has no channel A remote exploitable crash
+	  vulnerability exists in the SIP channel driver if an ACK with SDP
+	  is received after the channel has been terminated. The handling
+	  code incorrectly assumed that the channel would always be
+	  present. This patch adds a check such that the SDP will only be
+	  parsed and applied if Asterisk has a channel present that is
+	  associated with the dialog. Note that the patch being applied was
+	  modified only slightly from the patch provided by Walter Doekes
+	  of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
+	  Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
+	  issueA21064_fix.patch uploaded by wdoekes (License 5674)
+
+2013-08-23 15:34 +0000 [r397525]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/astmm.h, main/logger.c, main/utils.c,
+	  include/asterisk/lock.h, main/astmm.c, channels/sig_pri.c,
+	  main/astobj2.c, include/asterisk/logger.h, main/lock.c,
+	  include/asterisk/utils.h: Fix memory corruption when trying to
+	  get "core show locks". Review
+	  https://reviewboard.asterisk.org/r/2580/ tried to fix the
+	  mismatch in memory pools but had a math error determining the
+	  buffer size and didn't address other similar memory pool
+	  mismatches. * Effectively reverted the previous patch to go in
+	  the same direction as trunk for the returned memory pool of
+	  ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols()
+	  when BETTER_BACKTRACES is defined. * Fixed some formatting in
+	  ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated
+	  by libpri when MALLOC_DEBUG is enabled. * Fixed
+	  __dump_backtrace() freeing memory from ast_bt_get_symbols() when
+	  MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of
+	  compile issues with the utils directory. (closes issue
+	  ASTERISK-22221) Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/2778/
+
+2013-08-22 08:19 +0000 [r397377]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* default.exports, main/asterisk.exports.in: Add _IO_stdin_used in
+	  version-script to fix SIGBUSes on Sparc. The
+	  --version-script,asterisk.exports linker flag (and the module
+	  exports) didn't provide _IO_stdin_used in the list of exported
+	  symbols. That causes some kind of libc compatibility mode to kick
+	  in, where stdio file structures (stdout/stderr) land somewhere
+	  else. In the case of the Sparc, they landed on misaligned memory.
+	  This became apparent first after r376428 (Reorder startup
+	  sequence) when a lot of ast_log's were replaced with fprintf's.
+	  Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
+	  architectures, the Sparc is very picky about memory alignment.)
+	  (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
+	  Kister Review: https://reviewboard.asterisk.org/r/2760/
+
+2013-08-21 17:00 +0000 [r397308]  David M. Lee <dlee at digium.com>
+
+	* main/http.c: Complete http_shutdown. This patch frees up some
+	  resources allocated in http.c. * tcp listeners stopped * tls
+	  settings freed * uri redirects freed * unregister internal http.c
+	  uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
+	  Patches: http.patch uploaded by Corey Farrell (license 5909)
+
+2013-08-21 14:56 +0000 [r397256]  Matthew Jordan <mjordan at digium.com>
+
+	* include/asterisk/frame.h: Set 14400 as the default max bit rate
+	  if T38MaxBitRate is not specified If an endpoint fails to include
+	  the T38MaxBitRate attribute during negotiation, Asterisk will
+	  negotiate a bit rate of 2400 instead of the ITU recommended bit
+	  rate of 14400. This patch fixes this by making AST_T38_RATE_14400
+	  the 'default' value of the enum by assigning it a value of 0,
+	  such that if an endpoint fails to include the attribute, the
+	  default will be 14400. Note that Walter Doekes included the nice
+	  comment in frame.h about why we are purposefully assigning
+	  AST_T38_RATE_14400 a value of 0. (closes issue ASTERISK-22275)
+	  Reported by: Andreas Steinmetz patches: fax-fix.patch uploaded by
+	  anstein (License 6523)
+
+2013-08-21 02:09 +0000 [r397204]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_sip.c: Fix Not Storing Current Incoming Recv
+	  Address In 1.8, r384779 introduced a regression by retrieving an
+	  old dialog and keeping the old recv address since recv was
+	  already set. This has caused a problem when a proxy is involved
+	  since responses to incoming requests from the proxy server, after
+	  an outbound call is established, are never sent to the correct
+	  recv address. In 11, r382322 introduced this regression. The fix
+	  is to revert that change and always store the recv address on
+	  incoming requests. Thank you Walter Doekes for helping to point
+	  out this error and Mark Michelson for your input/review of the
+	  fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
+	  Tested by: Alex Zarubin, Karsten Wemheuer Patches:
+	  asterisk-22071-store-recvd-address.diff by Michael L. Young
+	  (license 5026)
+
+2013-08-20 17:40 +0000 [r397112-397156]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Remove REF_DEBUG definition.
+
+	* channels/sip/dialplan_functions.c, channels/chan_sip.c: Fix
+	  refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
+	  the list of pvts. (closes issue ASTERISK-22248) reported by Corey
+	  Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
+	  (license #5909)
+
+2013-08-20 15:26 +0000 [r397033-397106]  Kinsey Moore <kmoore at digium.com>
+
+	* main/threadstorage.c, main/astfd.c: Unregister CLI commands on
+	  exit This patch ensures that CLI commands enabled by
+	  DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
+	  exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
+	  Tested by: Corey Farrell Patches: debug_cli_unregister.patch
+	  uploaded by Corey Farrell
+
+	* main/xmldoc.c: Fix xmldoc memory leak This fixes a
+	  single-attribute memory leak that was occurring when the
+	  "required" attribute was not true. (closes issue ASTERISK-22249)
+	  Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+	  xmldoc-free_attr_required.patch uploaded by Corey Farrell
+
+	* main/cel.c: Protect CEL from an invalid config on reload This
+	  patch fixes CEL to properly handle an invalid config on reload.
+	  (closes issue ASTERISK-22259) Reported by: Corey Farrell Tested
+	  by: Corey Farrell Patches: cel-config.patch uploaded by Corey
+	  Farrell
+
+2013-08-20 11:46 +0000 [r396994]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* configs/h323.conf.sample, configs/sip.conf.sample: Add
+	  "autoframing" option to sip.conf.sample and h323.conf.sample. The
+	  autoframing option was added to chan_sip.c in r43243 (mogorman,
+	  2006-09-19 01:32:57), but never made its way into the sample
+	  configs. Review: https://reviewboard.asterisk.org/r/2768/
+
+2013-08-20 01:17 +0000 [r396958]  Matthew Jordan <mjordan at digium.com>
+
+	* main/data.c: Fix invalid access to disposed memory in main/data
+	  unit test It is not safe to iterate over a macro'd list of ao2
+	  objects, deref them such that the item's destructor is called,
+	  and leave them in the list. The list macro to iterate over items
+	  requires the item to be a valid allocated object in order to
+	  proceed to the next item; with MALLOC_DEBUG on the corruption of
+	  the linked list is caught in the crash. This patch fixes the
+	  invalid access to free'd memory by removing the ao2 item from the
+	  list before de-refing it. Note that this is a backport of r396915
+	  from Asterisk trunk.
+
+2013-08-15 16:21 +0000 [r396745]  Kinsey Moore <kmoore at digium.com>
+
+	* main/cli.c, main/asterisk.c: Remove leading spaces from the CLI
+	  command before parsing If you've mistakenly put a space before
+	  typing in a command, the leading space will be included as part
+	  of the command, and the command parser will not find the
+	  corresponding command. This patch rectifies that situation by
+	  stripping the leading spaces on commands. Review:
+	  https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
+	  Lesher
+
+2013-08-14 19:05 +0000 [r396619-396656]  Joshua Colp <jcolp at digium.com>
+
+	* tests/test_hashtab_thrash.c: Tweak comment for why usleep is
+	  used.
+
+	* tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
+	  allow the critical threads to execute. Depending on certain
+	  conditions it was possible for the hashtab counting thread to
+	  starve other threads, preventing them from executing in the
+	  expected fashion. This change adds a sleep to allow the others to
+	  do what they need to do. While this doesn't thrash the hashtab as
+	  much as previously, it at least works. (closes issue
+	  ASTERISK-22276) Reported by: Matt Jordan
+
+2013-08-13 18:44 +0000 [r396579-396582]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: chan_sip: Convert 'just did sched_add
+	  waitid...' from warning to debug message. Patches:
+	  reviewboard-2377.patch uploaded by Paul Belanger Review:
+	  https://reviewboard.asterisk.org/r/2377/
+
+	* channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
+	  rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
+	  by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
+
+2013-08-08 20:14 +0000 [r396427]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* main/logger.c, main/utils.c, main/astobj2.c,
+	  include/asterisk/logger.h: Consistent memory allocation by
+	  ast_bt_get_symbols. Always use ast_alloc/ast_free. This is
+	  handled differently in trunk (r391012). Review:
+	  https://reviewboard.asterisk.org/r/2580/
+
+2013-08-06 08:14 +0000 [r396279]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* pbx/pbx_dundi.c, utils/extconf.c, apps/app_stack.c,
+	  apps/app_playback.c, funcs/func_global.c, main/cdr.c,
+	  pbx/pbx_loopback.c, main/pbx.c, funcs/func_strings.c: Check
+	  result of ast_var_assign() calls for memory allocation failure.
+	  We try to keep the system running even when all available memory
+	  is spent. Review: https://reviewboard.asterisk.org/r/2734/
+
+2013-08-05 20:17 +0000 [r396196-396240]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* channels/chan_sip.c: Fix Registration Failure When A Peer And TLS
+	  Are Used If a peer is used in a register line and TLS is defined
+	  as the transport, the registration fails since the transport on
+	  the dialog is never set properly resulting in UDP being used
+	  instead of TLS. This patch sets the dialog's transport based on
+	  the transport that was defined in the register line. If the
+	  register line does not specify a transport, the parsing function
+	  for the register line always defaults back to UDP. (closes issue
+	  ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey
+	  Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael
+	  L. Young (license 5026)
+
+	* channels/chan_sip.c: Restore Extra Line Break Between Peers When
+	  Running AMI Action SIPPeers The commit (r387133) for fixing
+	  ASTERISK-21466 accidentally removed an extra line break between
+	  the peers returned by the AMI action SIPPeers. This results in
+	  some parsers breaking because they expect this extra line break.
+	  This patch restores that extra line break. (closes issue
+	  ASTERISK-22239) Reported by: Jacek Konieczny Tested by: Jacek
+	  Konieczny, Michael L. Young Patches:
+	  asterisk-ami_sippeers_separator.patch by Jacek Konieczny (license
+	  6298)
+
+	* UPGRADE.txt: Adding a note to UPGRADE.txt about a change made to
+	  res_agi in order to indicate when streaming an audio file fails
+	  like it is done in other parts of the code to indicate an error.
+	  Note was requested by Paul Belanger:
+	  http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
+	  (related to issue ASTERISK-21903)
+
+2013-07-22 13:49 +0000 [r394886-395032]  Matthew Jordan <mjordan at digium.com>
+
+	* main/asterisk.c: Update copyright year to 2013 in asterisk.c;
+	  some whitespace fixes (closes issue ASTERISK-22179) Reported by:
+	  Malcolm Davenport
+
+	* funcs/func_channel.c: Clean up documentation This patch cleans up
+	  documentation in func_channel for the following items: *
+	  rtpsource * secure_signaling * secure_media (closes issue
+	  ASTERISK-20969) Reported by: snuffy patches:
+	  func_chan-update.diff uploaded by snuffy (License 5024)
+
+	* configs/indications.conf.sample: Provide proper ring tone in
+	  indications.conf for Malaysia The ring tone provided in the
+	  sample indications.conf was incorrect. This patch modifies the
+	  sample ring tone to be what it should: ring =
+	  425/400,0/200,425/400,0/2000 This brings it in line with the tone
+	  definition in DAHDI 2.7.0. (zonedata.c) (closes issue
+	  ASTERISK-21997) Reported by: Filip Jenicek patches:
+	  malaysia_ring.patch uploaded by phill (License 6277)
+
+	* main/http.c: Tolerate presence of RFC2965 Cookie2 header by
+	  ignoring it This patch modifies parsing of cookies in Asterisk's
+	  http server by doing an explicit comparison of the "Cookie"
+	  header instead of looking at the first 6 characters to determine
+	  if the header is a cookie header. This avoids parsing "Cookie2"
+	  headers and overwriting the previously parsed "Cookie" header.
+	  Note that we probably should be appending the cookies in each
+	  "Cookie" header to the parsed results; however, while clients can
+	  send multiple cookie headers they never really do. While this
+	  patch doesn't improve Asterisk's behavior in that regard, it
+	  shouldn't make it any worse either. Note that the solution in
+	  this patch was pointed out on the issue by the issue reporter,
+	  Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
+	  Stuart Henderson Tested by: mjordan, Stuart Henderson
+
+	* contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
+	  realtime scripts with schema for queue_log table This patch
+	  updates the realtime SQL scripts with an entry that will create
+	  the queue_log table. This brings the PostgreSQL scripts inline
+	  with the MySQL scripts, with respect to what tables they will
+	  create. (closes issue ASTERISK-21021) Reported by: Eugene
+	  patches: queue_log.sql uploaded by varnav (license 6360)
+
+	* configs/iax.conf.sample: Document connectedline parameter for
+	  chan_iax2 The connectedline parameter for a chan_iax2 peer was
+	  undocumented. This patch documents the options in the sample
+	  configuration file. (closes issue ASTERISK-21953) Reported by:
+	  Birger "WIMPy" Harzenetter
+
+2013-07-18 12:51 +0000 [r394640]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* res/res_agi.c: Properly indicate failure to open an audio stream
+	  in res_agi If there is an error streaming an audio file, the
+	  current return status makes it difficult for an AGI script to
+	  determine that there was an error with the audio file. This
+	  patches changes the result to return -1 and the function returns
+	  RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
+	  parts of res_agi, this would appear to be the proper way to
+	  handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
+	  Wainer Tested by: Ariel Wainer Patches:
+	  asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
+	  (license 5026) Review: https://reviewboard.asterisk.org/r/2625/
+
+2013-07-14 01:53 +0000 [r394302]  Matthew Jordan <mjordan at digium.com>
+
+	* funcs/func_strings.c: Clarify documentation for function PASSTHRU
+	  It is not apparent to the average user that the PASSTHRU function
+	  should not be passed as ${PASSTHRU(string)} but just as
+	  PASSTHRU(string) to functions which take a variable name and not
+	  its contents. This patch clarifies the behavior in the
+	  documentation and provides an example. (closes issue
+	  ASTERISK-21717) Reported by: Richard Miller patches:
+	  func_strings.diff uploaded by Richard Miller (license 5685)
+
+2013-07-11 16:25 +0000 [r394106]  Moises Silva <moises.silva at gmail.com>
+
+	* channels/chan_dahdi.c: Fix a longstanding issue with MFC-R2
+	  configuration that prevented users from mixing different variants
+	  or general MFC-R2 settings within the same E1 line. Most users do
+	  not have a problem with this since MFC-R2 lines are usually
+	  fractional E1s, or the whole E1 has the same country variant and
+	  R2 settings. In Venezuela however is common to have inbound
+	  MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
+	  properly parses the chan_dahdi.conf file to generate a new openr2
+	  context every time a new channel => section is found and the
+	  configuration was changed. (closes issue ASTERISK-21117) Reported
+	  by: Rafael Angulo Related Elastix issue:
+	  http://bugs.elastix.org/view.php?id=1612
+
+2013-07-10 01:41 +0000 [r393928]  Russell Bryant <russell at russellbryant.com>
+
+	* configs/sla.conf.sample, include/asterisk/utils.h,
+	  apps/app_meetme.c: astobj2-ify the SLA code The SLA code within
+	  app_meetme was written before asotbj2 had been merged into
+	  Asterisk. Worse, support for reloads did not exist at first and
+	  was added later as a bolt-on feature. I knew at the time that
+	  reloading was not safe at all while SLA was in use, so the reload
+	  would be queued up to execute when the system was idle.
+	  Unfortunately, this approach was still prone to errors beyond the
+	  fact that this was the only place in Asterisk where configuration
+	  was not reloaded instantly when requested. This patch converts
+	  various SLA objects to be reference counted objects using
+	  astobj2. This allows reloads to be processed while the system is
+	  in use. The code ensures that the objects will not disappear
+	  while one of the other threads is using them. However, they will
+	  be immediately removed from the global trunk and station
+	  containers so no new calls will use them if removed from
+	  configuration. Review: https://reviewboard.asterisk.org/r/2581/
+
+2013-07-03 23:27 +0000 [r393627]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: chan_dahdi: Fix segfault reloading
+	  chan_dahdi when round robin is used. * Clear round_robin[] in
+	  dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
+	  Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
+	  patch uploaded by rmudgett
+
+2013-06-14 16:14 +0000 [r391778]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_mixmonitor.c: app_mixmonitor: Fix crashes caused by
+	  unloading app_mixmonitor Unloading app_mixmonitor while active
+	  mixmonitors were running would cause a segfault. This patch fixes
+	  that by making it impossible to unload app_mixmonitor while
+	  mixmonitors are active. Review:
+	  https://reviewboard.asterisk.org/r/2624/
+
+2013-06-12 02:19 +0000 [r391489]  Matthew Jordan <mjordan at digium.com>
+
+	* main/loader.c: Fix memory leak while loading priority modules
+	  When we load a module with the LOAD_PRIORITY flag, we remove its
+	  entry from the load order list. Unfortunately, we don't free the
+	  memory associated with entry in the list. This patch corrects
+	  that and properly frees the memory for the module in the list.
+
+2013-06-11 08:03 +0000 [r391333]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_iax2.c: IAX2: Transfer Reject: Lock bridgecallno
+	  before touching it, refactor 1). When touching the bridgecallno,
+	  we need to lock it. 2). Remove magic number '0' and replace with
+	  TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
+	  indentation. Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2613/
+
+2013-07-15  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.23.0 Released.
+
+2013-07-12  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.23.0-rc2 Released.
+
+	* Properly lock and safely handle a transfer failure in IAX2
+
+	  When touching the bridgecallno, we need to lock it - otherwise a
+	  race condition can occur. This patch does the proper locking
+	  of the bridgecallno before modifying its state.
+
+2013-06-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.23.0-rc1 Released.
+
+2013-06-10 14:15 +0000 [r391215]  Matthew Jordan <mjordan at digium.com>
+
+	* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Add
+	  announce-to-first-user option for app_queue In r386792, the
+	  ability to play prompts to the first caller in a call queue was
+	  added. While this is arguably a bug fix for those who expect the
+	  first caller to continue receiving prompts while the agent is
+	  dialed, it has the side effect of preventing the first caller
+	  from hearing the agent immediately upon bridging. This may not be
+	  a problem for those who really want this option, but for those
+	  who didn't care whether or not the first caller in queue heard
+	  their position, it was an issue. This patch disables the ability
+	  for the first caller in the queue to hear prompts and adds a new
+	  option, announce-to-first-user, to queues.conf. Those who the
+	  behavior can enable it by setting this value to True. Note that
+	  if we ever implement the ability to have the prompts be stopped
+	  upon bridging, this option can be removed. (closes issue
+	  ASTERISK-21782) Reported by: Remi Quezada
+
+2013-06-10 09:30 +0000 [r391062-391143]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+	  unlock bridgecallno
+
+	* channels/chan_iax2.c: fix bad edit after conflict resolution
+
+	* channels/chan_iax2.c: IAX2: refactor nativebridge transfer remove
+	  triple checking of iaxs[fr->callno]->transferring reduce
+	  indentation. Reported by: alecdavis Tested by: alecdavis
+	  alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2602/
+
+	* channels/chan_iax2.c: IAX2: fix race condition with nativebridge
+	  transfers. 1). When touching the bridgecallno, we need to lock
+	  it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes
+	  the lock on the pvt is already held, when iax2_destroy_helper()
+	  is called. Thus we need to lock the bridgecallno pvt before we
+	  call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When
+	  evaluating the state of 'callno->transferring' of the current
+	  leg, we can't change it to READY unless the bridgecallno is
+	  locked. Why, if we are interrupted by the other call leg before
+	  'transferring = TRANSFER_RELEASED', the interrupt will find that
+	  it is READY and that the bridgecallno is also READY so Releases
+	  the legs. (closes issue ASTERISK-21409) Reported by: alecdavis
+	  Tested by: alecdavis alecdavis (license 585) Review
+	  https://reviewboard.asterisk.org/r/2594/
+
+2013-05-31 08:10 +0000 [r390181]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: reject call attempts when gatekeeper is
+	  configured but not registered (closes issue ASTERISK-21800)
+	  Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+	  Tested by: Dmitry Melekhov
+
+2013-05-29 20:10 +0000 [r390044]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: Fix segfault when dealing with chan_agent
+	  channels. Check the returned bridged pointer for NULL to avoid a
+	  crash. It looks like chan_agent is returning a NULL pointer when
+	  it probably should be returning a pointer to the channel the
+	  Agent channel is pretending to be. (closes issue ASTERISK-21793)
+	  Reported by: Rodrigo P. Telles Patches:
+	  jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: Rodrigo P. Telles
+
+2013-05-28 17:35 +0000 [r389895]  Jonathan Rose <jrose at digium.com>
+
+	* main/slinfactory.c: Fix a memory copying bug in slinfactory which

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