2013-01-09  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.2.0-rc2 Released.

	* Fix pjproject compilation in certain circumstances.

	  On a fresh checkout of Asterisk 11, running make before ./configure
	  could cause the pjproject subdirectory to get in an odd state that
	  would prevent compilation. This patch by Tilghman prevents that from
	  occurring.

	  (closes issue ASTERISK-20681)
	  Patch-by: Tilghman Lesher

	* AST-2012-014: Resolve crashes due to large stack allocations when
	  using TCP

	  Asterisk had several places where messages received over various
	  network transports may be copied in a single stack allocation. In
	  the case of TCP, since multiple packets in a stream may be
	  concatenated together, this can lead to large allocations that
	  overflow the stack.

	  This patch modifies those portions of Asterisk using TCP to either
	  favor heap allocations or use an upper bound to ensure that the
	  stack will not overflow:
	  * For SIP, the allocation now has an upper limit
	  * For HTTP, the allocation is now a heap allocation instead of a
	    stack allocation
	  * For XMPP, the allocation has been eliminated since it was
	    unnecessary.

	  This patch contains the fix for both res_jabber and res_xmpp.

	* AST-2012-015: Prevent exhaustion of system resources through
	  exploitation of event cache 

	  Asterisk maintains an internal cache for devices in the event
	  subsystem. The device state cache holds the state of each device
	  known to Asterisk, such that consumers of device state information
	  can query for the last known state for a particular device, even if
	  it is not part of an active call. The concept of a device in
	  Asterisk can include entities that do not have a physical
	  representation. One way that this occurred was when anonymous calls
	  are allowed in Asterisk. A device was automatically created and
	  stored in the cache for each anonymous call that occurred; this was
	  possible in the SIP and IAX2 channel drivers and through channel
	  drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk,
	  Jingle, and Motif). These devices are never removed from the system,
	  allowing anonymous call to potentially exhaust a system's resources.

	  This patch changes the event cache subsystem and device state
	  management to no longer cache devices that are not associated with a
	  physical entity.

	* Revert bad ringinuse=no patch.

	  With the option ringinuse=no set, the patch committed previous for
	  ASTERISK-16115 causes non-SIP queue members to never be called
	  because the device state is checked after a channel is created to
	  determine if the member is busy. These queue members always get the
	  "Member %s is busy, cannot dial" message.

	  Most channel drivers other than chan_sip use the default device
	  state handling. The default device state is considered in use or
	  unknown if the channel exists or not, respectively.

	* Fix multiple calls to a queue member that is only in queue.

	  When ringinuse=no queue members can receive more than one call if
	  these calls happen at nearly the same time. This patch fixes it so a
	  queu member does not receive more than one call from a queue. note
	  that this fix does not prevent multiple calls to a member if hte
	  member is in more than one queue (see ASTERISK-16115).

2012-12-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.2.0-rc1 Released.

2012-12-10 01:41 +0000 [r377505-377511]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/xmldoc.c, /: Improve documentation by making all of the
	  colors used readable, no matter what the background color is.
	  Dark blue on a black background is unreadable, as is yellow on a
	  light background. This patch turns on the bright attribute for
	  colors when on a dark background and turns *off* the bright
	  attribute when the -W command line option is used (indicating a
	  _light_ background). This ensures that text is readable in both
	  cases. Patch by: tilghman Review:
	  https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
	  377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 377510 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, addons/cdr_mysql.c: Remove some dead code and additionally
	  handle a case that wasn't handled. ........ Merged revisions
	  377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 377504 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-09 01:22 +0000 [r377462]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Add missing support for "who hung up" to
	  chan_motif. (closes issue ASTERISK-20671) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/2208/

2012-12-08 00:29 +0000 [r377401-377433]  Richard Mudgett <rmudgett@digium.com>

	* contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
	  allow/disallow in MySQL contrib script. Using the contrib
	  sippeers.sql script to create the sippeers MySQL table would
	  result in being unable to place calls if you set the disallow
	  value to all. (closes issue ASTERISK-20756) Reported by: Andre
	  Luis Patches: sippeers.patch patch uploaded by Andre Luis
	  ........ Merged revisions 377431 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377432 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
	  allocation dumps. ........ Merged revisions 377398 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377399 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-07 22:02 +0000 [r377383]  Kinsey Moore <kmoore@digium.com>

	* /, codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
	  show" CLI command. In r306010 "Asterisk media architecture
	  conversion - no more format bitfields", the logic for
	  incrementing encoders and decoders when opening transcoder
	  channels was changed without making the corresponding change when
	  decrementing encoder / decoder channels. The result being that
	  when a channel was destroyed, codec_dahdi couldn't properly tell
	  if it was an encoder or decoder, and the default case is to
	  assume it was a decoder. This could result in negative numbers
	  for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
	  encoders/decoders of 92 channels are in use. (closes issue
	  ASTERISK-19921) Patch-by: Shaun Ruffell ........ Merged revisions
	  377382 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-06 23:58 +0000 [r377355]  Richard Mudgett <rmudgett@digium.com>

	* apps/confbridge/conf_config_parser.c, /, apps/app_confbridge.c:
	  confbridge: Fix some resource leaks on conference teardown. *
	  Made destroy_conference_bridge() destroy a missed ast_mutex_t and
	  ast_cond_t. * Made join_conference_bridge() init the
	  ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
	  destroy them unconditionally. * Made join_conference_bridge()
	  abort if the new conference could not be added to the conferences
	  container. * Made leave_conference() discard any post-join
	  actions if join_conference_bridge() had to abort early. * Made
	  the join_conference_bridge() diagnostic messages better describe
	  what happened. * Renamed leave_conference_bridge() to
	  leave_conference() and made it only take a conference user
	  pointer. The conference pointer was redundant. * Made
	  conf_bridge_profile_copy() use struct copy instead of memcpy(). *
	  No need to lock the conference in start_conf_record_thread()
	  since all of the callers already have it locked. ........ Merged
	  revisions 377354 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-06 17:28 +0000 [r377340]  Russell Bryant <russell@russellbryant.com>

	* main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl
	  show' CLI command allows you to show the details about a specific
	  named ACL in acl.conf. This patch adds tab completion to the
	  command. Review: https://reviewboard.asterisk.org/r/2230/

2012-12-06 14:11 +0000 [r377319]  Matthew Jordan <mjordan@digium.com>

	* main/manager.c: Fix memory leak in 'manager show event' when
	  command entered incorrectly When the CLI command 'manager show
	  event' was run incorrectly and its usage instructions returned, a
	  reference to the event container was leaked. This would prevent
	  the container from being reclaimed when Asterisk exits. We now
	  properly decrement the count on the ao2 object using the nifty
	  RAII_VAR macro. Thanks to Russell for helping me stumble on this,
	  and Terry for writing that ridiculously helpful macro.

2012-12-05 17:08 +0000 [r377262]  Jonathan Rose <jrose@digium.com>

	* res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc
	  on an already dealloced session When srtp_create fails, the
	  session may be dealloced or just not alloced. At the same time
	  though, the session pointer might not be set to NULL in this
	  process and attempting to srtp_dealloc it again will cause a
	  segfault. This patch checks for failure of srtp_create and sets
	  the session pointer to NULL if it fails. (closes issue
	  ASTERISK-20499) Reported by: tootai Review:
	  https://reviewboard.asterisk.org/r/2228/ ........ Merged
	  revisions 377256 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377261 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-05 16:50 +0000 [r377259]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
	  connections. During the TLS re-work in chan_sip some TLS specific
	  code was moved into a separate function. This function operates
	  on a copy of the incoming SIP request. This copy was never
	  deinitialized causing a memory leak for each request processed.
	  This function is now given a SIP request structure which it can
	  use to copy the incoming request into. This reduces the amount of
	  memory allocations done since the internal allocated components
	  are reused between packets and also ensures the SIP request
	  structure is deinitialized when the TLS connection is torn down.
	  (closes issue ASTERISK-20763) Reported by: deti ........ Merged
	  revisions 377257 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377258 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-05 02:19 +0000 [r377213-377244]  Richard Mudgett <rmudgett@digium.com>

	* main/format.c, /: Fix registering core show codecs/codec CLI
	  commands twice. ........ Merged revisions 377241 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/confbridge/conf_config_parser.c, /: confbridge: Fix several
	  small issues. * Made func_confbridge_helper() allow an empty
	  value when setting options. You previously could not
	  Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
	  dialplan. * Made func_confbridge_helper() handle its datastore
	  better if multiple threads attempt to set the first CONFBRIDGE
	  option value on the channel. * Made the func_confbridge_helper()
	  only output one diagnostic message concerning the option. * Made
	  the bridge video_mode able to repeatedly change in the config
	  file and CONFBRIDGE dialplan function. The video_mode option
	  values are an enum and not independent of each other. * Made
	  handle_cli_confbridge_show_bridge_profile() better handle the
	  video_mode option. * Simplified datastore handling code in
	  conf_find_user_profile() and conf_find_bridge_profile(). (closes
	  issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter
	  ........ Merged revisions 377227 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_confbridge.c: confbridge: Update online XML
	  documentation. ........ Merged revisions 377212 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-04 12:59 +0000 [r377195]  Russell Bryant <russell@russellbryant.com>

	* contrib/scripts/install_prereq: Add libuuid to install_prereq for
	  Fedora. I ran this script and my build failed. pjproject requires
	  this.

2012-12-03 22:58 +0000 [r377039-377167]  Richard Mudgett <rmudgett@digium.com>

	* main/asterisk.c, /: Cleanup ast_run_atexits() atexits list. *
	  Convert atexits list to a mutex instead of a rd/wr lock. The lock
	  is only write locked. * Move CLI verbose Asterisk ending message
	  to where AMI message is output in really_quit() to avoid further
	  surprises about using stuff already shutdown. (issue
	  ASTERISK-20649) Reported by: Corey Farrell ........ Merged
	  revisions 377165 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377166 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/asterisk.c, /, include/asterisk/_private.h,
	  main/stdtime/localtime.c: Cleanup core main on exit. * Cleanup
	  time zones on exit. * Make exit clean/unclean report consistent
	  for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported
	  by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license
	  #5909) patch uploaded by Corey Farrell
	  core-cleanup-11-trunk.patch (license #5909) patch uploaded by
	  Corey Farrell Modified ........ Merged revisions 377135 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377136 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/config.c, /: Cleanup config cache on exit. (issue
	  ASTERISK-20649) Reported by: Corey Farrell Patches:
	  config-cleanup-all.patch (license #5909) patch uploaded by Corey
	  Farrell ........ Merged revisions 377104 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377105 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/cli.c, /: Cleanup CLI resources on exit and CLI command
	  registration errors. (issue ASTERISK-20649) Reported by: Corey
	  Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
	  uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
	  #5909) patch uploaded by Corey Farrell Modified ........ Merged
	  revisions 377073 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377074 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
	  do_reload() return handling since it never returned anything
	  other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
	  Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
	  Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
	  uploaded by Corey Farrell Modified ........ Merged revisions
	  377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 377070 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/ccss.c: Fix CCSS CLI commands and logger level not
	  unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
	  Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
	  Corey Farrell ........ Merged revisions 377037 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 377038 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-03 14:54 +0000 [r377021]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Fix an RTP instance reference count leak
	  in chan_motif. When setting up an RTP instance the RTCP portion
	  of the instance keeps a reference to the instance itself. In
	  order to release this reference and stop RTCP the stop API call
	  must be called before destroying the instance. (closes issue
	  ASTERISK-20751) Reported by: joshoa

2012-12-01 00:46 +0000 [r376983]  Joshua Colp <jcolp@digium.com>

	* configs/motif.conf.sample, channels/chan_motif.c: Tweak extension
	  used for incoming calls received on Motif. Based on feedback from
	  numerous individuals this patch tweaks incoming calls to first
	  look for an extension with the name of the endpoint. If no such
	  extension exists the call will silently fall back to the "s"
	  extension as it previously did.

2012-11-30 21:35 +0000 [r376952]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
	  RELEASE_COMPLETE in response to SETUP. Fix sending a
	  RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
	  have a B channel available to assign to the call. (closes issue
	  ABE-2869) Reported by: Guenther Kelleter Patches:
	  setup-reject_2.diff (license #6372) patch uploaded by Guenther
	  Kelleter Modified ........ Merged revision 376949 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 376950 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376951 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-30 17:07 +0000 [r376921]  Sean Bright <sean@malleable.com>

	* /, funcs/func_volume.c: Minor spelling fix to the VOLUME
	  documentation. ........ Merged revisions 376919 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376920 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-30 16:36 +0000 [r376917]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix potential crashes during SIP attended
	  transfers. The principal behind this patch is simple. During a
	  transfer, we manipulate channels that are owned by a separate
	  thread than the one we currently are running in, so it makes
	  sense that we need to grab a reference to the channels so that
	  they cannot disappear out from under us. In the wild, crashes
	  were sometimes seen when the transferring party would hang up the
	  call before the transfer target answered the call. The most
	  common place to see the crash occur was when attempting to send a
	  connected line update to the transferer channel. (closes issue
	  ASTERISK-20226) Reported by Jared Smith Patches:
	  ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
	  Tested by: Jared Smith ........ Merged revisions 376901 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376916 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-29 22:59 +0000 [r376866-376870]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
	  local_devicestate(). Regression introduced by ASTERISK-20390 fix.
	  (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
	  rmudgett ........ Merged revisions 376868 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376869 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
	  ........ Merged revisions 376864 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376865 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-29 21:57 +0000 [r376836]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Improve Code Readability And Fix Setting
	  natdetected Flag For 1.8, 10, 11 and trunk we are are improving
	  the code readability. For 11 and trunk, auto nat detection was
	  added. The natdetected flag was being set to 1 when the host
	  address in the VIA header did not specifiy a port. This patch
	  fixes this by setting the port on the temporary sock address used
	  to SIP_STANDARD_PORT in order for the sock address comparison to
	  work properly. (closes issue ASTERISK-20724) Reported by: Michael
	  L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2206/ ........ Merged
	  revisions 376834 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376835 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-29 17:17 +0000 [r376822]  Pedro Kiefer <pedro@kiefer.com.br>

	* channels/chan_sip.c: Fix chan_sip websocket payload handling
	  Websocket by default doesn't return an ast_str for the payload
	  received. When converting it to an ast_str on chan_sip the last
	  character was being omitted, because ast_str functions expects
	  that the given length includes the trailing 0x00. payload_len
	  only has the actual string length without counting the trailing
	  zero. For most cases this passed unnoticed as most of SIP
	  messages ends with \r\n. (closes issue ASTERISK-20745) Reported
	  by: Iñaki Baz Castillo Review:
	  https://reviewboard.asterisk.org/r/2219/

2012-11-29 00:46 +0000 [r376760-376790]  Richard Mudgett <rmudgett@digium.com>

	* main/asterisk.c, /, main/astmm.c: Add MALLOC_DEBUG atexit
	  unreleased malloc memory summary. * Adds the following CLI
	  commands to control MALLOC_DEBUG reporting of unreleased malloc
	  memory when Asterisk is shut down. memory atexit list on memory
	  atexit list off memory atexit summary byline memory atexit
	  summary byfunc memory atexit summary byfile memory atexit summary
	  off * Made check all remaining allocated region blocks atexit for
	  fence violations. * Increased the allocated region hash table
	  size by about three times. It still isn't large enough
	  considering the number of malloced blocks Asterisk uses. * Made
	  CLI "memory show allocations anomalies" use
	  regions_check_all_fences(). Review:
	  https://reviewboard.asterisk.org/r/2196/ ........ Merged
	  revisions 376788 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376789 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
	  "memory show allocations" misspelling of anomalies option. The
	  command will still accept the original misspelling. *
	  Miscellaneous tweaks to CLI "memory show allocations" command
	  output format. * Made CLI "memory show summary" summarize by line
	  number instead of by function if a filename is given. * Made CLI
	  "memory show summary" sort its output by filename or
	  function-name/line-number depending upon request. * Miscellaneous
	  tweaks to CLI "memory show summary" command output format.
	  ........ Merged revisions 376758 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376759 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-28 16:37 +0000 [r376727]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: manager: Make challenge work with
	  allowmultiplelogin=no Prior to this patch, challenge would yield
	  a multiple logins error if used without providing the username
	  (which isn't really supposed to be an argument to challenge) if
	  allowmultiplelogin was set to no because allowmultiplelogin finds
	  a user with a zero length login name. This check is simply
	  disabled for the challenge action when the username is empty by
	  this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
	  Patches: challenge_action_nomultiplelogin.diff uploaded by
	  Jonathan Rose (license 6182) ........ Merged revisions 376725
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 376726 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-28 00:08 +0000 [r376629-376690]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
	  char. The '-' char is supposed to be ignored by the dialplan
	  extension matching. Unfortunately, it's treatment is not handled
	  consistently throughout the extension matching code. * Made the
	  old exten matching code consistently ignore '-' chars. * Made the
	  old exten matching code consistently handle case in the matching.
	  * Made ignore empty character sets. * Fixed ast_extension_cmp()
	  to return -1, 0, or 1 as documented. The only user of it in
	  pbx_lua.c was testing for -1. It was originally returning the
	  strcmp() value for less than which is not usually going to be -1.
	  * Fix character set sorting if the sets have the same number of
	  characters and start with the same character. Character set [0-9]
	  now sorts before [02-9a] as originally intended. * Updated some
	  extension label and priority already in use warnings to also
	  indicate if the extension is aliased. (closes issue
	  ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
	  Harzenetter Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2201/ ........ Merged
	  revisions 376688 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376689 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
	  pbx/pbx_dundi.c: Remove unnecessary channel module references. *
	  Removed call to ast_module_user_hangup_all() in
	  res_config_mysql.c since it is effectively a noop. No channels
	  can attach a reference to that module. * Removed call to
	  ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
	  of unload_module() has already called it. * Removed redundant
	  channel module references in pbx_dundi.c. The registered dialplan
	  function callback dispatchers for the read/read2/write callbacks
	  already reference the module before calling. * pbx_dundi: Moved
	  unregistering CLI commands, DUNDi switch, and dialplan functions
	  to the first thing the unload_module() does. This will reduce the
	  chance of new channels using DUNDi services while the module is
	  being torn down. ........ Merged revisions 376657 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376658 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
	  and use better names. * Update doxygen of AST_LIST_REMOVE().
	  ........ Merged revisions 376627 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376628 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-22 23:58 +0000 [r376588]  Matthew Jordan <mjordan@digium.com>

	* main/lock.c, /, main/logger.c, include/asterisk/lock.h:
	  Re-initialize logmsgs mutex upon logger initialization to prevent
	  lock errors Similar to the patch that moved the fork earlier in
	  the startup sequence to prevent mutex errors in the recursive
	  mutex surrounding the read/write thread registration lock, this
	  patch re-initializes the logmsgs mutex. Part of the start up
	  sequence before forking the process into the background includes
	  reading asterisk.conf; this has to occur prior to the call to
	  daemon in order to read startup parameters. When reading in a
	  conf file, log statements can be generated. Since this can't be
	  avoided, the mutex instead is re-initialized to ensure a reset of
	  any thread tracking information. This patch also includes some
	  additional debugging to catch errors when locking or unlocking
	  the recursive mutex that surrounds locks when the DEBUG_THREADS
	  build option is enabled. DO_CRASH or THREAD_CRASH will cause an
	  abort() if a mutex error is detected. (issue ASTERISK-19463)
	  Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
	  376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 376587 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-20 21:58 +0000 [r376561]  David M. Lee <dlee@digium.com>

	* res/res_http_websocket.c: Added missing newlines to websocket
	  ast_logs. Without these newlines, log messages just continue
	  tacking onto the same line, and do not flush immediately.

2012-11-20 18:57 +0000 [r376550]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/include/sip.h, /, channels/chan_sip.c: Add "Require:
	  timer" to 200 OK responses when appropriate. The method by which
	  the Require header is added to 200 responses is inspired by the
	  method that Olle Johansson uses in his darjeeling-prack branch.
	  (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
	  behest of Olle Johansson Review:
	  https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
	  376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 376522 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-20 17:37 +0000 [r376540]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_sip.c: Reduce CLI spam of "Extension Changed"
	  device state messages. Asterisk 11 follows RFC3265 that states
	  that after every subscribe or resubscribe a notify should be
	  sent. Thus the console if filled continuously with the following
	  after every subscribe; == Extension Changed 8512[phones] new
	  state IDLE for Notify User cisco1 In Asterisk 1.8 only changes
	  would be sent. Thus only when a device state changed was anything
	  emitted to the console. fix: Only print to console when device
	  state isn't forced. (closes issue ASTERISK-20706) Reported by:
	  alecdavis Tested by: alecdavis alecdavis (license 585)

2012-11-19 19:54 +0000 [r376471]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c, main/security_events.c,
	  main/indications.c: Fix most leftover non-opaque ast_str uses.
	  Instead of calling str->str, one should use ast_str_buffer(str).
	  Same goes for str->used as ast_str_strlen(str) and str->len as
	  ast_str_size(str). Review:
	  https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
	  376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 376470 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-18 20:22 +0000 [r376415-376441]  Matthew Jordan <mjordan@digium.com>

	* main/asterisk.c, /, main/utils.c: Reorder startup sequence to
	  prevent lockups when process is sent to background Although it is
	  very rare and timing dependent, the potential exists for the call
	  to 'daemon' to cause what appears to be a deadlock in Asterisk
	  during startup. This can occur when a recursive mutex is obtained
	  prior to the daemon call executing. Since daemon uses fork to
	  send the process into the background, any threading primitives
	  are unsafe to re-use after the call. Implementations of pthread
	  recursive mutexes are highly likely to store the thread
	  identifier of the thread that previously obtained the mutex. If
	  the mutex was locked prior to the fork, a subsequent unlock
	  operation will potentially fail as the thread identifier is no
	  longer valid. Since the mutex is still locked, all subsequent
	  attempts to grab the mutex by other threads will block. This
	  behavior exhibited itself most often when DEBUG_THREADS was
	  enabled, as this compile time option surrounds the mutexes in
	  Asterisk with another recursive mutex that protects the storage
	  of thread related information. This made it much more likely that
	  a recursive mutex would be obtained prior to daemon and unlocked
	  after the call. This patch does the following: a) It backports a
	  patch from Asterisk 11 that prevents the spawning of the
	  localtime monitoring thread. This thread is now spawned after
	  Asterisk has fully booted. b) It re-orders the startup sequence
	  to call daemon earlier during Asterisk startup. This limits the
	  potential of threading primitives being accessed by
	  initialization calls before daemon is called. c) It removes calls
	  to ast_verbose/ast_log/etc. prior to daemon being called.
	  Developers should send error messages directly to stderr prior to
	  daemon, as calls to ast_log may access recursive mutexes that
	  store thread related information. d) It reorganizes when thread
	  local storage is created for storing lock information during the
	  creation of threads. Prior to this patch, the read/write lock
	  protecting the list of threads in ast_register_thread would
	  utilize the lock in the thread local storage prior to it being
	  initialized; this patch prevents that. On a very related note,
	  this patch will *greatly* improve the stability of the Asterisk
	  Test Suite. Review: https://reviewboard.asterisk.org/r/2197
	  (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
	  mjordan ........ Merged revisions 376428 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376431 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/confbridge/conf_state.c, /: Add a test event that reports
	  changes in ConfBridge state This patch adds a test event to
	  ConfBridge that reports transitions between states in ConfBridge.
	  This is used by tests in the Asterisk Test Suite that verify
	  state changes based on the entering/leaving of conference
	  participants. ........ Merged revisions 376414 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-16 19:59 +0000 [r376391]  Jonathan Rose <jrose@digium.com>

	* res/res_monitor.c, /: monitor: prevent attempts to move/remove
	  recordings skipped with 'i' and 'o'. The i and o options for
	  monitor skip the input and output sides of a recording
	  respectively. This patch addresses a problem in those options
	  when monitor is called without specifying a specific filename
	  where monitor will try to move the recording that was skipped.
	  Since this usually doesn't exist when these options are used, it
	  would produce a warning when it does this in most cases, but it
	  is conceivable that there are use cases where this could result
	  in moving/removing a file unintentionally. (closes issue
	  ASTERISK-20641) Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/2190/ ........ Merged
	  revisions 376389 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376390 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-16 00:09 +0000 [r376339-376343]  David M. Lee <dlee@digium.com>

	* /, utils/extconf.c: Fixed extconf.c breakage introduced in
	  r376306. To quote wdoekes: > Note that I'm not confirming
	  legitimacy of having that file in tree at > all. Is anyone using
	  aelparse/conf2ael? ........ Merged revisions 376340 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376342 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* utils/Makefile, tests/test_astobj2_thrash.c (added),
	  utils/utils.xml, /, utils/hashtest.c (removed),
	  tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
	  include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit
	  tests. Both hashtest and hashtest2 are manual testing apps that
	  thrash hash tables (hashtab and ao2 containers, respectively), by
	  spinning up several threads that randomly insert, delete, lookup
	  and iterate over the hash table. If the app doesn't crash, the
	  hash table probably passes the test. Those utils are not a part
	  of the typical Asterisk build, so they do not usually get
	  compiled. This all makes them less that useful. This patch
	  removes those manual test programs and replaces them with
	  Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
	  also attempts to make the tests more deterministic. * Rather than
	  spinning up some number of threads that operate on the hash table
	  randomly, spin up four threads that concurrenly add, remove,
	  lookup and iterate over the hash table. * Each thread checks the
	  state of the hash table both during and after execution, and
	  indicates a test failure if things are not as expected. * Each
	  thread times out after 60 seconds to prevent deadlocking the unit
	  test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
	  revisions 376306 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376315 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-15 23:03 +0000 [r376310]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: app_meetme: Fix channels lingering when
	  hung up under certain conditions Channels would get stuck and
	  MeetMe would repeatedly display an Unable to write frame to
	  channel error in the conf_run function if hung up during certain
	  sound prompts such as during user count announcements. This patch
	  fixes that by reintroducing a hangup check in the meetme's main
	  loop (also in conf_run). (closes issue ASTERISK-20486) Reported
	  by: Michael Cargile Review:
	  https://reviewboard.asterisk.org/r/2187/ Patches:
	  meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
	  Rose (license 6182) ........ Merged revisions 376307 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376308 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-15 02:08 +0000 [r376264]  Rusty Newton <rnewton@digium.com>

	* apps/app_voicemail.c, /: Patch to play correct sound file when a
	  voicemail's urgent status is removed We were attempting to play
	  "vm-urgent-removed", which didn't exist. Now we play
	  "vm-marked-nonurgent" which exists and is the correct sound file.
	  Previous behavior was silence and a warning on the CLI. (issue
	  ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
	  Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
	  uploaded by Rusty Newton (license 5829) ........ Merged revisions
	  376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 376263 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-14 19:53 +0000 [r376234]  Richard Mudgett <rmudgett@digium.com>

	* pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
	  Future dated call files are ignored when astspooldir is relative
	  to the current directory. The queue_file() assumed that the qdir
	  needed to be prepended if the given filename did not start with a
	  '/'. If astspooldir is relative it is not going to start from the
	  root directory obviously so it will not start with a '/'. The
	  filename used in queue_file() ultimately results in qdir
	  prepended multiple times. * Made queue_file() not prepend qdir if
	  the filename contains a '/'. (closes issue ASTERISK-20593)
	  Reported by: James Le Cuirot Patches:
	  0004-Fix-future-call-files-from-relative-directories.patch
	  (license #6439) patch uploaded by James Le Cuirot ........ Merged
	  revisions 376232 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376233 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-13 18:48 +0000 [r376217]  Brent Eagles <beagles@digium.com>

	* main/channel.c, /: Patch to prevent stopping the active generator
	  when it is not the silence generator. This patch introduces an
	  internal helper function to safely check whether the current
	  generator is the one that is expected before deactivating it. The
	  current externally accessible ast_channel_stop_generator()
	  function has been modified to be implemented in terms of the new
	  function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
	  ........ Merged revisions 376199 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376208 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-12 20:45 +0000 [r376168]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, /: Properly check if the "Context" and "Extension"
	  headers are empty in a ShowDialPlan action. The code which
	  handles the ShowDialPlan action wrongly assumed that a non-NULL
	  return value from the function which retrieves headers from an
	  action indicates that the header has a value. This is incorrect
	  and the contents must be checked to see if they are blank.
	  (closes issue ASTERISK-20628) Reported by: jkroon Patches:
	  asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
	  ........ Merged revisions 376166 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376167 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-12 20:16 +0000 [r376144]  Michael L. Young <elgueromexicano@gmail.com>

	* main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
	  Problem When adding a dynamic hint, if an extension contains an
	  underscore no variable subsitution is being performed. This patch
	  changes from checking if the extension contains an underscore to
	  checking if the extension begins with an underscore. (closes
	  issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
	  Steven T. Wheeler, Michael L. Young Patches:
	  asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
	  L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2188/ ........ Merged
	  revisions 376142 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376143 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-11 17:08 +0000 [r376130]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_sip.c,
	  configs/sip.conf.sample: Remove a fixed size limitation for
	  producing SDP and change how ICE support is disabled by default.
	  With ICE support enabled in chan_sip and a large number of
	  interfaces on the system it was possible for the produced SDP to
	  be truncated due to some fixed size buffers. These buffers have
	  now been changed so they will dynamically grow as needed. ICE
	  support is now also enabled by default in res_rtp_asterisk to
	  provide a smoother experience for chan_motif users where it is
	  required. To maintain the previous behavior in chan_sip it is no
	  longer enabled by default there. (closes issue ASTERISK-20643)
	  Reported by: coopvr

2012-11-08 22:08 +0000 [r376089]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
	  Turns out the "helpful" setting of ms and res in this macro is
	  completely useless after the timeout antipattern fix. If you're a
	  new guy looking to write code, don't write a macro like this one.
	  ........ Merged revisions 376087 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376088 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-08 21:10 +0000 [r376048-376060]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c, /: chan_dahdi/SS7: Made reject incoming call
	  for an in-alarm or blocked channel. If a SS7 call comes in
	  requesting a CIC that is in-alarm, the call is accepted and
	  connects if the extension exists in the dialplan. The call does
	  not have any audio. * Made release the call immediately with
	  circuit congestion cause. (closes issue ASTERISK-20204) Reported
	  by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
	  #5621) patch uploaded by rmudgett ........ Merged revisions
	  376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 376059 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/asterisk.c, include/asterisk/utils.h,
	  include/asterisk/astmm.h, /, main/utils.c, main/astmm.c: Add
	  MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc().
	  It will return a memory block filled with 0x55. A nonzero value.
	  * Makes free() fill the released memory block and boundary
	  fence's with 0xdeaddead. Any pointer use after free is going to
	  have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is
	  usually an invalid memory address so a crash is expected. * Puts
	  the freed memory block into a circular array so it is not reused
	  immediately. * When the circular array rotates out a memory block
	  to the heap it checks that the memory has not been altered from
	  0xdeaddead. * Made the astmm_log message wording better. * Made
	  crash if the DO_CRASH menuselect option is enabled and something
	  is found. * Fixed a potential alignment issue on 64 bit systems.
	  struct ast_region.data[] should now be aligned correctly for all
	  platforms. * Extracted region_check_fences() from
	  __ast_free_region() and handle_memory_show(). * Updated
	  handle_memory_show() CLI usage help. Review:
	  https://reviewboard.asterisk.org/r/2182/ ........ Merged
	  revisions 376029 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 376030 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-07 19:03 +0000 [r376014]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c,
	  main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c,
	  res/res_fax.c, apps/app_record.c, channels/chan_agent.c,
	  main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
	  channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
	  channels/chan_dahdi.c, apps/app_waitforring.c,
	  channels/sig_analog.c: Multiple revisions 375993-375994 ........
	  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov
	  2012) | 30 lines Fix misuses of timeouts throughout the code.
	  Prior to this change, a common method for determining if a
	  timeout was reached was to call a function such as
	  ast_waitfor_n() and inspect the out parameter that told how many
	  milliseconds were left, then use that as the input to
	  ast_waitfor_n() on the next go-around. The problem with this is
	  that in some cases, submillisecond timeouts can occur, resulting
	  in the out parameter not decreasing any. When this happens
	  thousands of times, the result is that the timeout takes much
	  longer than intended to be reached. As an example, I had a
	  situation where a 3 second timeout took multiple days to finally
	  end since most wakeups from ast_waitfor_n() were under a
	  millisecond. This patch seeks to fix this pattern throughout the
	  code. Now we log the time when an operation began and find the
	  difference in wall clock time between now and when the event
	  started. This means that sub-millisecond timeouts now cannot play
	  havoc when trying to determine if something has timed out. Part
	  of this fix also includes changing the function ast_waitfor() so
	  that it is possible for it to return less than zero when a
	  negative timeout is given to it. This makes it actually possible
	  to detect errors in ast_waitfor() when there is no timeout.
	  (closes issue ASTERISK-20414) reported by David M. Lee Review:
	  https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
	  mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
	  lines Remove some debugging that accidentally made it in the last
	  commit. ........ Merged revisions 375993-375994 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375995 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-06 18:59 +0000 [r375966]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/features.h, main/channel.c, /,
	  main/channel_internal_api.c, main/features.c,
	  include/asterisk/channel.h: Fix stuck DTMF when bridge is broken.
	  When a bridge is broken by an AMI Redirect action or the
	  ChannelRedirect application, an in progress DTMF digit could be
	  stuck sending forever. * Made simulate a DTMF end event when a
	  bridge is broken and a DTMF digit was in progress. (closes issue
	  ASTERISK-20492) Reported by: Jeremiah Gowdy Patches:
	  bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by
	  Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch
	  jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2169/ ........ Merged
	  revisions 375964 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375965 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-12-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.1.0 Released.

2012-12-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.1.0-rc3 Released.

	* chan_local: Fix local_pvt ref leak in local_devicestate().

	Regression introduced by ASTERISK-20390 fix.

	(closes issue ASTERISK-20769)
	Reported by: rmudgett

2012-12-05  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.1.0-rc2 Released.

	* Fix a SIP request memory leak with TLS connections.

	During the TLS re-work in chan_sip some TLS specific code was moved
	into a separate function. This function operates on a copy of the
	incoming SIP request. This copy was never deinitialized causing a
	memory leak for each request processed.

	This function is now given a SIP request structure which it can use
	to copy the incoming request into. This reduces the amount of memory
	allocations done since the internal allocated components are reused
	between packets and also ensures the SIP request structure is
	deinitialized when the TLS connection is torn down.

	(closes issue ASTERISK-20763)
	Reported by: deti

2012-11-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.1.0-rc1 Released.

2012-11-06 12:09 +0000 [r375925]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Fix a bug where our Motif ICE candidates
	  were not quite proper, and make us more forgiving. An issue was
	  reported on the mailing list where calling would result in an
	  "Incomplete ICE-UDP candidate received on session" error message.
	  This is the result of the ICE-UDP candidate code not placing a
	  "network" attribute within the candidates. This is now done. To
	  increase compatibility though I have removed the requirement for
	  the "network" attribute to exist within ICE-UDP candidates that
	  are received since we don't actually require the value. Reported
	  on the mailing list by Jean-Denis Girard.

2012-11-05 23:09 +0000 [r375895]  Matthew Jordan <mjordan@digium.com>

	* main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
	  res/res_timing_dahdi.c, res/res_timing_timerfd.c,
	  bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
	  include/asterisk/timing.h, res/res_musiconhold.c,
	  channels/chan_iax2.c, res/res_fax_spandsp.c,
	  res/res_timing_kqueue.c: Refactor ast_timer_ack to return an
	  error and handle the error in timer users Currently, if an
	  acknowledgement of a timer fails Asterisk will not realize that a
	  serious error occurred and will continue attempting to use the
	  timer's file descriptor. This can lead to situations where errors
	  stream to the CLI/log file. This consumes significant resources,
	  masks the actual problem that occurred (whatever caused the timer
	  to fail in the first place), and can leave channels in odd
	  states. This patch propagates the errors in the timing resource
	  modules up through the timer core, and makes users of these
	  timers handle acknowledgement failures. It also adds some
	  defensive coding around the use of timers to prevent using bad
	  file descriptors in off nominal code paths. Note that the patch
	  created by the issue reporter was modified slightly for this
	  commit and backported to 1.8, as it was originally written for
	  Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
	  (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
	  jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
	  6358) ........ Merged revisions 375893 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375894 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-05 21:41 +0000 [r375864]  Richard Mudgett <rmudgett@digium.com>

	* main/loader.c, /: Add safety NULL pointer check in module user
	  references. Made __ast_module_user_remove() check for NULL
	  pointers. ........ Merged revision 375860 from C.3 ........
	  Merged revisions 375862 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375863 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-05 17:59 +0000 [r375847]  Jonathan Rose <jrose@digium.com>

	* /, UPGRADE.txt: chan_sip: Document a change to user-field
	  encoding introduced with r303509 The change in question was added
	  to improve compliance with RFC3261, but at the time of commit, it
	  wasn't adequately documented in the UPGRADE notes. (closes issue
	  ASTERISK-20561) Reported by: Deniz Review:
	  https://reviewboard.asterisk.org/r/2177/ ........ Merged
	  revisions 375846 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-04 03:09 +0000 [r375729-375802]  Matthew Jordan <mjordan@digium.com>

	* main/manager.c, /: Don't attempt to purge sessions when no
	  sessions exist Manager's tcp/tls objects have a periodic function
	  that purge old manager sessions periodically. During shutdown,
	  the underlying container holding those sessions can be disposed
	  of and set to NULL before the tcp/tls periodic function is
	  stopped. If the periodic function fires, it will attempt to
	  iterate over a NULL container. This patch checks for whether or
	  not the sessions container exists before attempting to purge
	  sessions out of it. If the sessions container is NULL, we simply
	  return. Note that this error was also caught by the Asterisk Test
	  Suite. ........ Merged revisions 375800 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375801 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_fax.c: Only deref a reserved gateway session if we
	  actually reserved one Its perfectly acceptable to have a gateway
	  session unreserved when we go to first allocate one. Unreffing
	  the reserved gateway session - when its NULL - will result in an
	  assertion error. This problem was caught by the Asterisk Test
	  Suite (once we had enough of the debugging flags enabled)
	  ........ Merged revisions 375797 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/manager.c, /: Properly clean up manager resources on exit
	  This patch does two things: 1) It properly unregisters the
	  manager CLI commands 2) It cleans up AMI users on exit. Prior to
	  this patch, the AMI users were not being disposed of properly,
	  resulting in a memory leak. (closes issue ASTERISK-20646)
	  Reported by: Corey Farrell patches: manager_shutdown.patch
	  uploaded by Corey Farrell (license 5909) ........ Merged
	  revisions 375793 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375794 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/db.c, /: Properly finalize prepared SQLite3 statements to
	  prevent memory leak The AstDB uses prepared SQLite3 statements to
	  retrieve data from the SQLite3 database. These statements should
	  be finalized during Asterisk shutdown so that the SQLite3
	  database can be properly closed. Failure to finalize the
	  statements results in a memory leak and a failure when closing
	  the database. This patch fixes those issues by ensuring that all
	  prepared statements are properly finalized at shutdown. (closes
	  issue ASTERISK-20647) Reported by: Corey Farrell patches:
	  astdb-sqlite3_close.patch uploaded by Corey Farrell (license
	  5909) ........ Merged revisions 375761 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/xmldoc.c: Fix memory leaks in XML documentation This patch
	  fixes two memory leaks: 1) When building XML documentation items,
	  the 'name' attribute was extracted from XML elements but not
	  properly freed after being copied into the item being built. 2)
	  When unloading XML documentation, the doctree container objects
	  were not properly freed. This patch corrects these memory leaks.
	  Note that this patch was modified slightly for this commmit, as
	  the case where the 'name' attribute doesn't exist also wasn't
	  handled in the item construction. This patch also checks for that
	  attribute not existing. (closes issue ASTERISK-20648) Reported
	  by: Corey Farrell Tested by: mjordan patches:
	  xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)

	* main/cdr.c, /: Prevent multiple CDR batches from conflicting when
	  scheduling the CDR write The Asterisk Test Suite caught an error
	  condition where a scheduled CDR batch write can be deleted twice
	  if two channels attempt to post their CDRs at the same time. The
	  batch CDR mutex is locked while the CDRs are appended to the
	  current batch list; however, it is unlocked prior to actually
	  scheduling the CDR write. As such, two threads can attempt to
	  remove the currently scheduled batch write at the same time,
	  resulting in an assertion error. This patch extends the time that
	  the mutex is locked to encompass actually scheduling the write.
	  This prevents two threads from unscheduling the currently
	  scheduled write at the same time. ........ Merged revisions
	  375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 375728 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-03 03:17 +0000 [r375702]  Andrew Latham <lathama@gmail.com>

	* README, include/asterisk/doxyref.h: Doxygen Updates Replace links
	  to missing text files removed in the 1.6.x series with links to
	  the wiki. Doxygen can handle URLs fine, don't atempt to quote
	  them. Also update the wiki link in the Readme to get everyone on
	  the same page. (issue ASTERISK-20259) ........ Merged revisions
	  375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 375699 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-02 20:59 +0000 [r375661]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, channels/chan_misdn.c, /, main/ccss.c,
	  main/format_pref.c: Things don't need to be that const. ........
	  Merged revisions 375658 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375659 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-02 20:56 +0000 [r375660]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports
	  open Skinny wasn't closing RTP sockets. This patch includes
	  ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes
	  the problem. Also add destroy for VRTP (which I believe is
	  unused, but exists). Review:
	  https://reviewboard.asterisk.org/r/2176/

2012-11-02 18:44 +0000 [r375627]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Multiple
	  revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
	  16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
	  primitives must be handled first. The frm->addr is a different
	  "address space" than the stack/instance address of other Lx
	  primitives. The test for B channel instance address could fail.
	  Patches: patch01_timers.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
	  2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
	  chan_misdn: Free memory in error paths and other memory leaks.
	  The one line commented with BUG is not easily fixable because
	  there is no de-init function one can call. Patches:
	  patch02_memory.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
	  16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
	  L2 de-establish/establish * An NT-PTMP cannot de/establish L2
	  since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
	  is finally active in handle_l1. * L2 deactivation logging
	  cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
	  as "UNKN". * Removed unused functions and code for L2 handling.
	  Patches: patch03_L2estab.diff (license #6372) patch uploaded by
	  Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
	  rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
	  lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
	  prim via lower_id layer (3 or 1) simply does not work. For TE (3)
	  it returns an error (len=-6) which is not evaluated by
	  handle_l1(), so the L1 layer status ends up wrong. Instead PH
	  must be sent via L4, only then does it reach L1 without an error
	  message. And NT PH prims only reach L1 when they are sent to
	  layer 2 id. --> use upper_id to send PH primitives. * Check for
	  errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
	  improved. * The lower_id is now not used for anything, except:
	  Why is lower_id layer deleted when it wasn't created? I removed
	  this code since it looks very wrong. Patches:
	  patch04_l1activation.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
	  2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
	  chan_misdn: Fix loss of B channels if L1 is down. If you make 2
	  calls out an NT PTMP port which is not connected to any phone,
	  the B channel associated with that call becomes unusable until
	  Asterisk is restarted. The problem is the EVENT_SETUP is queued
	  when L1 is not up in misdn_lib_send_event(). If L1 cannot be
	  activated the event won't be dequeued. It gets even worse when
	  the call is hung up. The queued EVENT_SETUP will be overwritten
	  by an EVENT_DISCONNECT. The reserved B channel then will never be
	  freed. If later someone connects a phone to the port, L1 will
	  eventually activate and the queued EVENT_DISCONNECT is sent down
	  the stack. However, it is ignored because it is the wrong call
	  state. The real fix would be that activation and queueing for a
	  new SETUP is done by the NT stack. But since it doesn't, the
	  workaround must be removed because it doesn't always work. Fix:
	  The event is no longer queued but immediately sent to the stack.
	  If L1 cannot be activated, the L3 state machine that was started
	  by the EVENT_SETUP will do its work, i.e. a timeout will release
	  the B channel properly. The SETUP possibly cannot be sent the
	  first time but is resent by T303 in case L1 could be activated.
	  Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
	  by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
	  rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
	  lines chan_misdn: Remove some calls to exit(). Try proper cleanup
	  when something goes wrong in misdn_lib_init(). Especially do not
	  call exit()! * Fix memory leak because stack_destroy() does not
	  free the stack struct. Patches: patch06_cleanup-init.diff
	  (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
	  ABE-2888 ........ Merged revisions 375519-375524 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 375625 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375626 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-02 17:24 +0000 [r375613]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
	  Origin Processing While looking at some debug logs, I noticed
	  that it was being reported that the SDP origin line was
	  unsupported or failed. Upon looking into this on my local
	  machine, I found that I too was getting this debug message yet
	  everything seemed to be getting processed properly. What was
	  discovered is, that, the variable to determine what is displayed
	  in the debug message for the SDP line that was processed, was not
	  being set for the origin line when the result was successful.
	  This patch fixes this and was tested on local machine. ........
	  Merged revisions 375594 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375601 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-11-01 14:52 +0000 [r375575]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Fix a bug
	  causing SIP reloads to remove all entries from the registry A
	  regression was introduced in chan_sip by changes to sip reload
	  introduced by r349097. That patch moved peer purging from the
	  beginning of the reload to after the general configuration was
	  finished. This patch fixes that by undoing the repositioning of
	  the original peer purging code and using a similar function after
	  performing general configuration that purges only autocreated
	  peers that were created when persist mode isn't enabled. (closes
	  issue ASTERISK-20611) Reported by: Alisher Review:
	  https://reviewboard.asterisk.org/r/2171/

2012-10-31 18:00 +0000 [r375559]  Joshua Colp <jcolp@digium.com>

	* res/res_http_websocket.exports.in: Fix an issue with
	  res_http_websocket where the chan_sip WebSocket handler could not
	  be registered. On some systems the optional API support uses the
	  GCC compiler attribute "weakref" to provide its functionality.
	  This code changes the function names and prefixes "__" to the
	  front. The res_http_websocket exports file did not take this into
	  account, thereby not allowing those functions to be global and
	  ultimately found. (closes issue ASTERISK-20631) Reported by:
	  danjenkins

2012-10-31 14:49 +0000 [r375532]  Matthew Jordan <mjordan@digium.com>

	* res/res_calendar_ews.c, /: Properly extract the Body information
	  of an EWS calendar item Unlike all other calendar modules,
	  res_calendar_ews fails to extract the Body information for a
	  calendar item. This is due, in part, to a quirk in the schema in
	  the XML - not only does a CalendarItem contain a Body element,
	  but the CalendarItem exists as a descendant of a different Body
	  element. The neon parser was erroneously skipping all Body
	  elements. This patch fixes that by bypassing Body elements that
	  are not a child of CalendarItem, and parsing the Body element out
	  if it is a child. Note that the original patch by Terry Wilson
	  only needed slight modifications to make it properly pull the
	  Body information out; as such, while I've linked to the patch
	  that I uploaded for Dmitry, I've attributed the patch to Terry.
	  (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
	  by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
	  uploaded by Terry Wilson (license 6283) ........ Merged revisions
	  375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 375531 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-30 19:23 +0000 [r375506]  Richard Mudgett <rmudgett@digium.com>

	* /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing
	  module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
	  Patches: smfix.patch (license #6099) patch uploaded by feyfre
	  Modified for coding guidelines. ........ Merged revisions 375496
	  from http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-30 19:09 +0000 [r375471-375486]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_mixmonitor.c: mixmonitor: Add a test event This test
	  event is being used to fix the mixmonitor_audiohook_inherit test.
	  ........ Merged revisions 375484 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375485 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_confbridge.c: confbridge: Fix a bug which made
	  conferences not record with AMI/CLI commands When confbridge was
	  changed to handle conference status with a state machine in
	  r374658. The function responsible for starting recording for a
	  conference was refactored with the function actually responsible
	  for launching the recording thread being split into a function
	  with another name. The old function name was still used for
	  manually started recordings through AMI or CLI. This patch fixes
	  that by switching which function is used to start recording the
	  conference. (closes issue ASTERISK-20601) Reported by: Vilius
	  Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose
	  (license 6182) ........ Merged revisions 375470 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-30 02:22 +0000 [r375469]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_queue.c: Ensure that the Queue application tracks
	  busy members in off nominal situations There are a few code paths
	  where the Queue application fails to count a paused or in use
	  queue member as being 'busy'. This can cause callers to get stuck
	  in the Queue until a paused agent unpauses themselves. (closes
	  issue ASTERISK-20623) Reported by: Bryan Walters patches:
	  app_queue.patch uploaded by Bryan Walters (license 5851) ........
	  Merged revisions 375450 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375451 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-29 21:23 +0000 [r375437]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Prevent resetting of NATted realtime peer
	  address on reload. If a "sip reload" is issued for a SIP peer,
	  then his IP address will be cleared, thus resulting in forgetting
	  the public IP address. Asterisk will then attempt to route SIP
	  traffic to the private IP address. The fix here is to make "sip
	  reload" ignore realtime peers when "host = dynamic" is spotted.
	  Realtime peers can now only have their IP address reset if they
	  have gone from being not dynamic to being dynamic. (closes issue
	  ASTERISK-18203) reported by daren ferreira (closes issue
	  ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
	  uploaded by JoshE (license #6075) ........ Merged revisions
	  375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 375417 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-29 19:29 +0000 [r375363-375390]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix the Park 'r' option when a channel parks
	  itself. When a channel uses the Park appliation to park itself
	  with the 'r' option, the channel hears music-on-hold instead of
	  the requested ringing. * Added a missing check for the 'r' option
	  when a channel parks itself. (closes issue ASTERISK-19382)
	  Reported by: James Stocks Patches by: dsessions Review:
	  https://reviewboard.asterisk.org/r/2148/ ........ Merged
	  revisions 375388 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375389 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
	  a NULL tech_pvt. The tech support customer was using the AMI
	  Redirect action shortly after a call was placed. While the
	  channel tried to do an ast_read(), the masquerade resulting from
	  the channel redirect took place. The masquerade in the middle of
	  the ast_read() resulted in the segfault. (closes issue AST-1025)
	  Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
	  (license #5621) patch uploaded by rmudgett ........ Merged
	  revisions 375361 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375362 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-23 16:22 +0000 [r375288-375327]  Jonathan Rose <jrose@digium.com>

	* contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better
	  response for various exit conditions to openssl (closes issue
	  ASTERISK-20260) Reported by: Daniel O'Connor Patches:
	  ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
	  6419) ........ Merged revisions 375325 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375326 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/app.c: core: Fix a memory leak in app.c from an early
	  return ast_app_group_match_get_count allocates memory with the
	  regcomp function and we previously forgot to free it when bailing
	  out due to a regex compilation failure against category. (closes
	  issue AST-1018) Reported by: Guenther Kelleter Patches:
	  regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
	  ........ Merged revisions 375299 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375300 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM
	  (closes issue ASTERISK-20457) Reported by: Richard Miller
	  Patches: code.patch uploaded by Richard Miller (license 5685)
	  ........ Merged revisions 375272 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375273 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-18 21:44 +0000 [r375219-375247]  Jonathan Rose <jrose@digium.com>

	* UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE
	  notes describing behavioral changes to rrmemory strategy caused
	  by 375216 (issue AST-989) Reported by: Thomas Arimont

	* /, apps/app_queue.c: app_queue: Make ordering of
	  rrmemory/rrordered persist over add/remove members Prior to this
	  patch, adding, removing or reloading members to rrmemory would
	  cause the order to become completely jumbled. Now it behaves more
	  or less like rrordered other than the fact that it stores the
	  members on a hash table rather than a linked list. This patch
	  also prevents removal of members and member reloads from jumbling
	  rrordered queues. (issue AST-989) Reported by: Thomas Arimont
	  Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
	  revisions 375216 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375217 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-18 20:02 +0000 [r375191]  Richard Mudgett <rmudgett@digium.com>

	* Makefile, /, build_tools/make_version, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
	  build_tools: Allow Asterisk to report git SHAs in version string.
	  Make git more attractive for managing work-in-progress.
	  Especially convenient when a potential patch set needs to be
	  tested on multiple platforms since one can use git to keep all
	  the test environments in sync independent of a subversion server.
	  Now the Asterisk version will show the exact git SHA5 that was
	  used when building (still appended by "M" if there are local
	  modifications) from a git clone of the Asterisk repository so the
	  developer can more easily know what is actually under test. You
	  will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of
	  this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported
	  This has zero impact for those not using git with the exception
	  of an extra test in the configure script to gather git's path.
	  This is necessary to prevent "sudo make install" from failing
	  since git may not be in the path in make's shell environment.
	  (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches:
	  0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
	  (license #5417) patch uploaded by Shaun Ruffell Modified ........
	  Merged revisions 375189 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375190 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-17 19:00 +0000 [r375148]  Kinsey Moore <kmoore@digium.com>

	* main/tcptls.c, /: Ensure Asterisk fails TCP/TLS SIP calls when
	  certificate checking fails When placing a call to a TCP/TLS SIP
	  endpoint whose certificate is not signed by a configured CA
	  certificate, Asterisk would issue a warning and continue to
	  process the call as if there was not an issue with the
	  certificate. Asterisk now properly fails the call if the
	  certificate fails verification or if the certificate does not
	  exist when certificate checking is enabled (the default
	  behavior). (closes issue ASTERISK-20559) Reported by: kmoore
	  Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged
	  revisions 375146 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375147 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-16 21:44 +0000 [r375079-375113]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
	  Don't crash on large user input. Allow SIP headers without space.
	  Optimize code a bit. Review:
	  https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
	  375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 375112 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Update sip_request_call SIP dial string
	  documentation. This was missed when merging review r1859.
	  ........ Merged revisions 375074 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375078 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-16 14:08 +0000 [r375051]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Remove a log message that was left in
	  accidentally from call-id logging development.

2012-10-15 21:15 +0000 [r375027]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, /, main/ccss.c, include/asterisk/strings.h,
	  channels/chan_iax2.c: Fix some potential misuses of ast_str in
	  the code. Passing an ast_str pointer by value that then calls
	  ast_str_set(), ast_str_set_va(), ast_str_append(), or
	  ast_str_append_va() can result in the pointer originally passed
	  by value being invalidated if the ast_str had to be reallocated.
	  This fixes places in the code that do this. Only the example in
	  ccss.c could result in pointer invalidation though since the
	  other cases use a stack-allocated ast_str and cannot be
	  reallocated. I've also updated the doxygen in strings.h to
	  include notes about potential misuse of the functions mentioned
	  previously. Review: https://reviewboard.asterisk.org/r/2161
	  ........ Merged revisions 375025 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 375026 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-15 08:11 +0000 [r375016]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Fix underscreen buttons warnings apeared
	  while transfer process

2012-10-14 11:57 +0000 [r374995]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* config.guess, config.sub, /: Update config.guess and config.sub:
	  2012-10-10 Update config.guess and config.sub to revision
	  fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
	  savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
	  64bit). config.guess:timestamp='2012-09-25'
	  config.sub:timestamp='2012-10-10' ........ Merged revisions
	  374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 374991 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-12 21:57 +0000 [r374932]  Kinsey Moore <kmoore@digium.com>

	* apps/app_voicemail.c: Avoid a segfault on invalid format names If
	  a format name was not found by ast_getformatbyname, a NULL
	  pointer would be passed into ast_format_rate and immediately
	  dereferenced. This ensures that a valid pointer is used since the
	  structure is already allocated on the stack. (closes issue
	  DPH-523) Reported-by: Steve Pitts

2012-10-12 16:20 +0000 [r374914]  Mark Michelson <mmichelson@digium.com>

	* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
	  Do not use a FILE handle when doing SIP TCP reads. This is used
	  to solve an issue where a poll on a file descriptor does not
	  necessarily correspond to the readiness of a FILE handle to be
	  read. This change makes it so that for TCP connections, we do a
	  recv() on the file descriptor instead. Because TCP does not
	  guarantee that an entire message or even just one single message
	  will arrive during a read, a loop has been introduced to ensure
	  that we only attempt to handle a single message at a time. The
	  tcptls_session_instance structure has also had an overflow buffer
	  added to it so that if more than one TCP message arrives in one
	  go, there is a place to throw the excess. Huge thanks goes out to
	  Walter Doekes for doing extensive review on this change and
	  finding edge cases where code could fail. (closes issue
	  ASTERISK-20212) reported by Phil Ciccone Review:
	  https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
	  374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 374906 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-11 21:18 +0000 [r374850-374877]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Fix a bug where audio on Google Voice
	  would not work due to ignoring candidates. Instead of ignoring
	  parts of the message that are not known just ignore the ones we
	  know may be present and that would cause a problem.

	* res/res_rtp_asterisk.c: Remove code that should not have gotten
	  in. (issue ASTERISK-20554)

	* res/res_rtp_asterisk.c, channels/chan_motif.c: Fix an issue where
	  outgoing calls would fail to establish audio due to ICE
	  negotiation failures. This change removes the requirement for
	  ufrag and pwd in the transport stanza and also makes us the
	  controlling agent. (closes issue ASTERISK-20554) Reported by:
	  mmichelson

2012-10-11 15:44 +0000 [r374845]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, /: Fix incorrect billing duration reported when batch
	  mode is enabled Similar to r369351, the billing duration can be
	  skewed when batch mode is enabled. This happened much more rarely
	  than the duration, as it only occured when the call was answered
	  (thereby indicating an actual answer time) and immediately hung
	  up on (indicating a billsec of 0). Since a billing time of '0'
	  can either mean that the call immediately ended or that the CDR
	  was improperly answered, we have to use additional information to
	  know whether or not we can trust the CDR billsec value. Prior to
	  this patch, we looked to see if we had a valid answer time. If we
	  did, and billsec was zero, we used the current time to calculate
	  what billsec value we could from the CDR being written. If batch
	  mode is enabled, this will incorrectly report a billsec value
	  being much greater than the actual duration of the call. Instead
	  of relying on the presence of an answer time to know whether or
	  not we can re-calculate the billsec for the CDR, we now also use
	  the presence of the CDR's end time to know if we need to
	  re-calculate or whether we can trust the billsec value that we
	  have. This prevents erroneous jumps in the billsec value, while
	  still making sure that in the worst case, some billing time will
	  be calculated. (closes issue AST-1016) Reported by: Thomas
	  Arimont Tested by: Thomas Arimont ........ Merged revisions
	  374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 374844 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-11 15:31 +0000 [r374842]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, include/asterisk/sip_api.h,
	  channels/chan_sip.exports.in (removed), main/sip_api.c (added):
	  Don't make chan_sip export global symbols. During testing, it was
	  discovered that having chan_sip export global symbols was
	  problematic. The biggest problem was that load order was
	  affected. Trying to use realtime could be problematic since in
	  all likelihood the necessary realtime driver(s) would not be
	  loaded before chan_sip. In addition, it was found that it was
	  impossible to use the Digium Phone Module for Asterisk since it
	  must be loaded before chan_sip since it must hook into chan_sip's
	  configuration parsing. The solution is to use a virtual table in
	  the same manner that other modules in Asterisk do, like
	  app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore

2012-10-11 13:33 +0000 [r374833]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Consider the Google Talk content stanza
	  name (jin:content) valid.

2012-10-10 21:03 +0000 [r374804]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_queue.c: app_queue: Made pass connected line updates
	  from the caller to ringing queue members. Party A calls Party B
	  Party B puts Party A on hold. Party B calls a queue. Ringing
	  queue member D sees Party B identification. Party B transfers
	  Party A to the queue. Queue member D does not get a connected
	  line update for Party A. Queue member D answers the call and
	  still sees Party B information. However, if Party A later
	  transfers the call to Party C then queue member D gets a
	  connected line update for Party C. * Made pass connected line
	  updates from the caller to queue members while the queue members
	  are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
	  (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
	  rmudgett ........ Merged revisions 374801 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 374802 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374803 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-10 13:35 +0000 [r374792]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c: Fix segfault regression from r370681 Due to usage
	  of ast_hook_send_action, AMI action handling code should be able
	  to handle a NULL mansession->session. This would cause a crash on
	  NULL dereference if action_originate was called from
	  ast_hook_send_action. (closes issue ASTERISK-20544)

2012-10-09 22:21 +0000 [r374771]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /: Fix execution of 'i' extension due to
	  uninitialized variable. The fix for ASTERISK-18243 added code
	  that could potentially use dst_exten[] uninitialized. As a result
	  the 'i' exten may not be executed when it should. (closes issue
	  ASTERISK-20455) Reported by: Richard Miller Patches:
	  pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
	  Miller Made some cosmetic modifications. ........ Merged
	  revisions 374758 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374763 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-09 21:34 +0000 [r374755-374756]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Improve logging for DTLS-SRTP failure
	  situations. (closes issue ASTERISK-20487) Reported by: mjordan

	* channels/chan_sip.c: Add a log message for when DTLS-SRTP is
	  requested and the underlying engine does not support it. (closes
	  issue ASTERISK-20487) Reported by: mjordan

2012-10-08 22:30 +0000 [r374708-374729]  Richard Mudgett <rmudgett@digium.com>

	* configs/chan_dahdi.conf.sample, /: dahdi.conf.sample: Add
	  description for "buffers" setting. This contains an edited
	  version of the patch originally created by John Bigelow. (closes
	  issue ASTERISK-14435) Reported by: John Bigelow Patches:
	  buffers.patch (license #5091) patch uploaded by John Bigelow
	  0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
	  (license #5417) patch uploaded by Shaun Ruffell Modified ........
	  Merged revisions 374727 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374728 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* pbx/pbx_spool.c, /: Fix deletion of unopenable spool files. If
	  scan_service() cannot open the spool file, it logs a message
	  saying that it will delete the file and calls remove_from_queue()
	  to do it. However, remove_from_queue() fails to delete the spool
	  file because struct outgoing has not yet been fully initialized.
	  * Merged allocating a new struct outgoing and init_outgoing()
	  into new_outgoing(). Allocation is initialization. * Made
	  apply_outgoing() not initialize the spool filename in struct
	  outgoing. * Made apply_outgoing() call ast_trim_blanks() and
	  ast_skip_blanks() rather than manually inlining them. * Reduced
	  indentation levels in apply_outgoing(). * Fixed a garbled comment
	  in remove_from_queue(). * Reworked scan_service() to simplify it.
	  (closes issue ASTERISK-17231) Reported by: David Chappell
	  Patches: spool_open_failure.diff (license #4997) patch uploaded
	  by David Chappell Started with this patch. ........ Merged
	  revisions 374686 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some
	  memory leaks on off nominal paths in init_outgoing() when merging
	  into the new_outgoing() function dealing with o->capabilities.
	  ........ Merged revisions 374695 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-25  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.0.0 Released.

2012-10-17  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.0.0-rc2 Released.

	* [r374792] Fix segfault regression from r370681

	  Due to usage of ast_hook_send_action, AMI action handling code should
	  be able to handle a NULL mansession->session.  This would cause a
	  crash	on NULL dereference if action_originate was called from
	  ast_hook_send_action.

  	  (closes issue ASTERISK-20544)

	* [r374842] Don't make chan_sip export global symbols.

	  During testing, it was discovered that having chan_sip export global
	  symbols was problematic.

	  The biggest problem was that load order was affected.
	  Trying to use realtime could be problematic since in
	  all likelihood the necessary realtime driver(s) would
	  not be loaded before chan_sip.

	  In addition, it was found that it was impossible to
	  use the Digium Phone Module for Asterisk since it
	  must be loaded before chan_sip since it must hook
	  into chan_sip's configuration parsing.

	  The solution is to use a virtual table in the same
	  manner that other modules in Asterisk do, like
	  app_voicemail.

	  (closes issue ASTERISK-20545)
	  Reported by: kmoore

	* [r374850] Fix an issue where outgoing calls would fail to establish
	  audio due to ICE negotiation failures.

	  This change removes the requirement for ufrag and pwd in the transport
	  stanza and also makes us the controlling agent.

	  (closes issue ASTERISK-20554)
	  Reported by: mmichelson

	* [r374851] Remove code that should not have gotten in (r374850)

	  (issue ASTERISK-20554)

	* [r374877] Fix a bug where audio on Google Voice would not work due to
	  ignoring candidates.

	  Instead of ignoring parts of the message that are not known just
	  ignore the ones we know may be present and that would cause a problem.

	* [r375148] Ensure Asterisk fails TCP/TLS SIP calls when certificate
	  checking fails

	  When placing a call to a TCP/TLS SIP endpoint whose certificate is not
	  signed by a configured CA certificate, Asterisk would issue a warning
	  and continue to process the call as if there was not an issue with the
	  certificate.  Asterisk now properly fails the call if the certificate
	  fails verification or if the certificate does not exist when
	  certificate checking is enabled (the default behavior).

	  (closes issue ASTERISK-20559)
	  Review: https://reviewboard.asterisk.org/r/2163/

	* [r375051] Remove a log message that was left in accidentally from
	  call-id logging development.

2012-10-08  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 11.0.0-rc1 Released.

2012-10-08 20:38 +0000 [r374632-374676]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c, configs/rtp.conf.sample: Disable ICE
	  support by default Since there are a number of legacy devices out
	  there that fail to handle ICE candidates properly (which is a
	  nice way of saying something much uglier), disable it by default.
	  Support for ICE candidates can be enabled in rtp.conf using the
	  icesupport setting.

	* apps/confbridge/conf_state.c (added),
	  apps/confbridge/conf_state_single.c (added),
	  apps/confbridge/conf_state_inactive.c (added),
	  apps/confbridge/conf_state_single_marked.c (added), /,
	  apps/confbridge/include/confbridge.h,
	  apps/confbridge/include/conf_state.h (added),
	  apps/confbridge/conf_state_multi.c (added),
	  apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
	  (added), apps/confbridge/conf_state_empty.c (added): Resolve
	  issues in ConfBridge regarding marked, waitmarked, and unmarked
	  users Thank's to Neil Tallim (flan)'s tireless testing, issue
	  reporting, and patches it became clear that app_confbridge had
	  some complex logic in how it handled interactions between marked,
	  waitmarked, and unmarked users. In particular, there were some
	  areas in which the interactions between the users resulted in
	  inconsistent behavior, and app_confbridge was missing logic in
	  how to handle some corner cases. Some areas included: * Poor
	  handling of mixing unmarked and waitmarked users *
	  Inconsistencies in how MOH and muting was applied to various
	  users * Handling of various announcements for different user
	  profile options flan's patches seem to fix the various issues,
	  but highlighted how hard the code could be to maintain. In an
	  attempt to make things easier to maintain and to more fully
	  enumerate the various cases that exist, this patch breaks up the
	  logic into a state machine-like setup. Please note that the
	  various state transitioned are documented on the Asterisk wiki:
	  https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
	  Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
	  the following issues, mjordan uploaded the patch, although it was
	  written by twilson. Any contributor license discrepency is due to
	  that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
	  flan, mjordan, jrose patches:
	  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
	  twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
	  flan Tested by: flan patches:
	  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
	  twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
	  Jonathan White Tested by: Jonathan White patches:
	  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
	  twilson (license 6283) ........ Merged revisions 374652 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* res/pjproject/pjlib/include/pj/sock.h,
	  res/pjproject/pjlib/src/pj/sock_symbian.cpp,
	  res/pjproject/pjlib/src/pj/sock_bsd.c,
	  res/pjproject/pjlib/src/pj/sock_linux_kernel.c: pjproject: Fix
	  for Solaris builds. Do not undef s_addr. pjproject, in order to
	  solve build problems on Windows [1], undefines s_addr in one of
	  it's headers that is included in res_rtp_asterisk.c. On Solaris
	  s_addr is not a structure member, but defined to map to the real
	  strucuture member, therefore when building on Solaris it's
	  possible to get build errors like: [CC] res_rtp_asterisk.c ->
	  res_rtp_asterisk.o In file included from
	  /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
	  from res_rtp_asterisk.c:51:
	  /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In
	  function `inaddrcmp':
	  /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
	  error: structure has no member named `s_addr'
	  /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
	  error: structure has no member named `s_addr' res_rtp_asterisk.c:
	  In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706:
	  warning: dereferencing type-punned pointer will break
	  strict-aliasing rules res_rtp_asterisk.c:710: warning:
	  dereferencing type-punned pointer will break strict-aliasing
	  rules res_rtp_asterisk.c: In function
	  `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error:
	  structure has no member named `s_addr' make[2]: ***
	  [res_rtp_asterisk.o] Error 1 make[1]: *** [res] Error 2 make[1]:
	  Leaving directory `/export/home/admin/asterisk-11-svn' gmake: ***
	  [_cleantest_all] Error 2 Unfortunately, in order to make this
	  work, I also had to make sure pjproject only used the typdef
	  pj_in_addr and not the struct pj_in_addr so that when building
	  Asterisk I could "typedef struct in_addr pj_in_addr". It's
	  possible then that the library and users of those interfaces in
	  Asterisk have a different idea about the type of the argument,
	  while on the surface it looks like they are all 32 bit big endian
	  values. [1] http://trac.pjsip.org/repos/changeset/484 (issues
	  ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang,
	  mjordan patches:
	  0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch
	  uploaded by Shaun Ruffell (license 5417)

	* main/acl.c: Trivial patch to make 'best_score' defined for all
	  architectures. Fixes trivial build error on Solaris: acl.c: In
	  function `get_local_address': acl.c:196: error: `best_score'
	  undeclared (first use in this function) acl.c:196: error: (Each
	  undeclared identifier is reported only once acl.c:196: error: for
	  each function it appears in.) make[2]: *** [acl.o] Error 1 (issue
	  ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang
	  patches:
	  0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch
	  by Shaun Ruffell (license 5417)

2012-10-06 03:20 +0000 [r374611-374622]  Matthew Jordan <mjordan@digium.com>

	* res/res_xmpp.c: Handle capability stanzas that fail to provide
	  node or version information While XEP-0115 states that the node
	  and ver attributes are both required, some devices fail to
	  provide either field. Prior to this patch, failure to provide the
	  node or ver attribute would cause a crash in res_xmpp. While
	  failing to provide the node or ver attribute is technically
	  invalid, since this information is not utilized by Asterisk
	  except for reporting purposes, for interoperability reasons, we
	  continue to process the capability stanza anyways. (closes issue
	  ASTERISK-20495) Reported by: Martin W Tested by: Martin W
	  patches: 20495.patch uploaded by Martin W (license #6434)

	* res/res_xmpp.c, main/message.c: Update documentation for
	  MessageSend application/command's From field for XMPP When using
	  the channel technology agnostic application/AMI command
	  MessageSend, the "From" field is technically optional for the SIP
	  channel driver. However, if being sent by the XMPP resource
	  module (either res_xmpp or res_jabber), the "From" field is
	  necessary, and must correspond to a defined account. This patch
	  updates the documentation for this application/AMI command to
	  reflect this. (closes issue ASTERISK-20405) Reported by: Leif
	  Madsen

2012-10-05 20:32 +0000 [r374587]  dlee <dlee@localhost>:

	* main/manager.c, /: Multiple revisions 374570,374581 ........
	  r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
	  22 lines Improve AMI long line error handling In AMI's parser,
	  when it receives a long line (> 1024 characters), it discards
	  that line, but continues to process the message normally.
	  Typically, this is not a problem because a) who has lines that
	  long and b) usually a discarded line results in an invalid
	  message. But if that line is specifying an optional field, then
	  the message will be processed, you get a 'Response: Success', but
	  things don't work the way you expected them to. This patch
	  changes the behavior when a line-too-long parse error occurs. *
	  Changes the log message to avoid way-too-long (and truncated
	  anyways) log messages * Adds a 'parsing' status flag to Response:
	  Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line
	  is too long * Responds with an appropriate error if parsing !=
	  MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow
	  Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581
	  | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
	  I've committed too much. Reverting part of r374570. ........
	  Merged revisions 374570,374581 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374586 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-05 18:34 +0000 [r374538]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  Merged revisions 374515-374535 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
	  (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
	  Made setup_bc() static. Patches: patch1_unused-code.diff (license
	  #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
	  ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
	  (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
	  states Patches: patch2_unused-states.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
	  | 16 lines chan_misdn: Remove unnecessary null pointer checks and
	  checks for stack->nt * cleanup_bc() is always called with valid
	  bc (or it would've crashed before). * Value of stack->nt is known
	  in advance at some places. * Rename handle_event() to
	  handle_event_te(), handle_frm() to handle_frm_te(). Patches:
	  patch3_checks.diff (license #6372) patch uploaded by Guenther
	  Kelleter Modified JIRA ABE-2882 ................ r374518 |
	  rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Fix spelling in log messages Patches:
	  patch4_spelling.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
	  2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
	  chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
	  calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
	  emptied, cleaned and set not in use, although
	  misdn_lib_send_event() already did the same. This is bad. When
	  it's not in use we are not allowed to touch it. * Moved log
	  message in front of the resulting actions and fixed it to match
	  the case. Patches: patch5_bccleanup.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
	  | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
	  etc., really bad stuff. * Fix return codes of cb_events() for
	  EVENT_SETUP to use caller's cleanup mechanisms. * Move
	  cl_queue_chan() call after bearer check. Patches:
	  patch6_leaks.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
	  2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
	  chan_misdn: We must initialize cause on sending a DISCONNECT. We
	  must initialize cause on sending a DISCONNECT, so it is later
	  correctly indicated to ast_channel in case the answer
	  (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
	  patch7_hangupcause.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2882 ................ r374522 |
	  rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Remove unused code for upqueue Patches:
	  patch8_unused-upqueue.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2882 ................ r374523 |
	  rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Improve debugging (port number, messages fixed, dups
	  removed) Patches: patch9_debug.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
	  | 8 lines chan_misdn: Better debug: we can print_bc_info even if
	  there's no ast leg. Patches: patch10_debug-bc-2.diff (license
	  #6372) patch uploaded by Guenther Kelleter Modified. JIRA
	  ABE-2882 ................ r374534 | rmudgett | 2012-10-05
	  12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
	  setup_bc() is called too early for an incoming SETUP on TE. This
	  prevents the B channel from being setup for HDLC mode when
	  requested by the bearer capability and config option hdlc=yes. It
	  violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
	  connect to the channel until a CONNECT ACKNOWLEDGE message has
	  been received." * Call setup_bc() on receipt of
	  CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
	  PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
	  Guenther Kelleter Modified. JIRA ABE-2881 ................
	  r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
	  | 2 lines chan_misdn: Remove some more deadcode. ................
	  ........ Merged revisions 374536 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374537 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-04 20:18 +0000 [r374477-374485]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User
	  Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
	  a recompile, allow values to be adjusted in dsp.conf For binary
	  distributions allows easy adjustment for wobbly GSM calls, and
	  other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
	  DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
	  alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2144/ ........ Merged
	  revisions 374479 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374481 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's
	  always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if
	  hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2145/ ........ Merged
	  revisions 374475 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374476 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-04 15:42 +0000 [r374428]  dlee <dlee@localhost>:

	* main/db.c, /, res/res_agi.c: Fix DBDelTree error codes for AMI,
	  CLI and AGI The AMI DBDelTree command will return Success/Key
	  tree deleted successfully even if the given key does not exist.
	  The CLI command 'database deltree' had a similar problem, but was
	  saved because it actually responded with '0 database entries
	  removed'. AGI had a slightly different error, where it would
	  return success if the database was unavailable. This came from
	  confusion about the ast_db_deltree retval, which is -1 in the
	  event of a database error, or number of entries deleted
	  (including 0 for deleting nothing). * Changed some poorly named
	  res variables to num_deleted * Specified specific errors when
	  calling ast_db_deltree (database unavailable vs. entry not found
	  vs. success) * Fixed similar bug in AGI database deltree, where
	  'Database unavailable' results in successful result (closes issue
	  AST-967) Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2138/ ........ Merged
	  revisions 374426 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374427 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-04 04:43 +0000 [r374379-374386]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User
	  configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
	  Asterisk's DTMF Specifications are based on AT&T specs, which may
	  not be compatible in other countries. Various countries have
	  different specifications for the maximum power level differences
	  between the DTMF low group and high group of frequencies. Power
	  level difference between frequencies for different
	  Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
	  8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
	  = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
	  (2006-03) Now allow 4 variables to be individually configured in
	  dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
	  specifications Add's the following variables to dsp.conf
	  ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
	  ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
	  (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
	  tbsky,alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2141/ ........ Merged
	  revisions 374384 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374385 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /: _dsp_init: bring inline with trunk preparation for clean merge
	  of DTMF TWIST patch No functional changes, just style. alecdavis
	  (license 585) Reported by: Alec Davis Tested by: alecdavis
	  related https://reviewboard.asterisk.org/r/2141 ........ Merged
	  revisions 374365 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374370 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-04 02:15 +0000 [r374196-374337]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_jabber.c: Check for presence of buddy in info/dinfo
	  handlers The res_jabber resource module uses the ASTOBJ library
	  for managing its ref counted objects. After calling
	  ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
	  the object has to be checked to see if the buddy existed. Prior
	  to this patch, the buddy object was not checked for NULL; with
	  this patch in both aji_client_info_handler and aji_dinfo_handler
	  the pointer is checked before used and, if no buddy object was
	  found, the handlers return an error code. This patch does not
	  take the approach that our JID can be used to log in from another
	  resource. If that approach is desired, an improvement could be
	  made to this patch to create the buddy on the fly. This patch
	  seeks only to prevent Asterisk from crashing. FYI: In Asterisk
	  11+, you really should be using res_xmpp. It does not have this
	  problem, as it moved to the astobj2 library. Note that multiple
	  people have proposed patches for this issue; the patch being
	  committed here is based on those. (closes issue ASTERISK-19532)
	  Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
	  fix-jabber uploaded by Karsten Wemheuer (license #5930)
	  xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
	  (license #6157) (closes issue ASTERISK-19557) Reported by:
	  ulugutz ........ Merged revisions 374335 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374336 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/ccss.c: Destroy the generic_monitors container after the
	  core_instances in ccss For each item in core_instances disposed
	  of in the shutdown of ccss, any generic monitor instances
	  referenced by the objects will be removed from generic_monitors
	  during their destruction. Hilarity ensues if generic_monitors no
	  longer exists. Thanks to the Asterisk Test Suite's generic_ccss
	  test for complaining loudly when it ran into this. ........
	  Merged revisions 374300 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/asterisk.c, /: Ensure Shutdown AMI event is still fired
	  during Asterisk shutdown Richard pointed out that having the
	  manager dispose of itself gracefully during shutdown meant that
	  the Shutdown event will no longer get fired. This patch moves the
	  AMI event just prior to running the atexit callbacks. ........
	  Merged revisions 374230 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374231 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/message.c: Fix findings from check-in on r374177 Richard
	  pointed out two problems with the check-in from r374177: * The
	  ast_msg_shutdown function declaration doesn't match the prototype
	  in main/message.c. * The ref/alloc function usage in astobj2 (in
	  trunk) can use the ao2_t_* variants of the functions to allow the
	  REF_DEBUG flag to enable/disable their debug counterparts.
	  ........ Merged revisions 374210 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/db.c, main/asterisk.c, main/xmldoc.c, main/format.c,
	  main/udptl.c, main/pbx.c, /, main/ccss.c,
	  include/asterisk/astobj2.h, channels/chan_agent.c,
	  main/taskprocessor.c, res/res_musiconhold.c, res/res_xmpp.c,
	  main/cel.c, main/named_acl.c, main/indications.c,
	  main/format_pref.c, main/astobj2.c, main/channel.c, main/data.c,
	  main/manager.c, main/features.c, main/config_options.c,
	  main/event.c, main/message.c: Fix a variety of ref counting
	  issues This patch resolves a number of ref leaks that occur
	  primarily on Asterisk shutdown. It adds a variety of shutdown
	  routines to core portions of Asterisk such that they can reclaim
	  resources allocate duringd initialization. Review:
	  https://reviewboard.asterisk.org/r/2137 ........ Merged revisions
	  374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 374178 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-01 20:26 +0000 [r374133-374150]  Sean Bright <sean@malleable.com>

	* main/db.c, include/asterisk/astdb.h, /, tests/test_db.c,
	  apps/app_queue.c: app_queue: Support persisting and loading of
	  long member lists. Greenlight in #asterisk brought up that he was
	  receiving an error message "Could not create persistent member
	  string, out of space" when running app_queue in Asterisk 10.
	  dump_queue_members() made an assumption that 8K would be enough
	  to store the generated string, but with queues that have large
	  member lists this is not always the case. This patch removes the
	  limitation and uses ast_str instead of a fixed sized buffer. The
	  complicating factor comes from the fact that ast_db_get requires
	  a buffer and buffer size argument, which doesn't let us pull back
	  more than what we pass in, so I introduced a new
	  ast_db_get_allocated() which returns an ast_strdup()'d copy of
	  the value from astdb. As an aside, I did some testing on the
	  maximum size of data that we can store in the BDB library we
	  distribute and was able to store a 10MB string and retrieve it
	  with no problems, so I feel this is a safe patch. Review:
	  https://reviewboard.asterisk.org/r/2136/ ........ Merged
	  revisions 374108 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 374135 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/db.c, /: Use ast_copy_string instead of strncpy to guarantee
	  a NUL terminated string. ........ Merged revisions 374132 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-10-01 16:12 +0000 [r374106]  Mark Michelson <mmichelson@digium.com>

	* apps/confbridge/conf_config_parser.c: Don't destroy confbridge
	  config when error is encountered during a reload. Not panicking
	  means that the old config is kept. (closes issue ASTERISK-20458)
	  Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded
	  by Mark Michelson(license #5049) Tested by Leif Madsen

2012-09-29 03:54 +0000 [r374085]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Fix ref leak when adding ICE candidates to
	  an SDP There was a missing decrement to the reference count for
	  the current ICE candidate when local candidates are being added
	  to an outbound SDP. This patch corrects that.

2012-09-28 19:29 +0000 [r374059]  Jonathan Rose <jrose@digium.com>

	* /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
	  The opinion of development was that it is both improper to have
	  Matt's personal email address used in the source and that the
	  command wouldn't be useful without it. (closes issue AST-467)
	  Reported by: Malcolm Davenport ........ Merged revisions 374032
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 374045 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-28 13:02 +0000 [r374019]  beagles <beagles@localhost>:

	* res/res_xmpp.c, main/message.c: Reset hangup flags on channels
	  created through messages and cleanup globals in res_xmpp on
	  unload. This patch fixes an issue where hangup flags were not
	  being reset on a channel, affecting subsequent use of that
	  channel. The patch also adds some additional cleanup to res_xmpp
	  to fix an issue with reloading the module. (closes
	  ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles
	  Review: https://reviewboard.asterisk.org/r/2134/

2012-09-28 12:16 +0000 [r373991]  Joshua Colp <jcolp@digium.com>

	* /, res/res_agi.c: Update documentation to make it explicit that
	  "stream file" will not restart musiconhold. (issue
	  ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 373990 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-27 22:19 +0000 [r373954]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
	  leak using channel name parameter. The SendDTMF channel name
	  parameter has two issues. 1) Crashes if the channel name does not
	  exist. 2) Leaks a channel reference if the channel is the current
	  channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
	  documentation. * Renamed app to senddtmf_name and tweaked the
	  type. ........ Merged revisions 373945 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373946 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-27 17:05 +0000 [r373880-373914]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, include/asterisk/http_websocket.h,
	  res/res_http_websocket.c: Make res_http_websocket an optional
	  dependency on supported platforms for chan_sip. (closes issue
	  ASTERISK-20439) Reported by: sruffell Patches:
	  0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded
	  by sruffell (license 5417)

	* main/loader.c, /: loader: Ensure dependent modules are properly
	  initialized. If an Asterisk module specifies a dependency in
	  ast_module_info.nonoptreq, it is possible for Asterisk to skip
	  calling the modules's .load function. Asterisk was loading and
	  linking the module via load_dynamic_module() but was not adding
	  the module to the resource_heap. Therefore the module was not
	  initialized based on it's priority along with the other modules
	  in the heap. Now use load_resource() instead of
	  load_dynamic_module() for non-optional requirement. This will add
	  the module to the resource_heap so the module can be properly
	  initialized in the correct order. This is required if there are
	  any module global data structures initialized in the .load()
	  callback for the module on platforms which do not support weak
	  references. (issue ASTERISK-20439) Reported by: sruffell Patches:
	  0001-loader-Ensure-dependent-modules-are-properly-initial.patch
	  uploaded by sruffell (license 5417) ........ Merged revisions
	  373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373910 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_local.c, /: Fix an issue where Local channels
	  dialed by app_queue are considered in use immediately. The
	  chan_local channel driver returns a device state of in use even
	  if a created Local channel has not yet been dialed. This fix
	  changes the logic to return a state of not in use until the
	  channel itself has been dialed. (closes issue ASTERISK-20390)
	  Reported by: tim_ringenbach Review:
	  https://reviewboard.asterisk.org/r/2116/ ........ Merged
	  revisions 373878 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373879 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-26 21:16 +0000 [r373850]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Move handling of 408 response so there is
	  no misleading warning message. (closes issue ASTERISK-20060)
	  Reported by: Walter Doekes ........ Merged revisions 373848 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373849 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-26 18:18 +0000 [r373818]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_meetme.c: Fixed meetme tab completion and command
	  documentation. * Removed unnecessary case sensitivity in meetme
	  list, lock, unlock, mute, unmute, and kick commands. * Separated
	  meetme lock/unlock, mute/unmute, and kick commands into their own
	  registered commands to simplify tab completion and parameter
	  checking. meetme_lock_cmd(), meetme_mute_cmd(), and
	  meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
	  AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
	  Merged revisions 373815 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373816 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-26 08:29 +0000 [r373804]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_queue.c: app_queue: 'agent available' hint, cleanup
	  restart, and initial state Fix previously untested senarios; 1).
	  On queue initialisation set queue_avail devstate to INUSE.
	  Previously was unavailable, which indicated an agent was
	  available. 2). When removing members, if there are no other
	  members available, set queue_avail to INUSE. Previously, if a
	  member interface had become 'unavailable', they were never going
	  to be removed, particularly when persistant queues is enabled.
	  3). When adding a member, check that they are available, if they
	  are set queue_avail to NOT_INUSE. Previously on reloaded, members
	  may have been 'unavailable'. 4). When pausing or unpausing a
	  member, set appropriate queue availability. alecdavis (license
	  585) Reported by: Alec Davis Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/2129/

2012-09-25 23:09 +0000 [r373738-373775]  Mark Michelson <mmichelson@digium.com>

	* /, main/say.c: Fix saying of date in Dutch. The Dutch say the
	  date before the month. (closes issue ASTERISK-20353) Reported by:
	  Teun Ouwehand ........ Merged revisions 373773 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373774 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* configs/agents.conf.sample, /, channels/chan_agent.c: Remove dead
	  code and documentation for nonexistent feature. multiplelogin was
	  removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
	  was removed. (closes issue AST-948) reported by Steve Pitts
	  ........ Merged revisions 373768 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373769 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_voicemail.c, /: Fix error where improper IMAP greetings
	  would be deleted. (closes issue ASTERISK-20435) Reported by:
	  fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
	  uploaded by Michael L. Young (License #5026) (with suggested
	  modification made by me) ........ Merged revisions 373735 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373737 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 20:13 +0000 [r373707]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Fix T.38 support when used with
	  chan_local in between. Users of the T.38 API can indicate
	  AST_T38_REQUEST_PARMS on a channel to request that the channel
	  indicate a T.38 negotiation with the parameters present on the
	  channel. The return value of this indication is expected to be
	  AST_T38_REQUEST_PARMS upon success but with chan_local involved
	  this could never occur. This fix changes chan_local to always
	  return AST_T38_REQUEST_PARMS for this situation. If the
	  underlying channel technology on the other side does not support
	  T.38 this would have been determined ahead of time using
	  ast_channel_get_t38_state and an indication would not occur.
	  (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
	  ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
	  https://reviewboard.asterisk.org/r/2070/ ........ Merged
	  revisions 373705 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373706 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 19:35 +0000 [r373704]  Kinsey Moore <kmoore@digium.com>

	* /: Recorded merge of revisions 373703 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix an
	  issue where media would not flow for situations where the legacy
	  STUN code is in use. The STUN packets should *not* be blocked by
	  strict RTP. (closes issue ASTERISK-20415) Reported-by: Michele
	  Cicciotti Patch-by: Josh Colp (trunk r369817) ........ Merged
	  revisions 373702 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-09-25 18:52 +0000 [r373690]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/sip.h, /, channels/chan_sip.c,
	  configs/sip.conf.sample: Properly handle UAC/UAS roles for SIP
	  session timers The SIP session timer mechanism contains a
	  mandatory 'refresher' parameter (included in the Session-Expires
	  header) which is used in the session timer offer/answer signaling
	  within a SIP Invite dialog. It looks like asterisk is
	  interpreting the uac resp. uas role only as the initial role of
	  client and server (caller is uac, callee is uas). The standard
	  rfc 4028 however assigns the client role to the ((RE)-Invite)
	  requester, the server role to the ((RE)-Invite) responder. This
	  patch has Asterisk track the actual refresher as "us" or "them"
	  as opposed to relying on just the configured "uas" or "uac"
	  properties. (closes issue AST-922) Reported by: Thomas Airmont
	  Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
	  revisions 373652 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373665 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 18:24 +0000 [r373688]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_queue.c: "show" completion option for "queue"
	  shouldn't appear twice When tab-completing CLI commands starting
	  with "queue", "show" appeared twice in the list due to the way
	  that Asterisk's tab completion functions and the order in which
	  the commands were registered. The registration order has been
	  altered to resolve this issue. (closes issue AST-940)
	  Reported-by: Steve Pitts ........ Merged revisions 373666 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373675 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 17:21 +0000 [r373635-373650]  Richard Mudgett <rmudgett@digium.com>

	* /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix
	  valgrind found memcpy issues in codec_ilbc. Valgrind found
	  codec_ilbc using memcpy instead of memmove for overlapping memory
	  blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
	  Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
	  #5674) patch uploaded by Walter Doekes ........ Merged revisions
	  373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373645 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* codecs/Makefile, /: Make rebuild GSM, ilbc, or lpc10 codecs if
	  the respective sources change. ........ Merged revisions 373618
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 373633 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 16:31 +0000 [r373632]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Set Quality of Service for
	  video rtp instance (closes issue ASTERISK-20201) Reported by:
	  ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
	  6008) ........ Merged revisions 373617 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373631 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 14:12 +0000 [r373582]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_presencestate.c: "He who go through turnstile sideways
	  is going to Bangkok"

2012-09-25 13:29 +0000 [r373580]  Kinsey Moore <kmoore@digium.com>

	* configs/res_odbc.conf.sample, /: Fix documentation for default
	  username in res_odbc This was previously stated to be "root", but
	  is actually the name of the context if unspecified. (closes issue
	  ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
	  373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373579 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-25 12:07 +0000 [r373552]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_multicast.c, /: Fix an issue where a caller to
	  ast_write on a MulticastRTP channel would determine it failed
	  when in reality it did not. When sending RTP packets via
	  multicast the amount of data sent is stored in a variable and
	  returned from the write function. This is incorrect as any
	  non-zero value returned is considered a failure while a return
	  value of 0 is success. For callers (such as ast_streamfile) that
	  checked the return value they would have considered it a failure
	  when in reality nothing went wrong and it was actually a success.
	  The write function for the multicast RTP engine now returns -1 on
	  failure and 0 on success, as it should. (closes issue
	  ASTERISK-17254) Reported by: wybecom ........ Merged revisions
	  373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373551 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-24 22:17 +0000 [r373508]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c, /: Revert change to res_rtp_asterisk
	  committed in r373236 (1.8) The change committed in r373236
	  attempted to account for endpoints that increased their RTP
	  timestamp in DTMF end of event re-transmissions. This change
	  attempted to make Asterisk continue to work with endpoints that
	  failed to follow the RFC while maintaining the fix that allowed
	  for out of order DTMF to be handled. Unfortunately, there is no
	  free lunch, and this patch broke any system that sent DTMF
	  immediately after an RTP session was established or when an SSRC
	  is updated. As such, that patch is being reverted for the
	  previous behavior. Endpoints that erroneously increase the RTP
	  timestamp in DTMF end of event packets will not work properly
	  with Asterisk. (issue ASTERISK-20424) ........ Merged revisions
	  373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373505 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-24 22:12 +0000 [r373502]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Be consistent, send From: "Anonymous"
	  <sip:anonymous@anonymous.invalid> When setting
	  CALLERID(pres)=unavailable in the dialplan, the From header in
	  the SIP message contains "Anonymous"
	  <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
	  should use a lowercase a in the userpart of the URI. * Make the
	  From header use a lowercase A in the userpart of the anonymous
	  URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
	  Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
	  patch uploaded by Antti Yrjola ........ Merged revisions 373500
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 373501 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-24 21:12 +0000 [r373470]  Jonathan Rose <jrose@digium.com>

	* funcs/func_audiohookinherit.c, /, apps/app_mixmonitor.c:
	  func_audiohookinherit: Document some missed sources. This patch
	  also mentions that AUDIOHOOK_INHERIT can be used to transfer
	  MixMonitor audiohooks. There is also wiki that addresses
	  audiohooks and the use of AUDIOHOOK_INHERIT at the following
	  link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
	  (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........
	  Merged revisions 373467 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373468 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-24 21:08 +0000 [r373469]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix potential reentrancy problems in
	  chan_sip. Asterisk v1.8 and later was not as vulnerable to this
	  issue. * Made find_call() lock each private as it processes the
	  found dialogs. (Primary cause of ABE-2876) * Made the other
	  functions that traverse the dialogs container lock each private
	  as it examines them. * Fix race condition in sip_call() if the
	  thread that sent the INVITE is held up long enough for a response
	  to be processed. The p->initid for the INVITE retransmission
	  could be added after it was canceled by the response processing.
	  * Made __sip_destroy() clean up resource pointers after freeing.
	  This is primarily defensive in case someone has a stale private
	  pointer. * Removed redundant memset() in reqprep(). The call to
	  init_req() already does the memset() and is the first reference
	  to req in reqprep(). * Removed useless set of req.method in
	  transmit_invite(). The calls to initreqprep() and reqprep() have
	  to do this because they memset() the req. JIRA ABE-2876
	  .......... Merged -r373423 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 373424 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373466 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-24 19:21 +0000 [r373413-373454]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Fix a deadlock caused by a race condition
	  between removing a hint and reloading the dialplan and
	  subscribing to the removed hint. If conditions were right it was
	  possible for both the PBX core and chan_sip to deadlock by both
	  having a lock that the other wants. In the case of the PBX core
	  it had the contexts lock and wanted a SIP dialog lock, while in
	  the case of chan_sip it had the SIP dialog lock and wanted the
	  contexts lock. This fix unlocks the SIP dialog before getting the
	  extension state so that the other thread will not block on trying
	  to lock it. Once the extension state is retrieved the SIP dialog
	  is locked again and life carries on. As the SIP dialog is
	  reference counted it is not possible for it to go away after
	  unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
	  ........ Merged revisions 373438 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373440 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_sip.c, res/res_format_attr_h264.c: Fix an issue
	  with H.264 format attribute comparison and fix an issue with
	  improper SDP being produced. The H.264 format attribute module
	  compares two format attribute structures to determine if they are
	  compatible or not. In some instances it was possible for this
	  check to determine that both structures were incompatible when
	  they actually should be considered compatible. This check has now
	  been made even more permissive by assuming that if no attribute
	  information is available the two structures are compatible. If
	  both structures contain attribute information a base level
	  comparison of the H.264 IDC value is done to see if they are
	  compatible or not. The above issue uncovered a secondary issue in
	  chan_sip where the SDP being produced would be incorrect if the
	  formats were considered incompatible. This has now been fixed by
	  checking that all information required to produce the SDP is
	  available instead of assuming it is. (closes issue
	  ASTERISK-20464) Reported by: Leif Madsen

2012-09-24 12:33 +0000 [r373403]  beagles <beagles@localhost>:

	* res/res_rtp_asterisk.c, configs/rtp.conf.sample:
	  res_rtp_asterisk: Make TURN and STUN server configurations
	  consistent. This patch removes the turnport configuration
	  property and changes the turnaddr property to be a combined
	  host[:port] configuration string. The patch also modifies the
	  documentation in the example configuration to reflect the
	  property changes and adds some additional text indicating how the
	  STUN port is configured. (closes issue ASTERISK-20344) Reported
	  by: beagles Tested by: beagles Review:
	  https://reviewboard.asterisk.org/r/2111/

2012-09-21 19:29 +0000 [r373318-373368]  Jonathan Rose <jrose@digium.com>

	* /, channels/iax2-provision.c: iax2-provision: Fix improper return
	  on failed cache retrieval (closes issue ASTERISK-20337) reported
	  by: John Covert Patches: iax2-provision.c.patch uploaded by John
	  Covert (license 5512) ........ Merged revisions 373342 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373343 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: app_queue: Make queue reload members and
	  variants of that work Prior to this patch, 'queue reload members'
	  cli command did not work at all. This also affects the manager
	  function 'QueueReload' when supplied with the 'members: yes'
	  field. (closes issue AST-956) Reported by: John Bigelow ........
	  Merged revisions 373298 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373300 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-20 19:16 +0000 [r373246]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Fix incorrect MeetME conference bridge
	  reference count decrementing and sometimes premature destruction.
	  When using the 'e' or 'E' option to MeetMe the configured
	  conference bridges are loaded and examined to see if any are
	  empty. If no conference bridges are empty the caller is prompted
	  to enter the number of one. This operation left around a pointer
	  to the last created conference bridge still containing
	  participants. When the caller that was not able to find any empty
	  conference bridge hung up this pointer was disposed of and the
	  reference count of the conference bridge decremented. If there
	  was only a single participant in the conference bridge it was
	  ultimately destroyed prematurely. (closes issue AST-994) Reported
	  by: John Bigelow ........ Merged revisions 373242 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373245 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-20 18:59 +0000 [r373235-373240]  Matthew Jordan <mjordan@digium.com>

	* configs/extensions.conf.sample, CHANGES, apps/app_queue.c:
	  app_queue: Support an 'agent available' hint Sets INUSE when no
	  free agents, NOT_INUSE when an agent is free. modifes
	  handle_statechange() scan members loop to scan for a free agent
	  and updates the Queue:queuename_avial devstate. Previously exited
	  early if the member was found in the queue. Now Exits later when
	  both a member was found, and a free agent was found. alecdavis
	  (license 585) Reported by: Alec Davis Tested by: alecdavis
	  Review: https://reviewboard.asterisk.org/r/2121/ ~~~~ Support all
	  ways a member can be available for 'agent available' hints Alec's
	  patch in r373188 added the ability to subscribe to a hint for
	  when Queue members are available. This patch modifies the check
	  that determines when a Queue member is available by refactoring
	  the availability checks in num_available_members into a shared
	  function is_member_available. This should now handle the
	  ringinuse option, as well as device state values other than
	  AST_DEVICE_NOT_INUSE.

	* res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF,
	  accomodate increasing timestamps in End events While endpoints
	  should not be changing the source timestamp between DTMF event
	  packets, the fact is there exists those endpoints that do exactly
	  that. To work around this, we absorb timestamps within the
	  expected re-transmit period. Note that this period only affects
	  End of Event packets, so it should not prevent the detection of
	  new DTMF digits that happen to arrive right on top of each other.
	  (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson
	  Tested by: mjordan, Vladimir Mikhelson Review:
	  https://reviewboard.asterisk.org/r/2124 ........ Merged revisions
	  373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373237 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* configs/extensions.conf.sample, CHANGES, apps/app_queue.c: Add
	  queue monitoring hints This patch adds support for hints on a
	  queue. Hints can be added using the nomenclature 'Queue:name',
	  where name is the name of the queue being monitored. This nifty
	  feature was done by Alec Davis. Review:
	  https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
	  Tested by: alecdavis patches: review1619.diff2 by alecdavis
	  (license 585)

2012-09-20 18:18 +0000 [r373229]  Joshua Colp <jcolp@digium.com>

	* channels/sip/include/sip.h, res/res_rtp_asterisk.c,
	  main/rtp_engine.c, channels/chan_sip.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add
	  support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As
	  mentioned on the review for this, WebRTC has moved towards
	  choosing DTLS-SRTP as the mechanism for key exchange for SRTP.
	  This commit adds support for this but makes it available for
	  normal SIP clients as well. Testing has been done to ensure that
	  this introduces no regressions with existing behavior and also
	  that it functions as expected. Review:
	  https://reviewboard.asterisk.org/r/2113/

2012-09-20 17:15 +0000 [r373220]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/features.h, main/channel.c,
	  apps/app_directed_pickup.c, funcs/func_channel.c,
	  main/features.c, include/asterisk/channel.h: Named call pickup
	  groups. Fixes, missing functionality, and improvements. *
	  ASTERISK-20383 Missing named call pickup group features:
	  CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
	  CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() -
	  Needs to also select from named pickup groups. * ASTERISK-20384
	  Using the pickupexten, the pickup channel selection could fail
	  even though there was a call it could have picked up. In a call
	  pickup race when there are multiple calls to pickup and two
	  extensions try to pickup a call, it is conceivable that the loser
	  will not pick up any call even though it could have picked up the
	  next oldest matching call. Regression because of the named call
	  pickup group feature. * See ASTERISK-20386 for the implementation
	  improvements. These are the changes in channel.c and channel.h. *
	  Fixed some locking issues in CHANNEL(). (closes issue
	  ASTERISK-20383) Reported by: rmudgett (closes issue
	  ASTERISK-20384) Reported by: rmudgett (closes issue
	  ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2112/

2012-09-20 13:00 +0000 [r373211]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Correct handling of unknown SDP stream types
	  When the patch to handle arbitrary SDP stream arrangements went
	  into Asterisk, it also included an ability to transparently
	  decline unknown stream types. The scanf calls used were not
	  checked properly causing this part of the functionality to be
	  broken. (closes issue ASTERISK-20203)

2012-09-18 20:14 +0000 [r373133]  Sean Bright <sean@malleable.com>

	* main/manager.c, /: Don't crash when passing a NULL message to
	  __astman_get_header. Before this commit, __astman_get_header
	  would blindly dereference the passed in 'struct message *' to
	  traverse the header list. There are cases, however, such as
	  '*CLI> sip qualify peer foo' where the message pointer is NULL,
	  so we need to check for that. ........ Merged revisions 373131
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 373132 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-18 15:47 +0000 [r373119]  dlee <dlee@localhost>:

	* Makefile, include/asterisk/utils.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  -fnested-functions compile flag, if needed. 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. It also adds a comment to utils.h as to why the nested
	  function needs a prototype. (closes issue ASTERISK-20399)
	  Reported by: David M. Lee Review:
	  https://reviewboard.asterisk.org/r/2102/

2012-09-15 00:27 +0000 [r373107]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c, /: Made companding law for SS7 calls only
	  determined by SS7 signaling type. For SS7, the companding law for
	  a call was chosen inconsistently depending upon ss7type (ITU vs
	  ANSI) and the DAHDI companding default (T1 vs E1). For incoming
	  calls, the companding law was determined by ss7type. For outgoing
	  calls, the companding law was determined by the DAHDI default.
	  With the wrong combination you would get A-law/u-law conflicts.
	  An A-law/u-law conflict sounds like bad static on the line. SS7
	  ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
	  noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
	  with T1 line: ok * Fix the companding law used to be determined
	  by the SS7 signaling type only. ........ Merged revisions 373090
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 373101 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-14 19:50 +0000 [r373079]  Matthew Jordan <mjordan@digium.com>

	* main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.c:
	  Resolve memory leaks in TLS initialization and TLS client
	  connections This patch resolves two sources of memory leaks when
	  using TLS in Asterisk: 1) It removes improper initialization (and
	  multiple re-initializations) of portions of the SSL library.
	  Asterisk calls SSL_library_init and SSL_load_error_strings during
	  SSL initialization; collectively this obviates the need for
	  calling any of the following during initialization or client
	  connection handling: * ERR_load_crypto_strings (handled by
	  SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
	  SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
	  SSL_library_init) 2) Failure to completely clean up all memory
	  allocated by Asterisk and by the SSL library for TLS clients.
	  This included not freeing the SSL_CTX object in the SIP channel
	  driver, as well as not clearing the error stack when the TLS
	  client exited. Note that these memory leaks were found by Thomas
	  Arimont, and this patch was essentially written by him with some
	  minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
	  Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
	  Arimont (license 5525) Review:
	  https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
	  373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 373062 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-13 20:04 +0000 [r373029-373047]  dlee <dlee@localhost>:

	* main/Makefile: Fixed make clean when configured
	  --disable-asteriskssl

	* main/channel.c, /, include/asterisk/channel.h: Fix timeouts for
	  ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
	  its timeout to ast_waitfor_nandfds, expecting it to decrement the
	  timeout by however many milliseconds were waited. This is a
	  problem if it consistently waits less than 1ms. The timeout will
	  never be decremented, and we wait... FOREVER! This patch makes
	  ast_waitfordigit_full manage the timeout itself. It maintains the
	  previously undocumented behavior that negative timeouts wait
	  forever. (closes issue ASTERISK-20375) Reported by: Mark
	  Michelson Tested by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/2109/ ........ Merged
	  revisions 373024 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 373025 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-12 20:53 +0000 [r372995]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Skip any non-content information when
	  looking for and handling content. This fixes a bug with Jitsi and
	  conference calling. Jitsi implements XEP-0298 which places some
	  conference-info information in the session-initiate request which
	  chan_motif did not expect to occur.

2012-09-12 18:23 +0000 [r372984]  Jonathan Rose <jrose@digium.com>

	* res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless
	  messages (closes issue ASTERISK-20361) Reported by: Noah
	  Engelberth Review: https://reviewboard.asterisk.org/r/2108/

2012-09-12 15:19 +0000 [r372937]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Add channel name to a warning to make
	  debugging easier. The "autodestruct with owner in place" message
	  is typically indicative of a channel reference leak. Printing out
	  the name of the channel in the message may be helpful when trying
	  to debug the issue. ........ Merged revisions 372932 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372933 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-12 14:18 +0000 [r372930]  dlee <dlee@localhost>:

	* main/Makefile: Fixed r372696 when configured
	  --disable-asteriskssl; properly install libasteriskssl.dylib on
	  OS X. I didn't realize that libasteriskssl.c was still compiled,
	  even when you disable asteriskssl; it simple gets statically
	  linked into asterisk.

2012-09-11 22:32 +0000 [r372917]  Jonathan Rose <jrose@digium.com>

	* channels/chan_local.c, /: chan_local: Switch from using a random
	  4 digit hex identifier to unique id Changes chan_local channels
	  to use an 8 digit hex identifier generated atomically and
	  sequentially in order to eliminate the chance of having multiple
	  channels with the same name during high call volume situations.
	  (issue ASTERISK-20318) Reported by: Dan Cropp Review:
	  https://reviewboard.asterisk.org/r/2104/ ........ Merged
	  revisions 372902 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372916 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-11 21:15 +0000 [r372886-372888]  Mark Michelson <mmichelson@digium.com>

	* main/asterisk.c, /, include/asterisk/_private.h, main/message.c:
	  Fix inability to shutdown gracefully due to an unending channel
	  reference. message.c makes use of a special message queue channel
	  that exists in thread storage. This channel never goes away due
	  to the fact that the taskprocessor used by message.c does not get
	  shut down, meaning that it never ends the thread that stores the
	  channel. This patch fixes the problem by shutting down the
	  taskprocessor when Asterisk is shut down. In addition, the thread
	  storage has a destructor that will release the channel reference
	  when the taskprocessor is destroyed. (closes issue AST-937)
	  Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
	  Michelson (License #5049) Tested by Jason Parker ........ Merged
	  revisions 372885 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/features.c: Fix bad channel application data reference.
	  When channels get bridged due to an AMI bridge action or a DTMF
	  attended transfer, the two channels that get bridged have their
	  application data pointing to the other channel's name. This means
	  that if one channel is hung up but the other moves on, it means
	  that the channel that moves on will have its application data
	  pointing at freed memory. (issue ASTERISK-20335) Reported by:
	  aragon ........ Merged revisions 372840 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372841 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-11 17:16 +0000 [r372864]  dlee <dlee@localhost>:

	* Makefile, /: Corrects the astsbindir setting when installing the
	  sample asterisk.conf. (closes issue ASTERISK-20406) ........
	  Merged revisions 372863 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-10 20:59 +0000 [r372795-372806]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
	  when expected When IAX2 debug was changed from iax_showframe to
	  iax_outputframe, some instances were missed (or added afterward).
	  This was causing debug output to not be displayed when expected.
	  (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
	  John Covert ........ Merged revisions 372804 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372805 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_jingle.c, include/asterisk/doxygen/architecture.h,
	  main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c:
	  Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
	  chan_jingle, and res_jabber are now deprecated in favor of using
	  chan_motif and res_xmpp. They are a feature-equivalent
	  replacement and are written to be more easily maintainable.
	  (closes issue ASTERISK-20298) Review:
	  https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen

2012-09-10 19:19 +0000 [r372777]  dlee <dlee@localhost>:

	* res/res_rtp_asterisk.c: res_rtp_asterisk: Eliminate "type-punned
	  pointer" build warning. Removes "res_rtp_asterisk.c:706: warning:
	  dereferencing type-punned pointer will break strict-aliasing
	  rules" warning from the build on 32-bit platforms. The problem is
	  that 'size' was referenced aliased to both (pj_size_t *) and
	  (pj_ssize_t *). Now just make a copy of size that is the right
	  type so there isn't any pointer aliasing happening. It also adds
	  comments and asserts regarding what looks like an inappropriate
	  use of pj_sock_sendto, but is actually totally fine. (closes
	  issue ASTERISK-20368) Reported by: Shaun Ruffel Tested by:
	  Michael L. Young Patches:
	  0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
	  uploaded by Shaun Ruffel (license 5417) slightly modified by
	  David M. Lee.

2012-09-10 18:50 +0000 [r372768]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: app_meetme: Document that 'p' option will
	  continue in dialplan. (closes issue AST-991) Reported by John
	  Bigelow ........ Merged revisions 372765 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372767 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-10 18:37 +0000 [r372766]  Kinsey Moore <kmoore@digium.com>

	* /: Recorded merge of revisions 372764 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Warn on
	  CLI when UDPTL init fails This adds a CLI warning when a SDP
	  offer is rejected due to UDPTL initialization failure.
	  Previously, there was no indication of the reason for offer
	  rejection in this case. (closes issue ASTERISK-20357)
	  Reported-by: Francesco Usseglio Gaudi ........ Merged revisions
	  372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-09-10 17:33 +0000 [r372754]  Jonathan Rose <jrose@digium.com>

	* main/channel.c, /: Masquerade: Retain parkinglot settings made by
	  CHANNEL function. Prior to this patch, the user would have a
	  parkinglot set on a channel that was parked and when the channel
	  was retrieved, any attempt by that channel to park would simply
	  use the default. This patch makes parkinglot values set in this
	  way be retained through the masquerade. (closes issue AST-990)
	  Reported by: Nick Huskinson Patches:
	  masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
	  (license 6182) ........ Merged revisions 372736 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372737 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-09 01:25 +0000 [r372711]  Matthew Jordan <mjordan@digium.com>

	* channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
	  needed In r356604, SRTP handling was fixed to accomodate multiple
	  crypto keys in an SDP offer and the ability to re-create an SRTP
	  session when the crypto keys changed. In certain circumstances -
	  most notably when a phone is put on hold after having been
	  bridged for a significant amount of time - the act of re-creating
	  the SRTP session causes problems for certain models of phones.
	  The patch committed in r356604 always re-created the SRTP session
	  regardless of whether or not the cryptographic keys changed.
	  Since this is technically not necessary, this patch modifies the
	  behavior to only re-create the SRTP session if Asterisk detects
	  that the remote key has changed. This allows models of phones
	  that do not handle the SRTP session changing to continue to work,
	  while also providing the behavior needed for those phones that do
	  re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
	  by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
	  https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
	  372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 372710 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-08 05:51 +0000 [r372696]  dlee <dlee@localhost>:

	* /, main/Makefile: Recorded merge of revisions 372695 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
	  OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. Without
	  this flag, those files will compile with the system installed
	  OpenSSL headers (if they exist). This is a real bummer if a
	  different path was specified using --with-ssl= (closes issue
	  ASTERISK-20392) ........ Merged revisions 372682 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-09-07 23:07 +0000 [r372622-372657]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
	  (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
	  Merged revisions 372655 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372656 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, funcs/func_math.c: Remove annoying unconditional debug message
	  from INC/DEC functions. (closes issue AST-1001) Reported by:
	  Guenther Kelleter ........ Merged revisions 372628 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372629 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Fix exception path typo in app_queue.c
	  try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
	  Pepper Patches: fix-local-channel-locking.patch (license #6350)
	  patch uploaded by Jeremy Pepper ........ Merged revisions 372624
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 372625 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
	  ServerEmail and MailCommand reported values. The AMI action
	  VoicemailUsersList VoicemailUserEntry event headers ServerEmail
	  and MailCommand did not report the global values if they were not
	  overridden. The VoicemailUserEntry event header ServerEmail was
	  not populated with the global value if the voicemail user did not
	  override it. The VoicemailUserEntry event header MailCommand was
	  never populated with a value. * Removed unused struct ast_vm_user
	  member mailcmd[]. (closes issue AST-973) Reported by: John
	  Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372621 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-07 21:04 +0000 [r372609-372611]  dlee <dlee@localhost>:

	* res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
	  res/pjproject/third_party/bin, res/pjproject/third_party/gsm/lib,
	  res/pjproject/lib, res/pjproject/pjlib/lib,
	  res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
	  res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
	  res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
	  res/pjproject/pjmedia/lib, res/pjproject/third_party/lib,
	  codecs/ilbc: svn:ignore cleanup. * pjproject bin and lib
	  directories should pretty much ignore everything * Ignore *.o in
	  codecs/ilbc

	* res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a
	  build regression introduced in r369517 "Add support for
	  ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
	  http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
	  When compiling asterisk in parallel like: $ make -j 10 It's
	  possible to get errors like the following:
	  .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
	  separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
	  Error 1 make[2]: ***
	  [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
	  Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
	  `+' to parent make rule. This is because the build system is
	  trying to build each of the libraries in pjproject in parallel.
	  Now the build will build pjproject in a single job and link the
	  results into res_asterisk_rtp. Parallel builds, on one test
	  system, saves ~1.5 minutes from a default Asterisk build: Single
	  job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
	  2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
	  0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
	  ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
	  1m2.353s user 2m39.120s sys 0m18.850s (closes issue
	  ASTERISK-20362) Reported by: Shaun Ruffel Patches:
	  0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
	  uploaded by Shaun Ruffel (License #5417)

2012-09-07 02:26 +0000 [r372531-372583]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_minivm.c: Free ast_str objects when temp file fails
	  to be created in MiniVM The previous commit (r372554) was from a
	  patch that was written before r366880, which ensured that ast_str
	  objects allocated in the sendmail routine were free'd in off
	  nominal paths. This commit frees the string objects in the off
	  nominal path introduced in r372554. (issue ASTERISK-17133)
	  Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372582 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
	  issue in MiniVM when sending mail When MiniVM sends an e-mail and
	  it has the volgain option set, it will spawn sox in a separate
	  process to handle the manipulation of the sound file. In doing
	  so, it creates a temporary file. There are two problems here: 1)
	  The file descriptor returned from mkstemp is leaked 2) The
	  finalfilename character pointer points to a buffer that loses
	  scope once volgain processing is finished. Note that in r316265,
	  Russell fixed some gcc warnings by using the return value of the
	  mkstemp call. A warning was placed in minivm that the file
	  descriptor was going to be leaked. This patch reverts that
	  change, as it handles the leak and 'uses' the file descriptor
	  returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
	  Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
	  Cohen (license #5035) ........ Merged revisions 372554 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372555 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_queue.c: Update QueueMemberStatus event documentation to
	  include member status values The Status: header in a
	  QueueMemberStatus event (and other QueueMember* events) is the
	  numeric value of the device state corresponding to that Queue
	  Member. As those values are not exactly obvious, listing them in
	  the documentation is useful. Matt Riddell reported this
	  indirectly through the wiki page. (closes issue ASTERISK-20243)
	  Reported by: Matt Riddell

2012-09-06 22:12 +0000 [r372523]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
	  parking a call for the second time. Using the AMI redirect action
	  to take an ISDN call out of a parking lot causes the MOH state to
	  get confused. The redirect action does not take the call off of
	  hold. When the call is subsequently parked again, the call no
	  longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
	  repeated AST_CONTROL_HOLD frames if it is already in a state
	  where it is supposed to be sending MOH. The MOH may have been
	  stopped by other means. (Such as killing the generator.) This
	  simple fix is done rather than making the AMI redirect action
	  post an AST_CONTROL_UNHOLD unconditionally when it redirects a
	  channel and thus potentially breaking something with an
	  unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
	  jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
	  rmudgett ........ Merged revisions 372521 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 372522 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-06 21:42 +0000 [r372519]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_queue.c: Ensure listed queues are not offered for
	  completion When using tab-completion for the list of queues on
	  "queue reset stats" or "queue reload
	  {all|members|parameters|rules}", the tab-completion listing for
	  further queues erroneously listed queues that had already been
	  added to the list. The tab-completion listing now only displays
	  queues that are not already in the list. (closes issue AST-963)
	  Reported-by: John Bigelow ........ Merged revisions 372517 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372518 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-06 18:55 +0000 [r372500]  dsessions <dsessions@localhost>:

	* channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime
	  Peers Cannot Register Prior to 1.8, it was not necessary for an
	  explicit "type" to be set for an asterisk LDAP realtime peer. Now
	  the routine find_peer actually checks the type field during
	  registration and fails to find the peer if it is not set. The
	  attached patches make the realtime type equal whatever type is
	  being searched for if the type is 0 upon return from routine
	  build_peer. (closes issue ASTERISK-17222) Reported by: John
	  Covert Patch by: David Vossel Tested by: Darren Sessions Review:
	  https://reviewboard.asterisk.org/r/2095/

2012-09-06 15:56 +0000 [r372473]  Jonathan Rose <jrose@digium.com>

	* /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
	  directmediapermit/deny ACL works r366547 introduced a change to
	  the directmedia ACL for chan_sip which modified the behavior
	  significantly. Prior to the patch, this option would bridge peers
	  with directmedia if a peer's IP address matched its own
	  directmedia ACL. After that patch, the peer would check the
	  bridged peer's ACL instead. This change has been present since
	  1.8.14.0. That patched failed to document the change in
	  Upgrade.txt, so this patch adds mention of that change to
	  UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
	  ........ Merged revisions 372471 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372472 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-06 14:30 +0000 [r372446]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
	  show" Previously, tabbing at the end of "queue show" produced a
	  list of available queues about which information could be shown,
	  but did not include an alternative command, "rules", to access
	  information about queue rules. The "rules" item should now be
	  shown in the list of tab-completable items. (closes issue
	  AST-958) Reported-by: John Bigelow ........ Merged revisions
	  372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 372445 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-06 02:50 +0000 [r372392-372419]  Matthew Jordan <mjordan@digium.com>

	* /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
	  neighboring peer is unreachable Consider a scenario where DUNDi
	  peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
	  and where PBX2 and PBX3 are also neighbors. If the connection is
	  temporarily broken between PBX1 and PBX3, PBX1 should not include
	  PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
	  message, as it cannot send messages to PBX3. If it does, PBX2
	  will assume that PBX3 already received the message and fail to
	  forward the message on to PBX3 itself. This patch fixes this by
	  only including peers in a DPDISCOVER message that are reachable
	  by the sending node. This includes all peers with an empty
	  address (00:00:00:00:00:00) and that are have been reached by a
	  qualify message. This patch also prevents attempting to qualify a
	  dynamic peer with an empty address until that peer registers.
	  (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
	  dundi_routing.patch uploaded by Peter Racz (license 6290) The
	  patch uploaded by Peter was modified slightly for this commit.
	  ........ Merged revisions 372417 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372418 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_followme.c: Allow configured numbers for FollowMe to
	  be greater than 90 characters When parsing a 'number' defined in
	  followme.conf, FollowMe previously parsed the number in the
	  configuration file into a buffer with a length of 90 characters.
	  This can artificially limit some parallel dial scenarios. This
	  patch allows for numbers of any length to be defined in the
	  configuration file. Note that Clod Patry originally wrote a patch
	  to fix this problem and received a Ship It! on the JIRA issue.
	  The patch originally expanded the buffer to 256 characters.
	  Instead, the patch being committed duplicates the string in the
	  config file on the stack before parsing it for consumption by the
	  application. (closes issue ASTERISK-16879) Reported by: Clod
	  Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
	  by Clod Patry (license #5138) Slightly modified for this commit.
	  ........ Merged revisions 372390 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372391 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 19:43 +0000 [r372373]  Richard Mudgett <rmudgett@digium.com>

	* main/dsp.c, /: Fix compile error. ........ Merged revisions
	  372372 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 19:24 +0000 [r372365]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c, /: Correct documentation for ModuleLoad AMI
	  action The documentation incorrectly listed 'rtp' as a reloadable
	  subsystem and left out many other reloadable subsystems. It is
	  now also documented that subsystems may only be reloaded, not
	  loaded or unloaded. (closes issue AST-977) Reported-by: John
	  Bigelow ........ Merged revisions 372354 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372358 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 18:46 +0000 [r372342]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
	  goertzel samples to 160, should be MF_GSIZE Related
	  https://reviewboard.asterisk.org/r/2097/ ........ Merged
	  revisions 372339 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372341 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 18:36 +0000 [r372340]  Kinsey Moore <kmoore@digium.com>

	* main/pbx.c, /: Ensure counts generated in
	  manager_show_dialplan_helper are correct When
	  manager_show_dialplan_helper was written, the counter increment
	  for the total number of contexts was placed with the extensions
	  increment instead of in the enclosing loop. This function should
	  now generate correct context counts. (closes issue AST-970)
	  Reported-by: John Bigelow ........ Merged revisions 372337 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372338 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 17:35 +0000 [r372327-372328]  Richard Mudgett <rmudgett@digium.com>

	* res/res_rtp_asterisk.c: Fix coding guidelines issue with a recent
	  commit.

	* res/res_rtp_asterisk.c: Fix RTP/RTCP read error message
	  confusion. The RTP/RTCP read error message can report "fail:
	  success" when the read failure is because of an ICE failure. *
	  Changed __rtp_recvfrom() to generate a PJ ICE message when ICE
	  fails. * Changed RTP/RTCP read error message to indicate an
	  unspecified error when errno is zero. (closes issue
	  ASTERISK-20288) Reported by: Joern Krebs Patches:
	  jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded
	  by rmudgett (modified)

2012-09-05 16:04 +0000 [r372311]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, main/rtp_engine.c,
	  include/asterisk/rtp_engine.h: Re-fix sending unnegotiated
	  payloads during a P2P RTP bridge. The previous fix still would
	  look in the static_RTP_PT table, which is inappropriate since we
	  specifically want to find a codec that has been negotiated.
	  (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches:
	  codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)

2012-09-05 13:47 +0000 [r372289]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
	  using IMAP storage or realtime config This patch fixes two memory
	  leaks: 1. When find_user is called with NULL as its first
	  parameter, the voicemail user returned is allocated on the heap.
	  The inboxcount2 function uses find_user in such a fashion when
	  counting new messages, and fails to free the resulting voicemail
	  user object. 2. When populate_defaults is called on a voicemail
	  user, it wipes whatever flags have been set on the object by
	  copying over the global flags object. If the VM_ALLOCED flag was
	  ste on the voicemail user prior to doing so, that flag is
	  removed. This leaks the voicemail user when free_user is later
	  called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
	  patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
	  Patch slightly modified for this commit. Review:
	  https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
	  372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 372288 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 12:17 +0000 [r372266]  Michael L. Young <elgueromexicano@gmail.com>

	* res/res_rtp_asterisk.c: Fix breakage caused by last merge.
	  Missing a variable for 11 and trunk.

2012-09-05 07:41 +0000 [r372214-372241]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit
	  delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss
	  detector to original -r349249 method with some changes, remove
	  unnecessary; 1. reseting of hits=0, when no signal, only need to
	  set it once. 2. incrementing of hits, when the hit is the same as
	  the current hit. 3. setting of lasthit, when it's the same as
	  before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
	  spelling mistakes (closes issue ASTERISK-19610) alecdavis
	  (license 585) Reported by: Jean-Philippe Lord Tested by:
	  alecdavis Review: https://reviewboard.asterisk.org/r/2085/
	  ........ Merged revisions 372239 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372240 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/dsp.c, /: dsp.c: optimize goerztzel sample loops, in
	  dtmf_detect, mf_detect and tone_detect use a temporary short int
	  when repeatedly used to call goertzel_sample. alecdavis (license
	  585) Reported by: alecdavis Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/2093/ ........ Merged
	  revisions 372212 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372213 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 04:52 +0000 [r372199]  Michael L. Young <elgueromexicano@gmail.com>

	* res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For
	  Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
	  place to increment the sequence number for retransmitted DTMF end
	  packets. With the introduction of the RTP engine API in 1.8, the
	  sequence number was no longer being incremented. This patch fixes
	  this regression as well as cleans up a few lines that were not
	  doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
	  Bansal Tested by: Michael L. Young Patches:
	  01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
	  6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2083/ ........ Merged
	  revisions 372185 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372198 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-05 02:25 +0000 [r372175]  Matthew Jordan <mjordan@digium.com>

	* cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully
	  written to PostgreSQL database PQClear is not called when the
	  result object of a call to PQExec has a status of
	  PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
	  handled properly, so this memory leak only occurred when CEL
	  records were successfully written. This patch properly clears the
	  result in the nominal code path. (closes issue ASTERISK-19991)
	  Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
	  mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
	  #6394) ........ Merged revisions 372158 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372165 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-09-04 15:48 +0000 [r372135-372137]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix issue where SIP devices were not
	  notified when custom devices changed to "ringing". The problem
	  had to do with logic used when checking for what the oldest
	  ringing channel was. The problem was that if no channel was
	  found, then no notification would be sent. For custom device
	  states, there is no associated channel, so no notification would
	  get sent. This fixes the issue by still sending the notification
	  even if no associated channel can be found for a ringing device
	  state change. (closes issue ASTERISK-20297) Reported by Noah
	  Engelberth

	* main/config_options.c, apps/app_confbridge.c: Prevent crash from
	  using app_page with no confbridge.conf file provided. Also
	  prevents other potential crashes when using aco API with
	  uninitialized aco_info structs. (closes issue ASTERISK-20305)
	  reported by Noah Engelberth Tested by Noah Engelberth Review:
	  https://reviewboard.asterisk.org/r/2086

2012-08-31 21:14 +0000 [r372118]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c: Prevent local RTP bridges from sending
	  inappropriate formats to participants. A change for Asterisk 11
	  caused a check for failure to incorrectly check the return value.
	  This resulted in the possibility of transmitting media that a
	  party had not negotiated. If this media happened to be G.729,
	  then this could potentially result in one-way audio if no G.729
	  translators are installed. (closes issue ASTERISK-20296) reported
	  by NITESH BANSAL

2012-08-30 20:54 +0000 [r372050-372091]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Prevent crash on shutdown due to refcount
	  error on queues container. When app_queue is unloaded, the queues
	  container has its refcount decremented, potentially to 0. Then
	  the taskprocessor responsible for handling device state changes
	  is unreferenced. If the taskprocessor happens to be just about to
	  run its task, then it will create and destroy an iterator on the
	  queues container. This can cause the refcount on the queues
	  container to increase to 1 and then back to 0. Going back to 0 a
	  second time results in double frees. This failure was seen
	  periodically in the testsuite when Asterisk would shut down.
	  ........ Merged revisions 372089 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 372090 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Help prevent ringing queue members from
	  being rung when ringinuse set to no. Queue member status would
	  not always get updated properly when the member was called, thus
	  resulting in the member getting multiple calls. With this change,
	  we update the member's status at the time of calling, and we also
	  check to make sure the member is still available to take the call
	  before placing an outbound call. (closes issue ASTERISK-16115)
	  reported by nik600 Patches: app_queue.c-svn-r370418.patch
	  uploaded by Italo Rossi (license #6409) ........ Merged revisions
	  372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 372049 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-30 16:24 +0000 [r371963-372028]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
	  ignored during calls by some IAX2 peers When an IAX2 call is made
	  using the credentials of a peer defined in a dynamic Asterisk
	  Realtime Architecture (ARA) backend, the ACL rules for that peer
	  are not applied to the call attempt. This allows for a remote
	  attacker who is aware of a peer's credentials to bypass the ACL
	  rules set for that peer. This patch ensures that the ACLs are
	  applied for all peers, regardless of their storage mechanism.
	  (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
	  mjordan, Alan Frisch

	* /: Block r372020

	* main/manager.c, /, README-SERIOUSLY.bestpractices.txt:
	  AST-2012-012: Resolve AMI User Unauthorized Shell Access through
	  ExternalIVR The AMI Originate action can allow a remote user to
	  specify information that can be used to execute shell commands on
	  the system hosting Asterisk. This can result in an unwanted
	  escalation of permissions, as the Originate action, which
	  requires the "originate" class authorization, can be used to
	  perform actions that would typically require the "system" class
	  authorization. Previous attempts to prevent this permission
	  escalation (AST-2011-006, AST-2012-004) have sought to do so by
	  inspecting the names of applications and functions passed in with
	  the Originate action and, if those applications/functions matched
	  a predefined set of values, rejecting the command if the user
	  lacked the "system" class authorization. As noted by IBM X-Force
	  Research, the "ExternalIVR" application is not listed in the
	  predefined set of values. The solution for this particular
	  vulnerability is to include the "ExternalIVR" application in the
	  set of defined applications/functions that require "system" class
	  authorization. Unfortunately, the approach of inspecting fields
	  in the Originate action against known applications/functions has
	  a significant flaw. The predefined set of values can be bypassed
	  by creative use of the Originate action or by certain dialplan
	  configurations, which is beyond the ability of Asterisk to
	  analyze at run-time. Attempting to work around these scenarios
	  would result in severely restricting the applications or
	  functions and prevent their usage for legitimate means. As such,
	  any additional security vulnerabilities, where an
	  application/function that would normally require the "system"
	  class authorization can be executed by users with the "originate"
	  class authorization, will not be addressed. Instead, the
	  README-SERIOUSLY.bestpractices.txt file has been updated to
	  reflect that the AMI Originate action can result in commands
	  requiring the "system" class authorization to be executed. Proper
	  system configuration can limit the impact of such scenarios.
	  (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
	  X-Force Research ........ Merged revisions 371998 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371999 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to
	  doc folder In r294740, the CODING-GUIDELINES was removed from the
	  doc folder in favor of the content on the Asterisk wiki. Some
	  folks still look in the doc folder initially for coding guideline
	  suggestions; as such, this patch adds a CODING-GUIDELINES file
	  back into the doc folder. The content of the file merely points
	  to the correct page on the Asterisk wiki where the coding
	  guidelines currently live. (closes issue ASTERISK-20279) Reported
	  by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
	  Andrew Latham (license 5985) ........ Merged revisions 371961
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371962 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-29 22:38 +0000 [r371950]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_meetme.c: Fix compile errors.

2012-08-29 21:07 +0000 [r371921]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: app_meetme: Adding test events for
	  following activity in MeetMe. ........ Merged revisions 371919
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371920 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-29 19:56 +0000 [r371862-371893]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix theoretical compile error with HAVE_EPOLL.
	  Really shows how much epoll is used since it had not been
	  reported yet.

	* main/channel.c, /: Initialize file descriptors for dummy channels
	  to -1. Dummy channels usually aren't read from, but functions
	  like SHELL and CURL use autoservice on the channel. (closes issue
	  ASTERISK-20283) Reported by: Gareth Palmer Patches:
	  svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
	  (modified) ........ Merged revisions 371888 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371890 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_dial.c, /: Fix hangup cause passthrough regression. The
	  v1.8 -r369258 change to fix the F and F(x) action logic
	  introduced a regression in passing the hangup cause from the
	  called channel to the caller channel. (closes issue
	  ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
	  app_dial_hangupcause.patch (license #6421) patch uploaded by
	  Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged
	  revisions 371860 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371861 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-29 17:25 +0000 [r371845]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
	  instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
	  Doekes ........ Merged revisions 371824 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371825 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-27 21:50 +0000 [r371784-371790]  Mark Michelson <mmichelson@digium.com>

	* configs/agents.conf.sample, /: Fix misleading documentation in
	  agents.conf.sample regarding ackcall usage. The documentation
	  made it sound as if the DTMF acknowledgment was needed at the
	  time the agent logs in, rather than when the agent is called.
	  This is likely a relic from the days when there were multiple
	  ways of logging in agents. (closes issue AST-962) reported by
	  Steve Pitts ........ Merged revisions 371787 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371789 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/manager.c, /: Fix incorrect documentation of the
	  MailboxStatus manager command. The "Waiting" field was
	  misdocumented as reporting the number of messages waiting. In
	  reality, it simply indicated the presence or absence of waiting
	  messages. ........ Merged revisions 371782 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371783 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-27 18:14 +0000 [r371753]  dlee <dlee@localhost>:

	* res/pjproject/pjlib-util/bin, res/pjproject/pjnath/build/output,
	  res/pjproject/pjlib/bin, res/pjproject/pjlib-util/build/output,
	  res/pjproject/pjnath/bin, res/pjproject/pjlib/build/output:
	  svn:ignore pjproject bin & output for all platforms.

2012-08-27 17:51 +0000 [r371749-371750]  Mark Michelson <mmichelson@digium.com>

	* /, configs/queues.conf.sample: Fix incorrectly documented option
	  in queues.conf sharedlastcall defaults to "no" not "yes" (closes
	  issue AST-979) reported by Steve Pitts ........ Merged revisions
	  371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 371748 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /: Re-add merge and block properties.

2012-08-27 16:55 +0000 [r371720]  dlee <dlee@localhost>:

	* main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
	  variants. The original implementations simply wrap pthread
	  functions, which take absolute time as an argument. The spinlock
	  version for systems without those functions treated the argument
	  as a delta. This patch fixes the spinlock version to be
	  consistent with the pthread version. (closes issue
	  ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
	  uploaded by Egor Gorlin (license 6416) ........ Merged revisions
	  371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-08-27 14:07 +0000 [r371692]  Kinsey Moore <kmoore@digium.com>

	* /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash
	  When compiling with BETTER_BACKTRACES enabled, Asterisk will
	  sometimes crash when "core show locks" is run. This happens
	  regularly in the testsuite since several tests run "core show
	  locks" to help with debugging. This seems to be a fault with
	  libraries on certain operating systems (notably CentOS 6.2/6.3)
	  running on virtual machines and utilizing gcc 4.4.6. (closes
	  issue ASTERISK-20090) ........ Merged revisions 371690 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371691 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-26 23:07 +0000 [r371664]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of
	  MF_GSIZE ........ Merged revisions 371662 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371663 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-22 15:54 +0000 [r371619]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Add support for call-id logging to
	  chan_motif. Review: https://reviewboard.asterisk.org/r/2077/

2012-08-21 20:54 +0000 [r371592]  Mark Michelson <mmichelson@digium.com>

	* cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c,
	  channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
	  main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
	  res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
	  res/res_config_sqlite.c: Fix misuses of asprintf throughout the
	  code. This fixes three main issues * Change asprintf() uses to
	  ast_asprintf() so that it pairs properly with ast_free() and no
	  longer causes MALLOC_DEBUG to freak out. * When ast_asprintf()
	  fails, set the pointer NULL if it will be referenced later. * Fix
	  some memory leaks that were spotted while taking care of the
	  first two points. (Closes issue ASTERISK-20135) reported by
	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071
	  ........ Merged revisions 371590 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371591 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-20 20:09 +0000 [r371571]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c: Use thread-local storage to store
	  pj_thread_descs. pj_thread_register() takes a parameter of type
	  pj_thread_desc. It was assumed that pj_thread_register either
	  used this item temporarily or made a copy of it. Unfortunately,
	  all it does is keep a pointer to the structure in thread-local
	  storage. This means that if our pj_thread_desc goes out of scope,
	  then pjlib will be referencing bogus data quite often, most
	  commonly on operations involving a pj_mutex_t. In our case, our
	  pj_thread_desc was on the stack and went out of scope very
	  shortly after registering our thread with pjlib. With this
	  change, the pj_thread_desc is stored in thread-local storage so
	  the pointer that pjlib keeps in thread-local storage will
	  reference legitimate memory. (closes issue ASTERISK-20237)
	  reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded
	  by Mark Michelson (license #5049) Tested by Jeremy Pepper

2012-08-20 15:34 +0000 [r371546]  Kinsey Moore <kmoore@digium.com>

	* main/udptl.c, /: Ignore recovered zero-length secondary UDPTL
	  packets In some cases, recovering lost packets using the
	  secondary packet recovery mechanism with UDPTL/T.38 can result in
	  the recovery of zero-length packets. These must be ignored or the
	  frame generated from them can cause segfaults and allocation
	  failures. (closes issue ASTERISK-19762) (closes issue
	  ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
	  Gagnon (rgagnon) ........ Merged revisions 371544 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371545 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-18 02:35 +0000 [r371492-371530]  Matthew Jordan <mjordan@digium.com>

	* /: Recorded merge of revisions 371529 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Remove
	  old debug code from http configuration loading (closes issue
	  ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff
	  uploaded by Andrew Latham (license #5985)

	* main/http.c: Remove old debug code from http configuration
	  loading (closes issue ASTERISK-20254) Reported by: Andrew Latham
	  Patches: http.diff uploaded by Andrew Latham (license #5985)

	* res/res_xmpp.c: Fix typo in JabberSend that looked for '2'
	  instead of '@' in recipient argument The summary says about all
	  there is to say. (closes issue ASTERISK-20239) Reported by:
	  Gregory Porras

	* funcs/func_hangupcause.c: Make the name of the "HangupCauseClear"
	  application consistent The name of the "HangupCauseClear"
	  application is "HangupCauseClear", not "HangupcauseClear". The
	  incorrect case of 'cause' caused the XML documentation to not
	  register properly. As an aside, this commit message felt very
	  awkward, but I'm not sure how else to note that "X", which has to
	  be "X", was referred to as "x". (closes issue ASTERISK-20253)
	  Reported by: Andrew Latham Patches: hangupcause.diff uploaded by
	  Andrew Latham (license #5985)

	* build_tools/cflags.xml, utils/utils.xml, res/res_fax.c,
	  sounds/sounds.xml, res/res_curl.c: Update module support level on
	  a variety of modules and compiler options Some core support
	  modules and compiler options were no longer tagged with a module
	  support level. This patch adds 'core' back to those options. Note
	  that this patch modifies a few of the patches provided by Andrew
	  Latham slightly. res_curl and res_fax are both 'core' supported
	  modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham
	  Tested by: mjordan Patches: astcanary.diff (license #5985)
	  uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded
	  by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew
	  Latham soundsxml.diff (license #5985) uploaded by Andrew Latham

	* main/xmldoc.c, /: Fix memory leak in XML documentation When
	  formatting documentation fields, the XML documentation parser
	  calls xmldoc_get_formatted. This function allocates a string
	  buffer at the beginning of its routine. Unfortunately, on certain
	  code paths, it also calls xmldoc_string_cleanup, which assumes
	  that it will create the string buffer. The previously allocated
	  string buffer is then leaked by the xmldoc_string_cleanup
	  routine. Now: we don't do that. (closes issue AST-932) Reported
	  by: Alexander Homig ........ Merged revisions 371469 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371491 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-17 19:49 +0000 [r371482]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When a peer registers using WebSocket do not
	  resolve the Contact provided. (closes issue ASTERISK-20238)
	  Reported by: james.mortensen

2012-08-17 15:58 +0000 [r371438]  Kinsey Moore <kmoore@digium.com>

	* main/loader.c, /: Add instrumentation to subsystem reloads When
	  Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
	  generate TestEvent AMI events on subsystem reloads such as cdr,
	  dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions
	  371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 371437 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-17 12:24 +0000 [r371426]  Joshua Colp <jcolp@digium.com>

	* res/res_format_attr_h264.c: Add some additional H.264 attributes,
	  "max-smbps" and "max-fps", for passthrough. (closes issue
	  ASTERISK-20206) Reported by: ddkprog Patches:
	  res_format_attr_h264.c.diff uploaded by ddkprog (license 6008)

2012-08-17 12:23 +0000 [r371425]  Russell Bryant <russell@russellbryant.com>

	* res/res_rtp_asterisk.c: rtp: Ensure defaults are set without
	  rtp.conf. While building up a new install to test chan_motif, I
	  ran into a failure due to icesupport being disabled. This was due
	  to me not having an rtp.conf. It was intended in the code for it
	  to be enabled by default, but it was only applied if rtp.conf
	  existed. This patch updates res_rtp_asterisk to be consistent in
	  how it handles defaults. A few options didn't have their default
	  values set globally, including icesupport. They are now set and
	  icesupport is enabled by default, even if you do not have an
	  rtp.conf.

2012-08-16 23:02 +0000 [r371399]  Terry Wilson <twilson@digium.com>

	* main/config.c, /: Handle integer over/under-flow in
	  ast_parse_args The strtol family of functions will return
	  *_MIN/*_MAX on overflow. To detect when an overflow has happened,
	  errno must be set to 0 before calling the function, then checked
	  afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
	  revisions 371392 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371398 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-16 22:44 +0000 [r371395]  Kinsey Moore <kmoore@digium.com>

	* main/loader.c, /: Add module reload instrumentation for
	  TEST_FRAMEWORK This adds AMI events for module reloads when
	  Asterisk is built with TEST_FRAMEWORK enabled and corrects
	  generation of the module load AMI event. (issue PQ-1126) ........
	  Merged revisions 371393 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371394 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-16 19:43 +0000 [r371355-371382]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable
	  to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING
	  flag was used instead, which will frequently flip during
	  reinvites. (closes issue AST-897) Reported by: Thomas Arimont
	  ........ Merged revisions 371357 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371358 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP
	  answer is included in the SIP ACK Under certain conditions, a SIP
	  transaction involving directmedia wouldn't trigger a re-invite
	  because the SDP answer was included in an ACK instead of in a
	  message that we would have triggered the invite with. This patch
	  just queues a source change control frame if the dialog is using
	  directmedia when we find sdp for an ACK. (closes issue AST-913)
	  Reported by: Thomas Arimont ........ Merged revisions 371337 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371338 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-15 23:28 +0000 [r371324]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Fix bug where final queue member would not
	  be removed from memory. If a static queue had realtime members,
	  then there could be a potential for those realtime members not to
	  be properly deleted from memory. If the queue's members were
	  loaded from realtime and then all the members were deleted from
	  the backend, then the queue would still think these members
	  existed. The reason was that there was a short- circuit in code
	  such that if there were no members found in the backend, then the
	  queue would not be updated to reflect this. Note that this only
	  affected static queues with realtime members. Realtime queues
	  with realtime members were unaffected by this issue. (closes
	  issue ASTERISK-19793) reported by Marcus Haas ........ Merged
	  revisions 371306 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371313 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-15 20:40 +0000 [r371295]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix Segfault When Registering SIP Over
	  WebSockets The helper function, get_address_family_filter, in
	  chan_sip for dns resolution by address family was not recognizing
	  the websockets transport and resulting in a null pointer being
	  sent to functions in netsock2, in an attempt to determine if we
	  are bound to ANY address ([::]) or not. This patch fixes this
	  issue by handling the transport types SIP_TRANSPORT_WS and
	  SIP_TRANSPORT_WSS which results in a sock address being set
	  properly for use in determining the address family. (closes issue
	  ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven
	  Beisiegel, James Mortensen Patches:
	  asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young
	  (license 5026)

2012-08-15 20:17 +0000 [r371258-371272]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
	  relatedpeer on SIP dialog destruction The other instance of this
	  bug was fixed by jcolp/file in r121496. If we are destroying a
	  dialog only set the MWI dialog pointer on the related peer to
	  NULL if it is the dialog currently being destroyed. (closes issue
	  ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged
	  revisions 371270 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371271 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_sip.c, channels/chan_iax2.c, channels/sig_pri.c:
	  Add HANGUPCAUSE information to callee channels This adds
	  HANGUPCAUSE information to called channels so that hangup
	  handlers can, in conjunction with predial dialplan execution,
	  access the hangupcause information when the dialed channel hangs
	  up on a one-to-one basis instead of a many-to-one basis as with
	  HANGUPCAUSE usage on the caller channel. Review:
	  https://reviewboard.asterisk.org/r/2069/ (closes issue
	  ASTERISK-20198)

2012-08-13 20:28 +0000 [r371227]  Kinsey Moore <kmoore@digium.com>

	* main/loader.c, /, apps/app_meetme.c: Add test instrumentation
	  This adds test instrumentation for loading and unloading of
	  modules and for certain actions in MeetMe to be used in the
	  testsuite or any other consumer of AMI events. These will only be
	  generated when Asterisk is built with TEST_FRAMEWORK enabled.
	  (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371203 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-13 19:52 +0000 [r371200]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix problem where incorrect pointer was
	  checked for nullity. ........ Merged revisions 371198 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371199 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-10 22:03 +0000 [r371146]  Richard Mudgett <rmudgett@digium.com>

	* CHANGES: Update CHANGES for private party ID.

2012-08-10 21:32 +0000 [r371143]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Fix a couple of documentation problems in
	  app_queue.c * The RemoveQueueMember app made mention of options
	  that could be passed in, but no options are supported. I have
	  removed the listing of options from the documentation. * The
	  RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
	  that could be set. (closes issue AST-949) reported by Steve Pitts
	  (closes issue AST-954) reported by Steve Pitts ........ Merged
	  revisions 371141 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371142 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-10 20:08 +0000 [r371121]  Matthew Jordan <mjordan@digium.com>

	* / (added): _ _ _ _ _ _ / \ ___| |_ ___ _ __(_)___| | __ / | / | /
	  _ \ / __| __/ _ \ '__| / __| |/ / | | | | / ___ \__ \| | __/ | |
	  \__ \ < | | | | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| |_|
	  Because it's one greater than 10.

2012-08-10 19:54 +0000 [r371120]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, channels/chan_misdn.c, channels/chan_sip.c,
	  main/channel_internal_api.c, main/features.c,
	  include/asterisk/channel.h, channels/sig_pri.c,
	  funcs/func_callerid.c, main/cli.c: Add private representation of
	  caller, connected and redirecting party ids. This patch adds the
	  feature "Private representation of caller, connected and
	  redirecting party ids", as previously discussed with us (DATUS)
	  and Digium. 1. Feature motivation Until now it is quite difficult
	  to modify a party number or name which can only be seen by
	  exactly one particular instantiated technology channel
	  subscriber. One example where a modified party number or name on
	  one channel is spread over several channels are supplementary
	  services like call transfer or pickup. To implement these
	  features Asterisk internally copies caller and connected ids from
	  one channel to another. Another example are extension
	  subscriptions. The monitoring entities (watchers) are notified of
	  state changes and - if desired - of party numbers or names which
	  represent the involving call parties. One major feature where a
	  private representation of party names is essentially needed, i.e.
	  where a party name shall be exclusively signaled to only one
	  particular user, is a private user-specific name resolution for
	  party numbers. A lookup in a private destination-dependent
	  telephone book shall provide party names which cannot be seen by
	  any other user at any time. 2. Feature Description This feature
	  comes along with the implementation of additional private party
	  id elements for caller id, connected id and redirecting ids
	  inside Asterisk channels. The private party id elements can be
	  read or set by the user using Asterisk dialplan functions. When a
	  technology channel is initiating a call, receives an internal
	  connected-line update event, or receives an internal redirecting
	  update event, it merges the corresponding public id with the
	  private id to create an effective party id. The effective party
	  id is then used for protocol signaling. The channel technologies
	  which initially support the private id representation with this
	  patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
	  (chan_dahdi). Once a private name or number on a channel is set
	  and (implicitly) made valid, it is generally used for any further
	  protocol signaling until it is rewritten or invalidated. To
	  simplify the invalidation of private ids all internally generated
	  connected/redirecting update events and also all
	  connected/redirecting update events which are generated by
	  technology channels -- receiving regarding protocol information -
	  automatically trigger the invalidation of private ids. If not
	  using the private party id representation feature at all, i.e. if
	  using only the 'regular' caller-id, connected and redirecting
	  related functions, the current characteristic of Asterisk is not
	  affected by the new extended functionality. 3. User interface
	  Description To grant access to the private name and number
	  representation from the Asterisk dialplan, the CALLERID,
	  CONNECTEDLINE and REDIRECTING dialplan functions are extended by
	  the following data types. The formats of these data types are
	  equal to the corresponding regular 'non-private' already existing
	  data types: CALLERID: priv-all priv-name priv-name-valid
	  priv-name-charset priv-name-pres priv-num priv-num-valid
	  priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
	  priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
	  priv-name priv-name-valid priv-name-pres priv-name-charset
	  priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
	  priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
	  REDIRECTING: priv-orig-name priv-orig-name-valid
	  priv-orig-name-pres priv-orig-name-charset priv-orig-num
	  priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
	  priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
	  priv-orig-subaddr-odd priv-orig-tag priv-from-name
	  priv-from-name-valid priv-from-name-pres priv-from-name-charset
	  priv-from-num priv-from-num-valid priv-from-num-pres
	  priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
	  priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
	  priv-to-name priv-to-name-valid priv-to-name-pres
	  priv-to-name-charset priv-to-num priv-to-num-valid
	  priv-to-num-pres priv-to-num-plan priv-to-subaddr
	  priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
	  priv-to-tag Reported by: Thomas Arimont Review:
	  https://reviewboard.asterisk.org/r/2030/

2012-08-10 17:56 +0000 [r371113]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a comparison that was causing presence
	  tests to fail. A recent change made it so that device state
	  changes that were not actual "changes" would not get reported to
	  subscribers. The problem was that this inadvertently blocked
	  presence updates as well.

2012-08-10 16:49 +0000 [r371059-371091]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call
	  data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
	  there really. This indication arrive from asterisk core not h.323
	  stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
	  Patches: ASTERISK-19308.patch ........ Merged revisions 371089
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371090 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by
	  GRQ (gatekeeper request) interval (close issue ASTERISK-20094)
	  Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371061 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* addons/ooh323c/src/ooTimer.c: restore calling cb functions by
	  timer expire this was broken in rev 369602

2012-08-10 02:07 +0000 [r371052]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix pickup extension channel reference error.
	  You cannot unref a pointer and then expect to ref it again later.
	  * Fix potential NULL pointer deref if the call pickup search
	  fails.

2012-08-09 21:35 +0000 [r371036-371043]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that
	  show status of connection to H.323 Gatekeeper (GkClient state)

	* addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ
	  (registration reject) is received (close issue ASTERISK-20094)
	  Patches: ASTERISK-20094.patch ........ Merged revisions 371011
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 371022 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-09 19:22 +0000 [r371030]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  channels/sig_pri.c, channels/sig_ss7.c: Use better libss7
	  detection test and move libpri compile test. ........ Merged
	  revisions 371012 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 371013 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-09 18:28 +0000 [r371010]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile
	  with append mode instead of overwrite ........ Merged revisions
	  370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 370989 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-09 17:40 +0000 [r370987]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_meetme.c: Correct documentation for the MeetMe x flag
	  The documentation for the x flag for MeetMe incorrectly described
	  its function as closing down the conference when the last marked
	  user left. It actually causes the users with that flag to leave
	  the conference when the last marked user exits. The functionality
	  of this flag is not changing. ........ Merged revisions 370985
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 370986 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-09 14:52 +0000 [r370979]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h,
	  channels/sip/include/sip.h: Extend extension state callbacks to
	  have more information. Quote from review board: This patch
	  extends the extension state callbacks so that monitoring channels
	  (as chan_sip) get more information of the devices which are
	  responsible for an extension state change. The additional
	  information is needed by chan_sip to present names/numbers of the
	  caller and callee in an early-state SIP notification. Users of
	  extenstion state callback not interested in the additional
	  information are not affected by the changes. Motivation: to
	  present the involved party's name/number in an early-state
	  nofification (used by the notified device as a pickup offer) one
	  after another so that a user can see which call he will pick up
	  in an undirected pickup. Such a pickup offer to a user shall
	  indicate the same call (number/name-A calls number/name-B) as the
	  call which would be picked up when an undirected pickup is
	  executed. Users interested in additional state info must use the
	  new functions ast_extension_state_add_extended() resp.
	  ast_extension_state_add_destroy_extended() to register an
	  extended state callback. When the callback is registered this
	  way, an extra member device_state_info of struct
	  ast_state_cb_info is passed to the callback in addition to the
	  aggregated extension state. This container holds an object for
	  every device of the monitored extension hint consisting of the
	  device name, the device state and a channel reference to the
	  channel which (presumably) caused the device state. The
	  information is used by chan_sip for early-state notifications.
	  When the state of a device changes and the new state contains
	  AST_EVENT_RINGING, an early-state notification is sent to the
	  subscribed devices with the caller/callee names/numbers of the
	  oldest ringing channel of the monitored extension. The notified
	  user may then invoke a direct pickup, which will pickup exactly
	  this channel. Users of the old non-extended callbacks will only
	  be called when the aggregated state did change (same behavior as
	  before). Users of the extended callback will also be called when
	  the state is unchanged but does contain AST_EVENT_RINGING. That
	  could be the case if two channels are ringing at one device and
	  one of them hangs up, so the aggregated state does not change.
	  This way the monitoring channel can create a new early-state
	  notification with the now ringing party-ids. Review:
	  https://reviewboard.asterisk.org/r/2048 This contribution comes
	  from Guenther Kelleter

2012-08-09 14:36 +0000 [r370978]  Jonathan Rose <jrose@digium.com>

	* pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show
	  cache and DUNDi show hints (closes issue ASTERISK-18390) Reported
	  by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by
	  Peter Racz (license #6290)
	  ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by
	  Jonathan Rose (license #6182)

2012-08-08 22:45 +0000 [r370955]  Michael L. Young <elgueromexicano@gmail.com>

	* /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When
	  a channel hangs up while being spied upon and the option to exit
	  the ChanSpy application when the spied on channel hangs up is
	  set, ast_autochan_destroy is not being called and therefore a
	  reference to the spied upon channel is not removed. The symptom
	  being reported was that when using func_group in the dialplan and
	  calling "group show channels" at the cli, the spied upon channel
	  was still being shown while "core show channels" showed that the
	  channel was not up. This patch calls ast_autochan_destroy when a
	  spied upon channel hangs up and the option to exit the ChanSpy
	  application is set, removing the reference to the channel
	  allowing the count for the group that the spied channel was part
	  of to be decremented. (closes issue ASTERISK-17515) Reported by:
	  Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
	  Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
	  L. Young (license 5026) ........ Merged revisions 370952 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370954 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-08 22:41 +0000 [r370951-370953]  Mark Michelson <mmichelson@digium.com>

	* CHANGES: Move a SIP change up to the other SIP changes in the
	  CHANGES file.

	* main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c,
	  apps/app_originate.c, include/asterisk/channel.h,
	  include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow
	  support for early media on AMI originates and call files. This is
	  based on the work done by Olle Johansson on review board. The
	  idea is that the channel specified in an AMI originate or call
	  file is typically not connected to the outgoing extension until
	  the channel has been answered. With this change, an EarlyMedia
	  header can be specified for AMI originates and an early_media
	  option can be specified in call files. With this option set, once
	  early media is received on a channel, it will be connected with
	  the outgoing extension. (closes issue ASTERISK-18644) Reported by
	  Olle Johansson Review: https://reviewboard.asterisk.org/r/1472

2012-08-08 21:22 +0000 [r370943]  Terry Wilson <twilson@digium.com>

	* main/manager.c, CHANGES: Add AMI_CLIENT dialplan function
	  Implementation of a dialplan function for checking manager
	  accounts. Right now it only returns the number of logged in
	  sessions for a manager account, but other attributes can be added
	  later. Patch by: Olle Johansson Review:
	  https://reviewboard.asterisk.org/r/421/

2012-08-08 20:47 +0000 [r370927]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c: Create the payload type if it does not exist
	  when setting information based on the 'm' line. An rtpmap
	  attribute is not required for defined payload numbers.

2012-08-08 20:32 +0000 [r370926]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Convert sig_analog to use a global
	  callback table.

2012-08-08 20:30 +0000 [r370925]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c, /: Do not define a cause that doesn't actually
	  exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there
	  is no cause information. As such, it should not be defined and
	  translatable as a cause. ........ Merged revisions 370923 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370924 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-08 20:17 +0000 [r370887-370902]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/sig_analog.h: Fix the analog dial *0 flash-hook of
	  bridged peer feature. The flash-hook the bridged peer feature now
	  correctly determines if the bridged peer is another chan_dahdi
	  channel, that it is an analog channel, and that it has the
	  correct signaling for an FXO port. It now also flash-hooks the
	  correct channel. ........ Merged revisions 370900 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370901 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Convert sig_pri to use a global callback table.

	* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
	  Convert sig_ss7 to use a global callback table.

2012-08-07 21:58 +0000 [r370881]  Damien Wedhorn <voip@facts.com.au>

	* build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite
	  of skinny debugging. Debugging messages and associated controls
	  only compiled in if configured with --enable-dev-mode. Debug
	  messages provide more detail (including thread id) and are
	  grouped so the user/dev can limit the type of messages displayed.
	  Functionally no real change to chan_skinny. Review:
	  https://reviewboard.asterisk.org/r/2040/

2012-08-07 19:59 +0000 [r370860]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP
	  code are must remain separate since in non-Asterisk format cases
	  they differ.

2012-08-07 19:26 +0000 [r370851-370859]  Kinsey Moore <kmoore@digium.com>

	* /: Recorded merge of revisions 370858 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
	  missing AST_CAUSE_* -> text translations ........ Merged
	  revisions 370856 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8

	* main/channel.c: Add missing AST_CAUSE_* -> text translations A
	  few of these were missing from the list and are necessary for the
	  Who Hung Up? functionality.

2012-08-07 17:47 +0000 [r370832-370845]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c: Fix a bug uncovered by the test suite where
	  the RTP payload number was not getting set.

	* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
	  channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce
	  memory consumption significantly for users of the RTP engine API
	  by storing only the payloads present and in use instead of every
	  possible one. Review: https://reviewboard.asterisk.org/r/2052/

2012-08-07 12:46 +0000 [r370820-370831]  Matthew Jordan <mjordan@digium.com>

	* main/channel.c, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, channels/chan_misdn.c,
	  channels/chan_sip.c, main/channel_internal_api.c,
	  channels/misdn/chan_misdn_config.h, main/features.c,
	  configs/misdn.conf.sample, include/asterisk/channel.h,
	  configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
	  channels/misdn_config.c: Add named callgroups/pickupgroups This
	  patch adds named calledgroups/pickupgroups to Asterisk. Named
	  groups are implemented in parallel to the existing numbered
	  callgroup/pickupgroup implementation. However, unlike the
	  existing implementation, which is limited to a maximum of 64
	  defined groups, the number of defined groups allowed for named
	  callgroups/pickupgroups is effectively unlimited. Named groups
	  are configured with the keywords "namedcallgroup" and
	  "namedpickupgroup". This corresponds to the numbered group
	  definitions of "callgroup" and "pickupgroup". Note that as the
	  implementation of named groups coexists with the existing
	  numbered implementation, a defined named group of "4" does not
	  equate to numbered group 4. Support for the named groups has been
	  added to the SIP, DAHDI, and mISDN channel drivers. Review:
	  https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther
	  Kelleter(license #6372)

	* contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That
	  change is wrong, wrong, wrong.

	* contrib/realtime/mysql/voicemail_data.sql: Update the MySQL
	  voicemail_data contrib script to reflect Asterisk 11 changes All
	  voicemails now have a 'msg_id' included in their metadata. The
	  ODBC message storage backend now requires this column; as such,
	  the MySQL contrib script that creates the voicemail_data table
	  has been updated with the appropriate column information.

2012-08-06 15:18 +0000 [r370801]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Improve debug message for temporary
	  outbound proxies. Thanks to Paul Belanger for pointing this out.
	  ........ Merged revisions 370797 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370798 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-03 21:52 +0000 [r370773]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c, channels/sip/config_parser.c,
	  channels/sip/include/sip.h: Multiple revisions 370769-370771
	  ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri,
	  03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a
	  SIP dialstring. This is based on the review request posted by
	  Walter Doekes (referenced lower in the commit message) The main
	  fix here is to treat the IPorHost portion of the dial string as a
	  temporary outbound proxy. This ensures requests get sent to the
	  proper location. Due to the age of the request, some parts were
	  no longer relevant. For instance, the request moved outbound
	  proxy parsing code into a single method. This is done in a
	  previous commit, so it was not necessary to do again. Also, the
	  review request fixed some errors with regards to request routing
	  for CANCEL and ACK requests. This has also been fixed in more
	  recent commits. (closes issue ASTERISK-19677) reported by Walter
	  Doekes Review https://reviewboard.asterisk.org/r/1859 ........
	  r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug
	  2012) | 3 lines Remove unused variable. ........ r370771 |
	  mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5
	  lines Seriously? Another compilation error fixed. Somebody beat
	  me. ........ Merged revisions 370769-370771 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370772 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-08-02 15:51 +0000 [r370740]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Fix regression from r370636 When the
	  chan_sip cleanup went in, a typo was included that caused some
	  subscriptions of non-Polycom phones to be limited to the same
	  capabilities as Polycom phones. This resolves the failures in the
	  test suite resulting from this regression.

2012-08-01 19:37 +0000 [r370726]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Fix a possible crash due to passing NULL to
	  ast_variables_dup()

2012-08-01 18:52 +0000 [r370720]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not
	  include linkedlists.h. Using astobj2 does not require
	  linkedlists.h be included even though astob2 uses linked lists
	  internally.

2012-08-01 02:26 +0000 [r370699]  Kinsey Moore <kmoore@digium.com>

	* /, utils/extconf.c: Revert alloca changes for utils These changes
	  were a tad overzealous in the utils directory. Unfortunately,
	  these don't compile with a "make". ........ Merged revisions
	  370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 370698 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-31 22:28 +0000 [r370681-370691]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: Add headers from SIPAddHeader to
	  outbound REFER requests. This is a patch from kkm from review
	  board. This is useful for adding headers to REFER requests that
	  emanate from a Transfer() dialplan application call. This also
	  fixes some uses of the Referred-by header, removing an extra set
	  of angle brackets. I've modified the reporter's original patch to
	  not require any additions to the sip_refer header and to just
	  remove the referred_by_name from sip_refer since it is no longer
	  needed or used. (closes Issue ASTERISK-17639) reported by Kirill
	  Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff
	  uploaded by Kirill Katsnelson (license #5845) Review:
	  https://reviewboard.asterisk.org/r/1159

	* main/manager.c, configs/manager.conf.sample, CHANGES: Add
	  "setvar" option to manager.conf. With this option set, channel
	  variables can be set on every manager originate. The Variable
	  header can still be used to set additional channel variables for
	  individual calls if desired. This work was completed by Olle
	  Johansson on review board. I have applied the review feedback and
	  am committing it in order to get this into trunk before Asterisk
	  11 is branched. Review: https://reviewboard.asterisk.org/r/1412

2012-07-31 21:20 +0000 [r370677]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Schedule pokes of registered SIP peers
	  within a given timespan after SIP reload With a large number of
	  SIP peers registered, performing a SIP reload causes a flood of
	  SIP OPTIONS request packets. These are immediately sent out, and,
	  as responses come back, can cause peers to be flagged as 'lagged'
	  due to handling of the many response messages. This fix prevents
	  this "packet storm" and schedules the pokes for a random time.
	  That time varies between 1 ms and the peer's qualify time, or, if
	  the qualify time is unknown, the global qualifyfreq setting. The
	  committed patch has some very small modifications to the patch
	  schmidts wrote for the review. (closes issue ASTERISK-19154)
	  Reported by: Nicolo Mazzon patches: issue19154.patch license
	  #6034 uploaded by schmidts Review:
	  https://reviewboard.asterisk.org/r/1652 ........ Merged revisions
	  370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 370672 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-31 20:33 +0000 [r370664]  Russell Bryant <russell@russellbryant.com>

	* main/event.c: Move event cache updates into event processing
	  thread. Prior to this patch, updating the device state cache was
	  done by the thread that originated the event. It would update the
	  cache and then queue the event up for another thread to dispatch.
	  This thread moves the cache updating part to be in the same
	  thread as event dispatching. I was working with someone on a
	  heavily loaded Asterisk system and while reviewing backtraces of
	  the system while it was having problems, I noticed that there
	  were a lot of threads contending for the lock on the event cache.
	  By simply moving this into a single thread, this helped
	  performance *a lot* and alleviated some deadlock-like symptoms.
	  Review: https://reviewboard.asterisk.org/r/2066/

2012-07-31 20:21 +0000 [r370655]  Kinsey Moore <kmoore@digium.com>

	* /, main/say.c, main/threadstorage.c, funcs/func_strings.c,
	  channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c,
	  pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c,
	  main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c,
	  channels/chan_alsa.c, channels/sig_ss7.c, main/db.c,
	  include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c,
	  tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c,
	  apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c,
	  addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
	  main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
	  channels/chan_gtalk.c, res/res_jabber.c,
	  res/res_http_websocket.c, res/ael/pval.c, main/channel.c,
	  main/manager.c, apps/app_osplookup.c, res/res_agi.c,
	  apps/app_minivm.c, main/logger.c, main/app.c,
	  addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c,
	  channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c,
	  channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c,
	  channels/chan_jingle.c, main/tcptls.c,
	  apps/app_directed_pickup.c, main/file.c, main/callerid.c,
	  apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c,
	  include/asterisk/strings.h, utils/extconf.c, main/dsp.c,
	  addons/res_config_mysql.c: Clean up and ensure proper usage of
	  alloca() This replaces all calls to alloca() with ast_alloca()
	  which calls gcc's __builtin_alloca() to avoid BSD semantics and
	  removes all NULL checks on memory allocated via ast_alloca() and
	  ast_strdupa(). (closes issue ASTERISK-20125) Review:
	  https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes
	  (wdoekes) ........ Merged revisions 370642 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370643 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-31 19:57 +0000 [r370644]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and
	  modify "dialplan add include" From corruptor's review board
	  posting: "I've noticed that we can remove particular extension
	  from context with dialplan remove extension command but in order
	  to remove all extensions in the context we should delete them on
	  by one. I've created dialplan remove context command which uses
	  ast_context_destroy to destroy the whole context with all
	  extensions. I've created to functions for in pbx_config.c:
	  handle_cli_dialplan_remove_context which actually removes context
	  and complete_dialplan_remove_context which completes input. They
	  are based on other similar functions and pretty trivial but I can
	  be mistaken somewhere. "I've also modified dialplan add include
	  <context2> into <context1>. I've made it similar dialplan add
	  extension ... command. It creates <context1> if it doesn't exist
	  and I've also modified complete_dialplan_add_include and removed
	  check for existance of <context2> because we can include
	  non-existent context into another one. (I usually include empty
	  (non-existent) contexts in advance). Should we raise warning in
	  this case as it's raised while reading extensions.conf? "I use
	  those functions with AMI. I think manager commands should be
	  created in addition to those CLI commands." I've addressed the
	  latest comments on review board and have made some other coding
	  guidelines-related cleanup. I also have modified the CHANGES file
	  to mention these new commands. (closes issue ASTERISK-19292)
	  reported by Andrey Solovyev Patches: dialplan_add_include.patch
	  uploaded by Andrey Solovyev (license #5214)
	  dialplan_remove_context.patch uploaded by Andrey Solovyev
	  (license #5214) Review: https://reviewboard.asterisk.org/r/2042

2012-07-31 19:10 +0000 [r370636]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c, channels/sip/security_events.c,
	  channels/sip/include/sip.h: Clean up chan_sip This clean up was
	  broken out from https://reviewboard.asterisk.org/r/1976/ and
	  addresses the following: - struct sip_refer converted to use the
	  stringfields API. - sip_{refer|notify}_allocate ->
	  sip_{notify|refer}_alloc to match other *alloc functions. -
	  Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
	  get_pidf_msg_text_body3 but get_content, to match add_content. -
	  get_body doesn't get the request body, renamed to
	  get_content_line. - get_body_by_line doesn't get the body line,
	  and is just a simple if test. Moved code inline and removed
	  function. - Remove camelCase in struct sip_peer peer state
	  variables, onHold -> onhold, inUse -> inuse, inRinging ->
	  ringing. - Remove camelCase in struct sip_request rlPart1 ->
	  rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata
	  to pvt->nonce because that is what it is, no need to update
	  struct sip_pvt because _it already has a nonce field_. - Removed
	  struct sip_pvt randdata stringfield. - Remove useless (and
	  inconsistent) 'header' suffix on variables in
	  handle_request_subscribe. - Use ast_strdupa on Event header in
	  handle_request_subscribe to avoid overly complicated strncmp
	  calls to find the event package. - Move get_destination check in
	  handle_request_subscribe to avoid duplicate checking for packages
	  that don't need it. - Move extension state callback management in
	  handle_request_subscribe to avoid duplicate checking for packages
	  that don't need it. - Remove duplicate append_date prototype. -
	  Rename append_date -> add_date to match other add_xxx functions.
	  - Added add_expires helper function, removed code that manually
	  added expires header. - Remove _header suffix on
	  add_diversion_header (no other header adding functions have
	  this). - Don't pass req->debug to request handle_request_XXXXX
	  handlers if req is also being passed. - Don't pass req->ignore to
	  check_auth as req is already being passed. - Don't create a
	  subscription in handle_request_subscribe if p->expiry == 0. -
	  Don't walk of the back of referred_by_name when splitting string
	  in get_refer_info - Remove duplicate check for no dialog in
	  handle_incoming when sipmethod == SIP_REFER, handle_request_refer
	  checks for that. Review: https://reviewboard.asterisk.org/r/1993/
	  Patch-by: gareth

2012-07-30 23:26 +0000 [r370565-370598]  Richard Mudgett <rmudgett@digium.com>

	* main/test.c: Tweak unit test warning message.

	* funcs/func_presencestate.c, main/test.c: Fix some presence-state
	  unit test typos.

	* apps/app_confbridge.c: DECLINE to load confbridge if the config
	  fails to load.

	* channels/chan_misdn.c, /: Release B channel allocation on error
	  path in chan_misdn. ........ Merged revisions 370563 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370564 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-30 14:52 +0000 [r370548]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: app_meetme: Change app_meetme support level
	  to extended from deprecated (closes issue ASTERISK-20134)
	  Reported by: Leif Madsen ........ Merged revisions 370547 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-30 13:45 +0000 [r370534-370541]  Russell Bryant <russell@russellbryant.com>

	* tests/test_event.c: Fix ast_event_new unit test. One of my recent
	  commits broke this test. The error was:
	  [test_event.c:event_new_test:214]: Events expected to be
	  identical have different size: 69 != 59 The difference in size
	  occurred because the first event had the EID IE added to the
	  event twice. ast_event_new() now always adds it automatically.
	  Previously it only added it if there were no IEs specified, which
	  was kind of weird.

	* include/asterisk/event_defs.h, res/res_corosync.c, main/event.c:
	  Add a "corosync ping" CLI command. This patch adds a new CLI
	  command to the res_corosync module. It is primarily used as a
	  debugging tool. It lets you fire off an event which will cause
	  res_corosync on other nodes in the cluster to place messages into
	  the logger if everything is working ok. It verifies that the
	  corosync communication is working as expected. I didn't put
	  anything in the CHANGES file for this, because this module is new
	  in Asterisk 11. There is already a generic "res_corosync new
	  module" entry in there so I figure that covers it just fine.

	* addons/app_mysql.c, CHANGES: Allow specifying a port number for
	  the MySQL server. This patch allows you to specify a port number
	  for the MySQL server. It's useful if a MySQL server is running on
	  a non-standard port. Even though this module is deprecated in
	  favor of func_odbc, someone asked for this feature and it seems
	  pretty harmless to add. It has been tested using a number of
	  combinations of with/without a port number specified in the
	  dialplan and changing the port number for mysqld.

2012-07-26 15:31 +0000 [r370510-370518]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command
	  to AMI This patch was submitted by mnicholson a while back. It
	  adds a new AMI action which allows users to request SIP peer
	  status on demand similar to existing PeerStatus events and to the
	  output you would see from CLI with sip show peer Review:
	  https://reviewboard.asterisk.org/r/1098/

	* /, res/res_agi.c: res_agi: Add message indicating need for \n
	  character in verbose message The while loop responsible for
	  reading AGI messages from a fastAGI service can end up looping
	  indefinitely when an AGI script fails to indicate the end of a
	  message with a \n character. This patch adds an indication that
	  we are expecting a \n character to end the message to make it
	  more clear to users that this is necessary if they are receiving
	  this warning over and over. (issue ASTERISK-20061) Reported by:
	  Eike Kuiper ........ Merged revisions 370494 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370495 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-25 14:27 +0000 [r370481-370488]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile: Repair editline builds using in-tree editline
	  sources. The previous change to the build system for using a
	  system-provided editline library was missing a crucial include
	  directory for building against the copy of the library in the
	  Asterisk source tree.

	* main/Makefile: Use an absolute path when referring to the
	  embedded editline directory. This patch changes the build system
	  to refer to the embedded editline directory using an absolute
	  path, which will resolve a problem seen on the CentOS automated
	  build agents.

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile,
	  main/editline/configure, configure.ac, main/editline/readline
	  (removed), main/editline/readline.c, main/editline/configure.in,
	  CHANGES, makeopts.in, main/editline/readline.h (added),
	  main/asterisk.c, contrib/scripts/install_prereq, main/cli.c:
	  Enable usage of system-provided NetBSD editline library if
	  available. This patch changes the Asterisk configure script and
	  build system to detect the presence of the NetBSD editline
	  library (libedit) on the system. If it is found, it will be used
	  in preference to the version included in the Asterisk source
	  tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie
	  Review: https://reviewboard.asterisk.org/r/1528/ Patches:
	  0001-Allow-linking-building-against-an-external-editline.patch
	  uploaded by jcollie (license #5373) (heavily modified by
	  kpfleming)

2012-07-25 03:51 +0000 [r370474]  Terry Wilson <twilson@digium.com>

	* main/pbx.c, /: Revert a change that broke compilation 1) There is
	  no such function as ast_ref() 2) The patch was originally
	  credited as the one uploaded by Guenther Kelleter (license 6372)
	  via issue AST-921, but the patch committed was not the patch
	  referenced on the issue. 3) Guenther Kelleter's patch was
	  actually correct. It moved the ast_free above the
	  presencechange_cleanup label. I am not committing his change as
	  it is not technically necesary--calling ast_free(NULL) is
	  perfectly safe and I worry that moving the ast_free outside of
	  the label could lead to future bugs if someone ever adds another
	  failure conditional and expects 'goto presencechange_cleanup;' to
	  clean up after everything.

2012-07-24 21:30 +0000 [r370466]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
	  handle_presencechange (closes issue AST-921) Reported by:
	  Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
	  Kelleter (license 6372)

2012-07-24 19:12 +0000 [r370453]  Kevin P. Fleming <kpfleming@digium.com>

	* tests/test_acl.c: Silence a warning message from older versions
	  of GCC. Revision 370426 introduced the use of a nested function
	  in tests/test_acl.c, but the lack of the 'auto' scope specifier
	  on the function and a forward declaration resulted in compilation
	  errors on the automated test systems.

2012-07-24 17:16 +0000 [r370433]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, channels/chan_oss.c: chan_oss: fix "sample rate" error message
	  Merged revisions 370428 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged
	  revisions 370432 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-24 16:54 +0000 [r370426-370431]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c, /: Rewrite a comment that didn't adequately explain
	  the code it was documenting. ........ Merged revisions 370429
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 370430 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* CHANGES: Update CHANGES for list/negation ACL feature.

	* tests/test_acl.c, main/acl.c: Allow permit/deny ACL lines to
	  contain multiple items and negated entries. Rules in ACLs
	  (specified using 'permit' and 'deny') can now contain multiple
	  items (separated by commas), and items in the rule can be negated
	  by prefixing them with '!'. This simplifies Asterisk Realtime
	  configurations, since it is no longer necessray to control the
	  order that the 'permit' and 'deny' columns are returned from
	  queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial
	  patch contributed by Tilghman Lesher Unit tests written by Kevin
	  P. Fleming

2012-07-24 16:15 +0000 [r370419-370420]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Build is underway so logging can go away.

	* res/res_rtp_asterisk.c: Temporarily enable pj logging to console
	  for debugging pjnath issue exposed by build slave.

2012-07-24 08:53 +0000 [r370413]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Remove code, that operate with cdr in
	  attempt_transfer(). That was removed somewhere between 1.2 and
	  1.4 and acidentaly put back in chan_unistim. (closes issue
	  ASTERISK-19628) Reported by: Igor Olhovskiy

2012-07-23 21:27 +0000 [r370407]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/Makefile, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  codecs/codec_ilbc.c, CHANGES, makeopts.in: Enable usage of
	  system-provided iLBC library. The WebRTC version of the iLBC
	  codec is now package as a library and is available on some
	  platforms. This patch allows codec_ilbc to be built against that
	  library if it is present. Review:
	  https://reviewboard.asterisk.org/r/1964/

2012-07-23 21:15 +0000 [r370387]  Matthew Jordan <mjordan@digium.com>

	* tests/test_abstract_jb.c (added), main/abstract_jb.c,
	  funcs/func_jitterbuffer.c, include/asterisk/abstract_jb.h: Unit
	  tests for the Jitter Buffer API; remove unnecessary resync This
	  patch includes the following: * Unit tests for the abstract
	  Jitter Buffer API. This includes both fixed and adaptive flavors,
	  testing nominal creation, frame input, frame retrieval,
	  resyncing; off nominal frame input overflow, out of order, and
	  others. * Tweaks to the abstract_jb API to remove the unnecessary
	  resync_threshold parameter from the create function
	  (resync_threshold is already in the struct passed into the create
	  function) * Ensure the fixed jitter buffer is empty before
	  destroying it, to avoid an ASSERT * Don't "resync" the adaptive
	  jitter buffer. The mechanism that was being used actually causes
	  the jitter buffer to think its being overflowed by going around
	  the jitterbuf API and attempting to 'resynch' it improperly. If a
	  resync is needed, the jitter buffer will do it properly by
	  itself. Note that this is only an optimization needed for trunk,
	  as the worst that happens is the loss of three voice packets
	  before the adaptive jitter buffer will resync anyway. Review:
	  https://reviewboard.asterisk.org/r/2035

2012-07-23 21:10 +0000 [r370386]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  separate configuration options for subscription and registration
	  minexpiry and maxexpiry. This offers more fine-grained control
	  over how long subscriptions last without negatively affecting the
	  expiration range for registrations. Uploaded by: Guenther
	  Kelleter(license #6372) Review:
	  https://reviewboard.asterisk.org/r/2051

2012-07-23 21:10 +0000 [r370385]  Kevin P. Fleming <kpfleming@digium.com>

	* /, funcs/func_shell.c: Improve documentation for the SHELL()
	  dialplan function. ........ Merged revisions 370383 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370384 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-23 21:02 +0000 [r370382]  Mark Michelson <mmichelson@digium.com>

	* UPGRADE.txt: Add notes to UPGRADE.txt about addition of msg_id to
	  VoiceMails.

2012-07-23 00:15 +0000 [r370354]  Joshua Colp <jcolp@digium.com>

	* UPGRADE.txt: Update UPGRADE.txt with notes about ICE support and
	  res_xmpp.

2012-07-22 23:37 +0000 [r370353]  Matthew Jordan <mjordan@digium.com>

	* CHANGES: Update CHANGES for Asterisk 11 This updates the CHANGES
	  file with things that were committed for Asterisk 11, but were
	  not noted in that file.

2012-07-22 17:03 +0000 [r370347]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_sip.c,
	  configs/sip.conf.sample, channels/sip/include/sip.h: Prevent
	  multiple local candidates from being added with the same
	  information and add support for disabling ICE on a per-peer
	  basis. (closes issue ASTERISK-20088) Reported by: wimpy Review:
	  https://reviewboard.asterisk.org/r/2044/

2012-07-21 13:25 +0000 [r370341]  Terry Wilson <twilson@digium.com>

	* main/config_options.c, apps/app_confbridge.c,
	  apps/confbridge/conf_config_parser.c: Fix segfault introduced by
	  conversion to ACO API The value "none" is specified in the config
	  file as a valid value for the "video_mode" option. The code prior
	  to the ACO conversion did not check for "none", but just ignored
	  it and relied on the default zero value. The parsing with ACO is
	  more strict, so without handling "none" specifically, parsing
	  would fail. When parsing failed, but the module loaded anyway,
	  the config info would never be stored, and one place in the code
	  did not check for this case and would segfault. It was also
	  possible that the aco_info struct's internals would be destroyed
	  and used as well. This patch keeps the module from loading after
	  parse failures, adds the "none" option to "video_mode", registers
	  CLI functions only after parsing has completed, checks the config
	  data for NULL before accessing it, and returns -1 on some
	  allocation failures when initializing. (closes issue
	  ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested
	  by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt
	  uploaded by Terry Wilson

2012-07-20 19:36 +0000 [r370335]  Jonathan Rose <jrose@digium.com>

	* channels/chan_iax2.c: chan_iax2: Fix a segfault introduced by
	  call ID logging Didn't previously check that a non NULL IAX
	  channel was stored in the array at the requested position before
	  attempting iax_pvt_callid_get (closes issue ASTERISK-20145)
	  Reported by: Birger "WIMPy" Harzenetter

2012-07-20 19:08 +0000 [r370329]  Matthew Jordan <mjordan@digium.com>

	* apps/app_dial.c: Clean up ManagerEvent Dial documentation The
	  paragraph describing the SubEvent belongs with the SubEvent
	  parameter itself, and not with its enum values. The order of
	  parsing was placing the description after the last enum, which
	  isn't correct.

2012-07-20 18:37 +0000 [r370328]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_misdn.c: Fix build error in chan_misdn from commit
	  370316 chan_misdn was not updated properly to account for a
	  change in parameters for HANGUPCAUSE functionality. It now builds
	  properly.

2012-07-20 16:25 +0000 [r370322]  Joshua Colp <jcolp@digium.com>

	* res/res_http_websocket.exports.in: Export the
	  ast_websocket_set_nonblock function for use by other modules.

2012-07-20 15:48 +0000 [r370316]  Kinsey Moore <kmoore@digium.com>

	* funcs/func_hangupcause.c (added), main/channel.c,
	  channels/chan_dahdi.c, channels/sig_analog.c, main/rtp_engine.c,
	  channels/chan_sip.c, main/channel_internal_api.c, UPGRADE.txt,
	  include/asterisk/channel.h, channels/chan_iax2.c,
	  channels/sig_pri.c, include/asterisk/frame.h, channels/sig_ss7.c:
	  Add hangupcause translation support The HANGUPCAUSE hash (trunk
	  only) meant to replace SIP_CAUSE has now been replaced with the
	  HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better
	  facilitate access to the AST_CAUSE translations for
	  technology-specific cause codes. The HangupCauseClear application
	  has also been added to remove this data from the channel. (closes
	  issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/

2012-07-20 15:40 +0000 [r370309-370315]  Richard Mudgett <rmudgett@digium.com>

	* CHANGES: Update CHANGES about adding the AccountCode header to
	  the AMI Hangup event. (issue ASTERISK-19963)

	* main/channel.c: Add the AccountCode header to the AMI Hangup
	  event. It's harder to correlate the Newchannel and Hangup AMI
	  events without specifying "AccountCode" in both. (closes issue
	  ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches:
	  hangup_acctcode.diff (license #6397) patch uploaded by Oleg A.
	  Arkhangelsky

2012-07-19 23:21 +0000 [r370303]  Terry Wilson <twilson@digium.com>

	* include/asterisk/config_options.h,
	  apps/confbridge/include/confbridge.h, main/config_options.c,
	  apps/confbridge/conf_config_parser.c: Convert app_confbridge to
	  use the config options framework Review:
	  https://reviewboard.asterisk.org/r/2024/

2012-07-19 22:25 +0000 [r370298]  Richard Mudgett <rmudgett@digium.com>

	* /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has
	  problems casting away constness. ........ Merged revisions 370275
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 370277 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-19 22:17 +0000 [r370272-370278]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c, res/res_xmpp.c, doc/appdocsxml.dtd,
	  main/message.c, main/xmldoc.c: Add the ability to specify
	  technology specific documentation A number of applications/AMI
	  commands in Asterisk have specific behavioral differences
	  depending on the resource or channel technology those
	  applications are executed on. For example, the MessageSend
	  application/ command is technology agnostic, but how the channel
	  drivers that support that functionality behave is dependant on
	  the protocols and channel driver implementation. Prior to this
	  patch, those details were either documented in the
	  application/command documentation itself, or were left
	  undocumented. This patch adds a new element to the documentation
	  schema, <info/>. An info node is essentially a piece of
	  technology specific reference information that can be included by
	  any top level XML documentation node. For example, the
	  MessageSend application can now include XMPP/SIP specific
	  information, where that technology specific information can be
	  defined in chan_motif/res_xmpp/ chan_sip. Likewise, that
	  information can also be included in the MessageSend AMI command.
	  Review: https://reviewboard.asterisk.org/r/2049

	* /, main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled
	  To fix a memory leak in CEL, a channel datastore was introduced
	  whose destruction function pointer was pointed to the ast_free
	  macro. Without MALLOC_DEBUG enabled this compiles as fine, as
	  ast_free is defined as free. With MALLOC_DEBUG enabled, however,
	  ast_free takes on a definition from a different place then
	  utils.h, and became undefined. This patch resolves this by using
	  a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
	  calls ast_free; when MALLOC_DEBUG is not enabled, this is defined
	  to be ast_free, which is defined to be free. (issue AST-916)
	  Reported by: Thomas Arimont ........ Merged revisions 370273 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370274 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* res/res_rtp_asterisk.c, /: Handle extremely out of order RFC 2833
	  DTMF The current implementation of RFC 2833 DTMF handling in
	  res_rtp_asterisk will, if a packet arrives out of order, drop the
	  packet. This is to prevent duplicate ton generation in the
	  Asterisk core. Since the RTP layer does not buffer data itself,
	  this is the only option the RTP layer currently has for handling
	  packets that arrive out of order. For the most part, this doesn't
	  matter. For a particular digit, so long as a BEGIN packet arrives
	  before the first END packet, the digit will be produced. If
	  subsequent BEGIN packets arrive interleaved with the ENDs, they
	  will be dropped; likewise, if the BEGIN or END packets themselves
	  are out of order, those packets are dropped but sufficient
	  information is conveyed to the Asterisk core to produce the
	  appropriate digit. For certain sequences of DTMF packets - most
	  notably when, for a particular digit, an END packet arrives
	  before any BEGIN packet for that digit - this is a real problem.
	  When an END arrives before any BEGINs, the END packet is dropped
	  - but at the same time, it causes subsequent BEGIN packets for
	  that digit to be ignored. When the next in order END packet
	  arrives, it too is dropped - Asterisk believes that there was no
	  initial BEGIN. The solution this patch provides is to trust the
	  END packet to convey the information needed for the Asterisk core
	  to produce the DTMF digit. If we receive an END packet, and it: *
	  Has a timestamp greater then the last timestamp received from an
	  END packet * Does not have the same sequence number as the last
	  received sequence number (and is thus not an END packet
	  retransmission) Then we send the END frame up to the Asterisk
	  core. It contains enough DTMF information for Asterisk to produce
	  the digit. On the other hand, if we receive a BEGIN or
	  continuation packet that occurs with a timestamp equal to or less
	  then the last END timestamp, then we've received something out of
	  order - but we already have received enough information to
	  produce the digit. These packets are dropped. Much thanks goes to
	  Olle Johansson (oej) for providing the idea for this solution.
	  Review: https://reviewboard.asterisk.org/r/2033/ (closes issue
	  ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt
	  Jordan ........ Merged revisions 370252 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370271 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-19 20:37 +0000 [r370246-370265]  Jonathan Rose <jrose@digium.com>

	* main/named_acl.c, configs/acl.conf.sample: named_acl: Remove
	  systemname option from acl.conf, use asterisk.conf value Review:
	  https://reviewboard.asterisk.org/r/2057/

	* main/channel_internal_api.c: CallID Logging: Remove new
	  line/carriage return from callID change test event

2012-07-19 12:14 +0000 [r370234-370240]  Joshua Colp <jcolp@digium.com>

	* res/Makefile, res/pjproject/build/os-auto.mak.in: Use the
	  bruteforce method to get debugging enabled for pjproject.

	* res/Makefile: Turn on debugging for pjproject so we can get a
	  better idea of what is causing the generic CCSS test crash.

2012-07-18 19:48 +0000 [r370225]  Jonathan Rose <jrose@digium.com>

	* main/channel_internal_api.c: callid logging: Issue test events
	  when the callid is changed for a channel Review:
	  https://reviewboard.asterisk.org/r/2054/

2012-07-18 19:18 +0000 [r370187-370211]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/cel.c: Resolve severe memory leak in CEL logging modules.
	  A customer reported a significant memory leak using Asterisk 1.8.
	  They have tracked it down to
	  ast_cel_fabricate_channel_from_event() in main/cel.c, which is
	  called by both in-tree CEL logging modules (cel_custom.c and
	  cel_sqlite3_custom.c) for each and every CEL event that they log.
	  The cause was an incorrect assumption about how data attached to
	  an ast_channel would be handled when the channel is destroyed;
	  the data is now stored in a datastore attached to the channel,
	  which is destroyed along with the channel at the proper time.
	  (closes issue AST-916) Reported by: Thomas Arimont Review:
	  https://reviewboard.asterisk.org/r/2053/ ........ Merged
	  revisions 370205 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370206 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/channel.c, addons/app_mysql.c, main/pbx.c,
	  funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c,
	  funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c,
	  apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
	  res/res_odbc.c: Ensure that all ast_datastore_info structures are
	  'const'. While addressing a bug, I came across a instance of
	  'struct ast_datastore_info' that was not declared 'const'. Since
	  the API already expects them to be 'const', this patch changes
	  the declarations of all existing instances that were not already
	  declared that way. ........ Merged revisions 370183 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370184 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-18 15:15 +0000 [r370171-370177]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Fix a crash in pjnath when starting an
	  ICE connectivity check and immediately destroying the ICE
	  session. The initial ICE connectivity check is scheduled as a
	  timer item that is to be executed immediately. It is possible for
	  this timer item to start executing while the ICE session it is
	  working on is destroyed. To reduce the chance of this any timer
	  items that need to be immediately executed will be executed
	  within the thread that has started the initial ICE connectivity
	  check.

	* channels/chan_sip.c, include/asterisk/rtp_engine.h: Fix a crash
	  occurring as a result of excess stack usage. This fix involves
	  moving the allocation of some temporary codec structures to the
	  heap and also reduces the number of maximum payloads to something
	  more sane for both regular and low memory builds. (closes issue
	  ASTERISK-20140) Reported by: jonnt

2012-07-18 07:17 +0000 [r370165]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c, configs/unistim.conf.sample, CHANGES:
	  Added option 'interdigit_timer' to unistim.conf to make able
	  controll hardcoded dial timeout constant.

2012-07-17 19:05 +0000 [r370152-370157]  Joshua Colp <jcolp@digium.com>

	* res/res_xmpp.c: Add pubsub unsubscription support so
	  subscriptions do not linger for MWI and device state progatation.
	  The pubsub code did not attempt to remove subscriptions at all.
	  This has now changed so that if a client is being disconnected it
	  will unsubscribe. It will also unsubscribe at connection time so
	  if it unexpectedly disconnected duplicate subscriptions will not
	  occur. (closes issue ASTERISK-19882) Reported by: mattvryan

	* include/asterisk/xmpp.h, res/res_xmpp.c: Fix a crash as a result
	  of propagating MWI or device state over XMPP when the client is
	  disconnected. The MWI and device state propagation code wrongly
	  assumes that an XMPP client connection will remain established at
	  all times. This fix corrects that by making the lifetime of the
	  subscription the same as the lifetime of the connection itself.
	  As the connection is established and disconnected the
	  subscription itself is created and destroyed. (closes issue
	  ASTERISK-18078) Reported by: elguero

2012-07-16 19:58 +0000 [r370133]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip
	  outboundproxy parsing. The bug was clearing the global
	  outboundproxy when a peer-specific outboundproxy was bad. The
	  cleanup reduces duplicate code. Review:
	  https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
	  Michelson ........ Merged revisions 370131 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370132 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-16 19:14 +0000 [r370111-370126]  Joshua Colp <jcolp@digium.com>

	* res/res_xmpp.c: Fix an issue where a service discovery request
	  could crash Asterisk. A server sending a service discovery
	  request to us may or may not put a from attribute in the message.
	  If the from attribute is present use it in the to attribute for
	  the result. If the from attribute is not present do not add a to
	  attribute. (issue ASTERISK-16203) Reported by: wubbla

	* res/res_xmpp.c: Fix a bug where some XMPP servers would reject
	  authentication. We need to use the user portion of the JID and
	  not the full configured username.

	* res/res_xmpp.c: Add missing namespace for old non-SASL based
	  authentication.

	* channels/chan_sip.c: Fix a bug exposed by the testsuite where
	  text streams would no longer be parsed correctly.

2012-07-16 14:02 +0000 [r370083]  Kinsey Moore <kmoore@digium.com>

	* /, UPGRADE-10.txt, CHANGES, UPGRADE-1.8.txt: Add comments about
	  the BUILD_NATIVE change This is a significant change and mention
	  of it should have gone into UPGRADE.txt and CHANGES. ........
	  Merged revisions 370081 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370082 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-16 12:58 +0000 [r370072-370073]  Joshua Colp <jcolp@digium.com>

	* res/res_xmpp.c: Fix an issue where specifying the resource in the
	  username would cause authentication to fail.

	* channels/sip/sdp_crypto.c, channels/chan_sip.c,
	  channels/sip/security_events.c,
	  include/asterisk/http_websocket.h, configs/sip.conf.sample,
	  CHANGES, res/res_http_websocket.c, channels/sip/include/sip.h:
	  Add support for SIP over WebSocket. This allows SIP traffic to be
	  exchanged over a WebSocket connection which is useful for rtcweb.
	  Review: https://reviewboard.asterisk.org/r/2008

2012-07-16 07:38 +0000 [r370066-370067]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Deactivate timer for dialing entered
	  number on hook switch hang up. (closes issue ASTERISK-19554)
	  Reported by: Stefano Villani

	* channels/chan_unistim.c, contrib/unistimLang/fr.po (added),
	  CHANGES: Add French translation for chan_unistim phones on-screen
	  menus.

2012-07-13 18:41 +0000 [r370055-370060]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/format.h, res/res_format_attr_h263.c (added),
	  res/res_format_attr_h264.c (added): Reduce memory consumption and
	  add the H.264 and H.263 modules I shamefully neglected to add.

	* main/format.c, channels/chan_sip.c, main/translate.c,
	  include/asterisk/format.h, res/res_format_attr_silk.c,
	  res/res_format_attr_celt.c: Add support for parsing SDP
	  attributes, generating SDP attributes, and passing it through.
	  This support includes codecs such as H.263, H.264, SILK, and
	  CELT. You are able to set up a call and have attribute
	  information pass. This should help considerably with video calls.
	  Review: https://reviewboard.asterisk.org/r/2005/

2012-07-13 00:05 +0000 [r370048]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/live_ast: live_ast: don't set working directory
	  contrib/scripts/live_ast currently assumes that it is being run
	  from the top-level directory of the source tree. It creates a
	  script that will also set the working directory. This fix avoids
	  the need to set the working directory if the caller sets
	  LIVE_AST_BASE_DIR instead. It relies on realpath for that. If
	  realpath is not available, it will fall back to the original
	  behaviour. Review: https://reviewboard.asterisk.org/r/2027/

2012-07-12 21:43 +0000 [r370043]  Terry Wilson <twilson@digium.com>

	* include/asterisk/config_options.h,
	  configs/config_test.conf.sample, main/config_options.c,
	  tests/test_config.c: Handle deprecated (aliased) option names
	  with the config options api Add a simple way to register
	  "deprecated" option names that alias to a different "current"
	  name. Review: https://reviewboard.asterisk.org/r/2026/

2012-07-12 20:28 +0000 [r370037]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing
	  ast_hangup() calls on some analog exception paths. Make starting
	  analog_ss_thread() or __analog_ss_thread() failure paths hangup
	  the channel. ........ Merged revisions 370017 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370025 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-12 20:06 +0000 [r369995-370016]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Include Expires header for SIP PUBLISH
	  requests RFC3903 requres SIP PUBLISH requests to have Expires
	  headers, so add them. Review:
	  https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
	  ........ Merged revisions 370014 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 370015 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Prevent double uri_escaping in chan_sip
	  when pedantic is enabled If pedantic mode is enabled, outbound
	  invites will have double-escaped contacts. This avoids setting an
	  already-escaped string into a field where it is expected to be
	  unescaped. (closes issue ASTERISK-20023) Reported by: Walter
	  Doekes ........ Merged revisions 369993 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369994 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-12 14:38 +0000 [r369972-369974]  Michael L. Young <elgueromexicano@gmail.com>

	* /, funcs/func_math.c: Correct Documentation For DEC Function The
	  documentation for DEC in func_math.c was incorrect. Looks like a
	  copy and paste error. (Closes issue ASTERISK-20095) Reported by:
	  Billy Chia Tested by: Michael L. Young Patches: func_math.patch
	  uploaded by Billy Chia (license 6381) ........ Merged revisions
	  369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369971 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* funcs/func_math.c: Reverting last merge since it wasn't completed
	  properly.

	* funcs/func_math.c: Correct Documentation For DEC Function The
	  documentation for DEC in func_math.c was incorrect. Looks like a
	  copy and paste error. (Closes issue ASTERISK-20095) Reported by:
	  Billy Chia Tested by: Michael L. Young Patches: func_math.patch
	  uploaded by Billy Chia (license 6381) ........ Merged revisions
	  369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-07-11 18:33 +0000 [r369959]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/acl.h, channels/chan_sip.c,
	  include/asterisk/config.h, main/acl.c,
	  include/asterisk/channel.h, configs/manager.conf.sample,
	  channels/chan_iax2.c, CHANGES, main/named_acl.c (added),
	  main/config.c, main/loader.c, configs/iax.conf.sample,
	  main/manager.c, include/asterisk/event_defs.h,
	  configs/extconfig.conf.sample, configs/sip.conf.sample,
	  channels/sip/include/sip.h, main/asterisk.c,
	  configs/acl.conf.sample (added): Named ACLs: Introduces a system
	  for creating and sharing ACLs This patch adds Named ACL
	  functionality to Asterisk. This allows system administrators to
	  define an ACL and refer to it by a unique name. Configurable
	  items can then refer to that name when specifying access control
	  lists. It also includes updates to all core supported consumers
	  of ACLs. That includes manager, chan_sip, and chan_iax2. This
	  feature is based on the deluxepine-trunk by Olle E. Johansson and
	  provides a subset of the Named ACL functionality implemented in
	  that branch. For more information on this feature, see acl.conf
	  and/or the Asterisk wiki. Review:
	  https://reviewboard.asterisk.org/r/1978/

2012-07-11 17:16 +0000 [r369940]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, main/ast_expr2.h, main/ast_expr2f.c, res/ael/ael_lex.c,
	  funcs/func_realtime.c, main/ast_expr2.c: Allow the REALTIME()
	  function to report errors back to the caller. Also, do more error
	  checking on the arguments specified to the REALTIME() function
	  and clarify the documentation. While I was editing the file, a
	  few coding guidelines fixups, as well. Review:
	  https://reviewboard.asterisk.org/r/2031/ ........ Merged
	  revisions 369937 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369938 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-11 17:14 +0000 [r369939]  Matthew Jordan <mjordan@digium.com>

	* main/features.c: Don't perform an XInclude to a document node
	  that may not always be present Because some of the manager events
	  are defined in the top of the source, due to the macro calls not
	  containing all necessary information to have the documentation
	  colocated with the call itself, several include statements were
	  failing when built with 'make'. While this did not cause any
	  problems in compilation or validation, it did result in a number
	  of warnings being dumped to stderr. This patch changes those
	  references such that they always resolve, regardless of the
	  documentation build options.

2012-07-11 16:42 +0000 [r369936]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Do not consider failure to read the
	  configuration file in chan_motif to be a show stopper for loading
	  Asterisk by returning decline instead of failure. (closes issue
	  ASTERISK-20103) Reported by: Terry Wilson

2012-07-11 02:06 +0000 [r369905-369910]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c, main/channel.c, channels/sig_analog.c, main/logger.c,
	  channels/sig_pri.c, main/asterisk.c, main/loader.c: Fix
	  validation errors when producing documentation using default
	  build script The awk script parses out the first instance of the
	  DOCUMENTATION tag that it finds within a file. If a file did not
	  previously have a DOCUMENTATION tag but received one due to it
	  having an AMI event, then the XML fragment associated with the
	  AMI event was erroneously placed in the resulting XML file.
	  Without the python scripts, these XML fragments will not
	  validate. This patch adds DOCUMENTATION tags at the top of those
	  files that did not previously have them to prevent the awk script
	  from pulling AMI event documentation.

	* main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
	  channels/chan_local.c, channels/sig_analog.c, main/manager.c,
	  channels/chan_agent.c, main/features.c, main/logger.c,
	  channels/sig_pri.c, doc/appdocsxml.dtd, main/asterisk.c,
	  main/loader.c: Add some additional documentation for core AMI
	  events This patch adds some basic documentation for a number of
	  modules. This includes core source files in Asterisk (those in
	  main), as well as chan_agent, chan_dahdi, chan_local, sig_analog,
	  and sig_pri. The DTD has also been updated to allow referencing
	  of AMI commands.

2012-07-10 15:36 +0000 [r369900]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Fix failing SDP_offer_answer test Asterisk
	  now generates image stream declinations with the same transport
	  case that it used to before the stream declination improvements.
	  (udptl vs UDPTL) (closes issue SWP-4736)

2012-07-10 15:25 +0000 [r369873-369898]  Joshua Colp <jcolp@digium.com>

	* channels/chan_motif.c: Add additional description stanza names
	  from the old Google Talk protocol which is used with Google
	  Voice. (closes issue ASTERISK-20114) Reported by: Malcolm
	  Davenport

	* channels/chan_motif.c: Respect codec preference order when adding
	  codecs to a media description. This change allows an endpoint in
	  motif.conf to be configured with a preference of G.722 and
	  fallback of ulaw. With Google this allows communication with
	  Google Talk clients to use G.722 while when using Google Voice
	  ulaw will be used. (closes issue ASTERISK-20114) Reported by:
	  Malcolm Davenport

2012-07-10 13:40 +0000 [r369872]  Kinsey Moore <kmoore@digium.com>

	* main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related
	  documentation Correct documentation on labeliftrue and
	  labeliffalse parameters of GotoIf() and update several other
	  locations that use the same syntax. (closes issue ASTERISK-20007)
	  Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
	  revisions 369869 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369871 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-10 13:34 +0000 [r369870]  Matthew Jordan <mjordan@digium.com>

	* main/libasteriskssl.c: Fix initial loading problem with res_curl
	  When the OpenSSL duplicate initialization issues were resolved in
	  r351447, res_curl could fail to load if it checked
	  SSL_library_init after SSL initialization completed. This is due
	  to the SSL_library_init stub returning a value of 0 for success,
	  as opposed to a value of 1. OpenSSL uses a value of 1 to indicate
	  success - in fact, SSL_library_init is documented to always
	  return 1. Interestingly, the CURL libraries actually checked the
	  return value - the fact that nothing else that depends on OpenSSL
	  was having problems loading probably means they don't check the
	  return value. (closes issue AST-924) Reported by: Guenther
	  Kelleter patches: (AST-924.patch license #6372 uploaded by
	  Guenther Kelleter)

2012-07-10 11:49 +0000 [r369837-369864]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_motif.c: Add required items
	  for Google video support. This adds legacy STUN support for RTCP
	  sockets, adds RTCP candidates to the Google transport
	  information, and adds required codec parameters. (closes issue
	  ASTERISK-20106) Reported by: Malcolm Davenport

	* main/stun.c: When receiving a STUN binding request send one out
	  as the Google Talk client uses this as a method to determine if
	  the remote party is still reachable or not. Failure to do this
	  results in the Google Talk client ignoring RTP packets after a
	  specific period of time. This is also done as a result of
	  receiving a STUN binding request so that the username information
	  can be used from the inbound request, thus not requiring it to be
	  stored on a per candidate basis. (closes issue ASTERISK-20107)
	  Reported by: Malcolm Davenport

	* channels/chan_sip.c: Add support for exposing the received
	  contact URI and also for setting the request URI in messages.
	  (closes issue AST-911)

	* channels/chan_motif.c: Force the clock rate of G.722 to be 16000
	  when using the Google transports as it is 8000 elsewhere. (closes
	  issue ASTERISK-20105) Reported by: Malcolm Davenport

	* configs/motif.conf.sample: Document that multiple endpoints using
	  the same connection is not supported. (closes issue
	  ASTERISK-20104) Reported by: Malcolm Davenport

2012-07-09 17:07 +0000 [r369820]  Jason Parker <jparker@digium.com>

	* configs/sip_notify.conf.sample, /: Add Digium phones context to
	  sip_notify sample config. This makes it so that they can be
	  reconfigured remotely. (closes issue ASTERISK-19910) ........
	  Merged revisions 369818 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369819 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-09 16:44 +0000 [r369811-369817]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Fix an issue where media would not flow
	  for situations where the legacy STUN code is in use. The STUN
	  packets should *not* be blocked by strict RTP. (closes issue
	  ASTERISK-20102) Reported by: Malcolm Davenport

	* res/res_xmpp.c: Add additional namespaces for Google Talk which
	  are used for the gmail client. (closes issue ASTERISK-20101)
	  Reported by: Malcolm Davenport

	* channels/chan_motif.c: Fix dependency to be on res_xmpp. Long ago
	  in a galaxy far far away it used to use res_jabber.

2012-07-09 14:54 +0000 [r369794]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Fix small behavioral change
	  accidentally introduced in r369750 When removing the warning for
	  AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
	  the return value, which would likely make the indication not be
	  sent in audio. This fixes that while still removing the warning
	  message. ........ Merged revisions 369792 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369793 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-07 17:06 +0000 [r369769]  Joshua Colp <jcolp@digium.com>

	* res/res_xmpp.exports.in (added), include/asterisk/xmpp.h,
	  channels/chan_motif.c (added), UPGRADE.txt,
	  channels/chan_gtalk.c, res/res_xmpp.c, CHANGES, res/res_jabber.c,
	  configs/motif.conf.sample (added): Add a new unified Jingle,
	  Google Jingle, and Google Talk channel driver written from
	  scratch called chan_motif. This channel driver is a replacement
	  for both chan_gtalk and chan_jingle but adds additional features
	  not found in either. These features include full configuration
	  reload, video, full codec support, bidirectional cause code
	  mapping, hold, unhold, and ringing indication. It is also
	  compliant with the current published Jingle and Google Jingle
	  specifications. The original Google Talk protocol is also
	  supported for Google Voice interoperability. You may ask yourself
	  though where the name motif comes from... and I would say to
	  you... music! motif: a perceivable or salient recurring fragment
	  or succession of notes Sorta like a jingle! Review:
	  https://reviewboard.asterisk.org/r/1917/

2012-07-06 22:03 +0000 [r369765]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c:
	  Remove unnecessary generation of informational cause frames It is
	  not necessary to generate information cause code frames on every
	  protocol event that occurs. This removes all the instances where
	  the frame was not conveying a cause code and was instead just
	  conveying a protocol-specific message. This also corrects the
	  generation of the message associated with disconnects for MFC/R2
	  to use the MFC/R2 specific text for the disconnect cause.

2012-07-06 21:28 +0000 [r369764]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: chan_sip: Add case for FLASH control
	  frames so that we don't display a warning. chan_sip channels can
	  receive flash control frames when connected to analog phones and
	  possibly for other reasons. There really isn't a reason to warn
	  when these frames are received, we can safely ignore them.
	  Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license
	  6182) ........ Merged revisions 369750 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369751 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-06 18:49 +0000 [r369710-369733]  Mark Michelson <mmichelson@digium.com>

	* main/tcptls.c, /: Remove a superfluous and dangerous freeing of
	  an SSL_CTX. The problem here is that multiple server sessions
	  share a SSL_CTX. When one session ended, the SSL_CTX would be
	  freed and set NULL, leaving the other sessions unable to
	  function. The code being removed is superfluous because the
	  SSL_CTX structures for servers will be properly freed when
	  ast_ssl_teardown is called. (closes issue ASTERISK-20074)
	  Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
	  by Mark Michelson (license #5049) Testers: Trevor Helmsley
	  ........ Merged revisions 369731 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369732 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/bridging.c: Fix bridging thread leak. The bridge thread
	  was exiting but was never being reaped using pthread_join(). This
	  has been fixed now by calling pthread_join() in
	  ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by
	  Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
	  ........ Merged revisions 369708 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369709 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-06 14:32 +0000 [r369703]  Joshua Colp <jcolp@digium.com>

	* res/pjproject/pjnath/include/pjnath/ice_session.h,
	  res/pjproject/pjnath/src/pjnath/ice_session.c: Import revision
	  4196 from pjproject trunk. Fix a crash issue when starting ICE
	  connectivity checks and immediately destroying the ICE session.
	  This was exposed by the SIP CCSS test. Full fix for this issue
	  will be worked on as a medium to long term roadmap item. pjroject
	  issue viewable at https://trac.pjsip.org/repos/ticket/1548

2012-07-05 21:36 +0000 [r369681]  Matthew Jordan <mjordan@digium.com>

	* res/res_stun_monitor.c, CHANGES: Add 'stun show status' command
	  This patch adds a new CLI command, 'stun show status'. This
	  command will show a table describing all known STUN servers and
	  statuses. (closes issue ASTERISK-18046) Reported by: Jeremy
	  Kister Tested by: Jeremy Kister patches:
	  (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy
	  Kister) Review: https://reviewboard.asterisk.org/r/2001

2012-07-05 19:36 +0000 [r369677]  Richard Mudgett <rmudgett@digium.com>

	* res/pjproject/pjmedia/include/pjmedia,
	  res/pjproject/pjsip/include/pjsip,
	  res/pjproject/pjlib/include/pj/compat,
	  res/pjproject/pjmedia/include/pjmedia-codec: Make res/pjproject
	  ignore more files.

2012-07-05 19:36 +0000 [r369676]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_voicemail.c: AST-2012-011: Resolve heap corruption
	  issue with voicemail The heard and deleted arrays in the
	  voicemail state structure were not handled properly following the
	  memory leak fix in r354890 and a fix for an invalid free in
	  r356797. This could result in accessing and writing into freed
	  memory. The allocation for these arrays has been reworked to
	  avoid the possibility of invalid frees, access of freed memory,
	  and crashes that were occurring as a result of this. Locking
	  around accesses and modifications of the voicemail state
	  structure members dh_arraysize, heard, and deleted has been added
	  to prevent simultaneous modification and access when IMAP storage
	  is in use. If IMAP storage is not in use, this locking is not
	  compiled in. Review: https://reviewboard.asterisk.org/r/1994/
	  (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by:
	  Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by
	  kmoore (license 6273) ........ Merged revisions 369652 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369653 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-05 19:32 +0000 [r369666-369673]  Richard Mudgett <rmudgett@digium.com>

	* res/pjproject/pjsip/src/pjsip-ua,
	  res/pjproject/pjsip-apps/src/ipjsystest/ipjsystest.xcodeproj,
	  res/pjproject/pjnath/src/pjnath-test,
	  res/pjproject/third_party/build/speex,
	  res/pjproject/third_party/build/gsm/output,
	  res/pjproject/pjmedia/include/pjmedia-codec,
	  res/pjproject/third_party/build/baseclasses,
	  res/pjproject/third_party/build/srtp,
	  res/pjproject/pjsip-apps/src/samples,
	  res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
	  res/pjproject/pjlib/include/pj++,
	  res/pjproject/tests/pjsua/scripts-call,
	  res/pjproject/third_party/srtp/doc,
	  res/pjproject/pjsip-apps/src/pocketpj/output,
	  res/pjproject/pjnath/bin,
	  res/pjproject/third_party/srtp/crypto/replay,
	  res/pjproject/pjsip/include/pjsip,
	  res/pjproject/third_party/build/speex/speex,
	  res/pjproject/build.symbian, res/pjproject/third_party/bin,
	  res/pjproject/pjsip/src/pjsua-lib,
	  res/pjproject/third_party/srtp/include,
	  res/pjproject/third_party/portaudio/doc, res/pjproject/lib,
	  res/pjproject/pjmedia/include/pjmedia-videodev,
	  res/pjproject/pjlib/bin,
	  res/pjproject/third_party/srtp/crypto/cipher,
	  res/pjproject/third_party/build/speex/output,
	  res/pjproject/pjlib-util/src/pjlib-util,
	  res/pjproject/third_party/portaudio/test,
	  res/pjproject/third_party/build/gsm,
	  res/pjproject/third_party/portaudio/include,
	  res/pjproject/pjsip-apps/src/pjsua_wince,
	  res/pjproject/pjsip/include/pjsip-simple,
	  res/pjproject/pjmedia/src/pjmedia-codec,
	  res/pjproject/tests/pjsua,
	  res/pjproject/pjsip-apps/src/pocketpj/res,
	  res/pjproject/pjsip-apps/src/3rdparty_media_sample,
	  res/pjproject/third_party/gsm/inc,
	  res/pjproject/pjsip-apps/build/wince-evc4,
	  res/pjproject/pjsip-apps/src/ipjsua/Resources-iPad,
	  res/pjproject/third_party/portaudio/src/hostapi,
	  res/pjproject/third_party/portaudio/build, res/pjproject/build,
	  res/pjproject/third_party/build/resample,
	  res/pjproject/third_party/speex/include,
	  res/pjproject/pjsip/src/pjsip,
	  res/pjproject/pjlib/build/wince-evc4,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/group,
	  res/pjproject/pjsip-apps/src/symbian_ua,
	  res/pjproject/tests/pjsua/wavs,
	  res/pjproject/third_party/portaudio/src/os/win,
	  res/pjproject/pjsip-apps/src/ipjsua/Classes,
	  res/pjproject/pjmedia/include/pjmedia,
	  res/pjproject/tests/pjsua/scripts-sendto,
	  res/pjproject/third_party/gsm/src,
	  res/pjproject/third_party/portaudio/build/msvc,
	  res/pjproject/pjsip-apps/src/confbot,
	  res/pjproject/pjnath/src/pjturn-client,
	  res/pjproject/pjlib-util/build/output,
	  res/pjproject/third_party/BaseClasses,
	  res/pjproject/third_party/portaudio/src/hostapi/wasapi,
	  res/pjproject/third_party/portaudio/src/hostapi/wdmks,
	  res/pjproject/pjlib/src/pj/compat,
	  res/pjproject/third_party/srtp/crypto/include,
	  res/pjproject/third_party/speex/include/speex,
	  res/pjproject/third_party/gsm/add-test,
	  res/pjproject/pjsip/build,
	  res/pjproject/pjsip-apps/src/pjsua_wince/output,
	  res/pjproject/third_party/gsm/lib, res/pjproject/pjsip,
	  res/pjproject/pjsip-apps/src/pjsystest,
	  res/pjproject/third_party/portaudio/src,
	  res/pjproject/third_party/speex/libspeex,
	  res/pjproject/pjsip/build/wince-evc4/output,
	  res/pjproject/pjlib-util/src/pjlib-util-test,
	  res/pjproject/pjsip-apps/src/symsndtest,
	  res/pjproject/third_party/srtp/tables,
	  res/pjproject/third_party/g7221, res/pjproject/pjmedia/include,
	  res/pjproject/pjlib/include/pj,
	  res/pjproject/third_party/build/portaudio/output,
	  res/pjproject/pjsip-apps/bin,
	  res/pjproject/pjsip-apps/src/ipjsua/ipjsua.xcodeproj,
	  res/pjproject/pjsip-apps/src/pjsua,
	  res/pjproject/third_party/srtp/test,
	  res/pjproject/pjsip/include/pjsip-ua,
	  res/pjproject/third_party/resample,
	  res/pjproject/third_party/build/ilbc,
	  res/pjproject/pjmedia/src/pjmedia-audiodev,
	  res/pjproject/pjsip-apps/src/ipjsua,
	  res/pjproject/third_party/srtp/srtp,
	  res/pjproject/third_party/build/milenage,
	  res/pjproject/pjmedia/src/pjmedia, res/pjproject/pjlib-util,
	  res/pjproject/third_party/portaudio/src/common,
	  res/pjproject/third_party/portaudio/bindings/cpp,
	  res/pjproject/pjlib-util/build/wince-evc4/output,
	  res/pjproject/third_party/srtp/crypto/kernel,
	  res/pjproject/tests/pjsua/scripts-pres, res/pjproject/pjnath,
	  res/pjproject/pjsip/build/output,
	  res/pjproject/pjsip-apps/build/output,
	  res/pjproject/pjsip-apps/build, res/pjproject/tests/automated,
	  res/pjproject/pjnath/build/wince-evc4/output,
	  res/pjproject/third_party/portaudio/src/hostapi/asio,
	  res/pjproject/pjnath/include/pjnath,
	  res/pjproject/pjsip/src/test,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/gfx,
	  res/pjproject/pjsip/bin,
	  res/pjproject/third_party/build/portaudio,
	  res/pjproject/pjlib/build/output, res/pjproject/pjmedia/src,
	  res/pjproject/pjlib/src/pj, res/pjproject/pjlib,
	  res/pjproject/pjlib/build/wince-evc4/output,
	  res/pjproject/pjmedia/src/test/vectors,
	  res/pjproject/third_party/portaudio/src/hostapi/jack,
	  res/pjproject/pjmedia/src/pjmedia-codec/g722,
	  res/pjproject/third_party/portaudio/src/hostapi/coreaudio,
	  res/pjproject/pjmedia/build/output,
	  res/pjproject/pjlib-util/include/pjlib-util,
	  res/pjproject/third_party/portaudio/src/hostapi/asihpi,
	  res/pjproject/third_party/milenage, res/pjproject/pjnath/src,
	  res/pjproject/tests/pjsua/scripts-run,
	  res/pjproject/pjlib-util/build/wince-evc4,
	  res/pjproject/pjmedia/lib, res/pjproject/pjmedia/src/test,
	  res/pjproject/third_party/speex/symbian,
	  res/pjproject/third_party/speex/win32,
	  res/pjproject/third_party/srtp/crypto/test,
	  res/pjproject/pjlib-util/bin,
	  res/pjproject/third_party/portaudio/build/scons,
	  res/pjproject/tests/cdash,
	  res/pjproject/tests/pjsua/scripts-media-playrec,
	  res/pjproject/third_party/build/portaudio/src,
	  res/pjproject/pjlib/src, res/pjproject/third_party/mp3,
	  res/pjproject/pjnath/lib, res/pjproject/third_party/build/g7221,
	  res/pjproject/third_party/gsm/man,
	  res/pjproject/third_party/portaudio/src/os/unix,
	  res/pjproject/third_party/portaudio/bindings,
	  res/pjproject/pjsip-apps/src/python,
	  res/pjproject/pjnath/src/pjnath, res/pjproject/third_party/lib,
	  res/pjproject/third_party/portaudio/src/os/mac_osx,
	  res/pjproject/third_party/srtp/crypto/ae_xfm,
	  res/pjproject/pjsip-apps/bin/samples,
	  res/pjproject/pjnath/src/pjturn-srv,
	  res/pjproject/third_party/portaudio/pablio,
	  res/pjproject/pjlib/lib, res/pjproject/third_party/g7221/decode,
	  res/pjproject/pjlib/include/pj/compat,
	  res/pjproject/third_party/gsm,
	  res/pjproject/third_party/build/baseclasses/output,
	  res/pjproject/third_party/build/srtp/output,
	  res/pjproject/third_party/srtp, res/pjproject/pjnath/build,
	  res/pjproject/tests/pjsua/scripts-sipp, res/pjproject/pjsip-apps,
	  res/pjproject/pjnath/build/wince-evc4,
	  res/pjproject/third_party/srtp/crypto/rng,
	  res/pjproject/pjsip/build/wince-evc4,
	  res/pjproject/pjsip-apps/build/wince-evc4/output,
	  res/pjproject/third_party/gsm/tst,
	  res/pjproject/third_party/portaudio/src/hostapi/dsound,
	  res/pjproject/third_party/portaudio/testcvs,
	  res/pjproject/pjsip-apps/src/ipjsystest/Classes,
	  res/pjproject/pjlib/build, res/pjproject/third_party/portaudio,
	  res/pjproject/third_party/portaudio/src/hostapi/wmme,
	  res/pjproject/pjlib-util/docs,
	  res/pjproject/pjmedia/include/pjmedia-audiodev,
	  res/pjproject/pjsip-apps/src/vidgui,
	  res/pjproject/pjlib/src/pjlib-test,
	  res/pjproject/pjsip-apps/src/py_pjsua,
	  res/pjproject/third_party/portaudio/src/os,
	  res/pjproject/pjsip/include,
	  res/pjproject/pjmedia/build/wince-evc4,
	  res/pjproject/pjmedia/src/pjmedia-videodev,
	  res/pjproject/pjsip-apps/src, res/pjproject/third_party/speex,
	  res/pjproject/third_party/gsm/tls,
	  res/pjproject/third_party/g7221/common,
	  res/pjproject/tests/pjsua/tools,
	  res/pjproject/third_party/resample/include,
	  res/pjproject/third_party/build/samplerate/output,
	  res/pjproject/third_party/build/samplerate,
	  res/pjproject/third_party/gsm/bin,
	  res/pjproject/pjsip/src/pjsip-simple,
	  res/pjproject/third_party/g7221/encode,
	  res/pjproject/pjlib/src/pjlib-samples,
	  res/pjproject/pjsip-apps/lib,
	  res/pjproject/pjsip-apps/src/ipjsystest,
	  res/pjproject/pjlib-util/include,
	  res/pjproject/third_party/build/resample/output,
	  res/pjproject/third_party/build/ilbc/output,
	  res/pjproject/third_party/srtp/crypto,
	  res/pjproject/pjsip-apps/src/python/samples, res/pjproject/tests,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/sis,
	  res/pjproject/pjnath/include,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui,
	  res/pjproject/pjmedia/build, res/pjproject/pjmedia,
	  res/pjproject/third_party/build/milenage/output,
	  res/pjproject/pjlib-util/build, res/pjproject/pjsip/src,
	  res/pjproject/pjmedia/build/wince-evc4/output,
	  res/pjproject/third_party/portaudio/src/hostapi/alsa,
	  res/pjproject/pjsip-apps/docs,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/inc,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/data,
	  res/pjproject/tests/pjsua/scripts-pesq,
	  res/pjproject/third_party/srtp/pjlib,
	  res/pjproject/pjlib/include, res/pjproject/pjnath/build/output,
	  res/pjproject/third_party/srtp/crypto/hash,
	  res/pjproject/build/vs, res/pjproject/pjlib/docs,
	  res/pjproject/third_party/build,
	  res/pjproject/third_party/resample/src,
	  res/pjproject/third_party, res/pjproject/pjlib/src/pjlib++-test,
	  res/pjproject/third_party/build/g7221/output,
	  res/pjproject/third_party/srtp/crypto/math,
	  res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/src/pocketpj,
	  res/pjproject/tests/pjsua/scripts-recvfrom,
	  res/pjproject/third_party/portaudio/build/dev-cpp,
	  res/pjproject/pjsip/include/pjsua-lib,
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/src, res/pjproject,
	  res/pjproject/third_party/portaudio/src/hostapi/oss,
	  res/pjproject/pjlib-util/src, res/pjproject/third_party/ilbc:
	  Make res/pjproject ignore some generated files.

	* include/asterisk/utils.h: Tweak some comments and whitespace in
	  utils.h

2012-07-05 18:11 +0000 [r369644]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c: app_mixmonitor: Fix a reference leak in
	  manager_mixmonitor function Manager_mixmonitor included an early
	  return on failed executions of mixmonitor that would result in a
	  leaked channel reference. (closes issue ASTERISK-19943) Reported
	  by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded
	  by Mark Murawski (license 5791)

2012-07-05 17:03 +0000 [r369628]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Do not send a BYE when a provisional
	  response arrives during a re-INVITE Commits r369557 and r369579
	  were done to improve handling of re-INVITEs when the UA that was
	  supposed to receive the re-INVITE fails to respond. A limitation
	  of those patches occurred when a UA sent a provisional response
	  to the re-INVITE. This triggered a sending of a BYE in
	  check_pending. This patch tweaks the handling of the re-INVITE
	  such that a BYE is not sent in response to those messages. (issue
	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
	  patches: (reinvite_tweak.diff license #5012 by Steve Davies)
	  ........ Merged revisions 369626 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369627 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-05 11:42 +0000 [r369602-369620]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooCmdChannel.c,
	  addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c:
	  Fix dev mode ooh323 warnings

	* addons/chan_ooh323.c, addons/ooh323c/src/ooq931.h,
	  addons/ooh323c/src/ooCalls.h, configs/chan_ooh323.conf.sample
	  (removed), addons/ooh323c/src/ooh323ep.c, CHANGES,
	  configs/ooh323.conf.sample (added),
	  addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
	  addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooLogChan.h,
	  addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/ooh245.c,
	  addons/ooh323cDriver.c, addons/ooh323c/src/ooh245.h,
	  addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c:
	  Added direct media support to ooh323 channel driver options are
	  documented in config sample sample config rename to proper name -
	  ooh323.conf To change media address ooh323 send empty TCS if
	  there was completed TCS exchange or send facility
	  forwardedelements with new fast start proposal if not. Then close
	  transmit logical channels and renew TCS exchange. If new fast
	  start proposal is received then ooh323 stack call back channel
	  driver routine to change rtp address in the rtp instance. If
	  empty TCS is received then close transmit logical channels and
	  renew TCS exchange Review:
	  https://reviewboard.asterisk.org/r/1607/

	* addons/ooh323cDriver.c: fix small mistake in the previous

	* addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
	  addons/ooh323c/src/decode.c, addons/ooh323c/src/perutil.c,
	  addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
	  addons/ooh323c/src/ooq931.c: Fix modern gcc warning Review:
	  https://reviewboard.asterisk.org/r/1767

2012-07-03 17:07 +0000 [r369559-369581]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: More improvements to re-INVITEs timing
	  out after a provisional response There is no need to call
	  check_pendings() on a final response to an INVITE when destroying
	  the scheduler entry as it will be done later during normal
	  processing. (issue ASTERISK-19992) ........ Merged revisions
	  369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369580 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle
	  re-INVITEs with provisional but no final repsonses A previous
	  attempt at fixing this issue had negative side effects related to
	  attended transfers which this patch should resolve. Many thanks
	  to Steve Davies for all of the good suggestions and testing.
	  (closes issue ASTERISK-19992) Reported by: Steve Davies Tested
	  by: Steve Davies, Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/2009/ ........ Merged
	  revisions 369557 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369558 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-07-02 14:06 +0000 [r369517-369527]  Joshua Colp <jcolp@digium.com>

	* configs/xmpp.conf.sample (added), include/asterisk/xmpp.h
	  (added), configs/cli_aliases.conf.sample, res/res_xmpp.c (added):
	  Add a cleaned up drop-in replacement for res_jabber called
	  res_xmpp. This provides the same externally facing functionality
	  but is implemented differently internally. This is currently not
	  built by default but this will be changed once chan_jingle2
	  (insert actual name in your head when reading this after it has
	  been merged) is in the tree. Review:
	  https://reviewboard.asterisk.org/r/1983/

	* res/res_rtp_asterisk.c: Ensure the timer heap is protected by a
	  lock.

	* res/pjproject/pjlib/include/pj/config_site.h: Enable IPv6 support
	  in pjproject.

	* res/res_rtp_asterisk.c: Don't try to send connectivity checks on
	  RTCP if RTCP is no longer present and don't do multiple ICE
	  connectivity checks at once.

	* res/pjproject/pjlib/src/pj/sock_qos_common.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/crc32.c (added),
	  res/pjproject/pjsip/src/pjsip-simple/xpidf.c (added),
	  res/pjproject/third_party/gsm/src/gsm_implode.c (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-cancel-no-final.xml
	  (added), res/pjproject/build.symbian/pjmedia.mmp (added),
	  res/pjproject/third_party/build/portaudio/src/pa_hostapi.h
	  (added), res/pjproject/pjlib/src/pjlib-test/fifobuf.c (added),
	  res/pjproject/pjlib/src/pj/file_access_unistd.c (added),
	  res/pjproject/third_party/gsm/src/toast_ulaw.c (added),
	  res/pjproject/pjsip/include/pjsip/sip_transport_tls.h (added),
	  res/pjproject/pjsip/include/pjsip/sip_multipart.h (added),
	  res/pjproject/pjmedia/src/pjmedia/errno.c (added),
	  res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.vcp (added),
	  res/pjproject/third_party/speex/COPYING (added),
	  res/pjproject/pjlib/src/pj/os_core_darwin.m (added),
	  res/pjproject/third_party/ilbc/packing.c (added),
	  res/pjproject/third_party/build/portaudio/src/pa_mac_core_internal.h
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/300_srtp_receive_crypto_tag_zero.py
	  (added), res/pjproject/third_party/ilbc/packing.h (added),
	  res/pjproject/pjlib/src/pj/pool_caching.c (added),
	  res/pjproject/pjnath/include/pjnath/errno.h (added),
	  res/pjproject/pjmedia/include/pjmedia-codec/h264_packetizer.h
	  (added), res/pjproject/pjmedia/include/pjmedia/sdp_neg.h (added),
	  res/pjproject/third_party/speex/libspeex/lsp_bfin.h (added),
	  res/pjproject/third_party/portaudio/aclocal.m4 (added),
	  res/pjproject/third_party/mp3/mp3_port.h (added),
	  res/pjproject/third_party/BaseClasses/ctlutil.cpp (added),
	  res/pjproject/pjsip-apps/src/pocketpj/PocketPJDlg.cpp (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/240_publish_scenarios.py
	  (added), res/pjproject/README-RTEMS (added),
	  res/pjproject/third_party/build/portaudio/output (added),
	  res/pjproject/pjsip-apps/build/Makefile (added),
	  res/pjproject/tests/pjsua/scripts-sipp/prack_fork.xml (added),
	  res/pjproject/pjlib-util/src/pjlib-util-test/stun.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/dns_dump.c (added),
	  res/pjproject/pjmedia/include/pjmedia/circbuf.h (added),
	  res/pjproject/pjlib/build/os-darwinos.mak (added),
	  res/pjproject/third_party/srtp/test/rtpw.c (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-answer-180-multiple-fmts.xml
	  (added),
	  res/pjproject/third_party/srtp/crypto/include/cryptoalg.h
	  (added), res/pjproject/third_party/portaudio/bindings/cpp
	  (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-answer-200-reinvite-without-sdp.xml
	  (added), res/pjproject/third_party/portaudio/configure.in
	  (added), res/pjproject/pjmedia/include/pjmedia-codec/g722.h
	  (added), res/pjproject/pjsip-apps/src/vidgui/pj-pkgconfig.mak
	  (added), res/pjproject/pjmedia/include/pjmedia-codec/speex.h
	  (added), res/pjproject/config.guess (added),
	  res/pjproject/tests/cdash/cfg_site_sample.py (added),
	  res/pjproject/third_party/portaudio/src/common/pa_skeleton.c
	  (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiSettingItemList.hrh
	  (added), res/pjproject/third_party/srtp/test/getopt_s.c (added),
	  res/pjproject/pjmedia/src/pjmedia-codec/g722 (added),
	  res/pjproject/tests/pjsua/scripts-pesq/201_codec_g722.py (added),
	  res/pjproject/pjnath/src/pjturn-client/client_main.c (added),
	  res/pjproject/third_party/gsm/src/short_term.c (added),
	  res/pjproject/build.symbian/libg7221codec.mmp (added),
	  res/pjproject/pjmedia/src/pjmedia/wsola.c (added),
	  res/pjproject/pjlib-util/include/pjlib-util/hmac_sha1.h (added),
	  res/pjproject/pjlib/include/pj++/list.hpp (added),
	  res/pjproject/third_party/ilbc/anaFilter.c (added),
	  res/pjproject/third_party/mp3 (added),
	  res/pjproject/pjmedia/src/pjmedia/tonegen.c (added),
	  res/pjproject/pjsip-apps/src/samples/stateful_proxy.c (added),
	  res/pjproject/third_party/ilbc/anaFilter.h (added),
	  res/pjproject/pjsip-apps/src/symsndtest/app_main.cpp (added),
	  res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.cpp (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-invite.xml (added),
	  res/pjproject/third_party/g7221/encode/sam2coef.c (added),
	  res/pjproject/pjlib/src/pj/compat/string.c (added),
	  res/pjproject/pjlib/include/pj/compat/cc_gcce.h (added),
	  res/pjproject/pjlib/include/pj/config_site_sample.h (added),
	  res/pjproject/third_party/build/srtp/output (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_8000.py
	  (added), res/pjproject/tests/pjsua/scripts-sipp/uac-options.xml
	  (added), res/pjproject/third_party/ilbc/iCBConstruct.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/153_err_sdp_unsupported_codec.py
	  (added), res/pjproject/pjsip/build/wince-evc4 (added),
	  res/pjproject/third_party/ilbc/iCBConstruct.h (added),
	  res/pjproject/pjsip-apps/src/py_pjsua/py_pjsua.def (added),
	  res/pjproject/pjnath/build/pjstun_srv_test.vcproj (added),
	  res/pjproject/pjlib/src/pjlib-test/util.c (added),
	  res/pjproject/pjmedia/include/pjmedia-audiodev (added),
	  res/pjproject/pjlib/src/pj/ctype.c (added),
	  res/pjproject/third_party/ilbc/enhancer.c (added),
	  res/pjproject/pjsip-apps/src/py_pjsua (added),
	  res/pjproject/third_party/speex/libspeex/modes_wb.c (added),
	  res/pjproject/third_party/gsm/tst/gsm2cod.c (added),
	  res/pjproject/third_party/ilbc/enhancer.h (added),
	  res/pjproject/pjsip-apps/src (added),
	  res/pjproject/build/m-arm.mak (added),
	  res/pjproject/third_party/gsm/src/add.c (added),
	  res/pjproject/pjsip/src/pjsip/sip_parser_wrap.cpp (added),
	  res/pjproject/pjlib/src/pj/timer_symbian.cpp (added),
	  res/pjproject/pjsip-apps/src/vidgui/vidwin.cpp (added),
	  res/pjproject/pjlib/include/pj/pool_buf.h (added),
	  res/pjproject/third_party/g7221/encode (added),
	  res/pjproject/pjmedia/src/pjmedia-audiodev/wmme_dev.c (added),
	  res/pjproject/tests/pjsua/scripts-call/300_ice_1_0.py (added),
	  res/pjproject/tests/pjsua/config_site.py (added),
	  res/pjproject/pjsip-apps/src/pjsua/main.c (added),
	  res/pjproject/pjlib/src/pj/os_timestamp_posix.c (added),
	  res/pjproject/pjmedia/include/pjmedia-videodev/videodev_imp.h
	  (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/230_reg_bad_fail_stale_true.py
	  (added), res/pjproject/third_party/srtp/config.h_win32vc7
	  (added), res/pjproject/tests/pjsua/scripts-pesq (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-glare.xml
	  (added), res/pjproject/pjmedia/src/pjmedia/dummy.c (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/209c_reg_handle_423_bad_min_expires2.py
	  (added), res/pjproject/pjlib/include/pj++/hash.hpp (added),
	  res/pjproject/pjmedia/include/pjmedia-audiodev/audiodev_imp.h
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/401_fmtp_g7221_with_bitrate_24000.py
	  (added), res/pjproject/pjsip-apps/src/pjsua/pjsua_app.c (added),
	  res/pjproject/pjsip-apps/src/samples/stereotest.c (added),
	  res/pjproject/build.symbian/pjstun_client.mmp (added),
	  res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.cpp (added),
	  res/pjproject/pjsip-apps/src/ipjsua/Classes/FirstViewController.h
	  (added), res/pjproject/pjlib-util/lib (added),
	  res/pjproject/pjsip-apps/src/samples (added),
	  res/pjproject/pjsip-apps/src/ipjsua/Classes/FirstViewController.m
	  (added), res/pjproject/tests/pjsua/scripts-call/150_srtp_1_1.py
	  (added), res/pjproject/pjmedia/include/pjmedia/vid_stream.h
	  (added), res/pjproject/pjsip/src/pjsip/sip_dialog.c (added),
	  res/pjproject/pjlib/include/pj/compat/cc_armcc.h (added),
	  res/pjproject/third_party/build/speex/speex (added),
	  res/pjproject/third_party/bin (added),
	  res/pjproject/pjsip/build/Makefile (added),
	  res/pjproject/pjlib-util/include/pjlib-util/stun_simple.h
	  (added), res/pjproject/pjsip/src/pjsip/sip_util_proxy_wrap.cpp
	  (added), res/pjproject/pjlib/include/pj/compat/m_m68k.h (added),
	  res/pjproject/third_party/srtp/srtp.def (added),
	  res/pjproject/pjlib/src/pjlib-test/rand.c (added),
	  res/pjproject/third_party/build/gsm/config.h (added),
	  res/pjproject/pjmedia/include/pjmedia/avi.h (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uac-bad-ack.xml (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_gsm.py (added),
	  res/pjproject/pjsip/src/pjsip-ua/sip_reg.c (added),
	  res/pjproject/pjsip/build/wince-evc4/pjsip_ua_wince.vcp (added),
	  res/pjproject/pjsip/include/pjsip-ua/sip_regc.h (added),
	  res/pjproject/tests/pjsua/mod_pesq.py (added),
	  res/pjproject/pjnath/src/pjnath/ice_session.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/scanner.c (added),
	  res/pjproject/pjmedia/src/pjmedia-audiodev/audiodev.c (added),
	  res/pjproject/pjsip-apps/src/confbot/confbot.py (added),
	  res/pjproject/tests/pjsua/scripts-call/150_srtp_0_3.py (added),
	  res/pjproject/pjsip-apps/src/3rdparty_media_sample/alt_pjsua_vid.c
	  (added), res/pjproject/tests/pjsua/tools/cmp_wav.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/320_srtp_with_unknown_media_2.py
	  (added), res/pjproject/pjsip-apps/src/symbian_ua (added),
	  res/pjproject/pjmedia/src/pjmedia-audiodev/alsa_dev.c (added),
	  res/pjproject/third_party/portaudio/build/msvc (added),
	  res/pjproject/pjmedia/src/pjmedia/sound_legacy.c (added),
	  res/pjproject/third_party/ilbc/lsf.c (added),
	  res/pjproject/pjsip/src/test/inv_offer_answer_test.c (added),
	  res/pjproject/pjsip-apps/src/confbot (added),
	  res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.c
	  (added), res/pjproject/third_party/speex/libspeex/ltp_bfin.h
	  (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/group/ABLD.BAT
	  (added), res/pjproject/pjlib/src/pj/ioqueue_winnt.c (added),
	  res/pjproject/third_party/ilbc/lsf.h (added),
	  res/pjproject/third_party/speex/libspeex/lsp_tables_nb.c (added),
	  res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.h
	  (added),
	  res/pjproject/third_party/portaudio/build/scons/SConscript_common
	  (added), res/pjproject/pjmedia/include/pjmedia/frame.h (added),
	  res/pjproject/pjmedia/src/pjmedia-codec/audio_codecs.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/xml_wrap.cpp (added),
	  res/pjproject/pjsip-apps/src/pocketpj/res/PocketPJ.rc2 (added),
	  res/pjproject/third_party/build/portaudio/src/pa_mac_core_utilities.c
	  (added),
	  res/pjproject/pjmedia/include/pjmedia-audiodev/audiotest.h
	  (added), res/pjproject/pjlib/src/pj/guid_win32.c (added),
	  res/pjproject/pjlib/build/os-sunos.mak (added),
	  res/pjproject/third_party/build/srtp/Makefile (added),
	  res/pjproject/third_party/speex/libspeex/gain_table.c (added),
	  res/pjproject/third_party/build/portaudio/src/pa_mac_core_utilities.h
	  (added), res/pjproject/third_party/BaseClasses/wxlist.h (added),
	  res/pjproject/tests/pjsua/scripts-sendto/122_sdp_with_unknown_dynamic_1.py
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_5.py
	  (added), res/pjproject/pjsip-apps/src/pjsua/gui.h (added),
	  res/pjproject/third_party/srtp/crypto/test/auth_driver.c (added),
	  res/pjproject/pjlib/include/pj/activesock.h (added),
	  res/pjproject/pjlib/src/pjlib-test/exception.c (added),
	  res/pjproject/pjlib/src/pjlib-test/main_rtems.c (added),
	  res/pjproject/third_party/build/portaudio/src/pa_linux_alsa.c
	  (added), res/pjproject/pjlib-util/src/pjlib-util/symbols.c
	  (added), res/pjproject/pjlib/include/pj/types.h (added),
	  res/pjproject/pjnath/src/pjnath/turn_sock.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/resolver_wrap.cpp
	  (added),
	  res/pjproject/third_party/build/portaudio/src/pa_linux_alsa.h
	  (added), res/pjproject/pjlib/include/pj/compat/errno.h (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/100_simple.py (added),
	  res/pjproject/pjsip-apps/src/ipjsystest/RootViewController.xib
	  (added), res/pjproject/pjlib/build/wince-evc4/output (added),
	  res/pjproject/pjlib/src/pjlib-test/echo_clt.c (added),
	  res/pjproject/third_party/portaudio/src/os/unix/pa_unix_util.c
	  (added), res/pjproject/pjsip/build/wince-evc4/pjsua_lib_wince.vcp
	  (added), res/pjproject/svn_add (added),
	  res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_old.c
	  (added), res/pjproject/pjlib-util/build/wince-evc4 (added),
	  res/pjproject/pjmedia/src/test (added),
	  res/pjproject/third_party/srtp/crypto/test (added),
	  res/pjproject/third_party/portaudio/src/os/unix/pa_unix_util.h
	  (added), res/pjproject/tests/pjsua/scripts-media-playrec (added),
	  res/pjproject/pjsip-apps/src/samples/vid_streamutil.c (added),
	  res/pjproject/pkgconfig.py (added),
	  res/pjproject/third_party/srtp/crypto/hash/sha1.c (added),
	  res/pjproject/pjlib/src/pj/addr_resolv_sock.c (added),
	  res/pjproject/pjnath/src/pjturn-srv (added),
	  res/pjproject/pjmedia/include/pjmedia/wav_playlist.h (added),
	  res/pjproject/pjsip/include/pjsip/sip_resolve.h (added),
	  res/pjproject/pjmedia/src/pjmedia-codec/ilbc.c (added),
	  res/pjproject/pjmedia/src/pjmedia/format.c (added),
	  res/pjproject/pjsip/src/pjsip/sip_dialog_wrap.cpp (added),
	  res/pjproject/third_party/speex/include/speex/speex_buffer.h
	  (added),
	  res/pjproject/pjmedia/src/pjmedia/transport_adapter_sample.c
	  (added), res/pjproject/pjsip-apps/src/vidgui/vidwin.h (added),
	  res/pjproject/pjlib/src/pjlib-test/main_symbian.cpp (added),
	  res/pjproject/pjlib/docs/doxygen.css (added),
	  res/pjproject/third_party/gsm/src/gsm_explode.c (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiContainer.hrh
	  (added), res/pjproject/third_party/speex/libspeex/kiss_fftr.c
	  (added), res/pjproject/pjlib/src/pj/addr_resolv_linux_kernel.c
	  (added), res/pjproject/third_party/gsm/tst/lin2cod.c (added),
	  res/pjproject/pjmedia/src/pjmedia-codec/l16.c (added),
	  res/pjproject/third_party/speex/libspeex/kiss_fftr.h (added),
	  res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.h (added),
	  res/pjproject/third_party/resample/src/stddefs.h (added),
	  res/pjproject/pjmedia/src/pjmedia/rtcp_xr.c (added),
	  res/pjproject/pjsip-apps/src/vidgui/vidgui.cpp (added),
	  res/pjproject/pjsip/src/pjsip/sip_resolve.c (added),
	  res/pjproject/pjsip/src/test/transport_tcp_test.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/121_sdp_with_video_static_2.py
	  (added), res/pjproject/build.symbian/libpassthroughcodec.mmp
	  (added), res/pjproject/third_party/srtp/crypto/rng/ctr_prng.c
	  (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-subscribe-notify-terminate.xml
	  (added), res/pjproject/third_party/portaudio/fixfile.bat (added),
	  res/pjproject/pjsip/src/test/multipart_test.c (added),
	  res/pjproject/pjsip-apps/lib (added),
	  res/pjproject/third_party/portaudio/pablio/pablio.c (added),
	  res/pjproject/pjmedia/src/pjmedia/rtp.c (added),
	  res/pjproject/pjmedia/src/pjmedia/stereo_port.c (added),
	  res/pjproject/pjsip/src/test/tsx_uas_test.c (added),
	  res/pjproject/third_party/portaudio/pablio/pablio.h (added),
	  res/pjproject/third_party/speex/libspeex/vq_bfin.h (added),
	  res/pjproject/pjmedia/include/pjmedia/bidirectional.h (added),
	  res/pjproject/third_party/BaseClasses/arithutil.cpp (added),
	  res/pjproject/third_party/build/milenage/output (added),
	  res/pjproject/pjlib-util/src/pjlib-util/http_client.c (added),
	  res/pjproject/third_party/srtp/crypto/hash/hmac.c (added),
	  res/pjproject/third_party/speex/libspeex/quant_lsp.c (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-forked-200.xml
	  (added), res/pjproject/pjmedia/src/pjmedia/mem_player.c (added),
	  res/pjproject/third_party/speex/libspeex/quant_lsp.h (added),
	  res/pjproject/third_party/speex/libspeex/filters_bfin.h (added),
	  res/pjproject/pjsip-apps/src/samples/confbench.c (added),
	  res/pjproject/pjmedia/src/pjmedia/resample_resample.c (added),
	  res/pjproject/third_party/build/g7221/output (added),
	  res/pjproject/pjsip/src/test/regc_test.c (added), res/pjproject
	  (added),
	  res/pjproject/pjmedia/include/pjmedia-videodev/videodev.h
	  (added), res/pjproject/pjsip-apps/build/Samples-vc.mak (added),
	  res/pjproject/pjlib/include/pj++/tree.hpp (added),
	  res/pjproject/pjmedia/src/pjmedia/g711.c (added),
	  res/pjproject/pjlib/include/pj/guid.h (added),
	  res/pjproject/pjlib/include/pj/compat/cc_codew.h (added),
	  res/pjproject/pjmedia/src/pjmedia/echo_port.c (added),
	  res/pjproject/pjlib/src/pj/activesock.c (added),
	  res/pjproject/third_party/BaseClasses/msgthrd.h (added),
	  res/pjproject/pjmedia/bin (added),
	  res/pjproject/third_party/portaudio/build/dev-cpp/Makefile-dll
	  (added), res/pjproject/pjmedia/src/test/main.c (added),
	  res/pjproject/pjsip-apps/src/samples/invtester.c (added),
	  res/pjproject/pjmedia/include/pjmedia/avi_stream.h (added),
	  res/pjproject/pjsip/src/test/tsx_bench.c (added),
	  res/pjproject/third_party/speex/libspeex/testdenoise.c (added),
	  res/pjproject/third_party/portaudio/src/hostapi/dsound/pa_win_ds_dynlink.c
	  (added),
	  res/pjproject/third_party/portaudio/src/hostapi/dsound/pa_win_ds_dynlink.h
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	  res/pjproject/third_party/speex/libspeex/fixed_bfin.h (added),
	  res/pjproject/third_party/speex/include/speex/speex_config_types.h.in
	  (added),
	  res/pjproject/third_party/speex/libspeex/exc_5_256_table.c
	  (added), res/pjproject/third_party/speex/libspeex/testresample.c
	  (added), res/pjproject/pjnath/src/pjturn-srv/listener_udp.c
	  (added), res/pjproject/pjsip/build/wince-evc4/output (added),
	  res/pjproject/pjsip-apps/src/samples/aectest.c (added),
	  res/pjproject/pjlib/src/pjlib-test/thread.c (added),
	  res/pjproject/pjlib/include/pj (added),
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	  (added), res/pjproject/third_party/build/ilbc (added),
	  res/pjproject/third_party/srtp/config_in.h (added),
	  res/pjproject/third_party/speex/libspeex/nb_celp.c (added),
	  res/pjproject/third_party/speex/libspeex/nb_celp.h (added),
	  res/pjproject/third_party/srtp/crypto/kernel (added),
	  res/pjproject/tests/pjsua/scripts-sendto/124_sdp_with_unknown_static_unknown_transport.py
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/310_srtp1_no_crypto.py
	  (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/202_reg_good_ok_wildcard.py
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	  (added), res/pjproject/tests/pjsua/mod_sendto.py (added),
	  res/pjproject/build/cc-auto.mak.in (added),
	  res/pjproject/pjmedia/build/output (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-auth.xml (added),
	  res/pjproject/pjlib/src/pjlib-test/errno.c (added),
	  res/pjproject/tests/automated/gnu-ipp.xml.template (added),
	  res/pjproject/pjmedia/include/pjmedia-audiodev/errno.h (added),
	  res/pjproject/build.symbian/pjsip_simpleU.def (added),
	  res/pjproject/pjsip/include/pjsip/sip_autoconf.h.in (added),
	  res/pjproject/third_party/speex/libspeex/fixed_debug.h (added),
	  res/pjproject/third_party/portaudio/depcomp (added),
	  res/pjproject/pjnath/src/pjturn-srv/auth.c (added),
	  res/pjproject/pjlib/src/pj/pool.c (added),
	  res/pjproject/pjlib-util/include/pjlib-util/sha1.h (added),
	  res/pjproject/pjnath/src/pjturn-srv/auth.h (added),
	  res/pjproject/pjmedia/src/pjmedia/stream_common.c (added),
	  res/pjproject/pjlib/include/pj/compat/m_auto.h.in (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000.py
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	  res/pjproject/pjlib/build/os-auto.mak.in (added),
	  res/pjproject/tests/cdash/cfg_msvc.py (added),
	  res/pjproject/third_party/gsm/src/lpc.c (added),
	  res/pjproject/third_party/resample/README.resample (added),
	  res/pjproject/pjsip-apps/src/ipjsua/MainWindow.xib (added),
	  res/pjproject/third_party/portaudio/include/pa_win_wmme.h
	  (added), res/pjproject/third_party/speex/symbian/config.h
	  (added), res/pjproject/pjnath/src/pjnath-test/test.c (added),
	  res/pjproject/pjsip-apps/src/samples/siprtp_report.c (added),
	  res/pjproject/pjnath/src/pjnath-test/test.h (added),
	  res/pjproject/third_party/srtp (added),
	  res/pjproject/third_party/build/g7221/libg7221codec.vcproj
	  (added),
	  res/pjproject/pjlib-util/build/wince-evc4/pjlib_util_test_wince.vcp
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	  res/pjproject/pjlib-util/include/pjlib-util/srv_resolver.h
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	  (added), res/pjproject/pjsip/include/pjsip_simple.h (added),
	  res/pjproject/pjmedia/src/test/audio_tool.c (added),
	  res/pjproject/pjlib/src/pj/exception_symbian.cpp (added),
	  res/pjproject/pjmedia/build/m-i386.mak (added),
	  res/pjproject/third_party/BaseClasses/wxutil.h (added),
	  res/pjproject/pjsip-apps/src/vidgui (added),
	  res/pjproject/pjsip/src/pjsua-lib/pjsua_media.c (added),
	  res/pjproject/pjlib-util/build/pjlib_util.vcproj (added),
	  res/pjproject/pjnath/include/pjnath/stun_transaction.h (added),
	  res/pjproject/third_party/portaudio/src/hostapi/oss/recplay.c
	  (added), res/pjproject/third_party/resample/include (added),
	  res/pjproject/pjmedia/include/pjmedia/transport.h (added),
	  res/pjproject/pjlib-util/src/pjlib-util/srv_resolver.c (added),
	  res/pjproject/build.symbian/pjsua_libU.def (added),
	  res/pjproject/pjsip/src/pjsip-simple/presence_body.c (added),
	  res/pjproject/pjmedia/include/pjmedia/stereo.h (added),
	  res/pjproject/tests/pjsua/scripts-call/301_ice_public_b.py
	  (added), res/pjproject/tests/automated/configure.py (added),
	  res/pjproject/pjsip-apps/src/symbian_ua/symbian_ua_reg.rss
	  (added), res/pjproject/pjsip-apps/src/pocketpj/PopUpWnd.h
	  (added),
	  res/pjproject/third_party/speex/libspeex/high_lsp_tables.c
	  (added), res/pjproject/pjlib/src/pj/ssl_sock_ossl.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/313_srtp1_unsupported_crypto.py
	  (added), res/pjproject/pjsip-apps/src/symbian_ua_gui (added),
	  res/pjproject/third_party/build/milenage/libmilenage.vcp (added),
	  res/pjproject/pjmedia/include/pjmedia/transport_loop.h (added),
	  res/pjproject/third_party/build/gsm/libgsmcodec.vcp (added),
	  res/pjproject/third_party/speex/libspeex/window.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/125_sdp_with_multi_audio_2.py
	  (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/data
	  (added), res/pjproject/pjsip/src/pjsip/sip_transport_wrap.cpp
	  (added), res/pjproject/pjmedia/include/pjmedia-videodev/errno.h
	  (added), res/pjproject/pjlib/src/pj/os_time_common.c (added),
	  res/pjproject/third_party/resample/src (added),
	  res/pjproject/pjlib/docs (added),
	  res/pjproject/tests/pjsua/scripts-sendto/161_err_replaces_dlg_not_found.py
	  (added), res/pjproject/pjsip-apps/src/pocketpj (added),
	  res/pjproject/pjsip-apps/src/samples/simple_pjsua.c (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/src (added),
	  res/pjproject/tests/pjsua/scripts-media-playrec/100_resample_lf_11_48.py
	  (added), res/pjproject/pjmedia/include/pjmedia/rtcp.h (added),
	  res/pjproject/tests/pjsua/scripts-sendto/300_srtp_invalid_crypto_tag_non_numeric.py
	  (added), res/pjproject/tests/pjsua/scripts-pres/100_peertopeer.py
	  (added), res/pjproject/pjmedia/src/pjmedia/vid_codec_util.c
	  (added), res/pjproject/third_party/gsm/MACHINES (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uac-subscribe.xml (added),
	  res/pjproject/third_party/build/baseclasses (added),
	  res/pjproject/third_party/srtp/include/srtp.h (added),
	  res/pjproject/tests/pjsua/scripts-sendto/173_timer_offer_refresher_uac.py
	  (added), res/pjproject/pjmedia/src/pjmedia/stream.c (added),
	  res/pjproject/tests/pjsua/scripts-recvfrom/209a_reg_handle_423_ok.py
	  (added), res/pjproject/pjlib/src/pjlib-samples/log.c (added),
	  res/pjproject/third_party/build/portaudio/src/pa_mac_core_old.c
	  (added), res/pjproject/pjsip/src/pjsip/sip_transport_tcp.c
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/150_err_extension.py
	  (added),
	  res/pjproject/pjlib-util/src/pjlib-util-test/encryption.c
	  (added), res/pjproject/lib (added),
	  res/pjproject/pjmedia/include/pjmedia/codec.h (added),
	  res/pjproject/pjmedia/src/pjmedia/converter_libswscale.c (added),
	  res/pjproject/pjlib/src/pj/ip_helper_win32.c (added),
	  res/pjproject/pjmedia/include/pjmedia-videodev/avi_dev.h (added),
	  res/pjproject/pjlib-util/src/pjlib-util/scanner_cis_bitwise.c
	  (added), res/pjproject/third_party/gsm/README (added),
	  res/pjproject/pjlib-util/src/pjlib-util (added),
	  res/pjproject/third_party/build/gsm (added),
	  res/pjproject/pjlib/include/pj/compat/cc_msvc.h (added),
	  res/pjproject/pjsip-apps/src/pjsua_wince (added),
	  res/pjproject/tests/pjsua (added),
	  res/pjproject/pjlib/include/pj++/timer.hpp (added),
	  res/pjproject/build.symbian/pjlib.mmp (added),
	  res/pjproject/pjsip/src/test/test.c (added),
	  res/pjproject/third_party/portaudio/build (added),
	  res/pjproject/pjsip/src/test/test.h (added),
	  res/pjproject/pjsip/include/pjsip_auth.h (added),
	  res/pjproject/pjlib/src/pj/errno.c (added),
	  res/pjproject/third_party/BaseClasses/wxdebug.cpp (added),
	  res/pjproject/pjsip/include/pjsip-simple/rpid.h (added),
	  res/pjproject/pjlib/include/pj/compat/os_sunos.h (added),
	  res/pjproject/third_party/portaudio/install-sh (added),
	  res/pjproject/pjlib/src/pj/os_info.c (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-no-media.xml
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/172_timer_supported_but_not_used.py
	  (added),
	  res/pjproject/third_party/build/resample/libresample_dll.vcproj
	  (added), res/pjproject/pjmedia/include (added),
	  res/pjproject/third_party/portaudio/src/hostapi/asio/ASIO-README.txt
	  (added), res/pjproject/pjsip-apps/src/python/samples/presence.py
	  (added),
	  res/pjproject/build/vs/pjproject-vs8-debug-static-defaults.vsprops
	  (added), res/pjproject/pjmedia/src/pjmedia/transport_srtp.c
	  (added),
	  res/pjproject/pjmedia/include/pjmedia-codec/amr_sdp_match.h
	  (added), res/pjproject/pjsip/src/pjsip-simple/rpid.c (added),
	  res/pjproject/pjlib-util/src/pjlib-util/dns_server.c (added),
	  res/pjproject/tests/pjsua/runall.py (added),
	  res/pjproject/pjlib/include/pj/compat/m_armv4.h (added),
	  res/pjproject/pjsip/src/pjsip/sip_util_proxy.c (added),
	  res/pjproject/pjlib-util/include/pjlib-util/crc32.h (added),
	  res/pjproject/pjlib-util/build/os-auto.mak.in (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-subscribe-multipart-notify.xml
	  (added), res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcp
	  (added), res/pjproject/pjmedia/include/pjmedia/sound.h (added),
	  res/pjproject/pjsip/build/output (added), res/pjproject/pjnath
	  (added), res/pjproject/INSTALL.txt (added),
	  res/pjproject/tests/pjsua/mod_call.py (added),
	  res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcw (added),
	  res/pjproject/pjsip/src/test/dlg_core_test.c (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_g711u.py
	  (added),
	  res/pjproject/tests/pjsua/scripts-sendto/411_fmtp_amrnb_offer_band_eff.py
	  (added), res/pjproject/third_party/build/resample/config.h
	  (added), res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.rc
	  (added), res/pjproject/pjlib/build/output (added),
	  res/pjproject/pjlib/include/pj/compat/m_powerpc.h (added),
	  res/pjproject/pjsip/src/test/msg_logger.c (added),
	  res/pjproject/pjsip-apps/src/pjsua_wince/resource.h (added),
	  res/pjproject/pjsip/src/pjsip/sip_auth_parser_wrap.cpp (added),
	  res/pjproject/aconfigure.ac (added),
	  res/pjproject/tests/pjsua/scripts-sendto/140_sdp_with_direction_attr_in_session_1.py
	  (added),
	  res/pjproject/pjsip-apps/src/pjsystest/pjsystest_wince.rc2
	  (added), res/pjproject/pjlib/include/pj/compat/os_win32.h
	  (added), res/pjproject/pjmedia/include/pjmedia/doxygen.h (added),
	  res/pjproject/pjsip/src/test/main_rtems.c (added),
	  res/pjproject/pjlib-util/include/pjlib-util/scanner_cis_bitwise.h
	  (added), res/pjproject/pjsip-apps/src/ipjsystest/main.m (added),
	  res/pjproject/build.symbian/pjsip.mmp (added),
	  res/pjproject/third_party/speex/include/speex/speex_jitter.h
	  (added), res/pjproject/tests/pjsua/run.py (added),
	  res/pjproject/third_party/speex/symbian (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000_stereo.py
	  (added), res/pjproject/pjsip-apps/src/samples/auddemo.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/300_srtp_crypto_case_insensitive.py
	  (added), res/pjproject/third_party/g7221/common/basic_op.c
	  (added),
	  res/pjproject/pjnath/build/wince-evc4/pjnath_test_wince.vcp
	  (added), res/pjproject/third_party/g7221/common/basic_op.h
	  (added), res/pjproject/third_party/portaudio/config.guess
	  (added), res/pjproject/third_party/portaudio/src/os/unix (added),
	  res/pjproject/third_party/speex/libspeex/cb_search_sse.h (added),
	  res/pjproject/tests/pjsua/tools/Makefile (added),
	  res/pjproject/pjlib/src/pj/compat/longjmp_i386.S (added),
	  res/pjproject/third_party/portaudio/pablio (added),
	  res/pjproject/build.symbian/symbian_ua_udeb.pkg (added),
	  res/pjproject/README.txt (added),
	  res/pjproject/third_party/srtp/srtp.vcproj (added),
	  res/pjproject/pjnath/build (added),
	  res/pjproject/third_party/portaudio/src/hostapi/dsound (added),
	  res/pjproject/tests/automated/prepare.xml.template (added),
	  res/pjproject/pjsip/src/pjsua-lib/pjsua_pres.c (added),
	  res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-and-ack(same-branch)-without-sdp.xml
	  (added), res/pjproject/pjlib/build (added),
	  res/pjproject/third_party/build/baseclasses/libbaseclasses.vcproj
	  (added),
	  res/pjproject/third_party/speex/include/speex/speex_preprocess.h
	  (added), res/pjproject/pjlib/src/pjlib-test (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/data/symbian_ua_gui.l01
	  (added), res/pjproject/pjlib/build/privkey.pem (added),
	  res/pjproject/pjmedia/src/pjmedia/alaw_ulaw_table.c (added),
	  res/pjproject/configure-legacy (added),
	  res/pjproject/tests/pjsua/scripts-sendto/200_ice_success_1.py
	  (added), res/pjproject/pjsip/include/pjsip/sip_transport.h
	  (added), res/pjproject/pjnath/src/pjturn-srv/server.c (added),
	  res/pjproject/pjmedia/build/os-linux.mak (added),
	  res/pjproject/pjlib/include/pj/compat/os_win32_wince.h (added),
	  res/pjproject/pjsip/src/pjsip-ua/sip_replaces.c (added),
	  res/pjproject/third_party/portaudio/src/common/pa_util.h (added),
	  res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiDocument.h
	  (added), res/pjproject/pjlib/src/pj/fifobuf.c (added),
	  res/pjproject/third_party/gsm/tls/sour1.dta (added),
	  res/pjproject/pjsip/include/pjsip/sip_types.h (added),
	  res/pjproject/pjlib/include/pj/compat/time.h (added),
	  res/pjproject/pjsip/src/pjsip/sip_auth_msg.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_1.py
	  (added), res/pjproject/pjsip/include/pjsip_ua.h (added),
	  res/pjproject/pjlib/build/Makefile (added),
	  res/pjproject/third_party/srtp/README (added),
	  res/pjproject/tests/pjsua/scripts-sendto/311_srtp1_recv_avp.py
	  (added), res/pjproject/pjsip-apps/src/pjsua/main_rtems.c (added),
	  res/pjproject/pjsip-apps/src/pocketpj/res/invisibl.bmp (added),
	  res/pjproject/pjlib/src/pjlib-test/rtems_network_config.h
	  (added), res/pjproject/third_party/srtp/crypto/math/stat.c
	  (added), res/pjproject/third_party/srtp/test/replay_driver.c
	  (added), res/pjproject/pjmedia/src/pjmedia-audiodev/audiotest.c
	  (added), res/pjproject/pjlib/src/pjlib++-test (added),
	  res/pjproject/pjsip-apps/src/samples/streamutil.c (added),
	  res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.c (added),
	  res/pjproject/tests/pjsua/scripts-sendto/500_pres_subscribe_with_bad_event.py
	  (added), res/pjproject/third_party/srtp/install-sh (added),
	  res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_16000.py
	  (added),
	  res/pjproject/third_party/srtp/crypto/cipher/null_cipher.c
	  (added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.h (added),
	  res/pjproject/pjlib-util/src (added),
	  res/pjproject/pjsip/include/pjsip/sip_config.h (added),
	  res/pjproject/pjlib/docs/doxygen.cfg (added): Add support for
	  ICE/STUN/TURN in res_rtp_asterisk and chan_sip. Review:
	  https://reviewboard.asterisk.org/r/1891/

2012-06-29 20:32 +0000 [r369512]  Mark Michelson <mmichelson@digium.com>

	* main/rtp_engine.c, /: Fix apparent copy and paste error where
	  incorrect "glue" is used. ........ Merged revisions 369511 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-29 17:02 +0000 [r369493]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, main/channel.c, main/autoservice.c, main/pbx.c,
	  channels/chan_local.c, funcs/func_channel.c,
	  main/channel_internal_api.c, main/features.c,
	  configs/cdr.conf.sample, include/asterisk/channel.h,
	  include/asterisk/pbx.h, CHANGES, apps/app_followme.c,
	  apps/app_queue.c: Hangup handlers - Dialplan subroutines that run
	  when the channel hangs up. Hangup handlers are an alternative to
	  the h extension. They can be used in addition to the h extension.
	  The idea is to attach a Gosub routine to a channel that will
	  execute when the call hangs up. Whereas which h extension gets
	  executed depends on the location of dialplan execution when the
	  call hangs up, hangup handlers are attached to the call channel.
	  You can attach multiple handlers that will execute in the order
	  of most recently added first. (closes issue ASTERISK-19549)
	  Reported by: Mark Murawski Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2002/

2012-06-29 16:56 +0000 [r369492]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: With some configurations a transport is
	  not actually specified so assume UDP in these cases. ........
	  Merged revisions 369490 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369491 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-29 16:42 +0000 [r369489]  Richard Mudgett <rmudgett@digium.com>

	* main/channel_internal_api.c, .cleancount: Remove obsolete struct
	  ast_channel note. The opaquing the ast_channel struct no longer
	  requires .cleancount to be changed when the struct is changed. *
	  Bump .cleancount value one last time because of struct
	  ast_channel for old times sake.

2012-06-29 15:33 +0000 [r369473]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Make the address family filter specific
	  to the transport. (closes issue ASTERISK-16618) Reported by: Leif
	  Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
	  Merged revisions 369471 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369472 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-28 01:12 +0000 [r369449-369454]  Terry Wilson <twilson@digium.com>

	* include/asterisk/config_options.h,
	  configs/config_test.conf.sample, main/config_options.c,
	  tests/test_config.c: Add the ability to set flags via the config
	  options api Allows the setting of flags via the config options
	  api. For example, code like this: #define OPT1 1 << 0 #define
	  OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int
	  flags; }; and a config like this: [blah] opt1=yes opt2=no
	  opt3=yes Review: https://reviewboard.asterisk.org/r/2004/

	* /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010:
	  Clean up after a reinvite that never gets a final response The
	  basic problem is that if a re-INVITE is sent by Asterisk and it
	  receives a provisional response, but no final response, then the
	  dialog is never torn down. In addition to leaking memory, this
	  also leaks file descriptors and will eventually lead to Asterisk
	  no longer being able to process calls. This patch just keeps
	  track of whether there is an outstanding re-INVITE, and if there
	  is goes ahead and cleans up everything as though there was no
	  outstanding reinvite. Review:
	  https://reviewboard.asterisk.org/r/2009/ (closes issue
	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
	  Davies, Terry Wilson ........ Merged revisions 369436 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369437 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-26 21:45 +0000 [r369414]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/logger.h, channels/chan_dahdi.c,
	  main/autoservice.c, main/pbx.c, channels/chan_local.c,
	  channels/sig_analog.c, main/channel_internal_api.c,
	  channels/chan_agent.c, main/features.c, main/logger.c,
	  channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
	  main/bridging.c, main/cli.c: Unique Call ID logging Phases III
	  and IV Adds call ID logging changes to specific channel drivers
	  that weren't handled handled in phase II of Call ID Logging. Also
	  covers logging for threads for threads created by systems that
	  may be involved with many different calls. Extra special thanks
	  to Richard for rigorous review of chan_dahdi and its various
	  signalling modules. review:
	  https://reviewboard.asterisk.org/r/1927/ review:
	  https://reviewboard.asterisk.org/r/1950/

2012-06-26 13:23 +0000 [r369370-369392]  Matthew Jordan <mjordan@digium.com>

	* /, main/adsi.c: Fix crash in unloading of res_adsi module When
	  res_adsi is unloaded, it removes the ADSI functions that it
	  previously installed by passing a NULL adsi_funcs pointer to
	  ast_adsi_install_funcs. This function was not checking whether or
	  not the adsi_funcs pointer passed in was NULL before
	  dereferencing it to check whether or not the version of the
	  functions matches what the core was expecting it. This patch
	  makes it so that the version is only checked if a potentially
	  valid adsi_funcs pointer was passed in. Passing in NULL removes
	  the installed functions, bypassing the version check. ........
	  Merged revisions 369390 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369391 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/manager.c: Update "manager show event" to support tab
	  completion Thank you rmudgett for pointing out that I was missing
	  this in the initial check-in for AMI event documentation
	  (r369346)

	* main/cdr.c, /: Fix incorrect duration reporting in CDRs created
	  in batch mode Certain places in core/cdr.c would, if the duration
	  value were 0, calculate the duration as being the delta between
	  the current time and the time at which the CDR record was
	  started. While this does not typically cause a problem in
	  non-batch mode, this can cause an issue in batch mode where CDR
	  records are gathered and written long after those calls have
	  ended. In particular, this affects calls that were never
	  answered, as those are expected to have a duration of 0. Often,
	  this would result in CDR logs with a significant number of calls
	  with lengthy durations, but dispositions of "BUSY". Note that
	  this does not affect cdr_csv, as that backend does not use
	  ast_cdr_getvar and instead directly reports the duration value.
	  The affected core backends include cdr_apative_odbc and
	  cdr_custom; other extended or deprecated CDR backends may
	  potentially still directly manipulate the duration values. (issue
	  ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
	  Reported by: Thomas Arimont Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1996/ ........ Merged
	  revisions 369351 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369369 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-25 19:26 +0000 [r369367]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how
	  local tag is generated when sending a 481 to an INVITE. Match our
	  local tag to whatever to-tag was sent in the initial INVITE.
	  Because the size of the to-tag may not fit in the buffer in the
	  sip_pvt, it has been changed to a string field. (closes issue
	  ASTERISK-19892) reported by Walter Doekes Review:
	  https://reviewboard.asterisk.org/r/1977 ........ Merged revisions
	  369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369353 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-25 17:59 +0000 [r369346]  Matthew Jordan <mjordan@digium.com>

	* apps/app_dial.c, apps/app_meetme.c, configure.ac,
	  apps/app_userevent.c, CHANGES, apps/app_queue.c, Makefile,
	  build_tools/get_documentation.py (added), main/manager.c,
	  configure, build_tools/post_process_documentation.py (added),
	  include/asterisk/xmldoc.h, apps/app_confbridge.c, makeopts.in,
	  apps/app_stack.c, apps/app_chanspy.c, doc/appdocsxml.dtd,
	  main/xmldoc.c, apps/app_voicemail.c: Add AMI event documentation
	  This patch adds the core changes necessary to support AMI event
	  documentation in the source files of Asterisk, and adds
	  documentation to those AMI events defined in the core application
	  modules. Event documentation is built from the source by two new
	  python scripts, located in build_tools: get_documentation.py and
	  post_process_documentation.py. The get_documentation.py script
	  mirrors the actions of the existing AWK get_documentation
	  scripts, except that it will scan the entirety of a source file
	  for Asterisk documentation. Upon encountering it, if the
	  documentation happens to be an AMI event, it will attempt to
	  extract information about the event directly from the manager
	  event macro calls that raise the event. The
	  post_process_documentation.py script combines manager event
	  instances that are the same event but documented in multiple
	  source files. It generates the final core-[lang].xml file. As
	  this process can take longer to complete than a typical 'make
	  all', it is only performed if a new make target, 'full', is
	  chosen. Review: https://reviewboard.asterisk.org/r/1967/

2012-06-25 16:07 +0000 [r369329]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix Bridge application occasionally returning
	  to the wrong location. * Fix do_bridge_masquerade() getting the
	  resume location from the zombie channel. The code must not touch
	  a clone channel after it has masqueraded it. The clone channel
	  has become a zombie and is starting to hangup. (closes issue
	  ASTERISK-19985) Reported by: jamicque Patches:
	  jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: jamicque ........ Merged revisions 369327
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 369328 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-25 15:55 +0000 [r369304-369326]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c,
	  main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple
	  revisions 369323-369324 ........ r369323 | mmichelson |
	  2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate
	  embedding of res_adsi.so module. The way this is done is to stop
	  using the optional API. Instead, res_adsi.so, when loaded fills
	  in a table of function pointers. Review:
	  https://reviewboard.asterisk.org/r/1991 ........ r369324 |
	  mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
	  lines Forgot to svn add this file in my last commit. ........
	  Merged revisions 369323-369324 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369325 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Be more consistent with the return code
	  for requests received from invalid domain. When Asterisk receives
	  an INVITE from an external domain when allowexternaldomains=no
	  send a 403 instead of a 404. This is consistent with Asterisk's
	  behavior when receiving a REGISTER in this situation. (Closes
	  issue ASTERISK-19601) Reported by Matthew Jordan Patches:
	  ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
	  #5049) ........ Merged revisions 369302 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369303 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-23 00:33 +0000 [r369237-369296]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix F and F(x) action logic in Bridge
	  application.

	* /, main/features.c: Fix Bridge application and AMI Bridge action
	  error handling. * Fix AMI Bridge action disconnecting the AMI
	  link on error. * Fix AMI Bridge action and Bridge application not
	  checking if their masquerades were successful. * Fix Bridge
	  application running the h-exten when it should not. * Made
	  do_bridge_masquerade() return if the masquerade was successful so
	  the Bridge application and AMI Bridge action could deal with it
	  correctly. * Made bridge_call_thread_launch() hangup the passed
	  in channels if the bridge_call_thread fails to start. Those
	  channels would have been orphaned. * Made builtin_atxfer() check
	  the success of the transfer masquerade setup. ........ Merged
	  revisions 369282 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369283 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Explicitly check caller hangup in app Queue
	  rather than a polluted res2 value. ........ Merged revisions
	  369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369263 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_queue.c: Fix F and F(x) action logic in Queue
	  application.

	* apps/app_dial.c, /: Check if PBX was started and fix F and F(x)
	  action logic in Dial application. ........ Merged revisions
	  369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369259 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/ccss.c: Check if PBX was started for generic CCSS recall.
	  ........ Merged revisions 369238 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369239 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup
	  debug message. They are all zombies now. ........ Merged
	  revisions 369235 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369236 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-22 20:05 +0000 [r369217]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't crash on a guest directmedia call A
	  sip_pvt may not have relatedpeer set if a call doesn't match up
	  with a peer. If there is no relatedpeer, there is no direct media
	  ACL to apply, so just return that it is allowed. (closes issue
	  ASTERISK-20040) Reported by: Terry Wilson ........ Merged
	  revisions 369214 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369215 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-22 19:54 +0000 [r369184-369216]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_dahdi.c: Fix wrong variable name in the R2
	  disconnect callback

	* /, channels/chan_sip.c: Don't parse media stream state for SIP
	  video streams The sendonly/recvonly/sendrecv/inactive media
	  stream attributes were parsed for video, but nothing was ever
	  done with them. With this code removed, an UNSUPPORTED message is
	  produced when these attributes are used in conjunction with a
	  video stream which is the better behavior since they were never
	  really supported in the first place. ........ Merged revisions
	  369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 369206 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c: Add HANGUPCAUSE hash implementation for
	  DAHDI MFC/R2 subtech This adds a minimal implementation of the
	  "Who Hung Up?" Asterisk 11 work to chan_dahdi.c for the MFC/R2
	  DAHDI subtech. Given the way that OpenR2 interfaces with
	  chan_dahdi, it is much harder to expose the type of protocol
	  information that is available in PRI, SS7, or other channel
	  technologies.

	* channels/sig_analog.c, channels/sig_pri.c: Add HANGUPCAUSE hash
	  support for analog and PRI DAHDI subtechs This is part of the
	  DAHDI support for the Asterisk 11 "Who Hung Up?" project and
	  covers the implementation for the technologies implemented in
	  sig_analog.c and sig_pri.c. Tested on a local machine to verify
	  protocol and cause information is available. Review:
	  https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222)

	* channels/sig_ss7.c: Add "Who Hung Up?" implementation for DAHDI
	  SS7 subtechnology Testing was done on a local machine to verify
	  that protocol and cause information was being sent properly.
	  Review: https://reviewboard.asterisk.org/r/1955/ (issue SWP-4222)

2012-06-20 21:33 +0000 [r369166-369167]  Richard Mudgett <rmudgett@digium.com>

	* main/logger.c: Don't waste time initializing the whole
	  call_identifer_str[]. The array is either setup with a callid
	  string or only the first element needs to be initialized.

	* channels/chan_misdn.c: Fix chan_misdn compile error.

2012-06-20 17:48 +0000 [r369148]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
	  locking issue on empty callList (issue ASTERISK-19298) Reported
	  by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........
	  Merged revisions 369146 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369147 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-20 11:47 +0000 [r369142]  Sean Bright <sean@malleable.com>

	* apps/app_externalivr.c: Remove declaration of eivr_connect_socket
	  because it no longer exists.

2012-06-20 11:20 +0000 [r369141]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: use right definition for channel name

2012-06-20 03:18 +0000 [r369110-369126]  Michael L. Young <elgueromexicano@gmail.com>

	* main/manager.c, CHANGES: Add IPv6 Support To Manager This patch
	  adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported
	  by: Michael L. Young Tested by: Michael L. Young Patches:
	  ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026)
	  Review: https://reviewboard.asterisk.org/r/1968/

	* main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer
	  segfault in ast_sockaddr_parse() While working with
	  ast_parse_arg() to perform a validity check, a segfault occurred.
	  The segfault occurred due to passing a NULL pointer to
	  ast_sockaddr_parse() from ast_parse_arg(). According to the
	  documentation in config.h, "result pointer to the result. NULL is
	  valid here, and can be used to perform only the validity checks."
	  This patch fixes the segfault by checking for a NULL pointer.
	  This patch also adds documentation to netsock2.h about why it is
	  necessary to check for a NULL pointer. (Closes issue
	  ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
	  L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
	  by Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1990/ ........ Merged
	  revisions 369108 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369109 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-19 23:36 +0000 [r369092]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: check rtptimeouts in ooh323 channels as
	  per config file (rtp voice, video, udptl except rtcp) (closes
	  issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
	  19179-ooh323-ast10.patch ........ Merged revisions 369091 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-19 21:13 +0000 [r369086]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c,
	  main/rtp_engine.c, include/asterisk/channel.h,
	  channels/chan_iax2.c: Ensure that pvt cause information does not
	  break native bridging Channel drivers that allow native bridging
	  need to handle AST_CONTROL_PVT_CAUSE_CODE frames and previously
	  did not handle them properly, usually breaking out of the native
	  bridge. This change corrects that behavior and exposes the
	  available cause code information to the dialplan while native
	  bridges are in place. This required exposing the HANGUPCAUSE hash
	  setter outside of channel.c, so additional documentation has been
	  added.

2012-06-19 15:44 +0000 [r369068]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix request routing issue when
	  outboundproxy is used. Asterisk was incorrectly setting the
	  destination of CANCELs and ACKs for error responses to the URI of
	  the initial INVITE. This resulted in further requests, such as
	  INVITEs with authentication credentials, to be routed
	  incorrectly. Instead, when these CANCEL or ACKs are to be sent,
	  we should simply keep the destination the same as what it
	  previously was. There is no need to alter it any. (closes issue
	  ASTERISK-20008) Reported by Marcus Hunger Patches:
	  ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
	  ........ Merged revisions 369066 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369067 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-18 22:56 +0000 [r369061]  Kinsey Moore <kmoore@digium.com>

	* main/features.c: Fix AST_CONTROL_PVT_CAUSE_CODE handling When the
	  IAX2 Who Hung Up? changes were added, they uncovered a bug in the
	  way AST_CONTROL_PVT_CAUSE_CODE was handled in
	  feature_request_and_dial(). This particular frame subtype was
	  being treated like more terminal control frames causing the
	  function to be exited prematurely.

2012-06-18 18:25 +0000 [r369057]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix monitoring calls put in a parking lot. *
	  Fix a regression that was introduced by -r366167 which
	  effectively disabled monitoring parked calls. (closes issue
	  ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
	  ........ Merged revisions 369043 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 369044 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-15 21:18 +0000 [r369034]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Various small chan_skinny fixes and
	  cleanup Added test to skinny_register to only allow device to
	  register against a device that is not already registered. Addback
	  l->device test for skinny_show_lines. Fixes segfault if a line is
	  configured but not configured to a device. Reverses part of
	  r368680. Removed redundant l->device tests in subsubstate and
	  dumpsub. l->device will always be valid if these routines are
	  called. Reverses 368948 - discussed with mjordan on irc. Some
	  indentation cleanup.

2012-06-15 17:13 +0000 [r369028]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c, channels/sip/include/sip.h: Allow chan_sip
	  to decline unwanted media streams This change replaces the static
	  array of four representable media streams with an AST_LIST so
	  that chan_sip can keep track of offered media streams. This
	  allows chan_sip to deal with offers containing multiple same-type
	  streams and many other situations without rejecting the SDP offer
	  in its entirety, yet still generating a valid response. This also
	  covers cases where Asterisk can not comprehend the offer if it is
	  in the correct format. Previously, chan_sip would reject SDP
	  offers or entirely ignore individual stream offers in an effort
	  to be more compatible which would often result in invalid SDP
	  responses. Review: https://reviewboard.asterisk.org/r/1988/

2012-06-15 16:30 +0000 [r369027]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Fix voicemail API tests by using the
	  correct argument order for create/destroy. ........ Merged
	  revisions 369024 from
	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
	  ........ Merged revisions 369026 from
	  http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones

2012-06-15 16:20 +0000 [r369013]  Kevin P. Fleming <kpfleming@digium.com>

	* main/format.c, main/udptl.c, main/netsock2.c, main/autoservice.c,
	  main/rtp_engine.c, main/frame.c, main/security_events.c, /,
	  main/say.c, main/threadstorage.c, channels/console_video.c,
	  main/devicestate.c, main/astfd.c, main/taskprocessor.c,
	  main/format_pref.c, main/astobj2.c, main/indications.c,
	  main/config.c, main/loader.c, main/term.c,
	  apps/confbridge/conf_config_parser.c, main/cli.c,
	  channels/sig_analog.c, main/framehook.c, main/strcompat.c,
	  main/plc.c, main/fskmodem_int.c, main/syslog.c,
	  main/stdtime/localtime.c, main/bridging.c, main/db.c,
	  channels/sig_ss7.c, main/datastore.c, main/sched.c,
	  channels/sip/sdp_crypto.c, main/strings.c, main/pbx.c,
	  channels/vcodecs.c, channels/sip/security_events.c,
	  main/libasteriskssl.c, channels/iax2-provision.c,
	  pbx/dundi-parser.c, main/aoc.c, main/cel.c, utils/astdb2bdb.c,
	  channels/iax2-parser.c, main/chanvars.c, main/netsock.c,
	  build_tools/find_missing_support_level (added), main/data.c,
	  main/srv.c, channels/chan_misdn.c, main/privacy.c,
	  main/fixedjitterbuf.c, channels/sip/dialplan_functions.c,
	  main/test.c, main/audiohook.c, codecs/codec_dahdi.c, main/alaw.c,
	  main/asterisk.c, main/timing.c, main/global_datastores.c,
	  main/fskmodem_float.c, main/ccss.c,
	  channels/sip/reqresp_parser.c, main/xml.c,
	  channels/misdn/isdn_msg_parser.c, main/utils.c, main/autochan.c,
	  channels/misdn/isdn_lib.c, main/enum.c, main/presencestate.c,
	  main/fskmodem.c, channels/misdn_config.c, main/io.c,
	  main/channel.c, main/cdr.c, res/ael/pval.c, main/ulaw.c,
	  main/dial.c, main/format_cap.c, main/tdd.c,
	  channels/console_gui.c, main/heap.c, channels/misdn/ie.c,
	  main/logger.c, main/app.c, channels/console_board.c,
	  main/image.c, main/message.c, main/dns.c, main/lock.c,
	  main/stun.c, channels/sip/srtp.c, main/dnsmgr.c,
	  main/slinfactory.c, main/channel_internal_api.c,
	  main/translate.c, main/jitterbuf.c, main/acl.c,
	  utils/astdb2sqlite3.c, channels/sip/utils.c, channels/sig_pri.c,
	  apps/app_system.c, funcs/func_realtime.c, main/tcptls.c,
	  main/hashtab.c, funcs/func_presencestate.c,
	  apps/app_celgenuserevent.c, main/abstract_jb.c, main/callerid.c,
	  main/file.c, main/config_options.c, res/snmp/agent.c,
	  main/astmm.c, main/event.c, channels/misdn/portinfo.c,
	  channels/sip/config_parser.c, channels/vgrabbers.c, main/dsp.c,
	  main/xmldoc.c: Multiple revisions 369001-369002 ........ r369001
	  | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11
	  lines Add support-level indications to many more source files.
	  Since we now have tools that scan through the source tree looking
	  for files with specific support levels, we need to ensure that
	  every file that is a component of a 'core' or 'extended' module
	  (or the main Asterisk binary) is explicitly marked with its
	  support level. This patch adds support-level indications to many
	  more source files in tree, but avoids adding them to third-party
	  libraries that are included in the tree and to source files that
	  don't end up involved in Asterisk itself. ........ r369002 |
	  kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3
	  lines Add a script to enable finding source files without
	  support-levels defined. ........ Merged revisions 369001-369002
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 369005 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-15 16:17 +0000 [r369007]  Kinsey Moore <kmoore@digium.com>

	* main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Add
	  HANGUPCAUSE hash support to IAX2 Continuing with the Who Hung Up?
	  project for Asterisk 11, this adds support to IAX2 for the
	  HANGUPCAUSE hash. Additionally, this breaks out some
	  functionality in frame.c for getting information about frame
	  types and subclasses. Review:
	  https://reviewboard.asterisk.org/r/1941/ (issue SWP-4222)

2012-06-15 15:33 +0000 [r369000]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.exports.in: Remove some symbol exports that
	  got missed in the removal of global symbols. (issue AST-807)
	  (issue AST-901) (issue AST-908) ........ Merged revisions 368998
	  from
	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
	  ........ Merged revisions 368999 from
	  http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones

2012-06-15 00:55 +0000 [r368972-368991]  Richard Mudgett <rmudgett@digium.com>

	* /: Remove remaining properties mmichelson left laying around from
	  phones branch merge.

	* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
	  main/ccss.c, main/app.c, apps/app_followme.c, apps/app_queue.c,
	  apps/app_stack.c: Allow non-normal execution routines to be able
	  to run on hungup channels. * Make non-normal dialplan execution
	  routines be able to run on a hung up channel. This is preparation
	  work for hangup handler routines. * Fixed ability to support
	  relative non-normal dialplan execution routines. (i.e., The
	  context and exten are optional for the specified dialplan
	  location.) Predial routines are the only non-normal routines that
	  it makes sense to optionally omit the context and exten. Setting
	  a hangup handler also needs this ability. * Fix Return
	  application being able to restore a dialplan location exactly.
	  Channels without a PBX may not have context or exten set. * Fixes
	  non-normal execution routines like connected line interception
	  and predial leaving the dialplan execution stack unbalanced.
	  Errors like missing Return statements, popping too many stack
	  frames using StackPop, or an application returning non-zero could
	  leave the dialplan stack unbalanced. * Fixed the AGI gosub
	  application so it cleans up the dialplan execution stack and
	  handles the autoloop priority increments correctly. * Eliminated
	  the need for the gosub_virtual_context return location. Review:
	  https://reviewboard.asterisk.org/r/1984/

	* main/pbx.c: Make the Hangup application set a softhangup flag.
	  The Hangup application used to just return -1 to cause normal
	  dialplan execution to hangup a channel. For the non-normal
	  execution routines like predial and connected-line interception
	  routines, the hangup request would exit the routine early but
	  otherwise be ignored. * Made the Hangup application not allow
	  setting a cause code of zero. A zero cause code is not defined.

	* include/asterisk/app.h: Move vm defines to group them better.

2012-06-14 19:40 +0000 [r368966]  Jason Parker <jparker@digium.com>

	* include/asterisk/app.h, /, tests/test_voicemail_api.c,
	  main/app.c, include/asterisk/app_voicemail.h (removed),
	  apps/app_voicemail.c: Multiple revisions 368963,368965 ........
	  r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) |
	  14 lines Remove global symbol requirement from app_voicemail.
	  This uses the existing "function installation" stuff that already
	  existed for other functions, like getting message counts. (closes
	  issue AST-807) (issue AST-901) (issue AST-908) Review:
	  https://reviewboard.asterisk.org/r/1965/ ........ Merged
	  revisions 368962 from
	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
	  ........ r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun
	  2012) | 11 lines These functions that were moved need to be
	  static. Also wrap test functions in a #ifdef. (issue AST-807)
	  (issue AST-901) (issue AST-908) ........ Merged revisions 368964
	  from
	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
	  ........ Merged revisions 368963,368965 from
	  http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones

2012-06-14 17:34 +0000 [r368948]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny
	  due to Key Pad Button Message handling AST-2012-008 (r367844)
	  fixed a denial of service attack exploitable in the Skinny
	  channel driver that occurred when certain messages are sent after
	  a previously registered station sends an Off Hook message.
	  Unresolved in that patch is an issue in the Asterisk 10 releases,
	  wherein, if a Station Key Pad Button Message is processed after
	  an Off Hook message, the channel driver will inappropriately
	  dereference a NULL pointer. This patch fixes those places where
	  the message handling or the channel callback functions would
	  attempt to dereference the line's pointer to the device. (issue
	  ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
	  mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
	  uploaded by mjordan (license 6283) ........ Merged revisions
	  368947 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-14 15:28 +0000 [r368929]  Mark Michelson <mmichelson@digium.com>

	* /, main/Makefile: Revert Makefile change to remove embedding
	  res_adsi.so The change has resulted in a linking error for
	  certain versions of GCC. This is much worse than the original
	  issue, so for now, temporarily revert the change. A more thorough
	  change will be sought out. ........ Merged revisions 368927 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368928 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-14 13:41 +0000 [r368920-368921]  Terry Wilson <twilson@digium.com>

	* include/asterisk/config_options.h, main/config_options.c: Add a
	  post_apply callback to the Config Options API This adds a
	  callback that only fires when changes have been successfully
	  applied via the Config Options API. Review:
	  https://reviewboard.asterisk.org/r/1980/

	* include/asterisk/config_options.h, main/config_options.c: Add
	  filename alias support to the Config Options API This adds the
	  ability to handle a single filename alias for a config file. This
	  is useful if a config filename has changed, but the old filename
	  should be supported for backwards compatibility. Review:
	  https://reviewboard.asterisk.org/r/1981/

2012-06-13 21:17 +0000 [r368900]  Mark Michelson <mmichelson@digium.com>

	* /, funcs/func_volume.c: Fix a deadlock that occurs when
	  func_volume is used on a local channel. This was discovered by
	  trying to perform a call forward to an extension that makes use
	  of func_volume. When the local channel is optimized away, the
	  datastore on the local;2 channel would have its audiohook
	  destroyed rather than detaching the audiohook from the channel
	  and then destroying it. With this patch, func_volume's datastore
	  destructor takes the proper route of detaching the audiohook and
	  then destroying it. (closes issue ASTERISK-19611) reported by
	  Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
	  Michelson (license #5049) ........ Merged revisions 368898 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368899 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-13 20:28 +0000 [r368896]  Matthew Jordan <mjordan@digium.com>

	* res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as
	  'core' supported modules Recently, various issues surrounding
	  weak symbols have caused problems with modules that rely on that
	  feature to be enabled in menuselect. This includes app_voicemail
	  and chan_dahdi, as they both rely upon res_smdi and res_adsi,
	  which, in certain circumstances, may not be enabled by default in
	  menuselect. Because res_smdi/res_adsi are dependencies for
	  chan_dahdi/app_voicemail, this patch marks both as 'core'
	  supported modules. This will allow both app_voicemail and
	  chan_dahdi to be enabled as well, regardless of whether or not
	  that system supports weak symbols. (issue AST-900) Reported by:
	  Thomas Arimont (issue AST-885) Reported by: Denis Alberto
	  Martinez ........ Merged revisions 368894 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368895 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-13 19:51 +0000 [r368886]  Mark Michelson <mmichelson@digium.com>

	* /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
	  the result is that Asterisk has a phantom module loaded at
	  startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
	  reported by Leif Madsen ........ Merged revisions 368873 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368885 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-13 14:55 +0000 [r368832-368855]  Matthew Jordan <mjordan@digium.com>

	* Makefile: Replace MODULES_DIR with ASTMODDIR in Makefile's
	  INSTALLDIRS Post Asterisk 10, the MODULES_DIR variable no longer
	  exists, and was replaced with ASTMODDIR.

	* Makefile, /: Do not install empty directories; add ASTLIBDIR
	  r368830 modified the installation script to only create a
	  directory if that directory does not exist. If some directory
	  variable was empty, it would attempt to create the empty
	  location. It also failed to create the ASTLIBDIR directory. This
	  patch fixes it such that the correct directories are made and
	  only created if a value specifying them actually exists. ........
	  Merged revisions 368852 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368853 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* Makefile, /: Do not perform install on existing directories If a
	  directory already exists, performing a 'make install' will remove
	  the permissions associated with the current directory and replace
	  them with the permissions of the user executing the install. This
	  patch changes this behavior to only perform an install on the
	  directory if the directory does not exist. Thus, if a user later
	  changes the permissions on that directory, those permissions will
	  be preserved in subsequent installs. Review:
	  https://reviewboard.asterisk.org/r/1986 Review:
	  https://reviewboard.asterisk.org/r/1864 (closes issue
	  ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
	  Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
	  by mjordan) ........ Merged revisions 368830 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368831 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-12 15:46 +0000 [r368809]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if
	  remote party information is present. On incoming calls, we were
	  setting the cid_tag on the dialog only if there was no remote
	  party information (Remote-Party-ID or P-Asserted-Identity)
	  present. The Caller ID tag is an invented parameter, though, and
	  should be set no matter the circumstance. (closes issue
	  ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
	  Reported by Trey Blancher ........ Merged revisions 368807 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368808 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-12 14:09 +0000 [r368793-368794]  Matthew Jordan <mjordan@digium.com>

	* /: Update merge property information

	* channels/chan_sip.c: Fix deadlock in SIP transfers that involve a
	  REFER request In r367163, "send to voicemail" functionality was
	  added to the SIP channel driver. This required updating the party
	  redirecting information for the channel based on the headers
	  provided in the REFER request. When the redirecting party
	  information is updated on the channel, a call to
	  ast_indicate_data occurs. Because handle_request_refer still had
	  the sip_pvt locked, a deadlock could occur between the pbx_thread
	  and the do_monitor thread servicing the REFER request. This patch
	  preserves the proper locking order between the channel and the
	  sip_pvt by ensuring that the sip_pvt is unlocked prior to
	  updating the party redirecting information on the channel.
	  (closes issue AST-903) Reported by: Matt Jordan patches:
	  jira_ast_903_trunk.patch by rmudgett (license 5621)

2012-06-12 04:03 +0000 [r368784]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c, UPGRADE.txt: Parse ANI2 information from SIP
	  From header parameters ANI2 information is now parsed out of SIP
	  From headers when present in the oli, isup-oli, and ss7-oli
	  parameters and is available via the CALLERID(ani2) dialplan
	  function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon
	  Review: https://reviewboard.asterisk.org/r/1947/

2012-06-11 17:34 +0000 [r368772]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/chan_sip.c, include/asterisk/channel.h,
	  channels/chan_iax2.c: Fix deadlock potential with
	  ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
	  the channel lock held can result in a deadlock because the
	  function also locks the bridged channel. (issue ASTERISK-19537)
	  (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
	  Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
	  Davis ........ Merged revisions 368759 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368760 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-11 15:23 +0000 [r368722-368751]  Kinsey Moore <kmoore@digium.com>

	* channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c,
	  res/res_fax.c, channels/sip/reqresp_parser.c, apps/app_queue.c,
	  main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c,
	  channels/sip/dialplan_functions.c, apps/app_directory.c,
	  pbx/pbx_config.c, res/res_odbc.c, res/res_speech.c,
	  apps/app_voicemail.c: Fix coverity UNUSED_VALUE findings in core
	  support level files Most of these were just saving returned
	  values without using them and in some cases the variable being
	  saved to could be removed as well. (issue ASTERISK-19672)
	  ........ Merged revisions 368738 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368739 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /: Recorded merge of revisions 368721 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix
	  compilation in dev-mode Backport a compilation fix in md5.c from
	  trunk that only showed up in dev-mode under certain compiler
	  versions. ........ Merged revisions 368719 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8

2012-06-08 21:08 +0000 [r368712-368714]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, main/utils.c, include/asterisk/strings.h: Fix
	  error paths in action_hangup() for AMI Hangup action. * Check
	  allocation function return values for failure. Crashing is bad. *
	  Tweak ast_regex_string_to_regex_pattern() parameters for proper
	  ast_str usage.

	* main/channel.c, include/asterisk/channel.h: Tweak
	  ast_channel_softhangup_withcause_locked() to take a typed
	  parameter.

2012-06-08 08:32 +0000 [r368688]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Fix MWI update so LED display correct
	  voicemail state after phone usage. Also fixes few warnings.
	  (closes issue #19675) Reported by: dbohling Patches: fixmwi.patch
	  uploaded by dbohling (license 6378)

2012-06-07 21:44 +0000 [r368680-368681]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Skinny cleanup (mwi_event_cb). Original
	  was testing for d->session, setting and testing again (all
	  nested). Removed duplicate testing and restructured function to
	  test/return and then the main code.

	* channels/chan_skinny.c: Skinny cleanup. Removed d->registered
	  which was mirroring d->session. Changed relevant references to
	  use d->session instead. Moved setting and unsetting of l->device
	  from session register to device configuration. As such, l->device
	  will always be valid unless it is has not been configured to a
	  device. Revised various test where checking if a device is
	  registered to use l->device->session.

2012-06-07 20:39 +0000 [r368674-368675]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Fix app_queue debug message use of args.options
	  after the string has been parsed.

	* apps/app_queue.c: Fix inverted test in app_queue for ringinuse.
	  Regression from -r367080 ringinuse commit. (issue ASTERISK-19536)

2012-06-07 20:32 +0000 [r368673]  Terry Wilson <twilson@digium.com>

	* main/udptl.c, include/asterisk/config_options.h, apps/app_skel.c,
	  main/config_options.c, tests/test_config.c: Fix reloading an
	  unchanged file with the Config Options API Adding multiple file
	  support broke reloading an unchanged file. This adds an enum for
	  return values for the aco_process_* functions and ensures that
	  the config is not applied if res is not ACO_PROCESS_OK. Review:
	  https://reviewboard.asterisk.org/r/1979/

2012-06-07 20:00 +0000 [r368668]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* formats/format_ogg_vorbis.c: Fix a typo in format_ogg_vorbis.c:
	  suport Review: https://reviewboard.asterisk.org/r/1970/

2012-06-07 15:43 +0000 [r368663]  Terry Wilson <twilson@digium.com>

	* include/asterisk/config_options.h, main/config_options.c,
	  tests/test_config.c: Add default handler documentation and
	  standardize acl handler Added documentation describing what flags
	  and arguments to pass to aco_option_register for default option
	  types. Also changed the ACL handler to use the flags parameter to
	  differentiate between "permit" and "deny" instead of adding an
	  additional vararg parameter. Review:
	  https://reviewboard.asterisk.org/r/1969/

2012-06-06 21:34 +0000 [r368646]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash
	  hook to orignate a second call deadlock. A deadlock can occur
	  when a POTS phone tries to flash hook to originate a second call
	  for 3-way or transfer. If another process is scanning the
	  channels container when the POTS line flash hooks then a deadlock
	  will occur. * Release the channel and private locks when creating
	  a new channel as a result of a flash hook. (closes issue
	  ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
	  ........ Merged revisions 368644 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368645 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-06 19:25 +0000 [r368637]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix a specific scenario where ACKs are
	  not matched. If a dialog-starting INVITE contains a to-tag, then
	  Asterisk will respond with a 481. In this case, the resulting
	  incoming ACK would not be matched, so Asterisk would continue
	  retransmitting the 481 until the transaction times out. There
	  were two issues. Asterisk, upon creating a sip_pvt would generate
	  a local tag. However, when the time came to transmit the 481,
	  since there was a to-tag in the INVITE, Asterisk would place this
	  original to-tag in the 481 response. When the ACK came in,
	  Asterisk would attempt to match the to-tag in the ACK to the
	  generated local tag. Unfortunately, Asterisk never actually
	  transmitted a response with the generated local tag, so the
	  to-tag in the ACK would not match. The other problem was that
	  when the 481 was sent, nothing was set on the sip_pvt to indicate
	  what CSeq is expected in the ACK. To fix the first problem, we
	  zero out the to-tag seen in the incoming INVITE. This way,
	  Asterisk, when time to send a response, will send its generated
	  local tag instead. To fix the second problem, we set the
	  sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
	  481. (closes issue ASTERISK-19892) Reported by Mark Michelson
	  ........ Merged revisions 368625 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368629 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-06 17:22 +0000 [r368606]  Matthew Jordan <mjordan@digium.com>

	* /, build_tools/make_version: Add feature modifier to versions
	  produced from branches Certain branches, such as Certified
	  Asterisk, may have a modifier added to them that specifies the
	  features available in that branch. For branches, this modifier is
	  expected to be reflected in the location of the branch in
	  subversion. For example, a subversion of URL of
	  /certified/branches/1.8.11 would have a feature modifier of
	  'certified'. This is slightly different then how features are
	  determined for tags, where the feature is part of the actual tag
	  name, e.g., "10.5.0-digiumphones". In keeping with the
	  nomenclature used for tags, the feature specifier for branches is
	  translated and placed after the revision numbers. For the example
	  given previously, this would result in a branch version of
	  "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged
	  revisions 368604 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368605 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-06 16:11 +0000 [r368588]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Ensure overlapping hold flags do not
	  conflict When changing between different modes of hold, the flags
	  were not being cleared out properly causing a failure to change
	  hold states. (closes issue ASTERISK-19919) Patch-by: Morten
	  Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions
	  368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 368587 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-06 01:11 +0000 [r368566-368569]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix parked call performing a DTMF blind
	  transfer after being retrieved. When a parked call was retrieved
	  from the parking lot, it could not do a blind transfer because it
	  caused the involved calls to be hung up unconditionally. * Made
	  the ParkedCall application return the ast_bridge_call() return
	  value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc
	  ........ Merged revisions 368567 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368568 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/features.c: Make builtin_blindtransfer() fully use
	  ast_async_goto() abilities.

2012-06-05 16:25 +0000 [r368550]  Jonathan Rose <jrose@digium.com>

	* CHANGES: Merge 'core' and 'core changes' sections in CHANGES
	  file.

2012-06-05 15:28 +0000 [r368519-368537]  Kinsey Moore <kmoore@digium.com>

	* /: Recorded merge of revisions 368536 from
	  http://svn.asterisk.org/svn/asterisk/branches/10 ........ Resolve
	  some build warnings My newly upgraded compiler caught these
	  usages of uninitialized values. They weren't actually used.
	  ........ Merged revisions 368533 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8

	* /, apps/app_voicemail.c: Ensure that pages and emails are sent
	  using RFC822-compliant date format When localization was added to
	  app_voicemail, these headers were altered when they should have
	  remained in en_US format for RFC compliance. This reverts the
	  changes to those two lines. (closes issue ASTERISK-19876)
	  ........ Merged revisions 368520 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368524 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c,
	  channels/chan_sip.c, main/channel_internal_api.c,
	  main/features.c, include/asterisk/channel.h, apps/app_queue.c:
	  Convert AST_FLAG_ANSWERED_ELSEWHERE usage to
	  AST_CAUSE_ANSWERED_ELSEWHERE This was essentially duplicated
	  functionality where normal channels used
	  AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
	  AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts
	  that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review:
	  https://reviewboard.asterisk.org/r/1944 (closes issue
	  ASTERISK-19865) Patch-by: Birger Harzenetter

2012-06-04 22:12 +0000 [r368500]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Relay proper SIP responses on calling
	  side. Revision 351130 broke corect HANGUPCAUSE setting for the
	  404 case in chan_sip. Other cases were also potentially broken.
	  This patch fixes the relaying of causes to be what they used to
	  be. (closes issue ASTERISK-19914) Reported by Pavel Troller
	  Tested by Walter Doekes (via a reviewboard test to be committed
	  later) Patches: chan_sip.diff uploaded by Pavel Troller (license
	  #6302) ........ Merged revisions 368498 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368499 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-04 21:18 +0000 [r368472]  Richard Mudgett <rmudgett@digium.com>

	* /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
	  ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
	  ........ Merged revisions 368469 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368470 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-04 20:53 +0000 [r368435-368467]  Mark Michelson <mmichelson@digium.com>

	* contrib/editors/asterisk.vim: Also have vim syntax-highlight
	  type=network.

	* contrib/editors/asterisk.vim: Add vim syntax highlighting for
	  type=line, type=phone, and type=application. (closes issue
	  ASTERISK-19800) Reported by: Billy Chia Patches:
	  asterisk.vim.patch uploaded by Billy Chia (license #6381)

	* main/channel.c, apps/app_mixmonitor.c: Remove some extra
	  debugging I forgot to remove in the merge of Digium phone
	  support.

	* /: Remove automerge properties.

	* /, contrib/realtime/mysql/voicemail_messages.sql,
	  main/presencestate.c (added), main/config.c, main/channel.c,
	  include/asterisk/callerid.h, include/asterisk/file.h,
	  main/manager.c, channels/chan_skinny.c,
	  include/asterisk/event_defs.h, include/asterisk/sip_api.h
	  (added), tests/test_voicemail_api.c (added), main/features.c,
	  apps/app_voicemail.exports.in, main/app.c, main/message.c,
	  channels/sip/include/sip.h, main/pbx.c, channels/chan_sip.c,
	  include/asterisk/presencestate.h (added),
	  include/asterisk/config.h, include/asterisk/app_voicemail.h
	  (added), configs/manager.conf.sample, apps/app_queue.c,
	  include/asterisk/manager.h, include/asterisk/app.h,
	  funcs/func_presencestate.c (added), include/asterisk/message.h,
	  main/file.c, main/callerid.c, main/event.c,
	  include/asterisk/pbx.h, tests/test_config.c,
	  channels/chan_sip.exports.in (added), apps/app_mixmonitor.c,
	  main/asterisk.c, apps/app_voicemail.c: Merge changes dealing with
	  support for Digium phones. Presence support has been added. This
	  is accomplished by allowing for presence hints in addition to
	  device state hints. A dialplan function called PRESENCE_STATE has
	  been added to allow for setting and reading presence. Presence
	  can be transmitted to Digium phones using custom XML elements in
	  a PIDF presence document. Voicemail has new APIs that allow for
	  moving, removing, forwarding, and playing messages. Messages have
	  had a new unique message ID added to them so that the APIs will
	  work reliably. The state of a voicemail mailbox can be obtained
	  using an API that allows one to get a snapshot of the mailbox. A
	  voicemail Dialplan App called VoiceMailPlayMsg has been added to
	  be able to play back a specific message. Configuration hooks have
	  been added. Configuration hooks allow for a piece of code to be
	  executed when a specific configuration file is loaded by a
	  specific module. This is useful for modules that are dependent on
	  the configuration of other modules. chan_sip now has a public
	  method that allows for a custom SIP INFO request to be sent
	  mid-dialog. Digium phones use this in order to display progress
	  bars when files are played. Messaging support has been expanded a
	  bit. The main visible difference is the addition of an AMI action
	  MessageSend. Finally, a ParkingLots manager action has been added
	  in order to get a list of parking lots.

2012-06-04 19:46 +0000 [r368421]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /: Fix potential deadlock between masquerade and
	  chan_local. * Restructure ast_do_masquerade() to not hold channel
	  locks while it calls ast_indicate(). * Simplify many calls to
	  ast_do_masquerade() since it will never return a failure now. If
	  it does fail internally because a channel driver callback
	  operation failed, the only thing ast_do_masquerade() can do is
	  generate a warning message about strange things may happen and
	  press on. * Fixed the call to ast_bridged_channel() in
	  ast_do_masquerade(). This change fixes half of the deadlock
	  reported in ASTERISK-19801 between masquerades and chan_iax.
	  (closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
	  rmudgett Review: https://reviewboard.asterisk.org/r/1915/
	  ........ Merged revisions 368405 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368407 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-02 21:13 +0000 [r368359]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/utils.h, res/res_http_websocket.exports.in
	  (added), include/asterisk/http_websocket.h (added), main/utils.c,
	  res/res_http_websocket.c (added): Add res_http_websocket module
	  which implements the WebSocket protocol according to RFC 6455.
	  Review: https://reviewboard.asterisk.org/r/1952/

2012-06-01 23:53 +0000 [r368311]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_stack.c: Fix deadlock when Gosub used with alternate
	  dialplan switches. Attempting to remove a channel from
	  autoservice with the channel lock held will result in deadlock. *
	  Restructured gosub_exec() to not call ast_parseable_goto() and
	  ast_exists_extension() with the channel lock held. (closes issue
	  ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
	  ........ Merged revisions 368308 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368310 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-01 20:42 +0000 [r368268-368269]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Improve SDP offer/answer RFC compliance
	  Asterisk should not accept SDP offers that contain unknown RTP
	  profiles (for audio/video streams) or unknown top-level media
	  types. When it does, it answers with an SDP that does not match
	  the offer properly, and this will nearly always result in a
	  broken call. This patch causes such offers to be rejected.
	  Review: https://reviewboard.asterisk.org/r/1811/

	* /, channels/chan_sip.c: Improve SDP parsing warning messages *
	  'Unsupported media type' is only reported when that is in fact
	  the case, not when a supported media type is included in an 'm'
	  line that has an invalid format. * All warning messages related
	  to parsing 'm' lines now include the 'm' line contents. * (minor
	  bugfix) newline added to port-number-zero warning messages. *
	  Warning messages improved to use RFC-specified terminology for
	  various items. * Warnings for offers that include more than one
	  port for a single media type now include the media type. Review:
	  https://reviewboard.asterisk.org/r/1811/ ........ Merged
	  revisions 368218 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368267 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-06-01 18:20 +0000 [r368181-368221]  Terry Wilson <twilson@digium.com>

	* configs/config_test.conf.sample (added): Add missing config for
	  config API test

	* main/udptl.c, include/asterisk/utils.h,
	  include/asterisk/astobj2.h, configure.ac,
	  include/asterisk/config.h, main/astobj2.c, main/config.c,
	  Makefile, include/asterisk/config_options.h (added), configure,
	  main/asterisk.exports.in, apps/app_skel.c, main/config_options.c
	  (added), tests/test_config.c, makeopts.in,
	  configs/app_skel.conf.sample (added),
	  include/asterisk/stringfields.h: Add new config-parsing framework
	  This framework adds a way to register the various options in a
	  config file with Asterisk and to handle loading and reloading of
	  that config in a consistent and atomic manner. Review:
	  https://reviewboard.asterisk.org/r/1873/

2012-06-01 13:04 +0000 [r368143]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample,
	  channels/sip/include/sip.h: Help mitigate potential reinvite
	  glare scenarios. When Asterisk servers are set up back-to-back,
	  and direct media is to be used betweeen endpoints, it is fairly
	  common for the two Asterisk servers to send direct media
	  reinvites to each other simultaneously. This results in 491s and
	  ACKs being exchanged between the servers. While the media
	  eventually gets set up properly, the problem is that there can be
	  a noticeable delay for the streams to stabilize. This patch adds
	  a new directmedia option called "outgoing". With this set, an
	  immediate direct media reinvite will only be sent if the call
	  direction is outgoing. For incoming dialogs, an immediate direct
	  media reinvite will not be sent, but further "reactionary" direct
	  media reinvites may be sent. Review:
	  https://reviewboard.asterisk.org/r/1954

2012-06-01 03:30 +0000 [r368094]  Michael L. Young <elgueromexicano@gmail.com>

	* /, funcs/func_channel.c: Add documentation to function CHANNEL
	  for options echocan_mode and buffers The ability to set
	  "echocan_mode" and "buffers" through the dialplan was added to
	  chan_dahdi some time ago. This patch adds some documentation to
	  func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
	  Noll Tested by: Michael L. Young Patches:
	  asterisk-19911-branch18.diff uploaded by Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/1949/
	  ........ Merged revisions 368092 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368093 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-31 18:39 +0000 [r368052]  Richard Mudgett <rmudgett@digium.com>

	* res/ael/pval.c, main/tcptls.c, main/manager.c,
	  res/res_config_odbc.c, /, channels/chan_sip.c,
	  channels/chan_agent.c, funcs/func_math.c, main/features.c,
	  apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c:
	  Coverity Report: Fix issues for error type REVERSE_INULL (core
	  modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue
	  ASTERISK-19648) Reported by: Matt Jordan ........ Merged
	  revisions 368039 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 368042 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-30 18:08 +0000 [r367908-367982]  Richard Mudgett <rmudgett@digium.com>

	* /: Use the DEADLOCK_AVOIDANCE() macro instead. (issue
	  ASTERISK-19854) ........ Merged revisions 367980 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367981 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
	  executing CLI "pri show channels" and "ss7 show channels"
	  commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
	  * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
	  deadlock properly. * Code ss7_grab() better. (closes issue
	  ASTERISK-19854) Reported by: Jaxon Patches:
	  jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
	  by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
	  Jaxon ........ Merged revisions 367976 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367978 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_meetme.c: Coverity Report: Fix issues for error type
	  REVERSE_INULL (deprecated modules) * Fix only issue pointed out
	  by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
	  * Change use of %i to %d in sscanf() in find_user(). The use of
	  %i gives unexpected parsing because it can accept hex, octal, and
	  decimal integer formats. * Changed other uses of %i in
	  app_meetme() to use %d for consistency. (issue ASTERISK-19648)
	  Reported by: Matt Jordan ........ Merged revisions 367906 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367907 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-29 18:40 +0000 [r367845]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_skinny.c: AST-2012-008: Fix remote crash
	  vulnerability in chan_skinny When a skinny session is
	  unregistered, the corresponding device pointer is set to NULL in
	  the channel private data. If the client was not in the on-hook
	  state at the time the connection was closed, the device pointer
	  can later be dereferened if a message or channel event attempts
	  to use a line's pointer to said device. The patches prevent this
	  from occurring by checking the line's pointer in message handlers
	  and channel callbacks that can fire after an unregistration
	  attempt. (closes issue ASTERISK-19905) Reported by: Christoph
	  Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
	  AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
	  AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........
	  Merged revisions 367844 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-25 16:33 +0000 [r367783]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
	  without suggested MOH class crash. * Made schedule_delivery() set
	  the received frame f->data.ptr to NULL if the datalen is zero. *
	  Fix queue_signalling() memcpy() size error. * Made
	  queue_signalling() not use C++ keyword variable names. (closes
	  issue ASTERISK-19597) Reported by: mgrobecker Patches:
	  jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett, Michael L. Young ........ Merged
	  revisions 367781 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367782 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-25 02:31 +0000 [r367732]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Fix pvt_sip for inbound call to use
	  peer's allowtransfer setting The pvt_sip allowtransfer was not
	  being set to that of the peer's setting. Therefore, the global
	  allowtransfer setting was being used instead which would lead to
	  calls not being transfered if the global setting was set to 'no'
	  despite the setting on the peer being 'yes' and vice versa, calls
	  would be allowed to transfer even if the peer's setting was 'no'
	  but the global setting was 'yes'. (Closes issue ASTERISK-19856)
	  Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
	  issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1923/ ........ Merged
	  revisions 367730 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367731 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-24 23:52 +0000 [r367693]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored
	  if dial forked and one fork redirects. The Dial and Queue I
	  option is intended to block connected line updates and
	  redirecting updates. However, it is a feature that when a call is
	  locally redirected, the I option is disabled if the redirected
	  call runs as a local channel so the administrator can have an
	  opportunity to setup new connected line information.
	  Unfortunately, the Dial and Queue I option is disabled for *all*
	  forked calls if one of those calls is redirected. * Make the Dial
	  and Queue I option apply to each outgoing call leg independently.
	  Now if one outgoing call leg is locally redirected, the other
	  outgoing calls are not affected. * Made Dial not pass any
	  redirecting updates when forking calls. Redirecting updates do
	  not make sense for this scenario. * Made Queue not pass any
	  redirecting updates when using the ringall strategy. Redirecting
	  updates do not make sense for this scenario. * Fixed deadlock
	  potential with chan_local when Dial and Queue send redirecting
	  updates for a local redirect. * Converted the Queue stillgoing
	  flag to a boolean bitfield. (closes issue ASTERISK-19511)
	  Reported by: rmudgett Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1920/ ........ Merged
	  revisions 367678 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367679 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-24 18:56 +0000 [r367640]  Jonathan Rose <jrose@digium.com>

	* main/rtp_engine.c, channels/chan_sip.c,
	  include/asterisk/rtp_engine.h: chan_sip: fix problem
	  directmediapermit/deny uses the wrong address When remotely
	  bridging calls with directmedia, Asterisk would check the address
	  of the peers/users holding directmedia ACLs (set via
	  directmediapermit/directmediadeny) instead of the bridged peer.
	  This is similar to r366547, but trunk specific and involves
	  changes to the rtpengine instead of just chan_sip. (closes issue
	  AST-876) review: https://reviewboard.asterisk.org/r/1924/

2012-05-24 13:33 +0000 [r367563]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_confbridge.c: Fix crash in ConfBridge when user
	  announcement is played for more than 2 users A patch introduced
	  in r354938 made it so that ConfBridge would not attempt to play
	  sound files if those files did not exist. Unfortunately,
	  ConfBridge uses the same underlying function, play_sound_helper,
	  to playback both sound files and numbers to callers. When a
	  number is being played back, the name of the sound file is
	  expected to be NULL. This NULL value was passed into a function
	  that tested for the existance of a sound file and is not tolerant
	  to NULL file names, causing a crash. This patch fixes the
	  behavior, such that if a sound file does not exist we do not
	  attempt to play it, but we only attempt that check if the a sound
	  file was specified in the first place. If a sound file was not
	  specified, we use the 'play number' logic in the helper function.
	  (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
	  by: Florian Gilcher patches: asterisk-19899.diff uploaded by
	  mjordan (license 6283) ........ Merged revisions 367562 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-24 00:36 +0000 [r367477-367520]  Richard Mudgett <rmudgett@digium.com>

	* channels/iax2-parser.c: Made use IAX frame cache only for
	  cacheable frame types.

	* main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination.
	  The AST_CONTROL_HOLD MOH class from the WaitExten application can
	  now be queued onto a channel, passed over local channels with the
	  /m option, and passed over IAX channels. ........ Merged
	  revisions 367469 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367470 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-23 20:39 +0000 [r367419]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c: logger: Fix a potential callid reference leak
	  discovered in development Uncovered a nasty reference leak while
	  I was writing some changes to chan_dahdi/sig_analog. Slapped
	  myself around a bit after seeing that I performed the unchecked
	  return causing this problem.

2012-05-23 20:30 +0000 [r367418]  Mark Michelson <mmichelson@digium.com>

	* main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined.
	  Thanks to Paul Belanger for pointing out this error. ........
	  Merged revisions 367416 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367417 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-23 13:46 +0000 [r367376]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Re-add
	  LastMsgsSent value for SIP peers Previously, MWI logic utilized a
	  counter called 'lastmsgssent' to know whether or not MWI NOTIFY
	  requests had been sent to a specific peer. When MWI notifications
	  were changed to use the internal event framework, this value was
	  no longer needed for its original purpose. Hence, it was no
	  longer updated with the new/old message counts for a peer. The
	  value was previously removed for Asterisk 10; however, since it
	  was still present in Asterisk 1.8 and still useful for reporting
	  purposes, it was decided to re-add the value. This patch re-adds
	  the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show
	  peer [peer]' command, and makes it so that the value of
	  lastmsgssent is updated appropriately. The value should now
	  display the new/old message counts for a particular peer. (closes
	  issue ASTERISK-17866) Reported by: Steve Davies patches by:
	  ast-17866-rb1272.patch (License #5041 by irroot) Modified
	  slightly for this commit Review:
	  https://reviewboard.asterisk.org/r/1939 ........ Merged revisions
	  367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 367369 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-22 17:29 +0000 [r367274-367309]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /, include/asterisk/cel.h,
	  main/channel_internal_api.c, include/asterisk/channel.h,
	  main/cel.c, main/asterisk.c: Fix race condition for CEL
	  LINKEDID_END event This patch fixes to situations that could
	  cause the CEL LINKEDID_END event to be missed. 1) During a core
	  stop gracefully, modules are unloaded when ast_active_channels ==
	  0. The LINKDEDID_END event fires during the channel destructor.
	  This means that occasionally, the cel_* module will be unloaded
	  before the channel is destroyed. It seemed generally useful to
	  wait until the refcount of all channels == 0 before unloading, so
	  I added a channel counter and used it in the shutdown code. 2)
	  During a masquerade, ast_channel_change_linkedid is called. It
	  calls ast_cel_check_retire_linkedid which unrefs the linkedid in
	  the linkedids container in cel.c. It didn't ref the new linkedid.
	  Now it does. Review: https://reviewboard.asterisk.org/r/1900/
	  ........ Merged revisions 367292 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367299 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Resolve crash in subscribing for MWI
	  notifications ASTOBJ_UNREF sets the variable to NULL after
	  unreffing it, so the variable should definitely not be used after
	  that. To solve this in the two cases that affect subscribing for
	  MWI notifications, we instead save the ref locally, and unref
	  them in the error conditions. (closes issue ASTERISK-19827)
	  Reported by: B. R Review:
	  https://reviewboard.asterisk.org/r/1940/ ........ Merged
	  revisions 367266 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367267 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-21 22:45 +0000 [r367227]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Made ast_queue_hangup() and
	  ast_queue_hangup_with_cause() lock instead of trylock. It made no
	  sense to trylock the channel and then unconditionally lock the
	  channel right after.

2012-05-21 20:35 +0000 [r367189]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_iax2.c: Make chan_iax2 reject cause code
	  indications correctly If chan_iax2 does not reject the
	  PVT_CAUSE_CODE frames, the cause will not be stored properly.

2012-05-21 20:31 +0000 [r367163-367183]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/callerid.h, channels/chan_sip.c,
	  main/callerid.c: Revert revision 367163. This should have been
	  committed to my team trunk-digiumphones branch instead of trunk.

	* include/asterisk/callerid.h, channels/chan_sip.c,
	  main/callerid.c: Add "send to voicemail" Digium phone
	  functionality to Asterisk. This change accommodates two methods
	  by which calls can be directed to a user's voicemail. * Incoming
	  calls can be redirected to any user's voicemail. * Established
	  calls can be blind transferred to any user's voicemail. Digium
	  phones indicate the desire to direct a call to voicemail by using
	  a Diversion header with a reason parameter of "send_to_vm". This
	  patch adds the "send_to_vm" reason as a valid redirecting reason.
	  In addition, chan_sip.c has been modified to update redirecting
	  information on the transferred channel by reading a Diversion
	  header on a REFER request. (closes issue AST-871) Reported by
	  Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925

2012-05-21 17:39 +0000 [r367124]  Terry Wilson <twilson@digium.com>

	* include/asterisk/astobj2.h: Minor documentation change

2012-05-18 19:39 +0000 [r367080]  Jonathan Rose <jrose@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: app_queue:
	  Per Member ringinuse option and deprecation of ignorebusy Adds a
	  number of methods for controlling the setting of 'ringinuse'
	  which is basically the same concept as the old ignorebusy
	  setting, only now the per member setting always controls whether
	  or not the member is actually ringed while in use. A CLI command
	  and a manager action have been added to change a given queue
	  member's ringinuse option while Asterisk is running and the an
	  argument has been added for adding members with deliberately set
	  ringinuse in queues.conf Some effort has been made to ensure
	  compatability with dialplans and databases still referring to
	  'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe
	  Lindheimer Review: https://reviewboard.asterisk.org/r/1919/

2012-05-18 17:54 +0000 [r367010-367029]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK
	  static analysis reports some more. This addresses core findings 4
	  and 6. Moises Silva helped me by stating that a break could be
	  safely added to the case where it is added in chan_dahdi.c In
	  say.c, I have added a comment indicating that static analysis
	  complains but that it is currently unknown if this is correct.
	  This fixes all core findings of this type. (closes issue
	  ASTERISK-19662) reported by Matthew Jordan ........ Merged
	  revisions 367027 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367028 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
	  Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
	  structures were allocated but never freed. This was a bigger
	  issue for clients than servers since new SSL_CTX structures could
	  be allocated for each connection. Servers, on the other hand,
	  typically set up a single SSL_CTX for their lifetime. This is
	  solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
	  ssl_ctx on it, it is freed so that a new one can take its place.
	  2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
	  been added so that servers can properly free their SSL_CTXs.
	  (issue ASTERISK-19278) ........ Merged revisions 367002 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 367003 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-18 15:51 +0000 [r366917-366955]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
	  main/cli.c: Fix more memory leaks This patch adds to what was
	  fixed in r366880. Specifically, it addresses the following: *
	  chan_sip: dispose of an allocated frame in off nominal code paths
	  in sip_rtp_read * func_odbc: when disposing of an allocated
	  resultset, ensure that any rows that were appended to that
	  resultset are also disposed of * cli: free the created return
	  string buffer in another off nominal code path * chan_dahdi: free
	  a frame that was allocated by the dsp layer if we choose not to
	  process that frame (issue ASTERISK-19665) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........
	  Merged revisions 366944 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366948 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c,
	  res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c,
	  apps/app_record.c, res/res_calendar_caldav.c, res/res_jabber.c,
	  apps/app_queue.c, channels/chan_iax2.c, main/enum.c,
	  main/editline/term.c, main/config.c, res/res_srtp.c, main/cli.c,
	  main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c,
	  funcs/func_odbc.c, main/features.c, apps/app_minivm.c,
	  main/editline/readline.c, channels/sip/config_parser.c,
	  main/xmldoc.c, res/res_calendar.c, apps/app_voicemail.c: Fix a
	  variety of memory leaks This patch addresses a number of memory
	  leaks in a variety of modules that were found by a static
	  analysis tool. A brief summary of the changes: * app_minivm: free
	  ast_str objects on off nominal paths * app_page: free the
	  ast_dial object if the requested channel technology cannot be
	  appended to the dialing structure * app_queue: if a penalty rule
	  failed to match any existing rule list names, the created rule
	  would not be inserted and its memory would be leaked * app_read:
	  dispose of the created silence detector in the presence of off
	  nominal circumstances * app_voicemail: dispose of an allocated
	  unique ID field for MWI event un-subscribe requests in off
	  nominal paths; dispose of configuration objects when using the
	  secret.conf option * chan_dahdi: dispose of the allocated frame
	  produced by ast_dsp_process * chan_iax2: properly unref peer in
	  CLI command "iax2 unregister" * chan_sip: dispose of the
	  allocated frame produced by sip_rtp_read's call of
	  ast_dsp_process; free memory in parse unit tests *
	  func_dialgroup: properly deref ao2 object grhead in nominal path
	  of dialgroup_read * func_odbc: free resultset in off nominal
	  paths of odbc_read * cli: free match_list in off nominal paths of
	  CLI match completion * config: free comment_buffer/list_buffer
	  when configuration file load is unchanged; free the same buffers
	  any time they were created and config files were processed *
	  data: free XML nodes in various places * enum: free context
	  buffer in off nominal paths * features: free ast_call_feature in
	  off nominal paths of applicationmap config processing * netsock2:
	  users of ast_sockaddr_resolve pass in an ast_sockaddr struct that
	  is allocated by the method. Failures in ast_sockaddr_resolve
	  could result in the users of the method not knowing whether or
	  not the buffer was allocated. The method will now not allocate
	  the ast_sockaddr struct if it will return failure. * pbx: cleanup
	  hash table traversals in off nominal paths; free ignore pattern
	  buffer if it already exists for the specified context * xmldoc:
	  cleanup various nodes when we no longer need them *
	  main/editline: various cleanup of pointers not being freed before
	  being assigned to other memory, cleanup along off nominal paths *
	  menuselect/mxml: cleanup of value buffer for an attribute when
	  that attribute did not specify a value * res_calendar*: responses
	  are allocated via the various *_request method returns and should
	  not be allocated in the various write_event methods; ensure
	  attendee buffer is freed if no data exists in the parsed node;
	  ensure that calendar objects are de-ref'd appropriately *
	  res_jabber: free buffer in off nominal path * res_musiconhold:
	  close the DIR* object in off nominal paths * res_rtp_asterisk: if
	  we run out of ports, close the rtp socket object and free the rtp
	  object * res_srtp: if we fail to create the session in libsrtp,
	  destroy the temporary ast_srtp object (issue ASTERISK-19665)
	  Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1922 ........ Merged revisions
	  366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 366881 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-18 14:27 +0000 [r366896]  Jonathan Rose <jrose@digium.com>

	* channels/sip/dialplan_functions.c: chan_sip: Fix a small
	  TEST_FRAMEWORK related error that prevents compiling Introduced
	  with r366842, a function call made only with TEST_FRAMEWORK
	  enabled was missing an argument since the function arguments were
	  changed.

2012-05-18 14:21 +0000 [r366843-366888]  Kinsey Moore <kmoore@digium.com>

	* /, channels/sip/config_parser.c: Reorder and renumber tests
	  appropriately It appears that a patch did not apply properly when
	  adding tests 12 and 13 and test 11 was duplicated. These tests
	  have been reordered and renumbered such that they make sense.
	  ........ Merged revisions 366882 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366884 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/channel.c: Make the new SIP_CAUSE backend behave more like
	  the original SIP_CAUSE There was a slight discrepancy in the
	  behaviors of the old SIP_CAUSE and the new SIP_CAUSE/HANGUPCAUSE
	  when a channel had been originated and had not yet been answered.
	  This caused the noload_res_srtp_attempt_srtp test to fail since
	  the SIP_CAUSE variable was never actually set. This behavior has
	  been restored.

2012-05-17 16:28 +0000 [r366842]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/logger.h, main/channel.c,
	  channels/sip/include/dialog.h, main/pbx.c, channels/chan_sip.c,
	  main/channel_internal_api.c, main/logger.c,
	  include/asterisk/channel.h, CHANGES, channels/sip/include/sip.h,
	  main/cli.c: logger: Adds additional support for call id logging
	  and chan_sip specific stuff This patch improves the handling of
	  call id logging significantly with regard to transfers and adding
	  APIs to better handle specific aspects of logging. Also, changes
	  have been made to chan_sip in order to better handle the creation
	  of callids and to enable the monitor thread to bind itself to a
	  particular call id when a dialog is determined to be related to a
	  callid. It then unbinds itself before returning to normal
	  monitoring. review: https://reviewboard.asterisk.org/r/1886/

2012-05-17 13:21 +0000 [r366746]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking
	  bounds of array index after using it; improper sizeof This patch
	  fixes two problems pointed out by a static analysis tool. * In
	  chan_dahdi, when an event is handled the index of the sub channel
	  is first obtained. In very off nominal cases, the method that
	  determines the index can return a negative value. In the event
	  handling code, whether or not the index returned is valid was
	  being checked after that value was used to index into an array.
	  This patch makes it so the value is checked before any indexing
	  is done. * In res_calendar_ews, sizeof was being passed a pointer
	  instead of the struct to determine the amount of memory to
	  allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
	  issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged
	  revisions 366740 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366741 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-16 18:00 +0000 [r366663-366700]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h: Remove missed idx parameter to some
	  ao2 global holder macros.

	* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
	  Change ao2 global array to ao2 global object holder. Review:
	  https://reviewboard.asterisk.org/r/1921/

2012-05-15 23:41 +0000 [r366599]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
	  getting a Diversion header's reason parameter. The use here was
	  assuming that the pointer would be updated, but the updated
	  string is actually returned by ast_strip_quoted() instead.
	  ........ Merged revisions 366597 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366598 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-15 19:36 +0000 [r366462-366546]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c: The predial routine must be run on the
	  local;1 channel. When ast_call() operates on a local channel, it
	  copies a lot of things from the local;1 channel to the local;2
	  channel. This includes among other things, channel variables and
	  party id information. Other reasons it was a bad idea to run
	  predial on the local;2 channel: 1) The channel has not been
	  completely setup. The ast_call() completes the setup. 2) The
	  local;2 caller and connected line party information is opposite
	  to any other channels predial runs on. (And it hasn't been setup
	  yet.) * Partially back out -r366183 by removing the chan_local
	  implementation of the struct ast_channel_tech.pre_call callback.

	* CHANGES, apps/app_followme.c: Add predial support to FollowMe.
	  Like the new predial feature for Dial. This adds the same b/B
	  options to FollowMe. Review:
	  https://reviewboard.asterisk.org/r/1910/

	* channels/chan_local.c: Make chan_local use the API call instead
	  of inlining its own version.

2012-05-14 20:15 +0000 [r366413]  Mark Michelson <mmichelson@digium.com>

	* /, pbx/dundi-parser.c: Fix two more coverity constant expression
	  result findings. These correspond to findings 0 and 1 in the core
	  findings of ASTERISK-19649. After contacting Mark Spencer, he was
	  unsure of what the intent behind these lines of code were, so
	  they are being axed. For Asterisk 1.8 and 10, the output of
	  debugging DUNDi frames will not be changed, but for trunk the
	  "Retry" portion will be omitted since it does not properly
	  distinguish retransmissions from initial frames. (closes issue
	  ASTERISK-19649) Reported by Matthew Jordan ........ Merged
	  revisions 366409 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366412 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-14 19:44 +0000 [r366408]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_unistim.c, apps/app_dial.c, main/rtp_engine.c,
	  channels/chan_vpb.cc, channels/chan_sip.c, UPGRADE.txt,
	  channels/chan_gtalk.c, channels/chan_console.c,
	  channels/chan_iax2.c, apps/app_queue.c, apps/app_followme.c,
	  channels/chan_oss.c, channels/chan_jingle.c, main/channel.c,
	  channels/chan_phone.c, main/dial.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, funcs/func_frame_trace.c,
	  main/features.c, channels/chan_h323.c, main/file.c,
	  channels/chan_alsa.c, configs/sip.conf.sample,
	  include/asterisk/frame.h, channels/chan_mgcp.c: Commit framework
	  for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting
	  point for the Asterisk 11: Who Hung Up work and provides a
	  framework which will allow channel drivers to report the types of
	  hangup cause information available in SIP_CAUSE without incurring
	  the overhead of the MASTER_CHANNEL dialplan function. The initial
	  implementation only includes cause generation for chan_sip and
	  does not include cause code translation utilities. This change
	  deprecates SIP_CAUSE and replaces its method of reporting cause
	  codes with the new framework. This change also deprecates the
	  'storesipcause' option in sip.conf. Review:
	  https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221)

2012-05-14 19:27 +0000 [r366401]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix broken reinvite glare scenario. To
	  make a long story short, reinvite glares were broken because
	  Asterisk would invert the To and From headers when ACKing a 491
	  response. The reason was because the initreq of the dialog was
	  being changed to the incoming glared reinvite instead of being
	  set to the outgoing glared reinvite. This change has three parts
	  * In handle_incoming, we never will reject an ACK because it has
	  a to-tag present, even if we think the request may be out of
	  dialog. * In handle_request_invite, we do not change the initreq
	  when receiving a reinvite to which we will respond with a 491. *
	  In handle_request_invite, several superflous settings up
	  pendinginvite have been removed since this is dones automatically
	  by transmit_response_reliable Review:
	  https://reviewboard.asterisk.org/r/1911 ........ Merged revisions
	  366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 366390 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-14 13:42 +0000 [r366351]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configure, configure.ac, autoconf/ast_pkgconfig.m4 (added): Macro
	  AST_PKG_CONFIG_CHECK to use chkconfig AST_PKG_CONFIG_CHECK:
	  Similar to AST_EXT_LIB_CHECK, but simply uses pkg-config data.
	  This simple version only uses pkg-config(1)'s tests. This commit
	  also uses the macro to test for GTK2 and GMIME (instead of the
	  current direct usage of pkg-config). Review:
	  https://reviewboard.asterisk.org/r/1906/

2012-05-12 00:03 +0000 [r366298]  Russell Bryant <russell@russellbryant.com>

	* /, addons/format_mp3.c: format_mp3: Fix a possible crash in
	  mp3_read(). This patch fixes a potential crash in mp3_read() by
	  not assuming that dbuf has enough data to finish filling up the
	  output buffer. The patch also makes sure that the dbuf state gets
	  reset after we know we read everything out of it already. In
	  passing, this patch includes some other cleanups of this module,
	  including stripping trailing whitespace, formatting fixes based
	  on coding guidelines, and removing a number of unused members
	  from the private state struct. (closes issue ASTERISK-19761)
	  Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk
	  ........ Merged revisions 366296 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366297 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-10 23:49 +0000 [r366183-366242]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /: * Made ast_change_name() hold the channels
	  container lock while changing the channel name. * Eliminate
	  redundant list not empty check in clone_variables(). ........
	  Merged revisions 366240 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366241 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_dial.c: Tweak app_dial predial documentation.

	* apps/app_dial.c, main/channel.c, channels/chan_local.c,
	  include/asterisk/channel.h: Run predial routine on local;2
	  channel where you would expect. Before this patch, the predial
	  routine executes on the ;1 channel of a local channel pair.
	  Executing predial on the ;1 channel of a local channel pair is of
	  limited utility. Any channel variables set by the predial routine
	  executing on the ;1 channel will not be available when the local
	  channel executes dialplan on the ;2 channel. * Create
	  ast_pre_call() and an associated pre_call() technology callback
	  to handle running the predial routine. If a channel technology
	  does not provide the callback, the predial routine is simply run
	  on the channel. Review: https://reviewboard.asterisk.org/r/1903/

2012-05-10 20:56 +0000 [r366169]  Kinsey Moore <kmoore@digium.com>

	* funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c, /,
	  channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
	  channels/sip/reqresp_parser.c, main/devicestate.c,
	  pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c,
	  main/config.c, res/res_monitor.c, main/channel.c, main/cdr.c,
	  res/ael/pval.c, main/data.c, channels/chan_dahdi.c,
	  main/tcptls.c, main/manager.c, main/features.c, main/app.c,
	  main/event.c, pbx/pbx_dundi.c, res/res_odbc.c, main/xmldoc.c,
	  apps/app_voicemail.c: Resolve FORWARD_NULL static analysis
	  warnings This resolves core findings from ASTERISK-19650 numbers
	  0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84,
	  87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers
	  26, 33, and 29 were already resolved. Those skipped were either
	  extended/deprecated or in areas of code that shouldn't be
	  disturbed. (Closes issue ASTERISK-19650) ........ Merged
	  revisions 366167 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366168 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-10 18:35 +0000 [r366126]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c,
	  funcs/func_lock.c, main/features.c, main/acl.c,
	  channels/iax2-provision.c, apps/app_queue.c,
	  channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
	  main/asterisk.c, main/xmldoc.c, apps/app_voicemail.c: Coverity
	  Report: Fix issues for error type CHECKED_RETURN for core (issue
	  ASTERISK-19658) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1905/ ........ Merged
	  revisions 366094 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366106 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-10 16:22 +0000 [r366062]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Close the proper tcptls_session when
	  session creation fails. (issue AST-998) Reported by: Thomas
	  Arimont Tested by: Thomas Arimont ........ Merged revisions
	  366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 366053 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-10 15:57 +0000 [r366007-366051]  Jonathan Rose <jrose@digium.com>

	* /, funcs/func_cdr.c, main/features.c, apps/app_disa.c,
	  apps/app_chanspy.c: Coverity Report: Fix issues for error type
	  UNINIT in Core supported modules (issue ASTERISK-19652) Reported
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/
	  ........ Merged revisions 366048 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 366049 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, codecs/codec_dahdi.c: Block on frameout if the hardware has
	  enough samples to complete a frame. Fixes some problems with
	  skipping audio in elaborate scenarios involving multiple codecs
	  by making codec_dahdi operate in a more synchronous fashion
	  similar to codec_g729. This change also fixes the use of file
	  conversion tools from Asterisk's CLI. This change may cause the
	  thread responsible for transcoding audio to block briefly (Shaun
	  Ruffell describes this as 'several milliseconds') while waiting
	  for the hardware transcoder. (closes issue ASTERISK-19643)
	  reported by: Shaun Ruffell Patches:
	  0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
	  uploaded by Shaun Ruffell (license 5417) ........ Merged
	  revisions 365989 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365990 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-09 19:26 +0000 [r366002]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* Makefile: pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect Allow
	  menuselect to get its set of CFLAGS and LDFLAGS through the
	  environment of Make: make BUILD_CFLAGS="whatever"
	  BUILD_LDFLAGS="whatever" Review:
	  https://reviewboard.asterisk.org/r/1907/

2012-05-09 17:58 +0000 [r365951]  Richard Mudgett <rmudgett@digium.com>

	* configs/followme.conf.sample, apps/app_followme.c: Improve
	  FollowMe accept/decline DTMF string matching. If you hit the
	  wrong DTMF digit trying to accept/decline a FollowMe call, you
	  had to wait for the prompt to repeat to try again. * Make
	  FollowMe compare the last DTMF digits received to the
	  accept/decline matching strings.

2012-05-09 16:36 +0000 [r365913]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Prevent sip_pvt refleak when an
	  ast_channel outlasts its corresponding sip_pvt. chan_sip was
	  coded under the assumption that a SIP dialog with an owner
	  channel will always be destroyed after the owner channel has been
	  hung up. However, there are situations where the SIP dialog can
	  time out and auto destruct before the corresponding channel has
	  hung up. A typical example of this would be if the 'h' extension
	  in the dialplan takes a long time to complete. In such cases,
	  __sip_autodestruct() would complain about the dialog being auto
	  destroyed with an owner channel still in place. The problem is
	  that even once the owner channel was hung up, the sip_pvt would
	  still be linked in its ao2_container because nothing would ever
	  unlink it. The fix for this is that if __sip_autodestruct() is
	  called for a sip_pvt that still has an owner channel in place,
	  the destruction is rescheduled for 10 seconds in the future. This
	  will continue until the owner channel is finally hung up. (closes
	  issue ASTERISK-19425) reported by David Cunningham Patches:
	  ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
	  (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
	  Dean Vesvuio ........ Merged revisions 365896 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365898 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-09 02:35 +0000 [r365766-365856]  Richard Mudgett <rmudgett@digium.com>

	* configs/followme.conf.sample, UPGRADE.txt, apps/app_followme.c:
	  Keep answered FollowMe calls until call accepted or last step
	  times out.

	* apps/app_followme.c: Put winning FollowMe outgoing call on hold
	  if the caller put it on hold. The FollowMe caller call leg is
	  usually answered and listening to MOH. The caller could put the
	  call on hold while FollowMe is looking for a winner. The winning
	  outgoing call is now immediately placed on hold if the caller has
	  put the call on hold before the winning call was selected.

	* apps/app_followme.c: Restructure how the FollowMe outgoing
	  channel list is handled.

	* apps/app_followme.c: Addendum to -r365766. Since it is no longer
	  allocated.

	* apps/app_followme.c: Make FollowMe findmeexec() put the list head
	  on the stack instead of mallocing it. Why this tiny struct was
	  malloced instead of the 28k struct in the last change is beyond
	  me. Just doing my part to help stamp out sillyness.

2012-05-08 21:46 +0000 [r365751]  Sean Bright <sean@malleable.com>

	* apps/app_externalivr.c: Add interrupt ('I') command to
	  ExternalIVR. Sending the 'I' command from an external process
	  will cause the current playlist to be cleared, including stopping
	  any audio file that is currently playing. This is useful when you
	  want to interrupt audio playback only when specific DTMF is
	  entered by the caller.

2012-05-08 21:41 +0000 [r365633-365749]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_followme.c: Make FollowMe app_exec() not declare a 28k
	  struct on the stack. Helping to stamp out stack abuse.

	* apps/app_followme.c: Simplify findmeexec() parameter passing.

	* /, apps/app_followme.c: * Fix FollowMe memory leak on error paths
	  in app_exec(). * Fix FollowMe leaving recorded caller name file
	  on error paths in app_exec(). * Use correct buffer dimension
	  define in struct fm_args.namerecloc[]. This fixes unexpected
	  namerecloc filename length restriction. ........ Merged revisions
	  365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 365701 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_followme.c: * Fix accept/decline DTMF buffer
	  overwrite in FollowMe. * Made use MAX_YN_STRING define to make
	  all accept/decline DTMF buffers the same size. Just using 20
	  isn't good enough when someone didn't get the memo. * Fix stupid
	  use of a global variable in FollowMe. (ynlongest) * Fix bit field
	  declarations in FollowMe. ........ Merged revisions 365631 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365632 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-08 15:57 +0000 [r365576]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Send more accurate identification
	  information in dialog-info SIP NOTIFYs. This uses the calling
	  channel's caller ID and connected line information to populate
	  the remote and local identities in the dialog-info NOTIFY when an
	  extension is ringing. There is a bit of an oddity here, and that
	  is that we seed the remote target with the To header of the
	  outbound call rather than the from header. This is because it was
	  reported that seeding with the from header caused hints to be
	  broken with certain SNOM devices. A comment has been added to the
	  code to explain this. (closes issue ASTERISK-16735) reported by
	  Maciej Krajewski patches: local_remote_hint2.diff uploaded by
	  Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
	  Michelson (license #5049) Tested by Niccolo Belli ........ Merged
	  revisions 365574 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365575 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-07 20:08 +0000 [r365532]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Change comment to use local channel name
	  designators in features.c

2012-05-07 18:58 +0000 [r365480]  Matthew Jordan <mjordan@digium.com>

	* main/pbx.c, apps/app_voicemail.c: Fix channel opaquification
	  slip-up in r365477 Those channels are opaque now...

2012-05-07 18:51 +0000 [r365479]  Richard Mudgett <rmudgett@digium.com>

	* /, tests/test_config.c: Fix type punned compiler warning in
	  test_config.c ........ Merged revisions 365476 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365478 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-07 18:42 +0000 [r365477]  Matthew Jordan <mjordan@digium.com>

	* main/pbx.c, /, apps/app_voicemail.c: Support VoiceMail d() option
	  when extension does not exist in channel's context The VoiceMail
	  d([c]) option is documented to accept digits for a new extension
	  in context <c>, if played during the greeting. This option works
	  fine if the extension being redirected to has an extension with
	  the same initial digit in the channel's current context. If that
	  digit did not happen to exist in some extension, a dialplan match
	  would fail and the user would not be redirected. This patch fixes
	  it such that if the <c> option is used, the extensions are
	  matched in that context as opposed to the caller's original
	  context. (closes issue ASTERISK-18243) Reported by: mjordan
	  Tested by: mjordan Review:
	  https://reviewboard.asterisk.org/r/1892 ........ Merged revisions
	  365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 365475 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-04 22:17 +0000 [r365400]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c, funcs/func_aes.c, main/features.c,
	  apps/app_followme.c, channels/chan_iax2.c,
	  channels/sip/config_parser.c, pbx/pbx_config.c,
	  apps/app_chanspy.c, apps/app_stack.c, main/config.c,
	  apps/app_voicemail.c: Fix many issues from the NULL_RETURNS
	  Coverity report Most of the changes here are trivial NULL checks.
	  There are a couple optimizations to remove the need to check for
	  NULL and outboundproxy parsing in chan_sip.c was rewritten to
	  avoid use of strtok. Additionally, a bug was found and fixed with
	  the parsing of outboundproxy when "outboundproxy=," was set.
	  (Closes issue ASTERISK-19654) ........ Merged revisions 365398
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 365399 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-04 17:38 +0000 [r365356]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c, /: Fix local channel chains optimizing
	  themselves out of a call. * Made chan_local.c:check_bridge()
	  check the return value of ast_channel_masquerade(). In long
	  chains of local channels, the masquerade occasionally fails to
	  get setup because there is another masquerade already setup on an
	  adjacent local channel in the chain. * Made the outgoing local
	  channel (the ;2 channel) flush one voice or video frame per
	  optimization attempt. * Made sure that the outgoing local channel
	  also does not have any frames in its queue before the masquerade.
	  * Made do the masquerade immediately to minimize the chance that
	  the outgoing channel queue does not get any new frames added and
	  thus unconditionally flushed. * Made block indication -1 (Stop
	  tones) event when the local channel is going to optimize itself
	  out. When the call is answered, a chain of local channels pass
	  down a -1 indication for each bridge. This blizzard of -1 events
	  really slows down the optimization process. (closes issue
	  ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
	  Davis Review: https://reviewboard.asterisk.org/r/1894/ ........
	  Merged revisions 365313 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365320 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-04 15:52 +0000 [r365300]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and
	  FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
	  These three all are in RTP code that attempts to print the number
	  of sequence number cycles in an RTCP RR report. The code was
	  masking out the upper 16 bits and then shifting the number right
	  by 16 bits. This led to an all zero result in all cases. The fix
	  is to do the shift without the bit masking. (issue
	  ASTERISK-19649) ........ Merged revisions 365298 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365299 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-03 19:36 +0000 [r365248]  Michael L. Young <elgueromexicano@gmail.com>

	* tests/test_security_events.c: Update security events unit tests
	  The security events framework API was changed in Asterisk 10 but
	  the unit tests were not updated at the same time. This patch does
	  the following: * Adds two more security events that were added to
	  the API * Add challenge, received_challenge and received_hash in
	  the inval_password security event unit test (Closes issue
	  ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael
	  L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1897/

2012-05-03 18:43 +0000 [r365213]  Sean Bright <sean@malleable.com>

	* CHANGES: Update documentation references in CHANGES to reflect
	  the correct pages on the wiki. The current CHANGES file refers to
	  doc/ in many places and those files no longer exist.

2012-05-03 15:05 +0000 [r365161]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, /,
	  addons/ooh323c/src/h323/H323-MESSAGES.h,
	  addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of
	  Coverity Static analysis, change H225ProtocolIdentifier from
	  value to pointer per functions that use this. (close issue
	  ASTERISK-19670) Reported by: Matt Jordan Patches:
	  ASTERISK-19670.patch (License #5415) ........ Merged revisions
	  365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 365160 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-03 14:47 +0000 [r365158]  Sean Bright <sean@malleable.com>

	* apps/app_externalivr.c, CHANGES: Add IPv6 support to ExternalIVR.
	  Review: https://reviewboard.asterisk.org/r/1896/

2012-05-03 14:35 +0000 [r365157]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/ooq931.c: Fix coverity static analysis
	  warning, allocate full ie structure instead of without data
	  buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
	  Patches: ASTERISK-19674.patch (License #5415) ........ Merged
	  revisions 365143 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365155 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-02 17:43 +0000 [r365084]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, /, main/cel.c: Multiple revisions
	  365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03
	  -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race
	  and local channel linkedids This patch has the ;2 channel inherit
	  the linkedid of the ;1 channel and fixes the race condition by no
	  longer scanning the channel list for "other" channels with the
	  same linkedid. Instead, cel.c has an ao2 container of linkedid
	  strings and uses the refcount of the string as a counter of how
	  many channels with the linkedid exist. Not only does this
	  eliminate the race condition, but it also allows us to look up
	  the linkedid by the hashed key instead of traversing the entire
	  channel list. Review: https://reviewboard.asterisk.org/r/1895/
	  ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02
	  May 2012) | 11 lines Don't leak a ref if out of memory and can't
	  link the linkedid If the ao2_link fails, we are most likely out
	  of memory and bad things are going to happen. Before those bad
	  things happen, make sure to clean up the linkedid references.
	  This patch also adds a comment explaining why linkedid can't be
	  passed to both local channel allocations and combines two ao2_ref
	  calls into 1. Review: https://reviewboard.asterisk.org/r/1895/
	  ........ Merged revisions 365006,365068 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 365083 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-02 15:59 +0000 [r365011]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Save the address on which a MESSAGE was
	  received, so it can be used in MESSAGE() This is useful in cases
	  where chan_sip may be listening on multiple addresses.

2012-05-02 02:51 +0000 [r364966]  Matthew Jordan <mjordan@digium.com>

	* /, main/audiohook.c: Only log a failure to get read/write samples
	  from factories if it didn't happen In audiohook_read_frame_both,
	  anytime samples are obtained from the read/write factories a
	  debug statement is logged stating that samples were not obtained
	  from the factories. This statement used to only occur if
	  option_debug was turned on and no samples were obtained; in some
	  refactoring when the option_debug statement was removed, the
	  "else" clause was removed as well. This patch makes it so that
	  those debug log statements only occur if the condition leading up
	  to them actually happened. ........ Merged revisions 364965 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 23:23 +0000 [r364915]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove a function that has been marked
	  unused since Asterisk 1.6.0. The reason I'm removing this is that
	  Coverity reported a STRAY_SEMICOLON issue here. Since the
	  function has been unused for so long, I just elected to remove it
	  altogether. (closes issue ASTERISK-19660)

2012-05-01 23:21 +0000 [r364910]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
	  (closes issue ASTERISK-19755) Reported by: Gunther Kelleter
	  Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
	  Kelleter ........ Merged revisions 364902 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364903 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 23:11 +0000 [r364901]  Mark Michelson <mmichelson@digium.com>

	* /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
	  error. As it turned out, this wasn't a huge deal. We were calling
	  ast_app_parse_options() for a set of options of which none took
	  arguments. The proper thing to do for this case is to pass NULL
	  for the "args" parameter here. We were instead passing a
	  seemingly-randomly chosen char * from the function. While this
	  would never get written to, you can rest assured things would
	  have gotten bad had new options (which took arguments) been added
	  to func_volume. (closes issue ASTERISK-19656) ........ Merged
	  revisions 364899 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364900 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 22:00 +0000 [r364846]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c, /: * Fix error path resouce leak in
	  local_request(). * Restructure local_request() to reduce
	  indentation. ........ Merged revisions 364840 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364845 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 21:49 +0000 [r364844]  Jason Parker <jparker@digium.com>

	* main/manager.c, /: Prevent a potential crash when using manager
	  hooks. Found by me while poking at DPMA-127. ........ Merged
	  revisions 364841 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364842 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 19:10 +0000 [r364788]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_confbridge.c: Play conf-placeintoconf message to the
	  correct channel Correct the code in app_confbridge to play the
	  conf-placeintoconf message to the marked user entering the bridge
	  instead of to the conference while the marked user hears silence.
	  (closes issue ASTERISK-19641) Reported-by: Mark A Walters
	  ........ Merged revisions 364786 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364787 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-05-01 18:29 +0000 [r364785]  Jonathan Rose <jrose@digium.com>

	* /, main/app.c: Fix bad check in voicemail functions for
	  ast_inboxcount2_func Check looks for ast_inboxcount_func instead
	  of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
	  issue ASTERISK-19718) Reported by: Corey Farrell Patches:
	  ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
	  (license 5909) ........ Merged revisions 364769 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364777 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-30 19:51 +0000 [r364708]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Revert revision 360862. Revision 360862
	  was intended to improve identities sent in dialog-info NOTIFY
	  requests. Some users reported that hint became broken once this
	  was done. It's not clear exactly what part of the patch has
	  caused this regression, but broken hints are bad. For now, this
	  revision is being reverted so that the next releases of Asterisk
	  do not have bad behavior in them. The original reported issue
	  will have to be fixed differently in the next version of
	  Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 364707 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-30 17:17 +0000 [r364654]  Mark Murawki <markm@intellasoft.net>

	* /, main/logger.c: Merged revisions 364635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
	  10 lines Sanatize result from bfd_find_nearest_line
	  (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
	  to null resulting in a crash when strrchr(file) runs (closes
	  issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
	  Murawski ........ ........ Merged revisions 364650 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-30 16:59 +0000 [r364652]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323cDriver.c: Fix use freed pointer in return value
	  from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
	  Patches: ASTERISK-19663-ooh323.patch (License #5415) ........
	  Merged revisions 364649 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364651 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-29 19:50 +0000 [r364580]  Matthew Jordan <mjordan@digium.com>

	* formats/format_ilbc.c, /, formats/format_sln.c,
	  formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
	  formats/format_g723.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_wav_gsm.c,
	  formats/format_siren14.c, formats/format_gsm.c,
	  formats/format_g719.c, formats/format_siren7.c,
	  formats/format_g729.c: Fix error that caused truncate operations
	  to fail Another very inappropriate placement of a ')' (again
	  introduced in r362151) caused the various truncate operations to
	  attempt to truncate the sound file at a position of '0'. (issue
	  ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810)
	  Reported by: colbec ........ Merged revisions 364578 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364579 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-29 02:23 +0000 [r364537]  Michael L. Young <elgueromexicano@gmail.com>

	* /, apps/confbridge/conf_config_parser.c: Fix configuring custom
	  sound_leader_has_left in confbridge.conf The configuration option
	  to specify a custom sound_leader_has_left file for a conference
	  bridge was not being parsed. This patch fixes it so that a custom
	  sound file will now be used. (closes issue ASTERISK-19771)
	  Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young
	  Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak
	  (license 6380) Review: https://reviewboard.asterisk.org/r/1884/
	  ........ Merged revisions 364536 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-28 20:24 +0000 [r364500]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: Add support for lightweight NAT
	  keepalive. If enabled using the keepalive option in sip.conf a
	  small packet will be sent at a regular interval to keep the NAT
	  mapping open. This is lightweight as the remote side does not
	  need to parse and handle a SIP message. (closes issue AST-783)
	  Review: https://reviewboard.asterisk.org/r/1756/

2012-04-28 01:33 +0000 [r364437-364462]  Russell Bryant <russell@russellbryant.com>

	* main/md5.c: md5: supress some compiler warnings. md5.c: In
	  function ‘MD5Final’: md5.c:154:2: error: dereferencing
	  type-punned pointer will break strict-aliasing rules
	  [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing
	  type-punned pointer will break strict-aliasing rules
	  [-Werror=strict-aliasing] There is an md5 unit test and it still
	  passes.

	* configure, include/asterisk/autoconfig.h.in, res/res_corosync.c,
	  configure.ac: res_corosync: Fix build against corosync 2.0.

	* apps/app_minivm.c: app_minivm: Fix a couple compiler warnings.
	  The warnings were about argv[0] being used uninitialized, which
	  is correct. Just remove setting username to this value, since
	  username is set again before it actually gets used.

	* main/features.c, CHANGES: features: Add FEATURE() and
	  FEATUREMAP() functions. Add two new dialplan functions: FEATURE()
	  and FEATUREMAP(). FEATURE() lets you set some of the
	  configuration options from the [general] section of features.conf
	  on a per-channel basis. FEATUREMAP() lets you customize the key
	  sequence used to activate built-in features, such as blindxfer,
	  and automon. See the built-in documentation for details. Review:
	  https://reviewboard.asterisk.org/r/1871/

2012-04-28 00:31 +0000 [r364436]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, CHANGES: PreDial - Ability to run dialplan on
	  callee and caller channels before Dial. Thanks to Mark Murawski
	  for the initial patch and feature definition. (closes issue
	  ASTERISK-19548) Reported by: Mark Murawski Review:
	  https://reviewboard.asterisk.org/r/1878/ Review:
	  https://reviewboard.asterisk.org/r/1229/

2012-04-27 22:54 +0000 [r364397]  Terry Wilson <twilson@digium.com>

	* /, tests/test_config.c (added), main/config.c: Multiple revisions
	  364365,364369 ........ r364365 | twilson | 2012-04-27 17:31:01
	  -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric
	  type range checking and add tests ast_parse_arg wasn't checking
	  for strto* parse errors or limiting the results by the actual
	  range of the numeric types. This patch fixes that and adds unit
	  tests as well. Review: https://reviewboard.asterisk.org/r/1879/
	  ........ Merged revisions 364340 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012)
	  | 2 lines Add missing test_config.c ........ Merged revisions
	  364365,364369 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-27 22:11 +0000 [r364343]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Don't attempt to make use of the
	  dynamic_exclude_static ACL if DNS lookup fails. (closes issue
	  ASTERISK-18321) Reported by Dan Lukes Patches:
	  ASTERISK-18321.patch by Mark Michelson (license #5049) ........
	  Merged revisions 364341 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364342 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-27 19:30 +0000 [r364287]  Matthew Jordan <mjordan@digium.com>

	* /, include/asterisk/time.h: Prevent overflow in calculation in
	  ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
	  attempts to calculate the difference, in milliseconds, between
	  two timeval structs, and return the difference in a 64-bit
	  integer. Unfortunately, it assumes that the long tv_sec/tv_usec
	  members in the timeval struct are large enough to hold the
	  calculated values before it returns. On 64-bit machines, this
	  might be the case, as a long may be 64-bits. On 32-bit machines,
	  however, a long may be less (32-bits), in which case, the
	  calculation can overflow. This overflow caused significant
	  problems in MixMonitor, which uses the method to determine if an
	  audio factory, which has not presented audio to an audiohook, is
	  merely late in providing said audio or will never provide audio.
	  In an overflow situation, the audiohook would incorrectly
	  determine that an audio factory that will never provide audio is
	  merely late instead. This led to situations where a MixMonitor
	  never recorded any audio. Note that this happened most frequently
	  when that MixMonitor was started by the ConfBridge application
	  itself, or when the MixMonitor was attached to a Local channel.
	  (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
	  Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
	  #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
	  Murawski Tested by: Michael L. Young Patches:
	  32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
	  (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
	  feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
	  https://reviewboard.asterisk.org/r/1889/ ........ Merged
	  revisions 364277 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364285 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-27 18:59 +0000 [r364260]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Allow SIP pvts involved in Replaces
	  transfers to fall out of reference sooner Unref the SIP pvt
	  stored in the refer structure as soon as it is no longer needed
	  so that the pvt and associated file descriptors can be freed
	  sooner. This change makes a reference decrement unnecessary in
	  code that handles SIP BYE/Also transfers which should not touch
	  the reference anyway. (Closes issue ASTERISK-19579) Reported by:
	  Maciej Krajewski Tested by: Maciej Krajewski ........ Merged
	  revisions 364258 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364259 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-27 14:45 +0000 [r364205]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Allow for reloading SRTP crypto keys
	  within the same SIP dialog As a continuation of the patch in
	  r356604, which allowed for the reloading of SRTP keys in
	  re-INVITE transfer scenarios, this patch addresses the more
	  common case where a new key is requested within the context of a
	  current SIP dialog. This can occur, for example, when certain
	  phones request a SIP hold. Previously, once a dialog was
	  associated with an SRTP object, any subsequent attempt to process
	  crypto keys in any SDP offer - either the current one or a new
	  offer in a new SIP request - were ignored. This patch changes
	  this behavior to only ignore subsequent crypto keys within the
	  current SDP offer, but allows future SDP offers to change the
	  keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested
	  by: Thomas Arimont Review:
	  https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
	  364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 364204 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-27 12:58 +0000 [r364164]  Stefan Schmidt <sst@sil.at>

	* res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c: fix a
	  wrong behavior of alarm timezones in caldav and icalendar when an
	  alarm doesnt use utc. This change uses the same timezone from the
	  start time. ........ Merged revisions 364163 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-26 21:11 +0000 [r364082-364110]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_directed_pickup.c: Update Pickup application
	  documentation. (With feeling this time.) ........ Merged
	  revisions 364108 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364109 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/features.c: Fix DTMF atxfer running h exten after the
	  wrong bridge ends. When party B does an attended transfer of
	  party A to party C, the attending bridge between party B and C
	  should not be running an h exten when the bridge ends. Running an
	  h exten now sets a softhangup flag to ensure that an AGI will run
	  in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the
	  party B channel for the attending bridge between party B and C.
	  (closes issue AST-870) (closes issue ASTERISK-19717) Reported by:
	  Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
	  Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
	  ........ Merged revisions 364060 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364065 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-26 19:33 +0000 [r364048]  Terry Wilson <twilson@digium.com>

	* /, main/asterisk.c: Add more constness to the end_buf pointer in
	  the netconsole issue ASTERISK-18308 Review:
	  https://reviewboard.asterisk.org/r/1876/ ........ Merged
	  revisions 364046 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 364047 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-26 13:59 +0000 [r363989]  Olle Johansson <oej@edvina.net>

	* apps/app_queue.c: Code formatting fixes.

2012-04-26 13:31 +0000 [r363988]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Fix reference leaks involving SIP
	  Replaces transfers The reference held for SIP blind transfers
	  using the Replaces header in an INVITE was never freed on success
	  and also failed to be freed in some error conditions. This caused
	  a file descriptor leak since the RTP structures in use at the
	  time of the transfer were never freed. This reference leak and
	  another relating to subscriptions in the same code path have now
	  been corrected. (closes issue ASTERISK-19579) ........ Merged
	  revisions 363986 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363987 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-26 09:48 +0000 [r363936]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, channels/chan_sip.c: chan_sip: [general] maxforwards, not
	  checked for a value greater than 255 The peer maxforwards is
	  checked for both '< 1' and '> 255', but the default 'maxforwards'
	  in the [general] section is only checked for '< 1' alecdavis
	  (license 585) Reported by: alecdavis Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1888/ ........ Merged
	  revisions 363934 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363935 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-26 03:12 +0000 [r363689-363877]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_directed_pickup.c: Update Pickup application
	  documentation. (Even better) ........ Merged revisions 363875
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 363876 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_directed_pickup.c: * Put more information in
	  pickup_exec() LOG_NOTICE. * Delay duplicating a string on the
	  stack in pickup_exec().

	* /, apps/app_directed_pickup.c: Update Pickup application
	  documentation. ........ Merged revisions 363788 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363789 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, /, channels/sig_pri.c: Make
	  DAHDISendCallreroutingFacility wait 5 seconds for a reply before
	  disconnecting the call. Some switches may not handle the
	  call-deflection/call-rerouting message if the call is
	  disconnected too soon after being sent. Asteisk was not waiting
	  for any reply before disconnecting the call. * Added a 5 second
	  delay before disconnecting the call to wait for a potential
	  response if the peer does not disconnect first. (closes issue
	  ASTERISK-19708) Reported by: mehdi Shirazi Patches:
	  jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett ........ Merged revisions 363730
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 363734 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
	  Clear ISDN channel resetting state if the peer continues to use
	  it. Some ISDN switches occasionally fail to send a RESTART
	  ACKNOWLEDGE in response to a RESTART request. * Made the second
	  SETUP received after sending a RESTART request clear the channel
	  resetting state as if the peer had sent the expected RESTART
	  ACKNOWLEDGE before continuing to process the SETUP. The peer may
	  not be sending the expected RESTART ACKNOWLEDGE. (issue
	  ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
	  jira_ast_815_v1.8.patch (license #5621) patch uploaded by
	  rmudgett (modified) ........ Merged revisions 363687 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363688 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-25 13:57 +0000 [r363480-363637]  Olle Johansson <oej@edvina.net>

	* apps/app_queue.c: Add documentation Thanks Tilghman!

	* apps/app_queue.c: Formatting changes only

	* apps/app_followme.c, apps/app_queue.c: Use the DEFINED value for
	  musicclass length. For some reason, features.c has it's own
	  definition. Should propably be fixed too.

	* main/channel.c, configs/asterisk.conf.sample, CHANGES,
	  include/asterisk/options.h, main/asterisk.c: Make it possible to
	  change the minimum DTMF duration in asterisk.conf Asterisk has a
	  setting for the minimum allowed DTMF. If we get shorter DTMF
	  tones, these will be changed to the minimum on the outbound call
	  leg. (closes issue ASTERISK-19772) Review:
	  https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested
	  by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for
	  this patch: agave-dtmf-duration-asterisk-conf-1.8

	* main/say.c: Formatting fixes Developer guidelines are important.

	* main/channel.c: Formatting fixes Found a small amount of curly
	  brackets in my hotel room here in Denmark. I hereby donate them
	  to the Asterisk project.

2012-04-25 01:26 +0000 [r363377-363430]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix recalled party B feature flags for a
	  failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF
	  atxfer to C 3) B hangs up 4) C does not answer 5) B is called
	  back 6) B answers 7) B cannot initiate transfers anymore * Add
	  dial features datastore to recalled party B channel that is a
	  copy of the original party B channel's dial features datastore. *
	  Extracted add_features_datastore() from
	  add_features_datastores(). * Renamed struct ast_dial_features
	  features_caller and features_callee members to my_features and
	  peer_features respectively. These better names eliminate the need
	  for some explanatory comments. * Simplified code accessing the
	  struct ast_dial_features datastore. (closes issue ASTERISK-19383)
	  Reported by: lgfsantos ........ Merged revisions 363428 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363429 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/features.c: Hangup affected channel in error paths of
	  bridge_call_thread(). ........ Merged revisions 363375 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363376 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-24 17:52 +0000 [r363335]  Terry Wilson <twilson@digium.com>

	* /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
	  (closes issue ASTERISK-19758) Reported by: Barry Miller Tested
	  by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
	  (license 5434) ........ Merged revisions 362868 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362869 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-23 17:05 +0000 [r363269]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, apps/app_queue.c: Make app_dial and app_queue
	  use new macro and gosub calls. * Simplify some code in app_dial
	  and app_queue by calling ast_app_exec_macro() and
	  ast_app_exec_sub(). * Fix minor locking issue in app_dial for
	  post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling.

2012-04-23 16:08 +0000 [r363215]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be
	  checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
	  specification does not mandate how these 3 flags must be
	  specified, only that one of the three must be specified in every
	  call. ........ Merged revisions 363209 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363212 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-23 14:48 +0000 [r363159]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: AST-2012-004: Fix an error that allows AMI
	  users to run shell commands sans authorization. As detailed in
	  the advisory, AMI users without write authorization for SYSTEM
	  class AMI actions were able to run system commands by going
	  through other AMI commands which did not require that
	  authorization. Specifically, GetVar and Status allowed users to
	  do this by setting their variable/s options to the SHELL or EVAL
	  functions. Also, within 1.8, 10, and trunk there was a similar
	  flaw with the Originate action that allowed users with originate
	  permission to run MixMonitor and supply a shell command in the
	  Data argument. That flaw is fixed in those versions of this
	  patch. (closes issue ASTERISK-17465) Reported By: David Woolley
	  Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) ........ Merged revisions 363117 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
	  Merged revisions 363141 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363156 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-23 14:10 +0000 [r363105-363108]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE
	  handling when no channel owner exists If Asterisk receives a SIP
	  UPDATE request after a call has been terminated and the channel
	  has been destroyed but before the SIP dialog has been destroyed,
	  a condition exists where a connected line update would be
	  attempted on a non-existing channel. This would cause Asterisk to
	  crash. The patch resolves this by first ensuring that the SIP
	  dialog has an owning channel before attempting a connected line
	  update. If an UPDATE request is received and no channel is
	  associated with the dialog, a 481 response is sent. (closes issue
	  ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt
	  Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt
	  Jordan (license 6283) ........ Merged revisions 363106 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363107 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
	  heap overflow in keypad button handling When handling a keypad
	  button message event, the received digit is placed into a fixed
	  length buffer that acts as a queue. When a new message event is
	  received, the length of that buffer is not checked before placing
	  the new digit on the end of the queue. The situation exists where
	  sufficient keypad button message events would occur that would
	  cause the buffer to be overrun. This patch explicitly checks that
	  there is sufficient room in the buffer before appending a new
	  digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
	  ........ Merged revisions 363100 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
	  Merged revisions 363102 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 363103 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-21 11:45 +0000 [r363045-363046]  Russell Bryant <russell@russellbryant.com>

	* res/res_corosync.c: res_corosync: Recover if corosync gets
	  restarted. If corosync gets restarted while Asterisk is running,
	  automatically recover.

	* res/res_corosync.c: res_corosync: reimplement "corosync show
	  members" command. Reimplement the "corosync show members" CLI
	  command using a CPG iterator instead of the cpg_membership_get
	  API call. This will also show all CPG members, including those in
	  groups other than 'asterisk', which may be useful at some point
	  for debugging purposes.

2012-04-21 01:46 +0000 [r362920-362999]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, /: Update app_dial M and U option GOTO return
	  value documentation. ........ Merged revisions 362997 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362998 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* include/asterisk/app.h, main/app.c, apps/app_stack.c: Fix
	  connected-line/redirecting interception gosubs executing more
	  than intended. * Redo ast_app_run_sub()/ast_app_exec_sub() to use
	  a known return point so execution will stop after the routine
	  returns there. (s@gosub_virtual_context:1) * Create
	  ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
	  gosub application respectively with the parameter string already
	  created.

	* main/rtp_engine.c: Move debug message in
	  ast_rtp_instance_early_bridge_make_compatible(). Move debug
	  message in ast_rtp_instance_early_bridge_make_compatible() to be
	  output when what it states has actually happened.

2012-04-20 16:50 +0000 [r362919]  Michael L. Young <elgueromexicano@gmail.com>

	* /, main/event.c: Add missing payload type to events API The
	  Security Events Framework API was changed while adding the
	  generation of security events in chan_sip. A payload type and
	  name was missed from being added to struct ie_maps. (closes issue
	  ASTERISK-19759) Reported by: Michael L. Young Patches:
	  issue-asterisk-19759.diff uploaded by Michael L. Young (license
	  5026) ........ Merged revisions 362918 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-20 16:23 +0000 [r362867-362888]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c,
	  channels/chan_misdn.c, main/rtp_engine.c: Use
	  ast_channel_lock_both() where it was inlined before. The
	  CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the
	  channel lock was originally obtained is overkill where
	  ast_channel_lock_both() was inlined.

	* main/pbx.c: * Add more information to some messages in
	  __ast_pbx_run(). * Simplify some dialplan priority setting code
	  in ast_explicit_goto() because of opaquification.

2012-04-20 14:50 +0000 [r362817]  Terry Wilson <twilson@digium.com>

	* /, apps/app_speech_utils.c: Document Speech* apps hangup on
	  failure and suggest TryExec The Speech API apps return -1 on
	  failure, which will hang up the channel. This may not be
	  desirable behavior for some, but it isn't something that can be
	  changed without breaking people's dialplans or writing an option
	  to all of the Speech apps that does what TryExec already does.
	  This patch documents the hangup behavior of the apps, and
	  suggests TryExec as the solution. (closes issue AST-813) ........
	  Merged revisions 362815 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362816 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-20 00:57 +0000 [r362779]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, UPGRADE.txt, include/asterisk/channel.h, CHANGES,
	  channels/sig_pri.c, funcs/func_callerid.c: Add original party id
	  and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future)
	  have support for reporting who was the original redirecting party
	  of a call. * Added support for the original redirecting party and
	  reason to the REDIRECTING function and the system core as well as
	  to the stubbed locations in sig_pri.c. Review:
	  https://reviewboard.asterisk.org/r/1829/

2012-04-19 22:01 +0000 [r362731]  Walter Doekes <walter+asterisk@wjd.nu>

	* funcs/func_version.c, /: Fix documentation for
	  ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions
	  362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 362730 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-19 21:14 +0000 [r362682]  Michael L. Young <elgueromexicano@gmail.com>

	* /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and
	  trailing backslashes A couple of unit tests did not have have
	  leading or trailing backslashes when setting their test category
	  resulting in a warning message being displayed. Added the
	  backslash where needed. ........ Merged revisions 362680 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362681 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-19 21:01 +0000 [r362679]  Richard Mudgett <rmudgett@digium.com>

	* /, configs/queues.conf.sample: Update membermacro and membergosub
	  documentation in queues.conf.sample. ........ Merged revisions
	  362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 362678 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-19 19:05 +0000 [r362635]  Terry Wilson <twilson@digium.com>

	* addons/chan_ooh323.c, apps/app_alarmreceiver.c,
	  channels/iax2-provision.c, res/snmp/agent.c: Convert some
	  strncpys to ast_copy_string Review:
	  https://reviewboard.asterisk.org/r/1732/

2012-04-19 16:10 +0000 [r362588]  Sean Bright <sean@malleable.com>

	* /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when
	  the 'S' command is sent first. If the first command sent from an
	  ExternalIVR client is an 'S' command, we were blindly removing
	  the first element from the play list and deferencing it, even if
	  it was NULL. This corrects that and also locks appropriately in
	  one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
	  ........ Merged revisions 362586 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362587 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-19 14:35 +0000 [r362538]  Terry Wilson <twilson@digium.com>

	* /, main/asterisk.c: Handle multiple commands per connection via
	  netconsole Asterisk would accept multiple NULL-delimited CLI
	  commands via the netconsole socket, but would occasionally miss a
	  command due to the command not being completely read into the
	  buffer. This patch ensures that any partial commands get moved to
	  the front of the read buffer, appended to, and properly sent.
	  (closes issue ASTERISK-18308) Review:
	  https://reviewboard.asterisk.org/r/1876/ ........ Merged
	  revisions 362536 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362537 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-19 02:40 +0000 [r362497]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c,
	  apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
	  addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a
	  variety of potential buffer overflows * chan_mobile: Fixed an
	  overrun where the cind_state buffer (an integer array of size 16)
	  would be overrun due to improper bounds checking. At worst, the
	  buffer can be overrun by a total of 48 bytes (assuming 4-byte
	  integers), which would still leave it within the allocated memory
	  of struct hfp. This would corrupt other elements in that struct
	  but not necessarily cause any further issues. * app_sms: The
	  array imsg is of size 250, while the array (ud) that the data is
	  copied into is of size 160. If the size of the inbound message is
	  greater then 160, up to 90 bytes could be overrun in ud. This
	  would corrupt the user data header (array udh) adjacent to ud. *
	  chan_unistim: A number of invalid memmoves are corrected. These
	  would move data (which may or may not be valid) into the ends of
	  these buffers. * asterisk: ast_console_toggle_loglevel does not
	  check that the console log level being set is less then or equal
	  to the allowed log levels of 32. * format_pref: In
	  ast_codec_pref_prepend, if any occurrence of the specified codec
	  is not found, the value used to index into the array pref->order
	  would be one greater then the maximum size of the array. *
	  jitterbuf: If the element being placed into the jitter buffer
	  lands in the last available slot in the jitter history buffer,
	  the insertion sort attempts to move the last entry in the buffer
	  into one slot past the maximum length of the buffer. Note that
	  this occurred for both the min and max jitter history buffers. *
	  tdd: If a read from fsk_serial returns a character that is
	  greater then 32, an attempt to read past one of the statically
	  defined arrays containing the values that character maps to would
	  occur. * localtime: struct ast_time and tm are not the same size
	  - ast_time is larger, although it contains the elements of tm
	  within it in the same layout. Hence, when using memcpy to copy
	  the contents of tm into ast_time, the size of tm should be used,
	  as opposed to the size of ast_time. * extconf: this treats
	  ast_timing's minmask array as if it had a length of 48, when it
	  has defined the size of the array as 24. pbx.h defines minmask as
	  having a size of 48. (issue ASTERISK-19668) Reported by: Matt
	  Jordan ........ Merged revisions 362485 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362496 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-18 17:03 +0000 [r362432]  Michael L. Young <elgueromexicano@gmail.com>

	* tests/test_security_events.c: Fix building security events test
	  The Security Events Framework API changed in trunk to support
	  IPv6. This broke the building of the security events test which
	  was based around IPv4. This patches fixes the build by changing
	  the test to conform to the new changes. (related to issue
	  ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/

2012-04-18 16:41 +0000 [r362430]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add
	  ability to ignore layer 1 alarms for BRI PTMP lines. Several
	  telcos bring the BRI PTMP layer 1 down when the line is idle.
	  When layer 1 goes down, Asterisk cannot make outgoing calls.
	  Incoming calls could fail as well because the alarm processing is
	  handled by a different code path than the Q.931 messages. * Add
	  the layer1_presence configuration option to ignore layer 1 alarms
	  when the telco brings layer 1 down. This option can be configured
	  by span while the similar DAHDI driver teignorered=1 option is
	  system wide. This option unlike layer2_persistence does not
	  require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA
	  ABE-2845 ........ Merged revisions 362428 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362429 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-17 21:23 +0000 [r362365-362380]  Matthew Jordan <mjordan@digium.com>

	* /, main/format_pref.c: Handle case where an unknown format is
	  used to get the preferred codec size In ast_codec_pref_getsize,
	  if an unknown format is passed to the method, no preferred codec
	  will be selected and a negative number will be used to index into
	  the format list. The method now logs an unknown format as a
	  warning, and returns an empty format list. (issue ASTERISK-19655)
	  Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/ ........ Merged
	  revisions 362377 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c:
	  Fix places in resources where a negative return value could
	  impact execution This patch addresses a number of modules in
	  resources that did not handle the negative return value from
	  function calls adequately. This includes: * res_agi.c: if the
	  result of the read function is a negative number, indicating some
	  failure, the result would instead be treated as the number of
	  bytes read. This patch now treats negative results in the same
	  manner as an end of file condition, with the exception that it
	  also logs the error code indicated by the return. *
	  res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
	  to srcfd, and instead assigns a negative value, that file
	  descriptor could later be passed to functions that require a
	  valid file descriptor. If spawn_mp3 fails, we now immediately
	  retry instead of continuing in the logic. * res_rtp_asterisk.c:
	  if no codec can be matched between two RTP instances in a peer to
	  peer bridge, we immediately return instead of attempting to use
	  the codec payload type as an index to determine the appropriate
	  negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
	  revisions 362362 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362364 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-17 21:10 +0000 [r362363]  Jonathan Rose <jrose@digium.com>

	* res/res_config_curl.c, res/res_config_pgsql.c,
	  res/res_config_odbc.c, /: Make use of va_args more appropriate to
	  form in various res_config modules plus utils. A number of
	  va_copy operations weren't matched with a corresponding va_end in
	  res_config_odbc. Also, there was a potential for va_end to be
	  invoked twice on the same va_arg in utils, which would mean
	  invoking va_end on an undefined variable... which is bad. va_end
	  is removed from various functions in config_pgsql and config_curl
	  since they aren't making their own copy. The invokers of those
	  functions are responsible for calling va_end on them. (issue
	  ASTERISK-19451) Reported by: Walter Doekes Review:
	  https://reviewboard.asterisk.org/r/1848/ ........ Merged
	  revisions 362354 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362357 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-17 21:08 +0000 [r362358-362361]  Matthew Jordan <mjordan@digium.com>

	* main/manager.c, /, main/asterisk.c: Fix places in main where a
	  negative return value could impact execution This patch addresses
	  a number of modules in main that did not handle the negative
	  return value from function calls adequately, or were not
	  sufficiently clear that the conditions leading to improper
	  handling of the return values could not occur. This includes: *
	  asterisk.c: A negative return value from the read function would
	  be used directly as an index into a buffer. We now check for
	  success of the read function prior to using its result as an
	  index. * manager.c: Check for failures in mkstemp and lseek when
	  handling the temporary file created for processing data returned
	  from a CLI command in action_command. Also check that the result
	  of an lseek is sanitized prior to using it as the size of a
	  memory map to allocate. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
	  Merged revisions 362359 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362360 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, funcs/func_env.c: Fix places where a negative return from
	  ftello could be used as invalid input In a variety of locations
	  in both reading and writing a file, the result from the C library
	  function ftello is used as input to other functions. For the
	  parameters and functions in question, a negative value is invalid
	  input. This patch checks the return value from the ftello
	  function to determine if we were able to determine the current
	  position in the file stream and, if not, fail gracefully. (issue
	  ASTERISK-19655) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/ ........ Merged
	  revisions 362355 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362356 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-17 18:57 +0000 [r362307]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
	  funcs/func_env.c, res/res_phoneprov.c, channels/chan_gtalk.c,
	  cdr/cdr_pgsql.c, res/res_http_post.c, res/res_musiconhold.c,
	  res/res_jabber.c, res/res_format_attr_celt.c,
	  channels/chan_dahdi.c, funcs/func_groupcount.c,
	  apps/app_osplookup.c, funcs/func_odbc.c, main/ast_expr2f.c,
	  apps/app_minivm.c, channels/chan_alsa.c, codecs/codec_resample.c,
	  formats/format_h264.c, res/res_format_attr_silk.c,
	  res/res_config_ldap.c, main/ast_expr2.fl,
	  res/res_config_sqlite3.c, channels/chan_sip.c,
	  channels/vcodecs.c, codecs/codec_g726.c, main/data.c,
	  res/res_corosync.c, channels/chan_h323.c, codecs/codec_dahdi.c,
	  funcs/func_callerid.c, main/asterisk.c, res/res_odbc.c: Avoid
	  cppcheck warnings; removing unused vars and a bit of cleanup.
	  Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/

2012-04-17 18:29 +0000 [r362306]  Matthew Jordan <mjordan@digium.com>

	* /, formats/format_sln.c, formats/format_vox.c,
	  formats/format_wav.c, formats/format_pcm.c,
	  formats/format_wav_gsm.c, formats/format_siren14.c,
	  formats/format_gsm.c, formats/format_g719.c,
	  formats/format_siren7.c: Fix error that caused seek format
	  operations to set max file size to '1' or '0' A very
	  inappropriate placement of a ')' (introduced in r362151) caused
	  the maximum size of a file to be set as the result of a
	  comparison operation, as opposed to the result of the ftello
	  operation. This resulted in seeking being restricted to the
	  beginning of the file, or 1 byte into the file. Thanks to the
	  Asterisk Test Suite for properly freaking out about this on at
	  least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
	  ........ Merged revisions 362304 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362305 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-17 15:00 +0000 [r362266]  Michael L. Young <elgueromexicano@gmail.com>

	* /, channels/chan_sip.c: Turn off warning message when bind
	  address is set to any. When a bind address is set to an ANY
	  address (udpbindport=::), a warning message is displayed stating
	  that "Address remapping activated in sip.conf but we're using
	  IPv6, which doesn't need it. Please remove 'localnet' and/or
	  'externaddr' settings." But if one is running dual stack, we
	  shouldn't be told to turn those settings off. This patch checks
	  if the bind address is an ANY address or not. The warning message
	  will now only be displayed if the bind address is NOT an ANY
	  address and IPv6 is being used. Also, updated the copyright year.
	  (closes issue ASTERISK-19456) Reported by: Michael L. Young
	  Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
	  uploaded by Michael L. Young (license 5026) ........ Merged
	  revisions 362253 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362264 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-16 21:58 +0000 [r362203-362206]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative
	  return handling in channel drivers In chan_agent, while handling
	  a channel indicate, the agent channel driver must obtain a lock
	  on both the agent channel, as well as the channel the agent
	  channel is using. To do so, it attempts to lock the other channel
	  first, then unlock the agent channel which is locked prior to
	  entry into the indicate handler. If this unlock fails with a
	  negative return value, which can occur if the object passed to
	  agent_indicate is an invalid ao2 object or is NULL, the return
	  value is passed directly to strerror, which can only accept
	  positive integer values. In chan_dahdi, the return value of
	  dahdi_get_index is used to directly index into the sub-channel
	  array. If dahd_get_index returns a negative value, it would use
	  that value to index into the array, which could cause an invalid
	  memory access. If dahdi_get_index returns a negative number, we
	  now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
	  Merged revisions 362204 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362205 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_voicemail.c: Fix handling of negative return code
	  when storing voicemails in ODBC storage When storing a voicemail
	  message using an ODBC connection to a database, the voicemail
	  message is first stored on disk. The sound file associated with
	  the message is read into memory before being transmitted to the
	  database. When this occurs, a failure in the C library's lseek
	  function would cause a negative value to be passed to the mmap as
	  the size of the memory map to create. This would almost certainly
	  cause the creation of the memory map to fail, resulting in the
	  message being lost. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863 ........
	  Merged revisions 362201 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362202 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-16 21:20 +0000 [r362200]  Michael L. Young <elgueromexicano@gmail.com>

	* main/manager.c, main/security_events.c,
	  channels/sip/security_events.c, CHANGES,
	  include/asterisk/security_events_defs.h: Add IPv6 address support
	  to security events framework. The current Security Events
	  Framework API only supports IPv4 when it comes to generating
	  security events. This patch does the following: * Changes the
	  Security Events Framework API to support IPV6 and updates the
	  components that use this API. * Eliminates an error message that
	  was being generated since the current implementation was treating
	  an IPv6 socket address as if it was IPv4. * Some copyright dates
	  were updated on files touched by this patch. (closes issue
	  ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael
	  L. Young Patches: security_events_ipv6v3.diff uploaded by Michael
	  L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1777/

2012-04-16 20:17 +0000 [r362153]  Matthew Jordan <mjordan@digium.com>

	* formats/format_ilbc.c, /, formats/format_sln.c,
	  formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
	  formats/format_g723.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_wav_gsm.c,
	  formats/format_siren14.c, formats/format_gsm.c,
	  formats/format_g719.c, formats/format_siren7.c,
	  formats/format_g729.c: Check for IO stream failures in various
	  format's truncate/seek operations For the formats that support
	  seek and/or truncate operations, many of the C library calls used
	  to determine or set the current position indicator in the file
	  stream were not being checked. In some situations, if an error
	  occurred, a negative value would be returned from the library
	  call. This could then be interpreted inappropriately as
	  positional data. This patch checks the return values from these
	  library calls before using them in subsequent operations. (issue
	  ASTERISK-19655) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/ ........ Merged
	  revisions 362151 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362152 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-13 16:12 +0000 [r362081-362085]  Jonathan Rose <jrose@digium.com>

	* apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of
	  the newly forked CDR log Prior to this patch, ForkCDR's e option
	  would immediately set the end time of the forked CDR to that of
	  the CDR that is being terminated. This resulted in the new CDR's
	  end time being roughly the same as it's beginning time (which is
	  in turn roughly the same as the original's end time). (closes
	  issue ASTERISK-19164) Reported by: Steve Davies Patches:
	  cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
	  ........ Merged revisions 362082 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362084 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_meetme.c: Send relative path named recordings to the
	  meetme directory instead of sounds Prior to this patch, no effort
	  was made to parse the path name to determine a proper destination
	  for recordings of MeetMe's r option. This fixes that. Review:
	  https://reviewboard.asterisk.org/r/1846/ ........ Merged
	  revisions 362079 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 362080 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-12 20:08 +0000 [r362043]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/srv.c: Convert SRV lookup message to debug level This helps
	  clean up the Asterisk CLI by converting the log message from
	  verbose to debug

2012-04-12 16:29 +0000 [r361998]  Richard Mudgett <rmudgett@digium.com>

	* configs/asterisk.conf.sample, UPGRADE.txt, pbx/pbx_config.c,
	  include/asterisk/options.h, main/asterisk.c: Add option to invoke
	  the extensions.conf stdexten using the legacy macro method.
	  ASTERISK-18809 eliminated the legacy macro invocation of the
	  stdexten in favor of the Gosub method without a means of
	  backwards compatibility. (issue ASTERISK-18809) (closes issue
	  ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett
	  Review: https://reviewboard.asterisk.org/r/1855/

2012-04-12 16:25 +0000 [r361968-361987]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_iax2.c: Make trunkfreq take effect when set
	  Previously, setting trunkfreq had no effect on initial load or on
	  reload and only ever used the default value. This causes
	  trunkfreq to be used appropriately on initial load and reload.
	  (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........
	  Merged revisions 361972 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361981 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* Makefile, build_tools/cflags.xml, /,
	  build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s,
	  configure, codecs/gsm/Makefile, configure.ac, Makefile.rules,
	  makeopts.in, codecs/lpc10/Makefile: Simplify build system
	  architecture optimization This change to the build system rips
	  out any usage of PROC along with architecture-specific
	  optimizations in favor of using -march=native where it is
	  supported. This fixes broken builds on 64bit Intel systems and
	  results in better optimized code on systems running GCC 4.2+.
	  Review: https://reviewboard.asterisk.org/r/1852/ (closes issue
	  ASTERISK-19462) ........ Merged revisions 361955 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361956 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-11 17:20 +0000 [r361909]  Jonathan Rose <jrose@digium.com>

	* /, configs/queues.conf.sample, apps/app_queue.c: Change default
	  value of 'ignorebusy' on Queue members so that behavior is more
	  like 1.8 Prior to this patch, in order to restore that behavior,
	  a function would have to be used on the QueueMember to make the
	  ringinuse option do anything, which is pretty unreasonable.
	  (closes issue ASTERISK-19536) reported by: Philippe Lindheimer
	  Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged
	  revisions 361907 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-10 21:50 +0000 [r361856]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
	  if DAHDI channel during an MWI event In the MWI processing loop,
	  when a valid event occurs the temporary caller ID information is
	  deallocated. If a new DAHDI channel is successfully created, the
	  event is passed up to the analog_ss_thread without error and the
	  loop exits. If, however, the DAHDI channel is not created, then
	  the caller ID struct has been free'd, and the gains reset to
	  their previous level. This will almost certainly cause an invalid
	  access to the free'd memory, either in subsequent calls to
	  callerid_free or calls to callerid_feed. * Rework the -r361705
	  patch to better manage the cs and mtd allocated resources. *
	  Fixed use of mwimonitoractive flag to be correct if the
	  mwi_thread() fails to start. ........ Merged revisions 361854
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 361855 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-10 19:58 +0000 [r361659-361805]  Matthew Jordan <mjordan@digium.com>

	* /, main/http.c: Fix crash caused by unloading or reloading of
	  res_http_post When unlinking itself from the registered HTTP
	  URIs, res_http_post could inadvertently free all URIs registered
	  with the HTTP server. This patch modifies the unregister method
	  to only free the URI that is actually being unregistered, as
	  opposed to all of them. ........ Merged revisions 361803 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361804 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* funcs/func_curl.c, /: Allow func_curl to exit gracefully if list
	  allocation fails during write If the global_curl_info data
	  structure could not be allocated, the datastore associated with
	  the operation would be free'd, but the function would not return.
	  This would later dereference the datastore, almost certainly
	  causing Asterisk to crash. With this patch, if the data structure
	  is not allocated the method will return an error code, and not
	  attempt any further operation. ........ Merged revisions 361753
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 361754 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
	  if DAHDI channel during an MWI event In the MWI processing loop,
	  when a valid event occurs the temporary caller ID information is
	  deallocated. If a new DAHDI channel is successfully created, the
	  event is passed up to the analog_ss_thread without error and the
	  loop exits. If, however, the DAHDI channel is not created, then
	  the caller ID struct has been free'd, and the gains reset to
	  their previous level. This will almost certainly cause an invalid
	  access to the free'd memory, either in subsequent calls to
	  callerid_free or calls to callerid_feed. This patch makes it so
	  that we only free the caller ID structure if a DAHDI channel is
	  successfully created, and we bump the gains back up if we fail to
	  make a DAHDI channel. ........ Merged revisions 361705 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361706 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, funcs/func_global.c: Change SHARED function to use a safe
	  traversal when modifying a variable When the SHARED function
	  modifies a variable, it removes it from its list of variables and
	  reinserts the new value at the head of the list of variables.
	  Doing this inside a standard list traversal can be dangerous, as
	  the standard list traversal does not account for the list being
	  changed. While the code in question should not cause a use after
	  free violation due to its breaking out of the loop after freeing
	  the variable, it could lead to a maintenance issue if the loop
	  was modified. This also fixes a violation reported by a static
	  analysis tool, which also makes this code easier to maintain in
	  the future. ........ Merged revisions 361657 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361658 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 22:00 +0000 [r361561-361608]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews
	  when event email address node is empty If the XML calendar data
	  returned by a Microsoft Exchange Web Service specifies an XML
	  Event E-Mail Address ("EmailAddress"), and no e-mail address is
	  provided, a condition existed where an ast_calendar_attendee
	  struct would be allocated but not appended to the list of
	  attendees. Because of that, the memory associated with the
	  attendee would never be freed. This patch frees the memory if no
	  e-mail address is provided. ........ Merged revisions 361606 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361607 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
	  option with user specified A memory leak/reference counting leak
	  occurs if the MeetMeAdmin 'e' command (eject last user that
	  joined) is used in conjunction with a specified user. Regardless
	  of the command being executed, if a user is specified for the
	  command, MeetMeAdmin will look up that user. Because the 'e'
	  option kicks the last user that joined, as opposed to the one
	  specified, the reference to the user specified by the command
	  would be leaked when the user variable was assigned to the last
	  user that joined. ........ Merged revisions 361558 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361560 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 19:58 +0000 [r361523]  Richard Mudgett <rmudgett@digium.com>

	* /, main/message.c: Don't add an empty MESSAGE_DATA(key) header if
	  it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add
	  an empty key header if the key header did not already exist. If
	  it already existed it would delete it. * Made msg_set_var_full()
	  exit early if the named variable did not already exist and the
	  value to set is empty. ........ Merged revisions 361522 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 18:19 +0000 [r361476]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c,
	  funcs/func_strings.c, formats/format_ogg_vorbis.c,
	  channels/console_video.c, apps/app_ices.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c, res/res_config_sqlite.c, res/res_srtp.c,
	  main/cdr.c, main/tcptls.c, channels/console_gui.c,
	  funcs/func_channel.c, apps/app_sms.c, addons/chan_mobile.c,
	  apps/app_chanspy.c, main/xmldoc.c, channels/chan_mgcp.c,
	  res/res_config_sqlite3.c, res/res_clioriginate.c,
	  apps/app_voicemail.c: Add missing newlines to CLI logging
	  ........ Merged revisions 361471 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361472 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 16:33 +0000 [r361429]  Paul Belanger <paul.belanger@polybeacon.com>

	* bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c,
	  bridges/bridge_multiplexed.c: Multiple revisions 361403,361412
	  ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri,
	  06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412
	  | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
	  lines Fix typo in svn:keywords ........ Merged revisions
	  361403,361412 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361422 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 15:50 +0000 [r361382]  Russell Bryant <russell@russellbryant.com>

	* /, configs/rpt.conf.sample (removed),
	  configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf
	  (removed): Remove a few more files related to chan_usbradio and
	  app_rpt. ........ Merged revisions 361380 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361381 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 14:02 +0000 [r361334]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Fix a typo in the warning messages for an
	  ignored media stream Added a '\n' to the warning messages when we
	  ignore a media stream due to the port number being '0'. (closes
	  issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........
	  Merged revisions 361332 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361333 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-06 13:32 +0000 [r361331]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c, /: Remove unnecessary error message in
	  app_dial.c The error message for failure to stop autoservice
	  after a gosub or macro call during a dial was removed for macro
	  while Asterisk 1.4 was still being actively developed. The
	  corresponding gosub error message was never removed. (closes
	  issue ASTERISK-19551) ........ Merged revisions 361329 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361330 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-05 17:22 +0000 [r361092-361279]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
	  uses the class if it's been defined There were a few instances of
	  restarting music on hold in meetme that would cause Asterisk to
	  revert to the default class of music on hold for no adequate
	  reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
	  Merged revisions 361269 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361270 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, addons/ooh323cDriver.c: Fix some stuff involving calls to
	  memcpy and memset The important parts of the patch were already
	  applied through other updates. (closes issue ASTERISK-19445)
	  Reported by: Makoto Dei Patches: memset-memcpy-length.patch
	  uploaded by Makoto Dei (license 5027) ........ Merged revisions
	  361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 361211 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, funcs/func_devstate.c: Make 'help devstate change' display
	  properly (get rid of excess comma) (closes issue ASTERISK-19444)
	  Reported by: Makoto Dei Patches:
	  devstate-change-usage-truncate.patch uploaded by Makoto Dei
	  (license 5027) ........ Merged revisions 361201 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361208 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
	  channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
	  apps/app_externalivr.c, channels/chan_iax2.c,
	  res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU
	  old-style field designator extensions to fix clang warnings
	  (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
	  clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
	  ........ Also add from the patch the portion in res_fax_spandsp
	  that didn't apply to 1.8 Merged revisions 361142 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
	  ASTERISK-19540) ........ Merged revisions 361143 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
	  nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
	  by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
	  ........ Merged revisions 361090 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361091 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-03 20:14 +0000 [r361042]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_transfer.c: Fix the display of documentation for
	  Transfer This came up while fixing documentation generation for
	  many other cases where the argument separator was not being
	  displayed properly. Now that it is displayed properly, it shows
	  up in the wrong place for Transfer since the '/' is only required
	  if Tech is present. (related to issue ASTERISK-18168) ........
	  Merged revisions 361040 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 361041 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-04-03 20:03 +0000 [r361038-361039]  Mark Murawki <markm@intellasoft.net>

	* include/asterisk/manager.h: Fix dev-mode compiler warning about
	  gnu_printf (related to ASTERISK-19575)

	* main/channel.c, main/manager.c, main/utils.c,
	  include/asterisk/channel.h, include/asterisk/strings.h, CHANGES,
	  include/asterisk/manager.h: Allow the Hangup manager action to
	  match channels by regex * Hangup now can take a regular
	  expression as the Channel option. If you want to hangup multiple
	  channels, use /regex/ as the Channel option. Existing behavior to
	  hanging up a single channel is unchanged, but if you pass a
	  regex, the manager will send you a list of channels back that
	  were hung up. (closes issue ASTERISK-19575) Reported by: Mark
	  Murawski Tested by: Mark Murawski

2012-04-02 22:27 +0000 [r360994]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
	  This change prevents Asterisk from sending RTCP receiver reports
	  during a remote bridge since it is no longer receiving media and
	  should not be reporting anything. (related to ASTERISK-19366)
	  ........ Merged revisions 360987 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360993 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-30 21:38 +0000 [r360935]  Richard Mudgett <rmudgett@digium.com>

	* /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
	  logger_thread() had an exit path that failed to release the
	  logmsgs list lock. * Make logger_thread() exit path unlock the
	  logmsgs list lock. * Made ast_log() not queue any messages to the
	  logmsgs list if the close_logger_thread flag is set. (issue
	  ASTERISK-19463) Reported by: Matt Jordan ........ Merged
	  revisions 360933 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360934 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-29 23:36 +0000 [r360872-360886]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: Fix potential race condition during call
	  pickup. Prior to this patch, a connected line update was queued
	  during call pickup and then an answer frame was queued. The
	  original caller would presumably then have his connected line
	  updated and then the call would be answered. In actuality, the
	  answer frame was not how the call ended up being answered.
	  Rather, an odd section in app_dial that checks if the called
	  channel's state is up. The result is that the order of the
	  connected line update and the answer were variable. In most
	  cases, this wasn't actually a bad thing. However, if the 'I'
	  option was passed to dial, the connected line update would be
	  inhibited. The fix is to queued the connected line after the
	  answer frame is queued. This way the race in app_dial is between
	  two conditions resulting in an answer. This way the connected
	  line update occurs after the answer every time. (closes issue
	  ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
	  Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
	  Mark Michelson (license 5049) ........ Merged revisions 360884
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 360885 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Improve accuracy of identifying
	  information sent in dialog-info SIP NOTIFY requests. This change
	  makes use of connected party information in addition to caller ID
	  in order to populate local and remote XML elements in the
	  dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by:
	  Maciej Krajewski Tested by: Maciej Krajewski Patches:
	  local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
	  ........ Merged revisions 360862 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360863 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-29 21:57 +0000 [r360827]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h, main/astobj2.c: Misc changes to make
	  astobj2 enhancement diffs easier to follow. * Rename astobj2 API
	  parameter funcname to func. * Rename astobj2 API iterator
	  parameter to iter. * Update some documentation for OBJ_MULTIPLE.

2012-03-29 20:01 +0000 [r360785-360787]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/logger.h, main/dial.c, main/pbx.c,
	  include/asterisk/bridging.h, main/features.c, main/logger.c,
	  CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
	  Introducing the log message unique call identifiers feature Log
	  messages will now display a call number that they are tied to
	  (ordered for calls based on when they started). This feature is
	  made to be minimally invasive without requiring changes to many
	  of the existing log messages. These IDs won't show up for verbose
	  messages on CLI (but they will in log files) This is currently in
	  phase II of production, see more about this feature on the wiki
	  --
	  https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
	  Review: https://reviewboard.asterisk.org/r/1823/

	* include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
	  include/asterisk/bridging.h, main/features.c, main/logger.c,
	  CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
	  undoing 360785 due to merging mistake

	* include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
	  include/asterisk/bridging.h, main/features.c, main/logger.c,
	  CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
	  Introducing the log message unique call identifiers feature Log
	  messages will now display a call number that they are tied to
	  (ordered for calls based on when they started). This feature is
	  made to be minimally invasive without requiring changes to many
	  of the existing log messages. These IDs won't show up for verbose
	  messages on CLI (but they will in log files) This is currently in
	  phase II of production, see more about this feature on the wiki
	  --
	  https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
	  Review: https://reviewboard.asterisk.org/r/1823/

2012-03-28 19:39 +0000 [r360724]  Terry Wilson <twilson@digium.com>

	* channels/chan_jingle.c, addons/chan_ooh323.c,
	  cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
	  channels/chan_gtalk.c, apps/confbridge/conf_config_parser.c: Fix
	  setting CDR variables in the hangup extension A previous CDR fix
	  for setting CDR variables during a bridge via custom dialplan
	  features broke setting CDR variables in the hangup extension.
	  This patch fixes the issue. Review:
	  https://reviewboard.asterisk.org/r/1794/ ........ Merged
	  revisions 358978 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358989 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-27 18:44 +0000 [r360673]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Make a debug message regarding
	  subscription changes more accurate. I was getting confused during
	  some testing why Asterisk was saying that a subscription was
	  being added when it was clearly being removed. This fixes that
	  confusion. ........ Merged revisions 360625 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360672 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-27 17:13 +0000 [r360626-360627]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
	  Add global ao2 array container. Global ao2 objects must always
	  exist after initialization because there is no access control to
	  obtain another reference to the global object. It is expected
	  that module configuration could use these new API calls to
	  replace an active configuration parameter object with an updated
	  configuration parameter object. With these new API calls, the
	  global object could be replaced, removed, or referenced without
	  the risk of someone using a stale global object pointer. Review:
	  https://reviewboard.asterisk.org/r/1824/

	* main/astobj2.c: Attempt to be more helpful when using a bad ao2
	  object pointer.

2012-03-27 14:43 +0000 [r360576]  Jonathan Rose <jrose@digium.com>

	* /, configure: Updates config with bootstrap where I changed
	  configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
	  Clark ........ Merged revisions 360574 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360575 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-26 21:22 +0000 [r360536]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/dnsmgr.c, /: Convert ast_verb() to ast_debug() and increase
	  log level Rather then flood the CLI with verbose messages, we've
	  changed the level to debug. This will help keep the CLI clean.

2012-03-26 19:49 +0000 [r360490]  Jonathan Rose <jrose@digium.com>

	* /, configure.ac: Fix BETTER_BACKTRACES library detection for
	  Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
	  Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
	  Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
	  uploaded by Bryon Clark (license 6157) ........ Merged revisions
	  360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 360489 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-24 23:49 +0000 [r360359-360415]  Russell Bryant <russell@russellbryant.com>

	* funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
	  handling code path. ........ Merged revisions 360413 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360414 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_iax2.c: chan_iax2: Use OBJ_NODATA to be a bit more
	  explicit. This is just a minor code cleanup change. These uses of
	  ao2_callback() would never return anything since the callbacks
	  always returned 0. However, be more explicit that no returned
	  results are wanted by specifying OBJ_NODATA.

	* /, apps/app_page.c: app_page: Fix a memory leak on every Page().
	  dial_list is a dynamically allocated array that is allocated at
	  the beginning of Page() based on how many devices will be dialed.
	  This was never being freed. ........ Merged revisions 360363 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360364 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_jack.c: app_jack: fix datastore memory leak in error
	  handling path. ........ Merged revisions 360360 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360361 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y,
	  main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
	  main/ast_expr2.c: Multiple revisions 360356-360357 ........
	  r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
	  | 6 lines expression parser: Fix (theoretical) memory leak. Fix a
	  memory leak that is very unlikely to actually happen. If a
	  malloc() succeeded, but the following strdup() failed, the memory
	  from the original malloc() would be leaked. ........ r360357 |
	  russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
	  Rebuild parsers. This is needed to include the last fix to
	  main/ast_expr2.y. The changes look much bigger as this
	  regeneration of the code was done with newer versions of flex and
	  bison. ........ Merged revisions 360356-360357 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360358 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-24 00:40 +0000 [r360264-360311]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /, channels/sig_pri.c: Make number not available
	  presentation also set screening to network provided. Q.951
	  indicates that when the presentation indicator is "Number not
	  available due to interworking" for a number then the screening
	  indicator field should be "Network provided". * Made
	  ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
	  when the presentation is "Number not available due to
	  interworking". This fix makes Asterisk consistent and it also
	  makes it consistent with earlier branches as far as this
	  presentation value is concerned. * Made pri_to_ast_presentation()
	  and ast_to_pri_presentation() conversions handle the "Number not
	  available due to interworking" case better in sig_pri.c. This
	  change is possible because the minimum required libpri version
	  (v1.4.11) has the necessary defines in libpri.h. ........ Merged
	  revisions 360309 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360310 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Add missing initialization of
	  update_redirecting in chan_sip.c ........ Merged revisions 360262
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 360263 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-22 21:25 +0000 [r360227]  Jonathan Rose <jrose@digium.com>

	* apps/app_dial.c, include/asterisk/utils.h, main/features.c,
	  main/utils.c, CHANGES, apps/app_queue.c: Adds F option to Bridge
	  application Similar to dial and queue F option. (Closes issue
	  ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff
	  uploaded by To (license 6347) Review:
	  https://reviewboard.asterisk.org/r/1825/

2012-03-22 19:51 +0000 [r360190]  Kinsey Moore <kmoore@digium.com>

	* main/udptl.c, main/stdtime/test.c, main/autoservice.c,
	  main/rtp_engine.c, main/frame.c, main/fskmodem_float.c,
	  main/sha1.c, main/say.c, main/ecdisa.h, main/utils.c,
	  main/devicestate.c, main/taskprocessor.c, main/indications.c,
	  main/enum.c, main/config.c, main/loader.c, main/term.c,
	  main/cli.c, main/io.c, main/ulaw.c, main/channel.c, main/dial.c,
	  main/manager.c, main/tdd.c, main/strcompat.c, main/plc.c,
	  main/features.c, main/logger.c, main/fskmodem_int.c, main/app.c,
	  main/stdtime/localtime.c, main/image.c, main/dns.c,
	  main/message.c, main/md5.c, main/sched.c, main/lock.c,
	  main/pbx.c, main/dnsmgr.c, main/slinfactory.c, main/translate.c,
	  main/jitterbuf.c, main/cel.c, main/chanvars.c, main/netsock.c,
	  main/srv.c, main/privacy.c, main/fixedjitterbuf.c, main/file.c,
	  main/callerid.c, main/event.c, main/astmm.c, main/audiohook.c,
	  main/cygload.c, main/fixedjitterbuf.h, main/asterisk.c,
	  main/xmldoc.c, main/dsp.c, main/timing.c: Kill off red blobs in
	  most of main/* Everything still compiled after making these
	  changes, so I assume these whitespace-only changes didn't break
	  anything (and shouldn't have).

2012-03-21 14:55 +0000 [r360140]  Jonathan Rose <jrose@digium.com>

	* /, contrib/scripts/install_prereq: Update install_prereq script
	  to include missing GSM library for debian amd move SQLite3.
	  (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
	  debian_install_prereq.diff uploaded by Andrew Latham (license
	  5985) ........ Merged revisions 360138 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360139 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-21 14:47 +0000 [r360137]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, configure, configure.ac: Also detect gmime 2.6 Also detect
	  gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
	  (License #5035) <tzafrir.cohen@xorcom.com> ........ Merged
	  revisions 360087 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360098 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-21 13:31 +0000 [r360089]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
	  on the final response to a re-INVITE When Asterisk detects a
	  hangup and cannot send a BYE due to a pending INVITE, it sets the
	  pendingbye flag and waits for the final response to that INVITE.
	  When the response is received, it transmits the BYE. If, however,
	  that INVITE request is a pending re-INVITE, it needs to first
	  send a CANCEL request to terminate the pending re-INVITE. In that
	  circumstance, Asterisk was, in some scenarios, clearing the
	  pendingbye flag after processing the CANCEL request and not
	  checking for a pending BYE when receiving the final 487 response
	  to the INVITE. This patch ensures that if the pendingbye flag is
	  set, it is honored regardless of the nature of the INVITE request
	  currently in flight. (closes issue ASTERISK-19365) Reported by:
	  Thomas Arimont Tested by: Thomas Arimont Patches:
	  bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
	  6283) Review: https://reviewboard.asterisk.org/r/1807 ........
	  Merged revisions 360086 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360088 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-20 20:42 +0000 [r360036]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_echo.c: Prevent Echo() from relaying control, null,
	  and modem frames Echo()'s description states that it echoes
	  audio, video, and DTMF except for # while it actually echoes any
	  frame that it receives other than DTMF #. This was causing frame
	  storms in the test suite in some circumstances where Echo() was
	  attached to both ends of a pair of local channels and control
	  frames were being periodically generated. Echo()'s behavior and
	  description have been modifed so that it only echoes media and
	  non-# DTMF frames. ........ Merged revisions 360033 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 360034 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-20 18:17 +0000 [r359983]  Sean Bright <sean@malleable.com>

	* /, UPGRADE.txt, channels/chan_iax2.c, include/asterisk/manager.h:
	  chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus
	  AMI Events The PeerStatus event for IAX2 channels currently
	  includes a header named Post which should have been Port. Post
	  was removed and the AMI version has been updated to 1.3. ........
	  Merged revisions 359982 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-20 17:31 +0000 [r359942-359981]  Richard Mudgett <rmudgett@digium.com>

	* main/data.c, main/pbx.c, main/manager.c, /, main/features.c,
	  include/asterisk/manager.h, main/db.c: Allow AMI action callback
	  to be reentrant. Fix AMI module reload deadlock regression from
	  ASTERISK-18479 when it tried to fix the race between calling an
	  AMI action callback and unregistering that action. Refixes
	  ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2
	  object guaranteed that there were no active callbacks that
	  mattered when ast_manager_unregister() was called. Unfortunately,
	  this causes the deadlock situation. The patch stops locking the
	  ao2 object to allow multiple threads to invoke the callback
	  re-entrantly. There is no way to guarantee a module unload will
	  not crash because of an active callback. The code attempts to
	  minimize the chance with the registered flag and the maximum 5
	  second delay before ast_manager_unregister() returns. The trunk
	  version of the patch changes the API to fix the race condition
	  correctly to prevent the module code from unloading from memory
	  while an action callback is active. * Don't hold the lock while
	  calling the AMI action callback. (closes issue ASTERISK-19487)
	  Reported by: Philippe Lindheimer Review:
	  https://reviewboard.asterisk.org/r/1818/ Review:
	  https://reviewboard.asterisk.org/r/1820/ ........ Merged
	  revisions 359979 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359980 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* res/res_mutestream.c: Convert MuteAudio documentation to XML. *
	  Added missing error exits with cause in manager_mutestream(). *
	  Cleaned up manager_mutestream() and func_mute_write(). * Some
	  whitespace and comment cleanup.

2012-03-16 21:00 +0000 [r359905]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
	  channels This patch addresses a bug with chanspy on local
	  channels which roughly 50% of the time would create a situation
	  where chanspy can latch onto a zombie channel, keeping the zombie
	  alive forever and causing the channel doing the spying to never
	  be able to hang up. (closes issue ASTERISK-19493) Reported by:
	  lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
	  Merged revisions 359892 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359898 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-16 20:37 +0000 [r359904]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/app.h, main/app.c: Simplify some code in
	  ast_app_run_sub(). * Remove unnnecessary const from const char *
	  const var declaration in the ast_app_run_macro() and
	  ast_app_run_sub() prototypes. The second const is unnecessary.

2012-03-16 15:38 +0000 [r359857]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES:
	  Revert the pre-dial addition. The code may be just fine, but it
	  had not received a "ship it!" on review board yet.

2012-03-16 08:27 +0000 [r359811]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
	  uint32_t change from Review:
	  https://reviewboard.asterisk.org/r/1699/ ........ Merged
	  revisions 359809 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359810 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-15 20:11 +0000 [r359772]  Mark Murawki <markm@intellasoft.net>

	* main/pbx.c: Fix warning from commit r359705 (predial options for
	  app_dial)

2012-03-15 19:11 +0000 [r359708]  Matthew Jordan <mjordan@digium.com>

	* /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
	  manager There exists a remotely exploitable stack buffer overflow
	  in HTTP digest authentication handling in Asterisk. The
	  particular method in question is only utilized by HTTP AMI. When
	  parsing the digest information, the length of the string is not
	  checked when it is copied into temporary buffers allocated on the
	  stack. This patch fixes this behavior by parsing out pre-defined
	  key/value pairs and avoiding unnecessary copies to the stack.
	  (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
	  by: Matt Jordan ........ Merged revisions 359706 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359707 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-15 18:58 +0000 [r359705]  Mark Murawki <markm@intellasoft.net>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
	  options PreDial options 'b' and 'B' to app_dial * Added 'b' and
	  'B' options to Dial. These options will allow you to run
	  last-minute dialplan on the caller and callee channels while the
	  Dial application is executing, but before the call is started.
	  For example you can use the 'b' option to run dialplan on the
	  callee channel to get the name of the newly created channel right
	  away. Review: https://reviewboard.asterisk.org/r/1229/ (closes
	  issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark
	  Murawski, Stefan Schmidt

2012-03-15 18:55 +0000 [r359704]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun
	  in Milliwatt Milliwatt is vulnerable to a remotely exploitable
	  stack overrun when using the 'o' option. This occurs due to the
	  milliwatt_generate function not accounting for
	  AST_FRIENDLY_OFFSET when calculating the maximum number of
	  samples it can put in the output buffer. This patch resolves this
	  issue by taking into account AST_FRIENDLY_OFFSET when determining
	  the maximum number of samples allowed. Note that at no point is
	  remote code execution possible. The data that is written into the
	  buffer is the pre-defined Milliwatt data, and not custom data.
	  (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
	  by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
	  Russell Bryant (license 6283) Note that this patch was written by
	  Russell, even though Matt uploaded it ........ Merged revisions
	  359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
	  ........ Merged revisions 359656 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359694 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-15 18:34 +0000 [r359651]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Remove unused variable ‘srch’ Missed on the
	  previous commit

2012-03-15 18:32 +0000 [r359644]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, /, apps/app_queue.c: Add missing connected line
	  macro calls to initial dial for Dial and Queue apps. The
	  connected line interception macros do not get executed when the
	  outgoing channel is initially created and that channel's
	  caller-id is implicitly imported into the incoming channel's
	  connected line data. If you are using the interception macros,
	  you would expect that they get run for every change to a
	  channel's connected line information outside of normal dialplan
	  execution. Review: https://reviewboard.asterisk.org/r/1817/
	  ........ Merged revisions 359609 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359620 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-15 17:36 +0000 [r359607]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Remove some dead code found in
	  _sip_show_peers() Review:
	  https://reviewboard.asterisk.org/r/1696/

2012-03-15 00:54 +0000 [r359456-359560]  Russell Bryant <russell@russellbryant.com>

	* /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
	  sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
	  try_transfer() so that the code isn't (potentially) trying to
	  read from it while uninitialized. ........ Merged revisions
	  359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 359559 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of
	  uninitialized variable. Avoid potential use of idroster in
	  gtalk_alloc() before it has been initialized. ........ Merged
	  revisions 359508 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359509 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_chanisavail.c: app_chanisavail: Fix use of
	  uninitialized variable. Ensure that status is set before it is
	  used by resetting it during each loop iteration. This could have
	  resulted in incorrect results from this app. ........ Merged
	  revisions 359486 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359491 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is
	  initialized. Scan results indicated that this array could be used
	  uninitialized. At a quick look, it looks correct. In any case,
	  initializing it is a Good Thing (tm). ........ Merged revisions
	  359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 359458 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* include/asterisk/app.h, /: app.h: Always initialize
	  AST_DECLARE_APP_ARGS(). This patch ensures that the struct
	  defined by AST_DECLARE_APP_ARGS() is always fully initialized.
	  I'm not sure if this fixes any real bugs, but it silences a bunch
	  of warnings from coverity, and is generally a good thing to do
	  anyway. ........ Merged revisions 359452 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359454 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-14 22:38 +0000 [r359455]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /, channels/chan_agent.c,
	  include/asterisk/channel.h: Fix deadlock potential with some
	  ast_indicate/ast_indicate_data calls. Calling
	  ast_indicate()/ast_indicate_data() with the channel lock held can
	  result in a deadlock with a local channel because of how local
	  channels need to avoid deadlock. ........ Merged revisions 359451
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 359453 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-14 18:56 +0000 [r359406]  Matthew Jordan <mjordan@digium.com>

	* tests/test_jitterbuf.c (added): Add tests for main/jitterbuf.c
	  This patch adds unit tests for main/jitterbuf.c. This includes
	  checking for the following: * Nominal insertion and retrieval of
	  frames * Insertion and retrieval of frames where the frames are
	  inserted out of order with respect to the previous frame *
	  Insertion and retrieval of frames where some number of frames
	  that would occur in the expected sequence are instead dropped *
	  Insertion and retrieval of frames with an arrival time that does
	  not occur at the same rate as the surrounding frames *
	  Resynchronization of the jitter buffer when an inserted frame
	  breaks the resynchronization threshold * Overfilling of the
	  jitter buffer For each of the tests, both JB_TYPE_VOICE and
	  JB_TYPE_CONTROL permutations exist. Review:
	  https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964)
	  Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan

2012-03-14 18:12 +0000 [r359360]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel_internal.h: Three copies of the file
	  contents in channel_internal.h are a bit excessive.

2012-03-14 17:48 +0000 [r359359]  Matthew Jordan <mjordan@digium.com>

	* /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
	  missed resynchronizations When a change in time occurs, such that
	  the timestamps associated with frames being placed into an
	  adaptive jitter buffer (implemented in jitterbuf.c) are
	  significantly different then the previously inserted frames, the
	  jitter buffer checks to see if it needs to be resynched to the
	  new time frame. If three consecutive packets break the threshold,
	  the jitter buffer resynchs itself to the new timestamps. This
	  currently only occurs when history is calculated, and hence only
	  on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
	  hand, are never passed to the history calculations. Because of
	  this, if the jump in time is greater then the maximum allowed
	  length of the jitter buffer, the JB_TYPE_CONTROL frames are
	  dropped and no resynchronization occurs. Alterntively, if the
	  overfill logic is not triggered, the JB_TYPE_CONTROL frame will
	  be placed into the buffer, but with a time reference that is not
	  applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
	  the overflow logic until reads from the jitter buffer reach the
	  errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
	  frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
	  are unlikely to occur in multiples, it perform the
	  resynchronization on any JB_TYPE_CONTROL frame that breaks the
	  resynch threshold. Note that this only impacts chan_iax2, as
	  other consumers of the adaptive jitter buffer use the abstract
	  jitter buffer API, which does not use JB_TYPE_CONTROL frames.
	  Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
	  ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
	  Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
	  (license 5722) ........ Merged revisions 359356 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359358 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-14 17:39 +0000 [r359357]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
	  forked calls generating warnings for voice frames. When connected
	  line support was added, the wait_for_answer() variable single
	  changed its meaning slightly. Unfortunately, the places where
	  single was used did not necessarily get updated to reflect that
	  change. Also audio/video frames were sent to all forked calls
	  when the endpoints were never made compatible. * Don't pass
	  audio/video media frames when the channels have not been made
	  compatible. * Added handling of AST_CONTROL_SRCCHANGE to
	  app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
	  because that frame can also pass a requested MOH class. (closes
	  issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
	  ASTERISK-17541) Reported by: clint Review:
	  https://reviewboard.asterisk.org/r/1805/ ........ Merged
	  revisions 359344 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359355 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-14 14:40 +0000 [r359306]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/astobj2.h: Force non-inlining of
	  ao2_iterator_destroy when TEST_FRAMEWORK is enabled In r357272,
	  astobj2 was changed to automatically enable REF_DEBUG when the
	  TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers
	  (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy
	  in handle_astobj2_test. This by itself is not a problem;
	  unfortunately, the compiler believes that there is a code path
	  wherein an object allocated on the stack will be free'd. As
	  warnings are treated as errors, this prevents compilation of
	  astobj2. This patch works around that by adding the noinline
	  attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK
	  flag is enabled. Preventing inlining is only needed for the test
	  method defined in astobj2, which is also only enabled if
	  TEST_FRAMEWORK is enabled.

2012-03-14 10:56 +0000 [r359052-359261]  Russell Bryant <russell@russellbryant.com>

	* include/asterisk/logger.h, /, main/logger.c: Fix bogus
	  reads/writes of console log levels in asterisk.c This patch
	  updates the NUMLOGLEVELS define in logger.h to 32, to match the
	  fact that logger.c implements 32 log levels (because of the
	  custom log level stuff). asterisk.c uses this define to size an
	  array of levels per remote console. This array is modified in
	  ast_console_toggle_loglevel(), which is called by the "logger set
	  level" CLI command. While the documentation for the CLI command
	  doesn't make it terribly obvious, you can use this CLI command to
	  toggle a custom log level on a remote console, as well. However,
	  doing so led to an invalid array index in asterisk.c. This array
	  is read from any time a log message is written to a console. So,
	  all custom log level messages resulted in a bogus read if a
	  remote console was connected. ........ Merged revisions 359259
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 359260 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
	  reads/writes due to incorrect sizeof(). These few places in the
	  code used sizeof() on h_addr in struct hostent. This is
	  sizeof(char *). The correct way to get the size of this address
	  is to use h_length. This error would result in reads/writes of 8
	  bytes instead of 4 on 64-bit machines. ........ Merged revisions
	  359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 359212 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/sched.c: Fix inaccurate sizeof() in sched.c. This code
	  just needed sizeof(int), not sizeof(int *). ........ Merged
	  revisions 359157 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359162 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, utils/astman.c: Fix incorrect sizeof() in astman. ........
	  Merged revisions 359116 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359117 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_crypto.c: Fix incorrect usage of sizeof() in
	  res_crypto. In this case, just remove the memset(). There was a
	  redundant memset that is done correctly just 2 lines later.
	  ........ Merged revisions 359110 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359114 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
	  ........ Merged revisions 359088 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359091 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/features.c: Fix incorrect sizeof() usage in features.c.
	  This didn't actually result in a bug anywhere, luckily. The only
	  place where the result of these memcpys was used is in app_dial,
	  and the only field that it read out of ast_call_feature was the
	  first one, which is an int, so these memcpys always copied just
	  enough to avoid a problem. ........ Merged revisions 359069 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359072 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
	  ........ Merged revisions 359059 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359060 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/pbx.c, /: Don't use a buffer after it goes out of scope. 's'
	  is set to 'workspace'. Make sure 'workspace' doesn't go out of
	  scope while the reference to it via 's' is still used. ........
	  Merged revisions 359056 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359057 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
	  apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These
	  modules are being maintained outside of the tree and have been
	  for a long time now, so it doesn't make sense to keep them here.
	  Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged
	  revisions 359050 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 359051 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-13 21:24 +0000 [r359011]  Terry Wilson <twilson@digium.com>

	* include/asterisk/channel_internal.h (added): Add missing
	  channel_internal.h ...again.

2012-03-13 21:18 +0000 [r358997]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add ability
	  for chan_dahdi ISDN to block connected line updates per span.
	  Added new chan_dahdi.conf colp_send option parameter to block
	  connected line updates per span. (closes issue ASTERISK-17025)
	  Reported by: Michael Smith

2012-03-13 20:43 +0000 [r358907-358993]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: Fix setting CDR variables in the hangup
	  extension A previous CDR fix for setting CDR variables during a
	  bridge via custom dialplan features broke setting CDR variables
	  in the hangup extension. This patch fixes the issue. Review:
	  https://reviewboard.asterisk.org/r/1794/ ........ Merged
	  revisions 358978 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358989 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* include/asterisk/devicestate.h, /, channels/chan_sip.c,
	  tests/test_devicestate.c, main/devicestate.c: Make hints for
	  invalid SIP devices return Unavail, not idle This patch
	  drastically simplifies the device state aggegation code. The old
	  method was not only overly complex, but also made it impossible
	  to return AST_DEVICE_INVALID from the aggregation code. The unit
	  test update is as a result of fixing that bug. The SIP change
	  stems from a bug introduced by removing a DNS lookup for
	  hostname-based SIP channels. (closes issue ASTERISK-16702)
	  Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged
	  revisions 358943 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358944 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_voicemail.c: Fix IMAP storage compilation after
	  opaquification changes (closes issue ASTERISK-19513)

	* channels/chan_unistim.c, main/autoservice.c,
	  channels/chan_vpb.cc, channels/chan_local.c, main/rtp_engine.c,
	  res/res_musiconhold.c, bridges/bridge_multiplexed.c,
	  apps/app_followme.c, main/indications.c, main/cli.c,
	  main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, main/manager.c, main/features.c,
	  apps/app_dumpchan.c, res/res_agi.c, main/app.c,
	  apps/app_confbridge.c, apps/app_externalivr.c, main/bridging.c,
	  apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c,
	  channels/chan_sip.c, channels/chan_bridge.c,
	  main/channel_internal_api.c, channels/chan_agent.c,
	  apps/app_disa.c, include/asterisk/channel.h,
	  apps/app_talkdetect.c, apps/app_queue.c, apps/app_speech_utils.c,
	  apps/app_channelredirect.c, main/file.c, res/snmp/agent.c,
	  apps/app_macro.c, apps/app_stack.c, apps/app_chanspy.c,
	  apps/app_mixmonitor.c: Finalize ast_channel opaquification
	  Review: https://reviewboard.asterisk.org/r/1786/

2012-03-13 17:01 +0000 [r358858-358861]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix crash caused by opaquification change
	  -r356042. The set_format() function was more subtle in how it
	  modified the struct ast_channel readtrans/writetrans values. *
	  Fixed ast_activate_generator() conversion correctly. (closes
	  issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by:
	  rmudgett

	* main/format.c: Use struct copy instead of memcpy().

2012-03-13 08:06 +0000 [r358812]  Tilghman Lesher <tilghman@meg.abyt.es>

	* res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
	  utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable
	  macros in 1.8 to find the next highest "h" extension in a
	  context, like in 1.4. This change restores functionality that was
	  present in 1.4, when AEL macros were implemented with the Macro
	  dialplan application. Macros are fraught with functionality
	  issues, because they consume a large portion of the underlying
	  application stack. This limits the ability of AEL users to call
	  many layers of subroutines, an issue which Gosub does not have
	  (originally tested to 100,000 levels deep). Therefore, starting
	  in 1.6.0, AEL macros were implemented with Gosub. However, there
	  were some implicit behaviors of Macro, which were not replicated
	  at the same time as with the transition to Gosub, one of which is
	  documented in the related issue. In particular, the "h" extension
	  is designed to execute not in the Macro context, but in the
	  topmost calling context. Due to legacy issues with a misapplied
	  bugfix many years ago, when a macro exited in 1.4, it looks in
	  all calling contexts, bubbling up from the deepest level until it
	  finds an "h" extension. Since AEL hides the complexity of the
	  underlying dialplan logic from the AEL programmer, it's
	  reasonable to assume that this behavior should not change in the
	  transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
	  break working AEL configurations in the transition to Asterisk
	  1.8 LTS. This fix is the result, which implements a search for
	  the "h" extension in all calling Gosub contexts. Fixes
	  ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
	  (License #5003) by Tilghman Lesher (with slight modifications for
	  1.8) Tested by: Johan Wilfer Review:
	  https://reviewboard.asterisk.org/r/1776/ ........ Merged
	  revisions 358810 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358811 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-12 17:01 +0000 [r358766]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c, contrib/unistimLang/ru.po (added),
	  contrib/unistimLang/ru.po.utf8 (added),
	  configs/unistim.conf.sample, UPGRADE.txt, CHANGES,
	  contrib/unistimLang/en.po (added), contrib/unistimLang (added):
	  Massive changes in chan_unistim channel driver. Include many
	  fixes in channel driver operation and add additional
	  functionality: * Added ability to use multiple lines on phone, so
	  for one device in configuration multiple lines can be defined, it
	  allows to have multiple calls on one phone, callwaiting and
	  switching between calls. * Added ability for translation
	  on-screen menu to multiple languages. Tested on Russian
	  languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO
	  8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by
	  'language' and on-screen menu of phone * Other described in
	  CHANGES file Testing done by issue tracker users: ibercom,
	  scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production
	  system by Jonn Taylor (jonnt) using phone models: Nortel i2004,
	  1120E and 1140E. (closes issue ASTERISK-16890) Review:
	  https://reviewboard.asterisk.org/r/1243/

2012-03-10 20:06 +0000 [r358730]  Joshua Colp <jcolp@digium.com>

	* configs/confbridge.conf.sample, main/dial.c, apps/app_page.c,
	  apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
	  include/asterisk/dial.h, CHANGES,
	  apps/confbridge/conf_config_parser.c: Transition app_page to
	  using app_confbridge internally for the conference bridge portion
	  of paging. This also adds a new 'announcement' option to
	  ConfBridge user profiles. Review:
	  https://reviewboard.asterisk.org/r/1754/

2012-03-08 17:48 +0000 [r358646-358691]  Sean Bright <sean@malleable.com>

	* apps/app_dial.c, apps/app_directory.c, apps/app_queue.c: Resolve
	  a few more cases of variable shadowing.

	* channels/chan_phone.c, channels/chan_skinny.c,
	  channels/chan_agent.c, pbx/pbx_lua.c, pbx/pbx_dundi.c,
	  channels/chan_gtalk.c, pbx/pbx_config.c, channels/chan_oss.c,
	  apps/confbridge/conf_config_parser.c: Eliminate a bunch of shadow
	  warnings.

	* include/asterisk/linkedlists.h: Add some underscores in a few of
	  our llist macros to reduce name collisions.

2012-03-08 16:59 +0000 [r358645]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Make transfer not ignore port information
	  with SIP. Attempting to transfer with SIP to an address like
	  1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
	  the host string and ignored. This simply keeps chan_sip from
	  cutting off the port number during these kinds of transfers.
	  (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
	  https://reviewboard.asterisk.org/r/1790/diff/#index_header
	  ........ Merged revisions 358643 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358644 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-08 16:21 +0000 [r358609-358622]  Sean Bright <sean@malleable.com>

	* Makefile, configure, configure.ac, makeopts.in: Add
	  --enable-dev-mode=strict to configure. Passing -Wshadow to gcc
	  enables shadow warnings. From the gcc manual: Warn whenever a
	  local variable or type declaration shadows another variable,
	  parameter, type, or class member (in C++), or whenever a built-in
	  function is shadowed. Asterisk will not currently compile with
	  this option set, but a number of bugs have been discovered by
	  enabling this flag on specific files. The long-term goal is to
	  eliminate all of the suspect code that causes this warning to be
	  emitted.

	* Makefile: Whitespace only change to the Makefile

2012-03-07 21:28 +0000 [r358576]  Terry Wilson <twilson@digium.com>

	* cel/cel_odbc.c, configs/cel_odbc.conf.sample: Handle numeric
	  columns for eventtype properly in cel_odbc Patch also implements
	  correct handling of datetime2 and datetimeoffset new datatypes in
	  SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548)
	  Review: https://reviewboard.asterisk.org/r/1160/ Review:
	  https://reviewboard.asterisk.org/r/1804/

2012-03-07 18:33 +0000 [r358532]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_ss7.c: Change directly setting _softhangup in
	  sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
	  ASTERISK-19372) ........ Merged revisions 358530 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358531 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-07 16:16 +0000 [r358486]  Sean Bright <sean@malleable.com>

	* /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
	  number of samples set properly. If the wctc4xxp returns more than
	  a single packet, we need to update the number of samples in the
	  returned frame accordingly. Acked-by: Shaun Ruffell
	  <sruffell@digium.com> ........ Merged revisions 358484 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358485 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-07 15:19 +0000 [r358437-358444]  Terry Wilson <twilson@digium.com>

	* /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
	  cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 358441 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for
	  ODBC WCHAR fields Without detecting these types, cel_odbc blows
	  up when the character set for the table is utf8. This also wraps
	  cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
	  #ifdef seen in other parts of the code. ........ Merged revisions
	  358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 358436 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-06 17:47 +0000 [r358262-358379]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
	  calls on FXS ports. * Fix referencing the wrong variable in
	  chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
	  compiling with -Wshadow and finding this bug. ........ Merged
	  revisions 358377 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358378 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
	  Add dialtone_detect option for analog incoming calls. For analog
	  lines, enables Asterisk to use dialtone detection per channel if
	  an incoming call was hung up before it was answered. If dialtone
	  is detected, the call is hung up. no: Disabled. (Default) yes:
	  Look for dialtone for 10000 ms after answer. <number>: Look for
	  dialtone for the specified number of ms after answer. always:
	  Look for dialtone for the entire call. Dialtone may return if the
	  far end hangs up first. dialtone_detect=yes dialtone_detect=5000
	  dialtone_detect=always (closes issue ASTERISK-19316) Reported by:
	  Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy
	  Pepper Review: https://reviewboard.asterisk.org/r/1737/

	* /, channels/sig_ss7.c: Drop SS7 call if not connected yet when
	  INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
	  clear a failed call as soon as possible. * Made SS7 hangup a call
	  immediately if it has not connected yet for
	  INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
	  inband tone. (closes issue ASTERISK-19372) Reported by: Igor
	  Nikolaev ........ Merged revisions 358278 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358284 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* include/asterisk/channel.h: Make usage of
	  DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.

	* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
	  Setup DSP when SS7 call is connected or early media is available.
	  Outgoing SS7 calls fail to detect incoming DTMF so any bridged
	  channel that requires out-of-band DTMF will not work. * Added
	  sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
	  The new call converts conditionaled out unconverted code and
	  shows that the code really did something useful. * Improved some
	  chan_dahdi DTMF debug messages to help track DTMF handling.
	  (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........
	  Merged revisions 358260 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358261 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-05 19:06 +0000 [r358216]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: Eliminate double close of file descriptor in
	  manager.c The process_output function in manager.c attempted to
	  call fclose and close immediately afterwards. Since fclose
	  implies close, this resulted in a potential double free on file
	  descriptors. This patch changes that behavior and also adds error
	  checking to fclose and close depending on which was deemed
	  necessary. Also error messages. Thanks to Rosen Iliev for
	  pointing out the location of the problem. (closes issue
	  ASTERISK-18453) Reported By: Jaco Kroon Review:
	  https://reviewboard.asterisk.org/r/1793/ ........ Merged
	  revisions 358214 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358215 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-05 16:44 +0000 [r358164]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Defer sending the connected line reinvite
	  if a reinvite is already in progress. (issue ASTERISK-19355)
	  Reported by: tomaso (closes issue AST-825) ........ Merged
	  revisions 358162 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358163 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-05 16:00 +0000 [r358117]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
	  on Replaces errors Asterisk was not setting pendinginvite in the
	  upper half of handle_request_invite such that the 4xx was
	  retransmitted repeatedly even though an ack was received for
	  every retransmission. (closes issue ASTERISK-19303) Reported by:
	  Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy
	  (license 6358) ........ Merged revisions 358115 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358116 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-05 11:20 +0000 [r358082]  Sean Bright <sean@malleable.com>

	* configs/iax.conf.sample: Tab to spaces and text change.

2012-03-02 23:29 +0000 [r357999-358038]  Terry Wilson <twilson@digium.com>

	* channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix
	  unused-but-set-variable warnings All of these were pretty
	  obviously unused. Some were unused because the code that used
	  them was #if 0'd. In those cases, I just commented out the
	  unused-but-set variables. ........ Merged revisions 358029 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358033 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /: Correct some set-but-unused variable warnings in the mISDN
	  library. (from kpfleming's commit to trunk r356292) ........
	  Merged revisions 358011 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 358017 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
	  mode x=++x and x=x=1? Really? ........ Merged revisions 357986
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 357987 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-02 21:06 +0000 [r357942]  Kinsey Moore <kmoore@digium.com>

	* /, main/ccss.c, tests/test_event.c, main/event.c,
	  include/asterisk/strings.h: Fix case-sensitivity for
	  device-specific event subscriptions and CCSS This change fixes
	  case-sensitivity for device-specific subscriptions such that the
	  technology identifier is case-insensitive while the remainder of
	  the device string is still case-sensitive. This should also
	  preserve the original case of the device string as passed in to
	  the event system. CCSS is the only feature affected as it is the
	  only consumer of device-specific event subscriptions. The second
	  part of this patch addresses similar case-sensitivity issues
	  within CCSS itself that prevented it from functioning correctly
	  after the fix to the events system. This adds a unit test to
	  verify that the event system works as expected. (closes issue
	  ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
	  ........ Merged revisions 357940 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357941 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-02 18:38 +0000 [r357896]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /, channels/sig_pri.c: Remove ISDN hold
	  restriction for non-bridged calls. The check if an ISDN call is
	  bridged before it could be placed on hold is not necessary and is
	  overly restrictive. The check was originally done to prevent
	  problems with call transfers in case a user tried to transfer a
	  call connected to an application to another call connected to an
	  application. The ISDN transfer code has not required this
	  restriction for quite some time because ECT could transfer any
	  two active calls to each other. * Remove ISDN hold restriction
	  for calls connected to applications. * Made
	  ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
	  AST_CONTROL_UNHOLD instead of generating a warning message.
	  (closes issue ASTERISK-19388) Reported by: Birger Harzenetter
	  Tested by: rmudgett ........ Merged revisions 357894 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357895 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-02 16:57 +0000 [r357861]  Jonathan Rose <jrose@digium.com>

	* apps/app_queue.c: Adds a transfer callee on hangup option (like
	  with Dial option F) to queues. This should (and does in my
	  testing) act just like the Dial option of the same name. This
	  allows a queue member to be transfered to the next priority (no
	  args), or to a context/extension/priority similar to goto (with
	  args context^extension^priority) when a caller hangs up on them.
	  (closes issue ASTERISK-19283) Reported by: To Patches:
	  queue_f-v3.diff uploaded by To (license 6347) Review:
	  https://reviewboard.asterisk.org/r/1785/

2012-03-02 16:26 +0000 [r357834]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_chanspy.c: Remove bad usage of goto in ChanSpy
	  next_channel().

2012-03-02 16:19 +0000 [r357821]  Sean Bright <sean@malleable.com>

	* configs/iax.conf.sample: Beef up the IAX2 sample configuration a
	  bit and fix some formatting issues.

2012-03-02 16:03 +0000 [r357814-357815]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. *
	  Fix next_channel() channel reference leak in ChanSpy. (closes
	  issue ASTERISK-19461) Reported by: Irontec Patches:
	  app_chanspy_iteartor_next_unref.patch (license #6213) patch
	  uploaded by Irontec (issue ASTERISK-17515) ........ Merged
	  revisions 357809 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357810 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_usbradio.c: Fix compile error from latest channel
	  opaquification change.

2012-03-02 16:00 +0000 [r357813]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: The default value for mohinterpret is
	  the empty string, so when resetting to default values don't
	  explicitly set the value to "default." ........ Merged revisions
	  357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 357812 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-03-02 01:33 +0000 [r357774-357775]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Fix race condition that can cause important
	  control frames (such as a hangup) to be missed. This takes two
	  actions. 1. Move the reading of the alertpipe in __ast_read() to
	  immediately before the removal of frames from the readq. This
	  means we won't do something silly like read from the alertpipe,
	  then ignore the fact that there's a frame to get from the readq
	  since channel's fdno is the AST_TIMING_FD. 2. When
	  ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
	  if the channel's fdno is the AST_TIMING_FD, then set the fdno to
	  -1. This is because if the rate is 0 and the timingfunc is NULL,
	  it means that the channel's timing fd is being invalidated, so
	  any pending reads should not occur. This may actually solve more
	  issues than the referenced one below, but it's not known at this
	  time for sure. (closes issue ASTERISK-19223) reported by
	  Frank-Michael Wittig Review:
	  https://reviewboard.asterisk.org/r/1779 ........ Merged revisions
	  357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 357762 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c: Fix compilation error due to typo during
	  channel opaquification.
	  s/ast_channel_fd_set/ast_channel_internal_fd_set/g

2012-03-01 22:09 +0000 [r357721]  Terry Wilson <twilson@digium.com>

	* channels/chan_unistim.c, apps/app_dahdibarge.c,
	  main/autoservice.c, addons/chan_ooh323.c, channels/chan_vpb.cc,
	  apps/app_meetme.c, channels/console_video.c,
	  channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
	  main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
	  apps/app_dumpchan.c, channels/sig_ss7.c, channels/chan_mgcp.c,
	  main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c,
	  channels/chan_agent.c, apps/app_dahdiras.c,
	  include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c,
	  channels/chan_jingle.c, channels/chan_misdn.c, apps/app_flash.c,
	  funcs/func_channel.c, apps/app_directed_pickup.c, main/file.c,
	  channels/chan_h323.c, res/snmp/agent.c, main/dsp.c: Opaquify
	  ast_channel typedefs, fd arrays, and softhangup flag Review:
	  https://reviewboard.asterisk.org/r/1784/

2012-03-01 14:22 +0000 [r357673]  Kinsey Moore <kmoore@digium.com>

	* /, main/acl.c: Prevent outbound SIP NOTIFY packets from
	  displaying a port of 0 In the change from 1.6.2 to 1.8,
	  ast_sockaddr was introduced which changed the behavior of
	  ast_find_ourip such that port number was wiped out. This caused
	  the port in internip (which is used for Contact and Call-ID on
	  NOTIFYs) to be 0. This change causes ast_find_ourip to be
	  port-preserving again. (closes issue ASTERISK-19430) ........
	  Merged revisions 357665 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357667 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-29 20:41 +0000 [r357621]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/utils.c, include/asterisk/stringfields.h: Update
	  stringfield documentation for removed second va_list in favor of
	  va_copy. In r320946, the second va_list that was passed to
	  ast_string_field_build_va and friends, was removed. This patch
	  updates the documentation to reflect that. ........ Merged
	  revisions 357620 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-29 20:31 +0000 [r357610]  Sean Bright <sean@malleable.com>

	* res/res_agi.c, CHANGES: Add IPv6 support to FastAGI. Review:
	  https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon
	  Perreault, Mark Michelson

2012-02-29 19:48 +0000 [r357577]  Walter Doekes <walter+asterisk@wjd.nu>

	* apps/app_dial.c, /: Fix copying of CDR(accountcode) to local
	  channels. In r203638, during the addition of the Channel Event
	  Logging, in mid-2009, this got broken in trunk and ended up in
	  asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
	  the calling channel is available to dialed channels again as well
	  as showing up properly in the CDR's. (closes issue
	  ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch
	  (License #6033) by jamicque Review:
	  https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard
	  Mudgett ........ Merged revisions 357575 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357576 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-29 16:52 +0000 [r357542]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, addons/chan_ooh323.c,
	  funcs/func_strings.c, channels/console_video.c,
	  apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_skinny.c, apps/app_dumpchan.c, main/features.c,
	  apps/app_amd.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c,
	  include/asterisk/utils.h, funcs/func_timeout.c,
	  apps/app_privacy.c, apps/app_fax.c, channels/chan_agent.c,
	  apps/app_disa.c, include/asterisk/channel.h,
	  apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c,
	  apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.c,
	  apps/app_voicemail.c, channels/chan_unistim.c,
	  tests/test_substitution.c, channels/chan_vpb.cc,
	  apps/app_meetme.c, main/ccss.c, apps/app_readexten.c,
	  channels/chan_gtalk.c, main/autochan.c, apps/app_followme.c,
	  main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c,
	  apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c,
	  bridges/bridge_builtin_features.c, apps/app_minivm.c,
	  res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c,
	  main/message.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
	  apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c,
	  res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c,
	  channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
	  channels/chan_oss.c, channels/chan_jingle.c,
	  channels/chan_usbradio.c, funcs/func_blacklist.c,
	  main/abstract_jb.c, channels/chan_h323.c, main/file.c,
	  res/snmp/agent.c, apps/app_sms.c, apps/app_stack.c,
	  funcs/func_callerid.c: Opaquify ast_channel structs and lists
	  Review: https://reviewboard.asterisk.org/r/1773/

2012-02-28 22:31 +0000 [r357460-357503]  Jonathan Rose <jrose@digium.com>

	* /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp
	  to sample sip.conf - Also changes version of Asterisk 1.8 in
	  UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches:
	  asterisk-19352-transport-warning-message-v1.patch uploaded by
	  Michael L. Young (license 5026) ........ Merged revisions 357490
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 357497 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, cdr/cdr_adaptive_odbc.c: Add additional character type types
	  to supported data types for cdr_adaptive_odbc The reporter was
	  uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
	  this patch adds those along with some other character types to
	  the list of types cdr_adaptive_odbc will work using the varchar
	  conditions. The problem wasn't really UTF8 characters as much as
	  it was a failure to respond to the exact type that was
	  declared/in use on that database. (closes issue ASTERISK-19334)
	  Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
	  uploaded by Igor Nikolaev (license 6236) ........ Merged
	  revisions 357455 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357458 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 21:26 +0000 [r357436]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, apps/app_stack.c: Correctly reset the dialplan priority. When
	  the stack frame is allocated, we save the address to which we
	  should return, when the Gosub returns. However, if we just want
	  to restore the priority, then we need to subtract 1 before
	  setting it. Otherwise, when a Gosub goes to a nonexistent
	  address, it will skip a priority in the dialplan. This is because
	  when we return from an application, the PBX increments the
	  priority for us. ........ Merged revisions 357416 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357421 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 21:01 +0000 [r357409]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Use more reasonable cause code when
	  rejecting incoming call waiting calls. (closes issue
	  ASTERISK-19397) Reported by: Birger Harzenetter Patches:
	  nochannel-cause.patch (license #5870) patch uploaded by Birger
	  Harzenetter ........ Merged revisions 357407 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357408 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 20:43 +0000 [r357406]  Jonathan Rose <jrose@digium.com>

	* /, UPGRADE-10.txt: revision 357386 -- oops, accidentally made it
	  10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352)
	  reported by: jamicque ........ Merged revisions 357405 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 20:34 +0000 [r357404]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, res/res_musiconhold.c, apps/app_queue.c: Fix
	  REF_DEBUG compile errors.

2012-02-28 20:33 +0000 [r357358-357403]  Jonathan Rose <jrose@digium.com>

	* /, UPGRADE-10.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from
	  r357356 to a new section specific to 1.8.12 (issue
	  ASTERISK-19352) reported by: jamicque ........ Merged revisions
	  357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 357400 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating
	  changes to transport option (issue ASTERISK-19352) Reported by:
	  jamicque ........ Merged revisions 357356 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357357 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 19:55 +0000 [r357355]  Sean Bright <sean@malleable.com>

	* include/asterisk/netsock2.h: Documentation update. There is no
	  AST_SOCKADDR_UNSPEC.

2012-02-28 19:37 +0000 [r357354]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_page.c: Remove dupliate 'i' option table entry in
	  app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
	  Patches: app_page-duplicate-i-option.patch (license #5027) patch
	  uploaded by Makoto Dei ........ Merged revisions 357352 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357353 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 18:52 +0000 [r357319]  Mark Michelson <mmichelson@digium.com>

	* /, channels/sip/security_events.c: Add a security event for the
	  case where fake authentication challenge is sent. ........ Merged
	  revisions 357318 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 18:46 +0000 [r357317]  Richard Mudgett <rmudgett@digium.com>

	* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
	  Convert struct ast_tcptls_session_instance to finally use the ao2
	  object lock.

2012-02-28 18:23 +0000 [r357288]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Changes transport option in sip.conf so
	  that using multiple instances doesn't stack. Prior to this patch,
	  Using "transport=" multiple times would cause them to add to one
	  another like allow/deny. This patch changes that behavior to
	  simply use the transport option specified last. Also, if no
	  transport option is applied now, the default will automatically
	  be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
	  asterisk-19352-transport-warning-message-v1.patch uploaded by
	  Michael L. Young (license 5026)
	  issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
	  (license 5674) Review:
	  https://reviewboard.asterisk.org/r/1745/diff/#index_header
	  ........ Merged revisions 357266 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357271 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-28 18:15 +0000 [r357272]  Richard Mudgett <rmudgett@digium.com>

	* main/format.c, main/format_cap.c, include/asterisk/astobj2.h,
	  include/asterisk/lock.h, main/astobj2.c: Astobj2 locking
	  enhancement. Add the ability to specify what kind of locking an
	  ao2 object has when it is allocated. The locking could be one of:
	  MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options()
	  ao2_alloc_options() ao2_t_container_alloc_options()
	  ao2_container_alloc_options() ao2_rdlock() ao2_wrlock()
	  ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and
	  AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no
	  longer mean that the object is protected by an external
	  mechanism. They mean the lock associated with the object has
	  already been manually obtained by one of the ao2_lock calls. This
	  change is necessary for RWLOCK support since they are not
	  reentrant. Also an operation on an ao2 container may require
	  promoting a read lock to a write lock by releasing the already
	  held read lock to re-acquire as a write lock. Replaced API calls:
	  ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock()
	  ao2_unlink_nolock() with the respective ao2_t_link_flags()
	  ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API
	  calls to be more flexible and to allow an anticipated enhancement
	  to control linking duplicate objects into a container. The
	  changes to format.c and format_cap.c are taking advantange of the
	  new ao2 locking options to simplify the use of the format
	  capabilities containers. Review:
	  https://reviewboard.asterisk.org/r/1554/

2012-02-28 14:47 +0000 [r357178-357214]  Kevin P. Fleming <kpfleming@digium.com>

	* /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
	  build system has some special magic to ensure that if Asterisk is
	  built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
	  source is still compiled with the optimizer enabled (even though
	  the result will be thrown away), because the compiler is able to
	  find a great deal of coding errors and bugs as a result of
	  running its optimizers. Unfortunately at some point this mode got
	  broken, and the 'throwaway' compile of the code was no longer
	  done with the optimizer enabled. This patch corrects that
	  problem. ........ Merged revisions 357212 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 357213 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/astobj2.c: Trailing whitespace cleanup.

2012-02-28 00:42 +0000 [r357096-357145]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
	  Add ability to clone ao2 containers. Occasionally there is a need
	  to put all objects in one container also into another container.
	  Some reasons you might need to do this: 1) You need to
	  reconfigure a container. You would do this by creating a new
	  container with the new configuration and ao2_container_dup the
	  old container into it. Then replace the old container with the
	  new. Then destroy the old container. 2) You need the contents of
	  a container to remain stable while operating on all of the
	  objects. You would do this by creating a cloned container of the
	  original with ao2_container_clone. The cloned container is a
	  snapshot of the objects at the time of the cloning. When done,
	  just destroy the cloned container. Review:
	  https://reviewboard.asterisk.org/r/1746/

	* main/channel.c: Fix ast_channel allocation init setting priority
	  to -1 instead of 1. * Fix opaquification conversion error.
	  (closes issue ASTERISK-19424) Reported by: Jeremy Pepper Patches:
	  asterisk-19424-initialize_priority_regression.diff (license
	  #5026) patch uploaded by Michael L. Young

	* main/channel.c, /: Fix callerid of Originated calls. Thanks to
	  Matt Riddell for tracking this down. (closes issue
	  ASTERISK-19385) Reported by: ornix ........ Merged revisions
	  357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 357095 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-27 19:55 +0000 [r357051]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/res_odbc.h, res/res_odbc.c: Converts locking for
	  odbc containers from ast_mutex_lock to ao2_locks.

2012-02-27 17:03 +0000 [r357014]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c, main/netsock.c: Address comments from Mark
	  Michelson

2012-02-27 16:50 +0000 [r357013]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
	  main/dial.c, main/rtp_engine.c, main/ccss.c, main/features.c,
	  UPGRADE.txt, main/app.c, include/asterisk/channel.h,
	  configs/ccss.conf.sample, apps/app_followme.c, apps/app_queue.c,
	  include/asterisk/ccss.h: Deprecated macro usage for connected
	  line, redirecting, and CCSS This commit adds GoSub alternatives
	  to connected line, redirecting, and CCSS macro hooks so that
	  macro can finally be deprecated. This also adds deprecation
	  warnings for those features when used and in documentation.
	  Review: https://reviewboard.asterisk.org/r/1760/ (closes issue
	  SWP-4256)

2012-02-27 16:31 +0000 [r357005]  Sean Bright <sean@malleable.com>

	* include/asterisk/netsock.h, channels/chan_iax2.c, main/netsock.c:
	  Convert netsock.h over to use ast_sockaddrs rather than
	  sockaddr_in and update chan_iax2 to pass in the correct types.
	  chan_iax2 is the only consumer for the various ast_netsock_*
	  functions in trunk at this point, so this feels like a safe
	  change to make.

2012-02-27 16:24 +0000 [r356987]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: Adds an option to sip.conf that
	  prevents diversion headers from being added. send_diversion=no
	  will prevent Diversion headers from being added to SIP requests.
	  This doesn't prevent Diversion from being added with dialplan
	  such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported
	  by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/

2012-02-27 16:12 +0000 [r356966]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: There isn't much point in saving off and
	  restoring a value that we never use again.

2012-02-27 16:08 +0000 [r356965]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: Copy CDR variables when set during a bridge
	  This patch makes sure amaflags, accountcode, and userfield get
	  copied to the bridge CDR when set during a bridge (like via a
	  custom feature). (closes issue ASTERISK-16990) Review:
	  https://reviewboard.asterisk.org/r/1721/ ........ Merged
	  revisions 356963 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356964 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-27 15:35 +0000 [r356962]  Jonathan Rose <jrose@digium.com>

	* /, res/res_odbc.c: Remove possible segfaults from res_odbc by
	  adding locks around usage of odbc handle (closes issue
	  ASTERISK-19011) Reported by: Walter Doekes Patches:
	  issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
	  uploaded by Walter Doekes (license 5674) review:
	  https://reviewboard.asterisk.org/r/1719/ review:
	  https://reviewboard.asterisk.org/r/1622/ ........ Merged
	  revisions 356917 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356961 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-27 14:57 +0000 [r356881-356916]  Sean Bright <sean@malleable.com>

	* include/asterisk/netsock.h, main/netsock.c: Make
	  ast_netsock_set_qos() delegate to ast_set_qos().

	* include/asterisk/netsock.h: Correct typo in deprecation comment.

	* channels/chan_unistim.c, main/udptl.c, channels/chan_skinny.c,
	  include/asterisk/netsock.h, pbx/pbx_dundi.c,
	  channels/chan_mgcp.c: Prefer ast_set_qos() over
	  ast_netsock_set_qos()

	* main/netsock.c: Remove trailing whitespace

2012-02-26 18:25 +0000 [r356848]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c: Add
	  support change gatekeeper mode or ip per ooh323 reload command
	  (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
	  change_gk_on_reload-1.patch (License #5415)

2012-02-25 17:22 +0000 [r356799]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_voicemail.c: Fix crash in app_voicemail during
	  close_mailbox In r354890, a memory leak in app_voicemail was
	  fixed by properly disposing of the allocated heard/deleted
	  pointers. However, there are situations, particularly when no
	  messages are found in a folder, where these pointers are not
	  allocated and not NULL. In that case, an invalid free would be
	  attempted, which could crash app_voicemail. As there are a number
	  of code paths where this could occur, this patch uses the number
	  of messages detected in the folder before it attempts to free the
	  pointers. This resolves the crash detected in the Asterisk Test
	  Suite's check_voicemail_nominal test. ........ Merged revisions
	  356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 356798 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-24 23:40 +0000 [r356697-356765]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astobj2.h: astobj2.h comment tweaks.

	* include/asterisk/astobj2.h, main/astobj2.c: astobj2.h
	  documentation updates.

	* /, channels/chan_sip.c, include/asterisk/tcptls.h,
	  channels/sip/include/sip.h: Fix worker thread resource leak in
	  SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable
	  but noone could join them if they died on their own. * Fix the
	  SIP TCP/TLS worker threads to not be created joinable. *
	  _sip_tcp_helper_thread() only needs one parameter since the pvt
	  parameter is only passed in as NULL and never used. (closes issue
	  ASTERISK-19203) Reported by: Steve Davies Review:
	  https://reviewboard.asterisk.org/r/1714/ ........ Merged
	  revisions 356677 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356690 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-24 17:43 +0000 [r356606-356652]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch
	  for ASTERISK-19253 included properly shutting down the libsrtp
	  library in the case of module unload. Unfortunately, not all
	  distributions have the srtp_shutdown call. As such, this patch
	  removes calling srtp_shutdown. ........ Merged revisions 356650
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 356651 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
	  main/rtp_engine.c, /, include/asterisk/rtp_engine.h,
	  res/res_srtp.c: Allow SRTP policies to be reloaded Currently,
	  when using res_srtp, once the SRTP policy has been added to the
	  current session the policy is locked into place. Any attempt to
	  replace an existing policy, which would be needed if the remote
	  endpoint negotiated a new cryptographic key, is instead rejected
	  in res_srtp. This happens in particular in transfer scenarios,
	  where the endpoint that Asterisk is communicating with changes
	  but uses the same RTP session. This patch modifies res_srtp to
	  allow remote and local policies to be reloaded in the underlying
	  SRTP library. From the perspective of users of the SRTP API, the
	  only change is that the adding of remote and local policies are
	  now added in a single method call, whereas they previously were
	  added separately. This was changed to account for the differences
	  in handling remote and local policies in libsrtp. Review:
	  https://reviewboard.asterisk.org/r/1741/ (closes issue
	  ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
	  Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
	  Jordan (license 6283) (with some small modifications for this
	  check-in) ........ Merged revisions 356604 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356605 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-24 00:32 +0000 [r356573]  Terry Wilson <twilson@digium.com>

	* channels/chan_unistim.c, channels/chan_local.c,
	  addons/chan_ooh323.c, channels/chan_multicast_rtp.c,
	  channels/chan_vpb.cc, main/rtp_engine.c, apps/app_meetme.c,
	  apps/app_dictate.c, apps/app_record.c, apps/app_test.c,
	  bridges/bridge_softmix.c, channels/chan_gtalk.c, apps/app_ices.c,
	  res/res_musiconhold.c, channels/chan_iax2.c,
	  bridges/bridge_multiplexed.c, main/indications.c, main/cli.c,
	  main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
	  channels/chan_skinny.c, res/res_agi.c, main/features.c,
	  apps/app_mp3.c, apps/app_dumpchan.c, main/app.c, apps/app_amd.c,
	  channels/chan_alsa.c, apps/app_confbridge.c,
	  addons/chan_mobile.c, main/bridging.c, channels/chan_mgcp.c,
	  apps/app_nbscat.c, main/pbx.c, channels/chan_sip.c,
	  res/res_fax.c, apps/app_festival.c, channels/chan_bridge.c,
	  main/channel_internal_api.c, apps/app_fax.c,
	  apps/app_waitforsilence.c, res/res_adsi.c, channels/chan_agent.c,
	  bridges/bridge_simple.c, include/asterisk/channel.h,
	  channels/chan_console.c, apps/app_talkdetect.c,
	  channels/chan_oss.c, apps/app_speech_utils.c,
	  channels/chan_usbradio.c, channels/chan_jingle.c,
	  channels/chan_misdn.c, funcs/func_channel.c, main/file.c,
	  channels/chan_nbs.c, apps/app_chanspy.c, apps/app_voicemail.c,
	  res/res_calendar.c: Opaquification for ast_format structs in
	  struct ast_channel Review:
	  https://reviewboard.asterisk.org/r/1770/

2012-02-23 20:14 +0000 [r356523]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c, main/features.c: Fix blind transfer
	  parking issues if the dialed extension is not recognized as a
	  parking extension. Custom parking extensions may not be coded
	  such that the first and only extension priority is the Park
	  application. These custom parking extensions will not be
	  recognized as parking extensions. When a call is blind
	  transferred to an extension that is not recognized as a parking
	  extension, the normal blind transfer code causes the transferred
	  channel to start executing dialplan. Calls that get parked in
	  this manner do not know the original channel name that parked the
	  call so the original parker could never be called back if the
	  parked call is not retrieved before the timeout time. The parking
	  space is also announced to the call being parked as a side effect
	  of not knowing the original parking channel. * Fix handling of
	  BLINDTRANSFER channel variable for call parking. * Fixed SIP
	  blind transfer using the wrong dialplan context variable to check
	  for the parking extension. (closes issue ASTERISK-19322) Reported
	  by: aragon Tested by: rmudgett, jparker Review:
	  https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........
	  Merged revisions 356521 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356522 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-23 15:49 +0000 [r356477]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
	  When we send an ACK for a 2xx response to an INVITE, we are
	  supposed to use the learned route set. However, when we receive a
	  non-2xx final response to an INVITE, we are supposed to send the
	  ACK to the same place we initially sent the INVITE. We had been
	  doing this up until the changes went in that would build a route
	  set from provisional responses. That introduced a regression
	  where we would use the learned route set under all circumstances.
	  With this change, we now will set the destination of our ACK
	  based on the invitestate. If it is INV_COMPLETED then that means
	  that we have received a non-2xx final response (INV_TERMINATED
	  indicates a 2xx response was received). If it is INV_CANCELLED,
	  then that means the call is being canceled, which means that we
	  should be ACKing a 487 response. The other change introduced here
	  is setting the invitestate to INV_CONFIRMED when we send an ACK
	  *after* the reqprep instead of before. This way, we can tell in
	  reqprep more easily what the invitestate is prior to sending the
	  ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
	  patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
	  (license #5049) (with some slight modifications prior to commit)
	  ........ Merged revisions 356475 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356476 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-23 03:27 +0000 [r356429]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
	  ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
	  22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
	  compiler error (gcc 4.6.2) Review:
	  https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
	  pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
	  lines Add back strsep() function for previous commit ........
	  r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
	  2012) | 2 lines Missed one strsep() function ........ Merged
	  revisions 356290,356335,356337 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356428 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-23 01:53 +0000 [r356397]  Terry Wilson <twilson@digium.com>

	* tests/test_substitution.c, tests/test_utils.c: Fix some tests
	  that didn't get opaquification changes Review:
	  https://reviewboard.asterisk.org/r/1766/

2012-02-23 00:56 +0000 [r356366]  Richard Mudgett <rmudgett@digium.com>

	* main/channel_internal_api.c: Revert some apparently accidental
	  spacing changes.

2012-02-22 21:22 +0000 [r356314]  Terry Wilson <twilson@digium.com>

	* /, include/asterisk/calendar.h, main/loader.c,
	  res/res_calendar.c: Track module use count for res_calendar If
	  the res_calendar module was followed immediately by one of the
	  calendar tech modules and "core stop gracefully" was run,
	  Asterisk would crash. This patch adds use count tracking for
	  res_calendar so that it is unloaded after the tech modules when
	  shutting down gracefully. It is now not possible to unload all
	  the of the calendar modules via "module unload res_calednar.so",
	  but it is still possible to unload them all via "module unload -h
	  res_calendar.so". Review:
	  https://reviewboard.asterisk.org/r/1752/ ........ Merged
	  revisions 356291 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 356297 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-22 21:10 +0000 [r356292]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  Correct some set-but-unused variable warnings in the mISDN
	  library.

2012-02-22 17:34 +0000 [r356259]  Terry Wilson <twilson@digium.com>

	* channels/chan_misdn.c: Fix chan_misdn after the lastest
	  opaquification changes It now compiles, but there are some
	  unrelated warnings for set but unused variables.

2012-02-22 14:54 +0000 [r356216]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Merged revisions 356215 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r356215 | mjordan | 2012-02-22 08:53:53 -0600
	  (Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
	  | 27 lines Fix potential buffer overrun and memory leak when
	  executing "sip show peers" The "sip show peers" command uses a
	  fix sized array to sort the current peers in the peers
	  ao2_container. The size of the array is based on the current
	  number of peers in the container. However, once the size of the
	  array is determined, the number of peers in the container can
	  change, as the peers container is not locked. This could cause a
	  buffer overrun when populating the array, if peers were added to
	  the container after the array was created. Additionally, a memory
	  leak of the allocated array would occur if a user caused the
	  _show_peers method to return CLI_SHOWUSAGE. We now create a
	  snapshot of the current peers using an ao2_callback with the
	  OBJ_MULTIPLE flag. This size of the array is set to the number of
	  peers that the iterator will iterate over; hence, if peers are
	  added or removed from the peers container it will not affect the
	  execution of the "sip show peers" command. Review:
	  https://reviewboard.asterisk.org/r/1738/ (closes issue
	  ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
	  Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
	  Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
	  (license 6283) ........ ................

2012-02-22 00:35 +0000 [r356152-356183]  Terry Wilson <twilson@digium.com>

	* main/channel.c, main/channel_internal_api.c,
	  include/asterisk/channel.h: Rename
	  ast_channel_emulate_dtmf_digit* funcs The accessors names for the
	  "emulate_dtmf_digit" field on the ast_channel are misleading.
	  Change them to ast_channel_dtmf_digit_to_emulate*.

	* main/channel.c, main/framehook.c, res/res_monitor.c: Fix some
	  opaquification-related compiler warnings (closes issue
	  ASTERISK-19419) PseudoReview - seanbright on IRC

2012-02-21 11:17 +0000 [r356111]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
	  make sense when an IP is passed in. ........ Merged revisions
	  356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 356108 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-21 04:31 +0000 [r356075]  Kinsey Moore <kmoore@digium.com>

	* /, main/ccss.c: Add missing newline to ccss state change
	  notification Move along, nothing to see here... ........ Merged
	  revisions 356074 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-20 23:43 +0000 [r356042]  Terry Wilson <twilson@digium.com>

	* main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
	  cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
	  main/rtp_engine.c, apps/app_playtones.c, apps/app_record.c,
	  apps/app_sayunixtime.c, apps/app_test.c, main/devicestate.c,
	  apps/app_alarmreceiver.c, apps/app_chanisavail.c,
	  apps/app_ices.c, channels/chan_iax2.c,
	  bridges/bridge_multiplexed.c, main/cli.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, main/framehook.c, channels/chan_skinny.c,
	  main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c,
	  channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
	  channels/sig_ss7.c, apps/app_milliwatt.c, cdr/cdr_manager.c,
	  apps/app_dial.c, main/pbx.c, funcs/func_timeout.c,
	  apps/app_privacy.c, channels/chan_bridge.c, apps/app_echo.c,
	  apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c,
	  channels/chan_agent.c, apps/app_disa.c, bridges/bridge_simple.c,
	  include/asterisk/channel.h, apps/app_talkdetect.c,
	  apps/app_transfer.c, main/cel.c, res/res_monitor.c,
	  apps/app_playback.c, apps/app_speech_utils.c,
	  channels/chan_misdn.c, apps/app_sendtext.c, funcs/func_channel.c,
	  funcs/func_cdr.c, channels/sip/dialplan_functions.c,
	  apps/app_macro.c, apps/app_zapateller.c, main/audiohook.c,
	  apps/app_chanspy.c, apps/app_voicemail.c, apps/app_cdr.c,
	  res/res_calendar.c, channels/chan_unistim.c,
	  channels/chan_multicast_rtp.c, channels/chan_vpb.cc,
	  apps/app_meetme.c, main/ccss.c, apps/app_dictate.c,
	  apps/app_authenticate.c, apps/app_readexten.c,
	  channels/chan_gtalk.c, res/res_musiconhold.c,
	  apps/app_followme.c, main/channel.c, main/cdr.c,
	  channels/chan_phone.c, main/dial.c, main/manager.c,
	  apps/app_osplookup.c, bridges/bridge_builtin_features.c,
	  res/res_agi.c, apps/app_minivm.c, main/app.c,
	  apps/app_confbridge.c, main/image.c, apps/app_directory.c,
	  main/message.c, apps/app_ivrdemo.c, addons/chan_mobile.c,
	  apps/app_rpt.c, cdr/cdr_custom.c, apps/app_parkandannounce.c,
	  channels/chan_mgcp.c, apps/app_while.c, res/res_rtp_asterisk.c,
	  apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
	  res/res_fax.c, cdr/cdr_syslog.c, apps/app_waitforsilence.c,
	  main/channel_internal_api.c, res/res_adsi.c, pbx/pbx_lua.c,
	  funcs/func_jitterbuffer.c, channels/chan_console.c,
	  apps/app_queue.c, channels/sig_pri.c, channels/chan_oss.c,
	  channels/chan_jingle.c, channels/chan_usbradio.c,
	  apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c,
	  main/abstract_jb.c, main/file.c, channels/chan_h323.c,
	  include/asterisk/sched.h, res/snmp/agent.c, apps/app_sms.c,
	  channels/chan_nbs.c, funcs/func_callerid.c, apps/app_verbose.c,
	  apps/app_stack.c: ast_channel opaquification of pointers and
	  integral types Review: https://reviewboard.asterisk.org/r/1753/

2012-02-20 18:40 +0000 [r355903-355999]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: Remove spurious warning when
	  'qualifyfreqnotok' is set successfully. (closes issue
	  ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
	  Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
	  Covert (license 5512) ........ Merged revisions 355997 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355998 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
	  ........ Merged revisions 355952 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355953 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_dahdi.c, /: Change some debug messages from
	  LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355950 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_iax2.c: Add some boilerplate documentation for
	  IAXVAR and IAXPEER. ........ Merged revisions 355904 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355905 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so
	  that we can set it's port later. Without this, the call to
	  ast_sockaddr_set_port a few lines later is a noop. ........
	  Merged revisions 355901 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355902 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-18 08:02 +0000 [r355852]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
	  channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to
	  chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
	  in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
	  flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
	  Now provides a callback for all the low level sig_XXX modules.
	  (issue ASTERISK-19316) alecdavis (license 585) Reported by:
	  Jeremy Pepper Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1747/ ........ Merged
	  revisions 355850 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355851 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-17 22:03 +0000 [r355795]  Sean Bright <sean@malleable.com>

	* configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
	  trunkfreq to be greater than 1000ms. ........ Merged revisions
	  355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 355794 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-17 19:56 +0000 [r355749]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/asterisk.c: Non-verbose output should always go to the
	  remote console, regardless of the previous level.

2012-02-17 19:35 +0000 [r355748]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: Pass the correct value to
	  ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
	  variable to determine how often to send trunk packets, but this
	  value is in milliseconds while ast_timer_set_rate() expects the
	  rate argument to be ticks per second. So we divide 1000 by
	  trunkfreq and pass that in instead. With a default of 20ms, this
	  change makes IAX2 send trunk packets every 20ms instead of every
	  50ms. Tracked down by myself and Bob Wienholt. ........ Merged
	  revisions 355746 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355747 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-17 19:22 +0000 [r355745]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Fix regressions with regards to route-set
	  creation on early dialogs. This fixes two main issues: 1.
	  Asterisk would send a CANCEL to the route created by the
	  provisional response instead of using the same destination it did
	  in the initial INVITE. 2. If a new route set arrives in a 200 OK
	  than was in the 1XX response (perfectly possible if our outbound
	  INVITE gets forked), then the route set in the 200 OK needs to
	  overwrite the route set in the 1XX response. (closes issue
	  ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
	  Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
	  (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
	  (license 6034) Review: https://reviewboard.asterisk.org/r/1749
	  ........ Merged revisions 355732 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355733 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-16 22:00 +0000 [r355667]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_rpt.c: Fix channel opaquification for app_rpt

2012-02-16 20:03 +0000 [r355624]  Sean Bright <sean@malleable.com>

	* /, main/audiohook.c: Revert a change to
	  audio_audiohook_write_list that had no affect. When I made this
	  change initially, I was under the false impression that the
	  audiohooks structure remained on the channel after all of the
	  hooks had been detached. This is not the case, ast ast_read takes
	  care of removing the audiohooks structure if the lists are empty.
	  ........ Merged revisions 355622 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355623 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-16 19:51 +0000 [r355576-355621]  Richard Mudgett <rmudgett@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in,
	  autoconf/ast_c_declare_check.m4 (added), configure.ac,
	  formats/format_ogg_vorbis.c: Fix compile problem when old version
	  of libvorbisfile v1.1.2 is used. The principle difference between
	  libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
	  of the predefined callbacks OV_CALLBACKS_xxx in
	  vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
	  configure script to detect if libvorbisfile.h declares
	  OV_CALLBACKS_NOCLOSE. * Copied the declaration of
	  OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
	  (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........
	  Merged revisions 355608 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355620 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_monitor.c: Fix AMI Monitor action without File header
	  converting channel name into filename. * Fix potential Solaris
	  crash if Monitor application has a urlbase and no fname_base
	  option. ........ Merged revisions 355574 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355575 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-15 19:29 +0000 [r355450-355531]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
	  sure to log 'apathetic' messages too. ........ Merged revisions
	  355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 355530 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* build_tools/cflags.xml, channels/chan_iax2.c: Remove IAX_OLD_FIND
	  from chan_iax2.

	* /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
	  intended. Back in r646, TRUNK_CALL_START was added and defined as
	  0x4000. That same value was also hard-coded in one part of the
	  IAX2 code instead of using the #define. TRUNK_CALL_START has
	  changed over the years (for dealing with LOW_MEMORY), but the
	  hard-coded usage was never updated to match. This patch fixes
	  that. ........ Merged revisions 355448 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355449 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 20:27 +0000 [r355413]  Tilghman Lesher <tilghman@meg.abyt.es>

	* utils/refcounter.c, main/pbx.c, funcs/func_timeout.c,
	  include/asterisk/autoconfig.h.in, utils/hashtest.c, UPGRADE.txt,
	  CHANGES, main/config.c, configs/logger.conf.sample,
	  main/loader.c, include/asterisk/logger.h, main/manager.c,
	  main/logger.c, utils/ael_main.c, utils/hashtest2.c,
	  codecs/codec_dahdi.c, main/stdtime/localtime.c, main/asterisk.c,
	  addons/res_config_mysql.c: Re-commit the verbose branch. This
	  change permits each verbose destination (consoles, logger) to
	  have its own concept of what the verbosity level is. The big
	  feature here is that the logger will now be able to capture a
	  particular verbosity level without condemning each console to
	  need to suffer that level of verbosity. Additionally, a stray
	  'core set verbose' will no longer change what will go to the log.
	  Review: https://reviewboard.asterisk.org/r/1599/

2012-02-14 19:29 +0000 [r355321-355376]  Richard Mudgett <rmudgett@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  formats/format_ogg_vorbis.c: Fix voicemail problems when using
	  ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
	  format because it did not implement the seek and tell format
	  callbacks among other problems. Since we were already using the
	  libvorbis and libvorbisenc libraries we can use libvorbisfile as
	  it is also part of the vorbis library package. * Made use the
	  libvorbisfile to handle the ogg/vorbis file stream. The
	  format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
	  (closes issue ASTERISK-16926) Reported by: sque Patches:
	  ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
	  by sque ........ Merged revisions 355365 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355375 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
	  in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
	  Reported by: Alex Villacis Lasso Patches:
	  asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
	  (license #5617) patch uploaded by Alex Villacis Lasso Review:
	  https://reviewboard.asterisk.org/r/1740/ ........ Merged
	  revisions 355319 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355320 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 16:28 +0000 [r355274]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Properly invert the return of a strncmp
	  call. This was causing identification that should have been made
	  private to be public. (closes issue AST-814) reported by Patrick
	  Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
	  (license 5430) ........ Merged revisions 355268 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355271 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 15:58 +0000 [r355230]  Jason Parker <jparker@digium.com>

	* /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
	  CDRs by default in sample configs. ........ Merged revisions
	  355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 355229 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 13:35 +0000 [r355184]  Sean Bright <sean@malleable.com>

	* /, channels/chan_iax2.c: Clear the high order bit from the
	  destination call number before sending. send_apathetic_reply
	  takes the incoming frame's source call number as the destination
	  call number for the outgoing frame. If the incoming frame was a
	  full frame, then the high order bit of the source call number is
	  set and will be interpreted as a retransmit when sent back out as
	  the destination call number. ........ Merged revisions 355182
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 355183 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 09:58 +0000 [r355138]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: call manager_event only if there is not
	  null channel structure (Closes issue ASTERISK-19298) Reported by:
	  robinfood Patches: issue19298.patch uploaded by may213 (License
	  #5415) ........ Merged revisions 355136 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355137 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-14 00:43 +0000 [r355102]  Russell Bryant <russell@russellbryant.com>

	* res/res_agi.c, CHANGES: res_agi: Add AGIEXITONHANGUP variable.
	  This patch adds a variable AGIEXITONHANGUP for res_agi. If this
	  variable is set to "yes" on a channel, AGI() will exit
	  immediately once a channel hangup has been detected. This was the
	  behavior of AGI() in Asterisk 1.4 and earlier and is still
	  desired by some people. Review:
	  https://reviewboard.asterisk.org/r/1734/

2012-02-13 22:04 +0000 [r355055-355058]  Richard Mudgett <rmudgett@digium.com>

	* pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
	  execution. Since the dir timestamp is available at one second
	  resolution, we cannot know if it was updated within the same
	  second after we scanned it. Therefore, we will force another scan
	  if the dir was just modified. * Changed to force another scan if
	  the directory was just modified. (closes issue ASTERISK-19081)
	  Reported by: Knut Bakke Review:
	  https://reviewboard.asterisk.org/r/1688/ ........ Merged
	  revisions 355056 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 355057 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/chan_misdn.c: Fix compile error from most recent
	  ast_channel opaquification installment.

2012-02-13 19:56 +0000 [r355011]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
	  at a time as otherwise they would share the same common local
	  context list. (closes issue AST-758) ........ Merged revisions
	  355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 355010 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-13 17:27 +0000 [r354968]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, addons/chan_ooh323.c,
	  channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
	  apps/app_dumpchan.c, pbx/pbx_realtime.c, channels/chan_alsa.c,
	  apps/app_dial.c, main/pbx.c, apps/app_fax.c,
	  channels/chan_agent.c, include/asterisk/channel.h,
	  apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c,
	  funcs/func_channel.c, apps/app_macro.c, apps/app_chanspy.c,
	  res/res_calendar.c, apps/app_voicemail.c,
	  channels/chan_unistim.c, tests/test_substitution.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
	  apps/app_readexten.c, channels/chan_gtalk.c, main/cdr.c,
	  main/channel.c, main/dial.c, channels/chan_phone.c,
	  main/manager.c, apps/app_osplookup.c,
	  bridges/bridge_builtin_features.c, res/res_agi.c,
	  apps/app_minivm.c, apps/app_confbridge.c, apps/app_directory.c,
	  addons/chan_mobile.c, apps/app_rpt.c, apps/app_parkandannounce.c,
	  channels/chan_mgcp.c, apps/app_while.c, funcs/func_dialplan.c,
	  channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c,
	  pbx/pbx_lua.c, channels/sig_pri.c, apps/app_queue.c,
	  channels/chan_oss.c, channels/chan_jingle.c,
	  apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c,
	  res/snmp/agent.c, pbx/pbx_dundi.c, channels/chan_nbs.c,
	  apps/app_stack.c, apps/app_verbose.c: Opaquify char * and char[]
	  in ast_channel Review: https://reviewboard.asterisk.org/r/1733/

2012-02-13 17:25 +0000 [r354964]  Richard Mudgett <rmudgett@digium.com>

	* res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
	  reconnecting to pgsql database after connection loss. There can
	  only be one database connection in res_config_pgsql just like
	  res_config_sqlite. If the connection is lost, the connection may
	  not get reestablished to the same database if the res_pgsql.conf
	  and extconfig.conf files are inconsistent. * Made only use the
	  configured database from res_pgsql.conf. * Fixed potential buffer
	  overwrite of last[] in config_pgsql(). (closes issue
	  ASTERISK-16982) Reported by: german aracil boned Review:
	  https://reviewboard.asterisk.org/r/1731/ ........ Merged
	  revisions 354953 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354959 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-13 16:42 +0000 [r354939]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_confbridge.c: Don't try to play sound files that do
	  not exist. (closes issue ASTERISK-19188) Reported by: slesru
	  ........ Merged revisions 354938 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-10 22:44 +0000 [r354903]  Jason Parker <jparker@digium.com>

	* /, apps/app_voicemail.c: Fix a voicemail memory leak with
	  heard/deleted messages. open_mailbox() was changed quite a long
	  time ago to allocate this memory. close_mailbox() should have
	  been changed to be responsible for freeing it. ........ Merged
	  revisions 354889 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354890 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-10 18:08 +0000 [r354837]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
	  to the same exten and context. The astman_get_header() never
	  returns NULL so the check by the code for NULL would never fail.
	  (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
	  0018325.patch (license #6116) patch uploaded by Nuno Borges
	  (modified) ........ Merged revisions 354835 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354836 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-10 14:51 +0000 [r354799]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Fix IMAP app_voicemail compilation issue
	  introduced in r354429 This simply fixes the compilation issue
	  introduced in r354429 by re-adding the 'quote' variable. (closes
	  issue ASTERISK-19337) Reported by: John Taylor

2012-02-09 22:06 +0000 [r354751]  Terry Wilson <twilson@digium.com>

	* /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
	  is torn down CDRs cannot be modified after a bridge is torn down,
	  (e.g. after Dial() returns) even though the CDR() function may be
	  called. Since modifying the CDR code to change this behavior
	  could very easily break all kinds of things, this patch just
	  documents this limitation. (closes issues ASTERISK-16923) Review:
	  https://reviewboard.asterisk.org/r/1720/ ........ Merged
	  revisions 354749 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354750 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-09 20:52 +0000 [r354657-354704]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Fix parsing of SIP headers where compact
	  and non-compact headers are mixed Change parsing of SIP headers
	  so that compactness of the header no longer influences which
	  header will be chosen. Previously, a non-compact header would be
	  chosen instead of a preceeding compact-form header. (closes issue
	  ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
	  ........ Merged revisions 354702 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354703 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/config.c: Make the config parser remove escaping
	  backslashes The config parser in Asterisk does not currently
	  remove a backslash that is used to escape a semicolon which would
	  otherwise be interpreted as the start of a comment. The change
	  here causes that backslash to be removed, but does not create a
	  real escape system in the config parser. The biggest complication
	  with a real escape system would be breaking existing configs
	  everywhere (parsing \\ as \ and breaking on escaped non-semicolon
	  characters) even though it would be the "right" way to do things.
	  (closes issue ASTERISK-17121) Review:
	  https://reviewboard.asterisk.org/r/1724/ ........ Merged
	  revisions 354655 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354656 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-09 18:14 +0000 [r354597]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c, channels/sip/include/config_parser.h,
	  channels/sip/utils.c (added), configs/sip.conf.sample, CHANGES,
	  channels/sip/config_parser.c, channels/sip/include/sip.h,
	  channels/sip/include/sip_utils.h: Add auto_force_rport and
	  auto_comedia NAT options This patch adds the auto_force_rport and
	  auto_comedia NAT options. It also converts the nat= setting to a
	  list of comma-separated combinable options: no, force_rport,
	  comedia, auto_force_rport, and auto_comedia. nat=yes remains as
	  an undocumented option equal to "force_rport,comedia". The first
	  instance of 'yes' or 'no' in the list stops parsing and overrides
	  any previously set options. If an auto_* option is specified with
	  its non-auto_ counterpart, the auto setting takes precedence.
	  This patch builds upon the patch posted to ASTERISK-17860 by JIRA
	  user pedro-garcia. (closes issue ASTERISK-17860) Review:
	  https://reviewboard.asterisk.org/r/1698/

2012-02-09 17:17 +0000 [r354552]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_fax.c: Adding reload support to res_fax.so (closes
	  issue ASTERISK-16712) reported by Frank DiGennaro Review:
	  https://reviewboard.asterisk.org/r/1713 ........ Merged revisions
	  354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 354546 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-09 17:09 +0000 [r354544-354549]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Clean-up of minor formatting issues in
	  r354542/3/4 rmudgett pointed out some formatting issues in the
	  check-in for ASTERISK-19290. This cleans those up. Review:
	  https://reviewboards.asterisk.org/r/1722/ ........ Merged
	  revisions 354547 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354548 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Fix SIP INFO DTMF handling for
	  non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as
	  changed to account for both lowercase alphatbetic DTMF events, as
	  well as uppercase alphabetic DTMF events. When this occurred, the
	  comparison of the character buffer containing the event code was
	  changed such that the buffer was first compared again '0' and '9'
	  to determine if it was numeric. Unfortunately, since the first
	  character in the buffer will typically be '1' in the case of
	  non-numeric event codes (10-16), this caused those codes to be
	  converted to a DTMF event of '1'. This patch fixes that, and
	  cleans up handling of both application/dtmf-relay and
	  application/dtmf content types. Review:
	  https://reviewboard.asterisk.org/r/1722/ (closes issue
	  ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........
	  Merged revisions 354542 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354543 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-09 03:09 +0000 [r354497-354498]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/chan_misdn.c: Fix some compile
	  problems from the 'cppcheck' patch.

	* /, apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce.
	  Well, thats embarrasing. I forgot to initialize the caller_id
	  storage. (closes issue ASTERISK-19311) Reported by: tootai Tested
	  by: rmudgett ........ Merged revisions 354495 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354496 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-09 02:28 +0000 [r354494]  Russell Bryant <russell@russellbryant.com>

	* main/channel.c, /: Remove some unnecessary locking from
	  ast_hangup(). This patch removes some unnecessary locking of the
	  channels container in ast_hangup(). The reason this came up is
	  that this lock can very quickly block the entire system. If any
	  of the channel cleanup code decides to block, it causes a problem
	  for the whole system. For example, when audiohooks get destroyed,
	  if that blocks for a while waiting on the mixmonitor thread to
	  exit because it's busy blocking on some I/O, it causes a problem
	  for many other threads in the meantime. Review:
	  https://reviewboard.asterisk.org/r/1712/ ........ Merged
	  revisions 354492 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354493 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-08 21:29 +0000 [r354459]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_ais.c (removed), contrib/scripts/install_prereq: Revision
	  354046 added res_corosync as a replacement for res_ais, but
	  didn't actually remove res_ais. This commit removes it. In
	  addition, the 'install_prereq' script has been updated to no
	  longer install AIS dependency packages, and instead install
	  Corosync packages instead.

2012-02-08 21:28 +0000 [r354458]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql,
	  CHANGES, channels/sip/include/sip.h: Add callbackextension
	  matching & realtime callbackextensions This patch is based on the
	  one by David Vossel, developer extrodinaire, at
	  https://reviewboard.asterisk.org/r/344/. If multiple peers are
	  defined with the same host/port, but differing
	  callbackextensions, it chooses the peer with the matching
	  callbackextension. Since callbackextension creates an outbound
	  registration with the callbackextension as the Contact address,
	  matching an incoming request by that (in addition to the
	  host/port) makes a lot of sense. This patch also adds support for
	  callbackextension to realtime by querying all peers with
	  callbackextensions on reload and adding registrations for them.
	  (closes issue ASTERISK-13456) Review:
	  https://reviewboard.asterisk.org/r/344/ Review:
	  https://reviewboard.asterisk.org/r/1717/

2012-02-08 21:25 +0000 [r354450]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: Restore some variables removed by the
	  'cppcheck' patch that were actually needed.

2012-02-08 20:49 +0000 [r354429]  Walter Doekes <walter+asterisk@wjd.nu>

	* apps/app_dial.c, main/udptl.c, main/pbx.c, addons/chan_ooh323.c,
	  funcs/func_env.c, funcs/func_strings.c, utils/astman.c,
	  main/acl.c, apps/app_disa.c, apps/app_alarmreceiver.c,
	  apps/app_queue.c, channels/chan_iax2.c,
	  addons/ooh323c/src/memheap.c, channels/chan_usbradio.c,
	  channels/chan_dahdi.c, apps/app_osplookup.c,
	  channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c,
	  main/ast_expr2f.c, apps/app_minivm.c, formats/format_h263.c,
	  addons/chan_mobile.c, apps/app_chanspy.c, main/ast_expr2.fl,
	  apps/app_voicemail.c: Avoid cppcheck warnings; removing unused
	  vars and a bit of cleanup. Patch by: Clod Patry Review:
	  https://reviewboard.asterisk.org/r/1651

2012-02-08 15:28 +0000 [r354395]  Kinsey Moore <kmoore@digium.com>

	* CHANGES: Add CHANGES documentation for the "pri set debug"
	  bitmask change (related to ASTERISK-17159)

2012-02-07 21:33 +0000 [r354360]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
	  Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
	  instead of "" 2. Don't set ipaddr or port to the string "(null)"
	  when they are empty 3. Add missing required fields, set default
	  for lastms to 0, and modify the length of the ipaddr field to 45
	  in the Postgresql realtime.sql file. (closes issue
	  ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
	  ........ Merged revisions 354348 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354349 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-07 18:07 +0000 [r354312-354314]  Sean Bright <sean@malleable.com>

	* contrib/scripts/live_ast: Continuation of last patch - since
	  LIVE_AST_LD_PATH_EXTRA will now never be empty, don't check for
	  it, instead of check if LD_LIBRARY_PATH is already set and if so,
	  append LIVE_AST_LD_PATH_EXTRA properly.

	* contrib/scripts/live_ast: Include live/usr/lib in the shared
	  library search path to that we pick up libasteriskssl.so at run
	  time when using live_ast.

	* contrib/scripts/live_ast: Whitespace only (remove trailing
	  spaces)

2012-02-07 15:29 +0000 [r354275]  Jonathan Rose <jrose@digium.com>

	* /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
	  for cdr_pgsql. Prior to this patch, attempts to reload
	  cdr_pgsql.so would cause the column list to keep its current data
	  and then add a second copy during the reload. This would cause
	  attempts to log the CDR to the database to fail. This patch also
	  cleans up some unnecessary null checks for ast_free and deals
	  with a few potential locking problems. (closes issue
	  ASTERISK-19216) Reported by: Jacek Konieczny Review:
	  https://reviewboard.asterisk.org/r/1711/ ........ Merged
	  revisions 354263 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354270 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-06 23:15 +0000 [r354174-354218]  Richard Mudgett <rmudgett@digium.com>

	* /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
	  extension" command. * Documented dialplan add extension
	  <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
	  of command without the app-data value. There are many
	  applications that do no need any parameters so it is silly to
	  require that field for all commands. * Fixed a couple
	  ast_malloc/ast_free mismatches with ast_add_extension2() calls.
	  (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
	  by: rmudgett ........ Merged revisions 354216 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354217 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sig_pri.h: Restore alternate SIG_PRI_DEBUG_DEFAULT
	  meaning.

2012-02-06 20:18 +0000 [r354165]  Kinsey Moore <kmoore@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c: Allow more control
	  over the output of pri debug This changes the debuglevel of 'pri
	  set debug' to a bit mask allowing the user to independently
	  select bits of output: 1 libpri internals including state machine
	  2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump
	  of the full frames Additionally, this ensures that the meaning of
	  "on" does not change and intrudces intense and hex to simplify
	  usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy

2012-02-06 17:33 +0000 [r354120]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Add missing headers to AMI UnParkedCall event
	  to uniquely identify the call. The AMI UnParkedCall event was
	  missing the Parkinglot and Uniqueid headers that the AMI
	  ParkedCall event contains. (closes issue ASTERISK-19240) Reported
	  by: Michael Yara ........ Merged revisions 354116 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354119 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-06 16:38 +0000 [r354084]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c, UPGRADE.txt: Make the 'c' option to MeetMe
	  work even if the 'q' option is used. (closes issue
	  ASTERISK-17053) Reported by: justdave

2012-02-05 10:58 +0000 [r354046]  Russell Bryant <russell@russellbryant.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, res/res_corosync.c (added),
	  configure.ac, configs/res_corosync.conf.sample (added), res/ais
	  (removed), UPGRADE.txt, configs/ais.conf.sample (removed),
	  CHANGES, makeopts.in: Replace res_ais with a new module,
	  res_corosync. This patch removes res_ais and introduces a new
	  module, res_corosync. The OpenAIS project is deprecated and is
	  now just a wrapper around Corosync. This module provides the same
	  functionality using the same core infrastructure, but without the
	  use of the deprecated components. Technically res_ais could have
	  been used with an AIS implementation other than OpenAIS, but that
	  is the only one I know of that was ever used. Review:
	  https://reviewboard.asterisk.org/r/1700/

2012-02-03 21:33 +0000 [r354001]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
	  due to r335976 Bad locking order was added to chan_agent to
	  prevent segfaults from having no locking in a patch by irroot.
	  This patch addresses the bad locking order by releasing locks
	  before getting the right locking order to stop deadlocks from
	  occuring when doing multiple interactions with agents. (closes
	  issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
	  https://reviewboard.asterisk.org/r/1708/ ........ Merged
	  revisions 353999 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 354000 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-03 16:50 +0000 [r353964]  Kinsey Moore <kmoore@digium.com>

	* UPGRADE.txt, cdr/cdr_adaptive_odbc.c,
	  configs/cdr_adaptive_odbc.conf.sample: Support schema selection
	  in cdr_adaptive_odbc Asterisk now supports using ODBC with
	  databases where a single schema must be selected. Previously,
	  INSERTs would fail because they did not take into account extra
	  fields cause by having multiple schemas. This also corrects some
	  SQL resource leaks. (closes issue ASTERISK-17106) Patch-by:
	  Alexander Frolkin Patch-by: Tilgnman Lesher

2012-02-03 16:23 +0000 [r353963]  Jonathan Rose <jrose@digium.com>

	* /, res/res_fax.c: Fixes a segfault occuring when performing
	  attended transfer with FAXOPT(gateway)=yes (closes issue
	  ASTERISK-19184) Reported by: Alexandr ........ Merged revisions
	  353962 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-02 22:28 +0000 [r353917]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
	  cause an infinite loop After R340970 Asterisk was still polling
	  the RTCP file descriptor after RTCP is shut down and removed. If
	  the descriptor happened to have data ready when the removal
	  occured then Asterisk would go into an infinite loop trying to
	  read data that it can never actually access. This change disables
	  the audio RTCP file descriptor for the duration of the T.38
	  transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
	  Vrban ........ Merged revisions 353915 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353916 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-02 20:18 +0000 [r353872]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
	  Restore the 'w' modifier support for ISDN spans.
	  Dial(DAHDI/g0/1234w888) This feature also causes the sending
	  complete ie to be sent for switch types that do not automatically
	  send the ie. (EuroISDN/ETSI) The main difference between dialing
	  Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
	  sending of the sending complete ie. (closes issue ASTERISK-19176)
	  Reported by: rmudgett Tested by: rmudgett ........ Merged
	  revisions 353867 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353868 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-02 18:55 +0000 [r353821]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /, main/http.c, configs/manager.conf.sample,
	  include/asterisk/manager.h, configs/http.conf.sample: Fix TLS
	  port binding behavior as well as reload behavior: * Removes
	  references to tlsbindport from http.conf.sample and
	  manager.conf.sample * Properly bind to port specified in
	  tlsbindaddr, using the default port if specified. * On a reload,
	  properly close socket if the service has been disabled. A note
	  has been added to UPGRADE.txt to indicate how ports must be set
	  for TLS. (closes issue ASTERISK-16959) reported by Olaf
	  Holthausen (closes issue ASTERISK-19201) reported by Chris
	  Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas
	  Review: https://reviewboard.asterisk.org/r/1709 ........ Merged
	  revisions 353770 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353820 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-02 17:07 +0000 [r353725-353772]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Fix sip show peers port output, align
	  columns, and fix ami port output. A previous patch I committed
	  from ASTERISK-16930 unexpectedly changed some output for the AMI
	  action "sippeers" which this patch changes back. Also, this
	  aligns the output for the cli command "sip show peers" and fixes
	  another issue that patch introduced by using
	  ast_sockaddr_stringify calls multiple times without immediately
	  using the pointer. I also went ahead and did a little janitorial
	  work to clean up whitespace in _sip_show_peers. (issue
	  ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
	  Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
	  Walter Doekes (license 5674) ........ Merged revisions 353769
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 353771 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers
	  for various functions in chan_sip There are a number of cleaner
	  looking wrappers for ast_sockaddr_stringify_fmt available which
	  are slightly more readable than using a direct call to
	  ast_sockaddr_stringify_fmt. This patch switches a number of those
	  calls in chan_sip to use those wrappers and is generally
	  harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
	  Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
	  Michael L. Young (license 5026) ........ Merged revisions 353720
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 353721 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-01 19:53 +0000 [r353647-353685]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_unistim.c, channels/chan_multicast_rtp.c,
	  channels/chan_local.c, addons/chan_ooh323.c,
	  channels/chan_vpb.cc, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/channel.c, channels/chan_phone.c,
	  channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c,
	  pbx/pbx_spool.c, channels/chan_skinny.c, main/features.c,
	  channels/sig_analog.h, channels/chan_alsa.c,
	  apps/app_confbridge.c, addons/chan_mobile.c, channels/sig_ss7.c,
	  channels/chan_mgcp.c, main/pbx.c, channels/sig_ss7.h,
	  channels/chan_sip.c, channels/chan_bridge.c,
	  channels/chan_agent.c, include/asterisk/channel.h,
	  channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
	  channels/chan_usbradio.c, channels/chan_jingle.c,
	  channels/sig_pri.h, channels/chan_misdn.c, channels/chan_h323.c,
	  channels/chan_nbs.c, include/asterisk/pbx.h: Constify some more
	  channel driver technology callback parameters. Review:
	  https://reviewboard.asterisk.org/r/1707/

	* cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample,
	  cel/cel_odbc.c, configs/cel.conf.sample, cel/cel_manager.c,
	  cel/cel_tds.c, configs/cel_pgsql.conf.sample,
	  configs/cel_odbc.conf.sample, main/cel.c,
	  configs/cel_custom.conf.sample: Remove inconsistency in CEL
	  eventtype for user defined events. The CEL eventtype field for
	  ODBC and PGSQL backends should be USER_DEFINED instead of the
	  user defined event name supplied by the CELGenUserEvent
	  application. If the field is output as a number, the user defined
	  name does not have a value and is always output as 21 for
	  USER_DEFINED and the userdeftype field would be required to
	  supply the user defined name. The following CEL backends
	  (cel_odbc, cel_pgsql, cel_custom, cel_manager, and
	  cel_sqlite3_custom) can be independently configured to remove
	  this inconsistency. * Allows cel_manager, cel_custom, and
	  cel_sqlite3_custom to behave the same way. (closes issue
	  ASTERISK-17189) Reported by: Bryant Zimmerman Review:
	  https://reviewboard.asterisk.org/r/1669/

	* main/channel.c, include/asterisk/channel.h: Fix ExtenSpy and
	  simplify the channel search functions. When ast_channel name was
	  opaquified, the channel search functions did not get converted
	  correctly. As a result ExtenSpy which uses a channel iterator
	  search by exten@context could never find anything. * Updated the
	  doxygen documentation for the search functions in channel.h.
	  Review: https://reviewboard.asterisk.org/r/1702/

2012-02-01 15:59 +0000 [r353600]  Sean Bright <sean@malleable.com>

	* /, include/asterisk/audiohook.h: Resolve an overlap in the
	  ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
	  AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
	  unintended side effects. This patch moves
	  AST_AUDIOHOOK_TRIGGER_WRITE, and updates
	  AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
	  This will affect existing modules that use these flags, so be
	  sure to recompile as necessary. (closes issue ASTERISK-19246)
	  Reported by: feyfre ........ Merged revisions 353598 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353599 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-01 15:07 +0000 [r353552]  Matthew Jordan <mjordan@digium.com>

	* /, contrib/init.d/etc_default_asterisk: Added clarification for
	  the VERBOSITY setting to etc_default_asterisk Clarified that
	  using the VERBOSITY setting in etc_default_asterisk is the same
	  as using the -v command line switch, which causes Asterisk to
	  launch in console mode. (closes issue ASTERISK-17030) Reported
	  by: Jonas ........ Merged revisions 353550 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353551 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-02-01 00:08 +0000 [r353504]  Terry Wilson <twilson@digium.com>

	* /, res/res_calendar.c: Allow res_calendar to be unloaded The
	  calendaring tech modules depend on res_calendar and initially
	  res_calendar just bumped the use count so that it couldn't be
	  unloaded. res_calendar can potentially create many threads and
	  I've seen issues where the Asterisk shutdown has failed where it
	  looked like these threads could be the culprit. This patch adds
	  unload support for res_calendar. Unloading res_calendar will also
	  unload the dependant tech modules as well. (closes issue
	  ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
	  ........ Merged revisions 353502 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353503 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-31 17:26 +0000 [r353466]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /, include/asterisk/channel.h: Fix memory leak in
	  error paths for action_originate(). * Fix memory leak of vars in
	  error paths for action_originate(). * Moved struct
	  fast_originate_helper tech and data members to stringfields. *
	  Simplified ActionID header handling for fast_originate(). * Added
	  doxygen note to ast_request() and ast_call() and the associated
	  channel callbacks that the data/addr parameters should be treated
	  as const char *. Review: https://reviewboard.asterisk.org/r/1690/
	  ........ Merged revisions 353454 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353463 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-30 23:58 +0000 [r353418]  Terry Wilson <twilson@digium.com>

	* main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
	  Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
	  currently takes a pointer to an ast_sockaddr and updates it
	  anytime an address resolves to something different. There are a
	  couple of issues with this. First, the ast_sockaddr is usually
	  the address of an ast_sockaddr inside a refcounted struct and we
	  never bump the refcount of those structs when using dnsmgr. This
	  makes it possible that a refresh could happen after the
	  destructor for that object is called (despite ast_dnsmgr_release
	  being called in that destructor). Second, the module using dnsmgr
	  cannot be aware of an address changing without polling for it in
	  the code. If an action needs to be taken on address update (like
	  re-linking a SIP peer in the peers_by_ip table), then polling for
	  this change negates many of the benefits of having dnsmgr in the
	  first place. This patch adds a function to the dnsmgr API that
	  calls an update callback instead of blindly updating the address
	  itself. It also moves calls to ast_dnsmgr_release outside of the
	  destructor functions and into cleanup functions that are called
	  when we no longer need the objects and increments the refcount of
	  the objects using dnsmgr since those objects are stored on the
	  ast_dnsmgr_entry struct. A helper function for returning the
	  proper default SIP port (non-tls vs tls) is also added and used.
	  This patch also incorporates changes from a patch posted by Timo
	  Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
	  ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
	  ........ Merged revisions 353371 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353397 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-30 22:44 +0000 [r353347-353370]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, channels/chan_sip.c: Merged revisions 353369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300
	  (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31
	  Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq
	  numbers. Missed in R353320 ........ ................

	* channels/sip/include/dialog.h, /, channels/chan_sip.c,
	  channels/sip/include/sip.h: Merged revisions 353321 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300
	  (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan
	  2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number
	  value MUST be expressible as a 32-bit unsigned integer * fix: use
	  %u instead of %d when dealing with CSeq numbers - to remove
	  possibility of -ve numbers. * fix: change all uses of seqno and
	  friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
	  Summary of CSeq numbers. An initial CSeq number must be less than
	  2^31 A CSeq number can increase in value up to 2^32-1 An
	  incrementing CSeq number must not wrap around to 0. Tested with
	  Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
	  Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1699/ ........
	  ................

2012-01-30 21:34 +0000 [r353262-353319]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Correct serious flaw in the top-level Makefile.

	* include/asterisk.h, /, main/Makefile, main/libasteriskssl.c
	  (added), configure.ac, Makefile.moddir_rules, main/ssl.c
	  (removed), addons, CHANGES, include/asterisk/optional_api.h,
	  Makefile, build_tools/mkpkgconfig, configure, main, makeopts.in,
	  build_tools/make_defaults_h, main/libasteriskssl.exports.in
	  (added): Address OpenSSL initialization issues when using
	  third-party libraries. When Asterisk is used with various
	  third-party libraries (CURL, PostgresSQL, many others) that have
	  the ability themselves to use OpenSSL, it is possible for
	  conflicts to arise in how the OpenSSL libraries are initialized
	  and shutdown. This patch addresses these conflicts by 'wrapping'
	  the important functions from the OpenSSL libraries in a new
	  shared library that is part of Asterisk itself, and is loaded in
	  such a way as to ensure that *all* calls to these functions will
	  be dispatched through the Asterisk wrapper functions, not the
	  native functions. This new library is optional, but enabled by
	  default. See the CHANGES file for documentation on how to disable
	  it. Along the way, this patch also makes a few other minor
	  changes: * Changes MODULES_DIR to ASTMODDIR throughout the build
	  system, in order to more closely match what is used during
	  run-time configuration. * Corrects some errors in the configure
	  script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. *
	  Adds a new variable for linker flags in the build system
	  (DYLINK), used for producing true shared libraries (as opposed to
	  the dynamically loadable modules that the build system produces
	  for 'regular' Asterisk modules). * Moves the Makefile bits that
	  handle installation and uninstallation of the main Asterisk
	  binary into main/Makefile from the top-level Makefile. * Moves a
	  couple of useful preprocessor macros from optional_api.h to
	  asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/

	* /, channels/chan_sip.c: Clarify log WARNING message when
	  port-zero SDP 'm' lines received. Previously, if an m-line in an
	  SDP offer or answer had a port number of zero, that line was
	  skipped, and resulted in an 'Unsupported SDP media type...'
	  warning message. This was misleading, as the media type was not
	  unsupported, but was ignored because the m-line indicated that
	  the media stream had been rejected (in an answer) or was not
	  going to be used (in an offer). ........ Merged revisions 353260
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 353261 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-29 22:33 +0000 [r353224]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Allow softkey reject while device onhook.
	  Fixes up softkey endcall. Previous code was a copy of onhook, now
	  allows for endcall softkey to be used while device is still
	  onhook.

2012-01-29 02:45 +0000 [r353177]  Russell Bryant <russell@russellbryant.com>

	* /, main/netsock.c: Find even more network interfaces. The
	  previous change made the code look for emN and pciN in addition
	  to what it did originally, which was search for ethN. However, it
	  needed to be looking for pciN#N, so that's what it does now. This
	  also moves the memset() to be before every ioctl(). ........
	  Merged revisions 353175 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353176 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-28 14:52 +0000 [r353128]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for
	  slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz
	  signed linear (PCM) audio for quite some time, but some endpoints
	  refer to it as 'L16-256'. This commit adds this as an alias for
	  the existing format. ........ Merged revisions 353126 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353127 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-28 04:31 +0000 [r353079]  Russell Bryant <russell@russellbryant.com>

	* /, main/netsock.c: Update ast_set_default_eid() to find more
	  network interfaces. As of Fedora 15, ethN is not the name of
	  ethernet interfaces. The names are emN or pciN. Update some code
	  that searched for interfaces named ethN to look for the new
	  names, as well. For more information about why this change was
	  made, see this page: http://domsch.com/blog/?p=455 ........
	  Merged revisions 353077 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 353078 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-27 21:38 +0000 [r352996-353040]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_queue.c: Audit of ao2_iterator_init() usage for v10.
	  Missed one. ........ Merged revisions 353039 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, tests/test_format_api.c: Audit of ao2_iterator_init() usage
	  for v10. Fix double format_cap iterator cleanup. ........ Merged
	  revisions 352992 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-27 19:26 +0000 [r352981]  Jonathan Rose <jrose@digium.com>

	* /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
	  with no valid channel not close AMI session. I also went ahead
	  and took a little time to make sure that the manager value
	  AMI_SUCCESS was used instead of just return 0 being thrown around
	  everywhere since that's how we handle this stuff these days.
	  (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
	  res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
	  (license 5766) ........ Merged revisions 352959 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352965 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-27 18:47 +0000 [r352957]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /, channels/chan_sip.c,
	  include/asterisk/indications.h, res/snmp/agent.c,
	  main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c,
	  apps/app_chanspy.c, main/indications.c, res/res_odbc.c,
	  res/res_srtp.c: Audit of ao2_iterator_init() usage for v1.8.
	  Fixes numerous reference leaks and missing ao2_iterator_destroy()
	  calls as a result. Review:
	  https://reviewboard.asterisk.org/r/1697/ ........ Merged
	  revisions 352955 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352956 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-27 15:57 +0000 [r352916]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_exchange.c, res/res_calendar_caldav.c,
	  res/res_calendar.c: Add aresult variable for CALENDAR_WRITE This
	  patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show
	  whether or not CALENDAR_WRITE has passed. This patch also adds
	  some debugging for caldav PUT responses and no longer treats
	  responses with no body as an error (as a PUT gets a 201 Created
	  with no body). (closes issue ASTERISK-16903) Reported by: Clod
	  Patry Tested by: Terry Wilson Patches: calendarstatus.diff
	  uploaded by Clod Patry (License #5138), slightly modified by
	  Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ -
	  This line, and those below, will be ignored-- M
	  res/res_calendar.c M res/res_calendar_exchange.c M
	  res/res_calendar_caldav.c

2012-01-27 00:11 +0000 [r352864]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
	  revisions 352863 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300
	  (Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan
	  2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be
	  representable using a non-negative 32 bit integer. If a BLF
	  subscription exists for long enough, using %d may print negative
	  version numbers. Unlikely, as 2^32 at 1 update per second is ~137
	  years, or half that before the versions number started going
	  negative. Tested with Asterisk 1.8.8.2 with Grandstream phones.
	  alecdavis (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1694/ ........
	  ................

2012-01-26 20:44 +0000 [r352821]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell
	  asterisk core to generate DTMF sounds). (Closes issue
	  ASTERISK-19233) Reported by: Matt Behrens Patches:
	  chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
	  ........ Merged revisions 352807 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352817 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-26 19:09 +0000 [r352757]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
	  create_addr_from_peer For whatever reason, we don't have a single
	  function for copying data like this from SIP peers to the SIP
	  pvt. This patch adds the copying of amaflags to the sip_pvt, but
	  it would probably be worth discussing this function along with
	  the others that essentially just copy some amount of data from a
	  peer to a private. (Closes issue ASTERISK-19029) Reported by:
	  Matt Lehner ........ Merged revisions 352755 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352756 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-26 06:36 +0000 [r352706]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, channels/chan_sip.c: Merged revisions 352705 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300
	  (Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan
	  2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make
	  similar to other Notify messages. sample output: <?xml
	  version="1.0"?> <dialog-info
	  xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
	  state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
	  <state>terminated</state> </dialog> </dialog-info> Tested with
	  Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
	  Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1693/ ........
	  ................

2012-01-25 22:25 +0000 [r352659]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
	  compiler error (gcc 4.6.2) ........ Merged revisions 352643 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352651 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 21:31 +0000 [r352626]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, include/asterisk/version.h (added), main/test.c,
	  build_tools/make_version_h (removed), include/asterisk: Remove
	  "asterisk/version.h" in favor of "asterisk/ast_version.h". A long
	  time ago, in a land far far away, we added
	  "asterisk/ast_version.h", which provides the ast_get_version()
	  and ast_get_version_num() functions. These were added so that
	  modules that needed the version information for the Asterisk
	  instance they were loaded in could actually get it (as opposed
	  the version that they were compiled against). We changed
	  everything in the tree to use the new mechanism (although later
	  main/test.c was added using the old method). However, the old
	  mechanism was never removed, and as a result, new code is still
	  trying to use it. This commit removes asterisk/version.h and
	  replaces it with a header that will generate a compile-time error
	  if you try to use it (the error message tells you which header
	  you should use instead). It also removes the Makefile and
	  build_tools bits that generated the file, and it updates
	  main/test.c to use the 'proper' method of getting the Asterisk
	  version information. This is an API change and thus is being
	  committed for trunk only, but it's a fairly minor one and
	  definitely improves the situation for out-of-tree modules.

2012-01-25 17:33 +0000 [r352565]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Remove some extraneous debugging from
	  registry memleak fix ........ Merged revisions 352551 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352556 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 17:23 +0000 [r352538]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c, CHANGES, main/message.c,
	  channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside
	  of calls. * Fix authenticate MESSAGE losing custom headers added
	  by the MESSAGE_DATA function in the authorization attempt. * Pass
	  up better From header contents for SIP to use. Now is in the
	  "display-name" <URI> format expected by MessageSend. (Note that
	  this is a behavior change that could concievably affect some
	  people.) * Block user from adding standard headers that are added
	  automatically. (To, From,...) * Allow the user to override the
	  Content-Type header contents sent by MessageSend. * Decrement
	  Max-Forwards header if the user transferred it from an incoming
	  message. * Expand SIP short header names so the dialplan and
	  other code only has to deal with the full names. * Documents what
	  SIP expects in the MessageSend(from) parameter. (closes issue
	  ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917)
	  Reported by: Shaun Clark Review:
	  https://reviewboard.asterisk.org/r/1683/ ........ Merged
	  revisions 352520 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 17:02 +0000 [r352519]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Clean up some SIP registry-related memory
	  leaks 1) Be sure and free at unload the epa_backend we allocate
	  at startup 2) Do the same sip_registry cleanup at unload we do at
	  reload Review: https://reviewboard.asterisk.org/r/1689/ ........
	  Merged revisions 352514 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352515 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 16:54 +0000 [r352517]  Kevin P. Fleming <kpfleming@digium.com>

	* main/format.c, /, main/format_cap.c, main/format_pref.c:
	  Eliminate unnecessary rebuilds of main/format*.c. These files
	  have no need to include "asterisk/version.h", and doing so forces
	  them to be rebuilt each time a Subversion checkout moves between
	  'modified' and 'unmodified' states. ........ Merged revisions
	  352516 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 16:42 +0000 [r352513]  Jonathan Rose <jrose@digium.com>

	* /, configs/sip.conf.sample: Redocuments sip types peer, user,
	  friend in sip.conf.sample There was faulty information in the
	  sample config describing user as a synonym for friend so it has
	  been changed to better elaborate on the differences between the
	  three entity types. (closes issue ASTERISK-15537) Reported by:
	  yarique ........ Merged revisions 352511 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352512 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-25 01:21 +0000 [r352475]  Terry Wilson <twilson@digium.com>

	* channels/chan_vpb.cc: Fix channel opaquification of stringfields
	  for chan_vpb

2012-01-24 22:28 +0000 [r352431]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Don't do a DNS lookup on an outbound
	  REGISTER host if there is an outbound proxy configured. (closes
	  issue ASTERISK-16550) reported by: Olle Johansson ........ Merged
	  revisions 352424 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352430 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-24 20:37 +0000 [r352377]  Jonathan Rose <jrose@digium.com>

	* /, sounds/Makefile: Set core sounds version to 1.4.22. Now that
	  we have the right license for the Russian 1.4.22 sounds as well
	  as the sounds for the Australian English 1.4.22 sounds, we can
	  finally set the sounds to use 1.4.22! (closes issue
	  ASTERISK-18978) Reported by: Cameron Twomey Patches:
	  confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
	  uploaded by Cameron Twomey ........ Merged revisions 352367 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352373 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-24 20:12 +0000 [r352348]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, addons/chan_ooh323.c, main/say.c,
	  apps/app_record.c, apps/app_sayunixtime.c, channels/chan_iax2.c,
	  main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
	  channels/chan_alsa.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
	  apps/app_dial.c, main/pbx.c, apps/app_page.c,
	  channels/chan_bridge.c, apps/app_privacy.c,
	  channels/chan_agent.c, apps/app_disa.c,
	  include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c,
	  main/cel.c, res/res_monitor.c, apps/app_playback.c,
	  apps/app_speech_utils.c, channels/chan_misdn.c,
	  funcs/func_channel.c, apps/app_chanspy.c, apps/app_voicemail.c,
	  channels/chan_unistim.c, channels/chan_multicast_rtp.c,
	  apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c,
	  apps/app_readexten.c, apps/app_userevent.c,
	  res/res_musiconhold.c, channels/chan_gtalk.c,
	  apps/app_followme.c, main/cdr.c, main/channel.c,
	  channels/chan_phone.c, main/dial.c, main/manager.c,
	  apps/app_minivm.c, res/res_agi.c, main/app.c,
	  apps/app_confbridge.c, main/image.c, apps/app_directory.c,
	  addons/chan_mobile.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  apps/app_parkandannounce.c, channels/chan_sip.c, res/res_fax.c,
	  main/channel_internal_api.c, channels/chan_console.c,
	  channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
	  funcs/func_global.c, channels/chan_jingle.c,
	  channels/chan_usbradio.c, channels/chan_h323.c, main/file.c,
	  res/snmp/agent.c, channels/chan_nbs.c, apps/app_stack.c,
	  addons/app_saycountpl.c: Opaquify channel stringfields Continue
	  channel opaque-ification by wrapping all of the stringfields.
	  Eventually, we will restrict what can actually set these
	  variables, but the purpose for now is to hide the implementation
	  and keep people from adding code that directly accesses the
	  channel structure. Semantic changes will follow afterward.
	  Review: https://reviewboard.asterisk.org/r/1661/

2012-01-24 17:04 +0000 [r352293]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_odbc.c: Fix locking issues with channel datastores
	  in func_odbc.c. * Fixed a potential memory leak when an existing
	  datastore is manually destroyed by inline code instead of calling
	  ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
	  Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
	  ........ Merged revisions 352291 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352292 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-23 20:31 +0000 [r352229-352232]  Mark Michelson <mmichelson@digium.com>

	* /, main/features.c: Fix grammar of comment. ........ Merged
	  revisions 352230 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352231 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/features.c: Fix blind transfers from failing if an 'h'
	  extension is present. This prevents the 'h' extension from being
	  run on the transferee channel when it is transferred via a native
	  transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
	  Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
	  ASTERISK-19173 by Mark Michelson (license 5049) Review:
	  https://reviewboard.asterisk.org/r/1685 ........ Merged revisions
	  352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 352228 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-23 19:22 +0000 [r352166]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17,
	  V27, V29) before starting spandsp layer While the FAXOPT function
	  could be used to set the modem capabilities, the input to that
	  function was not being applied correctly to the spandsp layer.
	  This patch applies the current model capabilities before starting
	  the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
	  Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
	  Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
	  5081) spandsp-modems-10.diff uploaded by mnicholson (license
	  5081) ........ Merged revisions 352144 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352149 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-23 18:34 +0000 [r352093-352134]  Jonathan Rose <jrose@digium.com>

	* configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
	  Add an announcement option to music-on-hold - plays sound when
	  put on hold/between songs This is a feature patch which allows an
	  'announcement' option to be specified in musiconhold.conf which
	  should be set to the name of a sound. If a valid sound is
	  specified for this option, then it will be played on that music
	  on hold class whenever a channel bound to that class is put on
	  hold as well as when Asterisk is able to detect that a song has
	  ended before starting the next song (excludes external players).
	  (closes ASTERISK-18977) Reported by: Timo Teräs Patches:
	  asterisk-moh-announcement.diff uploaded by Timo Teräs (license
	  5409)

	* CHANGES, apps/app_mixmonitor.c: Adds the ability to stop specific
	  mixmonitors by using unique IDs set at monitor launch. MixMonitor
	  receives a new option i(channel_variable) which stores the unique
	  id at said variable. StopMixMonitor now accepts ID as an optional
	  argument, which if included will make StopMixMonitor specifically
	  target the mixmonitor on that particular channel. CLI commands
	  and AMI actions have been ammended to work with the IDs as well.
	  In addition, monitors across a channel can now be listed be
	  listed via CLI command "mixmonitor list <channel>" which will
	  display all of the mixmonitors active on that channel along with
	  the files they each have open. Created by Sergio González Martín.
	  (closes issue ASTERISK-19096) Reported by: Sergio González Martín
	  Review: https://reviewboard.asterisk.org/r/1643/ Review:
	  https://reviewboard.asterisk.org/r/1682/

2012-01-23 17:36 +0000 [r352092]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
	  defined enum values. The invalid value used when notifycid was
	  enabled was benign. As far as the code was concerned -1 and 1 are
	  equivalent. (closes issue ASTERISK-19232) Reported by: Eike
	  Kuiper ........ Merged revisions 352090 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352091 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-21 00:23 +0000 [r352041]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time
	  unit inconsistency. Note: Noone calls ast_app_dtget() with the
	  timeout parameter of zero so the bad code normally will never get
	  executed. * Fix unnecessary floating point division in
	  func_timeout.c timeout_write() when all other values are
	  integers. (closes issue ASTERISK-16817) Reported by: Dmitry
	  Andrianov ........ Merged revisions 352029 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352035 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-21 00:11 +0000 [r352018-352019]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Remove XXX comment that is not necessary.
	  ........ Merged revisions 352016 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352017 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Fix RTP reference leak. If a blind
	  transfer were initiated using a REFER without a prior reINVITE to
	  place the call on hold, AND if Asterisk were sending RTCP
	  reports, then there was a reference for the RTP instance of the
	  transferer. This fixes the issue by merging two similar but
	  slightly conflicting sections of code into a single area. It also
	  adds a stop_media_flows() call in the case that the transferer's
	  UA never sends a BYE to us like it is supposed to. (issue
	  ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
	  ........ Merged revisions 352014 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 352015 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-20 23:05 +0000 [r351977]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Make CLI sip show channel list the complete
	  route set. (closes issue ASTERISK-16877) Reported by: klaus3000
	  Patches: show-complete-routeset-patch.txt (license #5054) patch
	  uploaded by klaus3000 (modified)

2012-01-20 21:26 +0000 [r351939]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c, UPGRADE.txt: SIP session timeout AMI event
	  Add an AMI event in the Call category that is issued when a call
	  is terminated due to either RTP stream inactivity or SIP session
	  timer expiration. Event description: Event: SessionTimeout
	  Source: source Channel: channel-name Uniqueid: channel-unique-id
	  `source` can be either RTPTimeout or SIPSessionTimer (closes
	  issue ASTERISK-16467) Patch-by: Kirill Katsnelson

2012-01-20 20:47 +0000 [r351900-351913]  Mark Michelson <mmichelson@digium.com>

	* main/features.c, UPGRADE.txt, CHANGES,
	  configs/features.conf.sample: Various parking improvements. *
	  Adds per-parking lot options comebackcontext and comebackdialtime
	  * Makes comebacktoorigin settable per parking lot * Sets a PARKER
	  channel variable when comebacktoorigin is disabled (closes issue
	  ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches:
	  asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff
	  by Mitch Sharp (bluecrow76) license 5231 with updates by me.
	  Review: https://reviewboard.asterisk.org/r/1674 Review:
	  https://reviewboard.asterisk.org/r/963 Reviewed by Richard
	  Mudgett

	* apps/app_mixmonitor.c: Prevent potential buffer overflow on AMI
	  MixMonitor command. Don't be alarmed. This only affected trunk,
	  and it would have required manager access to your system.

2012-01-20 19:36 +0000 [r351817-351862]  Kinsey Moore <kmoore@digium.com>

	* /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code
	  These changes are in a file that is not compiled by default, and
	  so were missed on earlier checks. ........ Merged revisions
	  351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 351861 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore
	  LSF_check function calls from set/unused variable removal These
	  functions are not noops and modify the array that is passed in.
	  Thanks for the catch Richard. ........ Merged revisions 351818
	  from http://svn.asterisk.org/svn/asterisk/branches/10

	* /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove
	  more set, but unused variables in the ilbc codec GCC 4.6.3 caught
	  these in dev mode as well. ........ Merged revisions 351816 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-20 16:00 +0000 [r351764]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Adds setting of mwi_from field to
	  check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
	  By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
	  5242) ........ Merged revisions 351759 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351762 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-20 16:00 +0000 [r351763]  Matthew Jordan <mjordan@digium.com>

	* /, codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from
	  helpfun in ilbc codec gcc version 4.6.2 caught an unused variable
	  in the ilbc codec library. This would prevent compilation with
	  --enable-dev-mode; variable removed. ........ Merged revisions
	  351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 351761 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-20 13:12 +0000 [r351709]  Stefan Schmidt <sst@sil.at>

	* /, contrib/asterisk-ng-doxygen: enable doxygen build for files in
	  the channels/sip folder like reqresp_parser.c ........ Merged
	  revisions 351707 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351708 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-19 23:31 +0000 [r351667]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
	  fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
	  get_calleridname() parsing and ensure that the output buffer is
	  nul terminated. * Make get_calleridname() truncate the name it
	  parses if the given buffer is too small rather than abandoning
	  the parse and not returning anything for the name. Adjusted
	  get_calleridname_test() unit test to handle the truncation
	  change. * Fix get_in_brackets_test() unit test to check the
	  results of get_in_brackets() correctly. * Fix
	  parse_name_andor_addr() to not return the address of a local
	  buffer. This function is currently not used. * Fix potential NULL
	  pointer dereference in sip_sendtext(). * No need to
	  memset(calleridname) in check_user_full() or tmp_name in
	  get_name_and_number() because get_calleridname() ensures that it
	  is nul terminated. * Reply with an accurate response if
	  get_msg_text() fails in receive_message(). This is academic in
	  v1.8 because get_msg_text() can never fail. ........ Merged
	  revisions 351618 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351646 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-19 22:44 +0000 [r351613]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c, /: Correct output of RTCP jitter
	  statistics in SR and RR reports Change the RTCP RR and SR
	  generation code to convert Asterisk's internal jitter statistics
	  to be represented in RTP timestamp units based on the rate of the
	  codec in use instead of in seconds. (closes issue ASTERISK-14530)
	  ........ Merged revisions 351611 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351612 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-19 21:55 +0000 [r351561]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates
	  doubling the :port part of SIP Notify Message-Account headers.
	  This patch prevents the domain string from getting mangled during
	  the initreqprep step by moving the initialization to before its
	  immediate use. It also documents this pitfall for the
	  ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
	  by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
	  ........ Merged revisions 351559 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351560 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-19 21:13 +0000 [r351506]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Prevent crash when an SDP offer is
	  received with an encrypted video stream when support for video is
	  disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
	  Reported by: Catalin Sanda ........ Merged revisions 351504 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351505 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-18 21:06 +0000 [r351452]  Matthew Jordan <mjordan@digium.com>

	* codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h
	  (added), codecs/ilbc/iLBC_test.c (added),
	  codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c
	  (added), codecs/ilbc/packing.c (added),
	  codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h
	  (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c
	  (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c
	  (added), codecs/ilbc/iLBC_encode.c (added),
	  codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
	  codecs/ilbc/iLBC_define.h (added), codecs/ilbc/FrameClassify.c
	  (added), codecs/ilbc/enhancer.h (added), codecs/ilbc/lsf.h
	  (added), codecs/ilbc/extract-cfile.awk (added),
	  codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
	  codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c
	  (added), codecs/ilbc/LICENSE_ADDENDUM (added),
	  codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added),
	  codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added),
	  codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added),
	  codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c
	  (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h
	  (added), codecs/ilbc/iLBC_decode.h (added),
	  codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added),
	  codecs/ilbc/gainquant.c (added), codecs/ilbc/hpInput.c (added),
	  codecs/ilbc/hpOutput.c (added), codecs/ilbc/iCBSearch.h (added),
	  codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added),
	  codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
	  codecs/ilbc/hpInput.h (added), codecs/ilbc/PATENTS (added),
	  codecs/ilbc/StateSearchW.c (added), codecs/ilbc/hpOutput.h
	  (added), codecs/codec_ilbc.c, contrib/scripts/get_ilbc_source.sh,
	  codecs/ilbc/LICENSE (added), codecs/ilbc/LPCencode.h (added),
	  codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c
	  (added): Include iLBC source code for distribution with Asterisk
	  This patch includes the iLBC source code for distribution with
	  Asterisk. Clarification regarding the iLBC source code was
	  provided by Google, and the appropriate licenses have been
	  included in the codecs/ilbc folder. Review:
	  https://reviewboard.asterisk.org/r/1675 Review:
	  https://reviewboard.asterisk.org/r/1649 (closes issue:
	  ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
	  ........ Merged revisions 351450 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351451 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-18 16:02 +0000 [r351409]  Stefan Schmidt <sst@sil.at>

	* /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't
	  recognized a proper callerid name and number from a
	  P-Asserted-Identity cause the header parsing logic was wrong.
	  Changing the parsing functions to the sip header parsing APIs in
	  reqresp_parser.h solves this problem. Review:
	  https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
	  Mark Michelson ........ Merged revisions 351396 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351408 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-17 19:45 +0000 [r351360]  Walter Doekes <walter+asterisk@wjd.nu>

	* Makefile: Fix support for parallel building with make (-j).
	  Previously make -j <N> would cause a race between doing cleanup
	  of certain files (defaults.h, menuselect, ...) and creating them
	  anew. Add a new target that depends on cleanup only and has a
	  submake doing the rest as command string. This way the cleanup
	  goes first. (closes issue ASTERISK-18751) Tested by: Jeremy
	  Kister Reviewed by: Paul Belanger Review:
	  https://reviewboard.asterisk.org/r/1660

2012-01-17 17:23 +0000 [r351311]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, /: Eliminate odd initialization of
	  probation variable. ........ Merged revisions 351306 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351308 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-17 17:15 +0000 [r351290]  Jonathan Rose <jrose@digium.com>

	* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds
	  pjmedia probation concepts to res_rtp_asterisk's learning mode.
	  In order to better handle RTP sources with strictrtp enabled
	  (which is now default in 10) using the learning mode to figure
	  out new sources when they change is handled by checking for a
	  number of consecutive (by sequence number) packets received to an
	  rtp struct based on a new configurable value called 'probation'.
	  Also, during learning mode instead of liberally accepting all
	  packets received, we now reject packets until a clear source has
	  been determined. Review: https://reviewboard.asterisk.org/r/1663/
	  ........ Merged revisions 351287 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351289 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-17 16:56 +0000 [r351288]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Use built-in parsing functions for
	  Contact and Record-Route headers. If a Contact or a Record-Route
	  header had a quoted string with an item in angle brackets, then
	  we would mis-parse it. For instance, "Bob <1234>"
	  <1234@example.org> would be misparsed as having the URI "1234"
	  The fix for this is to use parsing functions from
	  reqresp_parser.h since they are heavily tested and are awesome.
	  (issue ASTERISK-18990) ........ Merged revisions 351284 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351286 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-17 16:08 +0000 [r351235]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Fix udptl issue with initial INVITE
	  introduced by r351027 When an inital INVITE occurs that contains
	  image media, a channel is not yet associated with the SIP dialog.
	  The file descriptor associated with the udptl session needs to be
	  set in initialize_udptl or in sip_new to account for this
	  scenario. ........ Merged revisions 351233 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351234 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-17 01:48 +0000 [r351184]  Russell Bryant <russell@russellbryant.com>

	* /, channels/chan_sip.c: Merged revisions 351183 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r351183 | russell | 2012-01-16 20:43:19 -0500
	  (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012)
	  | 22 lines Add some missing locking in chan_sip. This patch adds
	  some missing locking to the function
	  send_provisional_keepalive_full(). This function is called from
	  the scheduler, which is processed in the SIP monitor thread. The
	  associated channel (or pbx) thread will also be using the same
	  sip_pvt and ast_channel so locking must be used. The
	  sip_pvt_lock_full() function is used to ensure proper locking
	  order in a safe manner. In passing, document a suspected
	  reference counting error in this function. The "fix" is left
	  commented out because when the "fix" is present, crashes occur.
	  My theory is that fixing it is exposing a reference counting
	  error elsewhere, but I don't know where. (Or my analysis of this
	  being a problem could have been completely wrong in the first
	  place). Leave the comment in the code for so that someone may
	  investigate it again in the future. Also add a bit of doxygen to
	  transmit_provisional_response(). (closes issue ASTERISK-18979)
	  Review: https://reviewboard.asterisk.org/r/1648 ........
	  ................

2012-01-16 21:50 +0000 [r351082-351143]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
	  response to INVITE When handling a non-2xx final response on an
	  INVITE transaction, we have to keep the transaction around after
	  we send an ACK in case we receive a retransmission of the
	  response so we can re-transmit the ACK, but also tear down the
	  ast_channel as soon as we transmit the ACK. Before this patch, we
	  could fail at both of these things. Calling
	  sip_alreadygone/needdestroy prevented us from keeping the
	  transaction up and retransmitting the ACK, and queueing
	  CONGESTION was not sufficient to cause the channel to be torn
	  down when originating calls via the CLI, for example. This patch
	  queues a hangup with CONGESTION instead of just queueing
	  CONGESTION for these responses and removes the sip_alreadygone
	  and sip_needdestroy calls from handle_response_invite on non-2xx
	  responses. It relies on the hangup calling sip_scheddestroy. For
	  more information, see section 17.1.1.1 of RFC 3261. (closes issue
	  ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
	  ........ Merged revisions 351130 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351131 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c: Don't prematurely stop SIP session timer
	  When Asterisk is the UAS (incoming call, endpoint is re-inviting)
	  the SIP session timer expires after half the time the sip
	  endpoint indicates in the Session-expires header in
	  proc_session_timer(). The session timer was being stopped totally
	  and being handled as an error case instead of running again until
	  the second expiry. This patch treats the half-time expiry as a
	  non-error case and continues the timer until the true expiry.
	  (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
	  by: Thomas Arimont Patches: session_timer_fix.diff by Terry
	  Wilson (License #5357) based on session_timer.patch by Thomas
	  Arimont (License #5525) ........ Merged revisions 351080 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351081 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-16 19:49 +0000 [r351079]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/ast_expr2.y, CHANGES, main/ast_expr2.c: Add ABS() absolute
	  value function to the expression parser.

2012-01-16 19:13 +0000 [r351029]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Create and initialize udptl only when
	  dialog negotiates for image media Prior to this patch, the udptl
	  struct was allocated and initialized when a dialog was associated
	  with a peer that supported T.38, when a new SIP channel was
	  allocated, or what an INVITE request was received. This resulted
	  in any dialog associated with a peer that supported T.38 having
	  udptl support assigned to it, including the UDP ports needed for
	  communication. This occurred even in non-INVITE dialogs that
	  would never send image media. This patch creates and initializes
	  the udptl structure only when the SDP for a dialog specifies that
	  image media is supported, or when Asterisk indicates through the
	  appropriate control frame that a dialog is to support T.38.
	  (closes issue ASTERISK-16698) Reported by: under Tested by:
	  Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
	  (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
	  Broad Tested by: Stefan Schmidt review:
	  https://reviewboard.asterisk.org/r/1668/ ........ Merged
	  revisions 351027 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 351028 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-16 17:12 +0000 [r350979]  Sean Bright <sean@malleable.com>

	* /, main/db.c: Sort the output of 'database showkey' as well. You
	  can pass wildcards (%) to the database CLI commands, so this will
	  sort the returned list of matches. ........ Merged revisions
	  350978 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-16 17:07 +0000 [r350977]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, /: Add missing code to set direct RTP setup
	  information during dialing. ........ Merged revisions 350975 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350976 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-16 14:31 +0000 [r350939]  Sean Bright <sean@malleable.com>

	* /, main/db.c: Sort the output of 'database show' by key. This
	  more closely mimics the behavior of 'database show' before the
	  conversion to sqlite3. ........ Merged revisions 350938 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-15 20:16 +0000 [r350887-350890]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/asterisk.c: Allow only one thread at a time to do
	  asterisk cleanup/shutdown. Add locking around the
	  really-really-quit part of the core stop/restart part. Previously
	  more than one thread could be called to do cleanup, causing
	  atexit handlers to be run multiple times, in turn causing
	  segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson
	  Review: https://reviewboard.asterisk.org/r/1662/ Review:
	  https://reviewboard.asterisk.org/r/1658/ ........ Merged
	  revisions 350888 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350889 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile
	  error in utils/extconf.c. Note that I'm not confirming legitimacy
	  of having that file in tree at all. Is anyone using
	  aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged
	  revisions 350885 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350886 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-14 16:43 +0000 [r350791-350839]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, autoconf/ast_gcc_attribute.m4, configure.ac,
	  autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the
	  configure script are properly quoted. Recent versions of autoconf
	  (2.68 on my system) won't properly process the configure script
	  unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
	  the script were, but many were not. This patch corrects the
	  unquoted calls. ........ Merged revisions 350837 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350838 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_h323.c, addons/chan_mobile.c,
	  res/res_pktccops.c, contrib/scripts/install_prereq: Multiple
	  revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14
	  09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two
	  prerequisites are properly installed on Debian-style
	  distributions. * Don't specify a specific version of libgmime;
	  newer versions are available now and acceptable. * Install
	  libsrtp so that res_srtp can be built. ........ r350789 |
	  kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3
	  lines Correct some 'set-but-not-used' variable warnings. ........
	  Merged revisions 350788-350789 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350790 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 22:17 +0000 [r350738]  Kinsey Moore <kmoore@digium.com>

	* /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for
	  the ASTERISK-18929 fix configure and autoconfig.h.in were not
	  regenerated when the fix was committed. ........ Merged revisions
	  350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 350737 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 21:52 +0000 [r350735]  Richard Mudgett <rmudgett@digium.com>

	* /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
	  Correct eventtype names in cel_odbc and cel_pgsql sample files
	  ........ Merged revisions 350733 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350734 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 21:42 +0000 [r350732]  Kinsey Moore <kmoore@digium.com>

	* /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure
	  asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it
	  returns a 'struct sockpeercred', not 'struct ucred', which causes
	  compilation of main/asterisk.c to fail in read_credentials().
	  This allows configure to check for sockpeercred and asterisk to
	  deal with it properly. (closes issue ASTERISK-18929) Reported-by:
	  Barry Miller Patch-by: Barry Miller ........ Merged revisions
	  350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 350731 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 20:32 +0000 [r350681]  Mark Michelson <mmichelson@digium.com>

	* /, channels/sip/config_parser.c: Set port to a default sane value
	  if a bogus one is provided when parsing hostnames. ........
	  Merged revisions 350679 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350680 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 18:52 +0000 [r350605-350644]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Remove some dead code in ast_bridge_call(). None
	  of the parameters to ast_bridge_call() can be NULL for the bridge
	  to work so no need to check for it.

	* configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
	  configs/cel.conf.sample, /, cel/cel_manager.c,
	  configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
	  main/cel.c, configs/cel_custom.conf.sample: Add missing CEL
	  logging fields to various CEL backends. Multiple revisions
	  350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51
	  -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging
	  fields to various CEL backends. * Add missing eventextra to
	  cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
	  EventExtra to cel_manager.c. * Add missing userdeftype support
	  for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
	  (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
	  ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13
	  Jan 2012) | 8 lines Use compatible names for event extra data for
	  various CEL backends. * Change eventextra to extra in cel_psql.c
	  and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c.
	  (issue ASTERISK-17190) ........ Merged revisions 350555,350571
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 350585 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-13 17:00 +0000 [r350551-350554]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_queue.c: Realtime queues failed to load queue
	  information without queue member table Previously, realtime
	  queues could be loaded without defining the queue member table.
	  This allowed for queue members to be dynamic, while the realtime
	  queue definitions could exist in some backing storage. Revision
	  342223 broke this when it changed the return value for
	  realtime_multientry to return NULL when no results are returned.
	  Previously, an empty ast_config object was expected. (closes
	  issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
	  Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
	  Jordan (license 6283) ........ Merged revisions 350552 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350553 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, bridges/bridge_builtin_features.c, channels/chan_bridge.c,
	  include/asterisk/bridging.h, apps/app_confbridge.c,
	  main/bridging.c: Fix crash from bridge channel hangup race
	  condition in ConfBridge This patch addresses two issues in
	  ConfBridge and the channel bridge layer: 1. It fixes a race
	  condition wherein the bridge channel could be hung up 2. It
	  removes the deadlock avoidance from the bridging layer and makes
	  the bridge_pvt an ao2 ref counted object Patch by David Vossel
	  (mjordan was merely the commit monkey) (issue ASTERISK-18988)
	  (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested
	  by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by
	  David Vossel (license 5628) (closes issue ASTERISK-19100)
	  Reported by: Matt Jordan Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1654/ ........ Merged
	  revisions 350550 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-12 16:10 +0000 [r350503]  Jonathan Rose <jrose@digium.com>

	* /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START
	  and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
	  Colledge Patches: features_18.patch uploaded by Nic Colledge
	  (license 6245) ........ Merged revisions 350501 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350502 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-11 22:53 +0000 [r350416-350454]  Richard Mudgett <rmudgett@digium.com>

	* /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
	  CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
	  Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
	  #5909) patch uploaded by Corey Farrell ........ Merged revisions
	  350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 350453 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe
	  optionally update connected line information when the accepting
	  endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
	  with AST_CONTROL_CONNECTED_LINE information so when the parties
	  are initially bridged, the connected line information will be
	  correct. * Added the 'I' option just like the app_dial and
	  app_queue 'I' option. * Made 'N' option ignored if the call is
	  already answered. (closes issue ASTERISK-18969) Reported by:
	  rmudgett Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1656/ ........ Merged
	  revisions 350364 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350415 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-11 19:19 +0000 [r350365]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Always treat arguments to get_by_name_cb as
	  strings Initially, support was left in for the old style of
	  searching, even though it wasn't actually used. In the case of
	  name_len != 0, the OBJ_KEY flag isn't passed because we aren't
	  matching on a full key and therefor can't use the hash function
	  to optimize. The code left in to support the old way of searching
	  unfortunately treated a prefix search like this as though an
	  ast_channel struct was passed as an arg and caused a crash. This
	  patch also adds needed parentheses around some matching
	  conditions. (closes issue ASTERISK-19182)

2012-01-10 22:10 +0000 [r350273-350313]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
	  function. The time passed by the LOCK function to an internal
	  function was relative time when the function expected absolute
	  time. * Don't use C++ keywords in get_lock(). (closes issue
	  ASTERISK-16868) Reported by: Andrey Solovyev Patches:
	  20101102__issue18207.diff.txt (license #5003) patch uploaded by
	  Andrey Solovyev (modified) ........ Merged revisions 350311 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350312 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/channel.c: Fix compiler warnings reported by gcc v4.2.4.

2012-01-09 22:15 +0000 [r350223]  Terry Wilson <twilson@digium.com>

	* main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
	  channels/chan_local.c, main/rtp_engine.c, main/say.c,
	  apps/app_record.c, apps/app_test.c, channels/console_video.c,
	  apps/app_alarmreceiver.c, apps/app_chanisavail.c,
	  bridges/bridge_multiplexed.c, channels/chan_iax2.c,
	  main/indications.c, main/cli.c, channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
	  apps/app_dumpchan.c, pbx/pbx_realtime.c, apps/app_amd.c,
	  channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
	  apps/app_milliwatt.c, channels/sig_ss7.c, apps/app_dial.c,
	  main/pbx.c, apps/app_page.c, apps/app_softhangup.c,
	  apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c,
	  apps/app_disa.c, include/asterisk/channel.h, main/aoc.c,
	  apps/app_talkdetect.c, main/cel.c, res/res_mutestream.c,
	  res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c,
	  funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c,
	  apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c,
	  channels/chan_unistim.c, channels/chan_vpb.cc, main/ccss.c,
	  apps/app_meetme.c, apps/app_readexten.c, res/res_musiconhold.c,
	  main/autochan.c, channels/chan_gtalk.c, apps/app_followme.c,
	  res/res_jabber.c, main/cdr.c, main/channel.c, main/dial.c,
	  channels/chan_phone.c, main/manager.c, funcs/func_groupcount.c,
	  funcs/func_audiohookinherit.c, funcs/func_frame_trace.c,
	  res/res_agi.c, apps/app_minivm.c, main/app.c,
	  apps/app_confbridge.c, apps/app_rpt.c, addons/chan_mobile.c,
	  apps/app_parkandannounce.c, channels/chan_mgcp.c,
	  apps/app_jack.c, apps/app_adsiprog.c, channels/chan_sip.c,
	  res/res_fax.c, apps/app_waitforsilence.c, funcs/func_lock.c,
	  main/channel_internal_api.c (added), res/res_adsi.c,
	  pbx/pbx_lua.c, channels/chan_console.c, apps/app_getcpeid.c,
	  channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
	  funcs/func_global.c, channels/chan_usbradio.c,
	  channels/chan_jingle.c, apps/app_flash.c,
	  apps/app_directed_pickup.c, main/abstract_jb.c, main/file.c,
	  channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c,
	  apps/app_sms.c, channels/chan_nbs.c, apps/app_stack.c,
	  main/dsp.c: Replace direct access to channel name with accessor
	  functions There are many benefits to making the ast_channel an
	  opaque handle, from increasing maintainability to presenting ways
	  to kill masquerades. This patch kicks things off by taking things
	  a field at a time, renaming the field to
	  '__do_not_use_${fieldname}' and then writing setters/getters and
	  converting the existing code to using them. When all fields are
	  done, we can move ast_channel to a C file from channel.h and lop
	  off the '__do_not_use_'. This patch sets up
	  main/channel_interal_api.c to be the only file that actually
	  accesses the ast_channel's fields directly. The intent would be
	  for any API functions in channel.c to use the accessor functions.
	  No more monkeying around with channel internals. We should use
	  our own APIs. The interesting changes in this patch are the
	  addition of channel_internal_api.c, the moving of the AST_DATA
	  stuff from channel.c to channel_internal_api.c (note: the
	  AST_DATA stuff will have to be reworked to use accessor functions
	  when ast_channel is really opaque), and some re-working of the
	  way channel iterators/callbacks are handled so as to avoid
	  creating fake ast_channels on the stack to pass in matching data
	  by directly accessing fields (since "name" is a stringfield and
	  the fake channel doesn't init the stringfields, you can't use the
	  ast_channel_name_set() function). I went with
	  ast_channel_name(chan) for a getter, and
	  ast_channel_name_set(chan, name) for a setter. The majority of
	  the grunt-work for this change was done by writing a semantic
	  patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review:
	  https://reviewboard.asterisk.org/r/1655/

2012-01-09 21:56 +0000 [r350222]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_iax2.c: Fix joinable thread terminating without
	  joiner memory leak in chan_iax.c. The iax2_process_thread() can
	  exit without anyone waiting to join the thread. If noone is
	  waiting to join the thread then a large memory leak occurs. *
	  Made iax2_process_thread() deatach itself if nobody is waiting to
	  join the thread. (closes issue ASTERISK-17339) Reported by:
	  Tzafrir Cohen Patches:
	  asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
	  (license #5617) patch uploaded by Alex Villacis Lasso (modified)
	  (closes issue ASTERISK-17825) Reported by: wangjin ........
	  Merged revisions 350220 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350221 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-09 19:37 +0000 [r350181]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, main/db.c: Fix shutdown handling of sqlite3 astdb. If a
	  db_sync was scheduled just before shutdown, the atexit code
	  calling db_sync would have no effect, causing the astdb commit
	  thread to stay alive. This caused the SIP/realtime_sipregs test
	  to fail. (The fallback kill would run the atexit code again and
	  that would wreak havoc.) This fixes that the atexit kill
	  condition is picked up properly. (closes issue ASTERISK-18883)
	  Reviewed by: Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/1659 ........ Merged revisions
	  350180 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-09 18:58 +0000 [r350077-350130]  Richard Mudgett <rmudgett@digium.com>

	* /, contrib/scripts/valgrind_compare (added): Multiple revisions
	  350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33
	  -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script
	  live_ast to invoke Asterisk with valgrind and suppression file. *
	  Added valgrind_compare script to compare two valgrind log files
	  for differences. (issue ASTERISK-17339) Reported by: Tzafrir
	  Cohen Patches: valgrind_compare (license #5035) script uploaded
	  by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch
	  uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license
	  #5185) patch uploaded by Paul Belanger ........ r350128 |
	  rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11
	  lines live_ast: valgrind: run asterisk under valgrind Adds a new
	  sub-command, "valgrind" to live_ast. It runs asterisk under
	  valgrind. The extra command-line parameters are passed to
	  Asterisk as usual, and parameters to valgrind are passed through
	  LIVE_AST_VALGRIND_ARGS in live.conf . Review:
	  https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
	  from http://svn.asterisk.org/svn/asterisk/branches/10 ........
	  Merged revisions 350127-350128 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350129 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/asterisk.c: Make Asterisk -x command line parameter imply
	  -r parameter presence. The Asterisk -x command line parameter is
	  documented inconsistently. * Made the -x documentation and
	  behavior consistent. * Since this is also a new year, updated the
	  copyright notices while here. (closes issue ASTERISK-19094)
	  Reported by: Eugene Patches:
	  issueA19094_correct_asterisk_option_x.patch (license #5674) patch
	  uploaded by Walter Doekes (modified) Tested by: Eugene ........
	  Merged revisions 350075 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 350076 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-09 15:40 +0000 [r350025]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_meetme.c: Prevent SLA settings from getting wiped out
	  on reload If SLA was reloaded without the config file being
	  changed, current settings got wiped out before the SLA reload
	  code decided it wasn't going to reload the file since nothing was
	  changed. Moving the settings reset later in the reload process
	  fixes this. (closes issue AST-744) ........ Merged revisions
	  350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 350024 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-06 23:31 +0000 [r349978]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't leak CID in From header when
	  presentation=unavailable When someone does
	  Set(CALLERPRES()=unavailable) (or
	  Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
	  header shows "Anonymous" <anonymous@anonymous.invalid>. When
	  sendrpid=yes/pai, the From header will still display the callerid
	  info, even though we supply an rpid header with the anonymous
	  info. It seems like we shouldn't leak that info in any case.
	  Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
	  seems to indicate that one shouldn't send identifying info in the
	  From in this case. This patch anonymizes the From header as well
	  even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
	  https://reviewboard.asterisk.org/r/1649/ ........ Merged
	  revisions 349968 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349977 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-06 21:26 +0000 [r349929]  Kinsey Moore <kmoore@digium.com>

	* /, pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected
	  behavior with confbridge A bug in the pbx_lua goto detection was
	  causing the dialplan to hangup unexpectedly after confbridge
	  exited if it had called lua dialplan code during execution.
	  Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue
	  ASTERISK-18976) ........ Merged revisions 349928 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-06 16:50 +0000 [r349874]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_followme.c: Fix memory leaks in app_followme
	  find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
	  Jordan ........ Merged revisions 349872 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349873 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-05 23:58 +0000 [r349823]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_fax.c: Fix premature free'ing of the frame committed
	  in r349608 Even though we set the frame to the ast_null_frame and
	  return that, the caller of the frame hook may still need the
	  frame. This now is a bit more careful about when it frees the
	  frame, i.e., only under the same conditions that applied when we
	  duplicated it in the first place. ........ Merged revisions
	  349822 from http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-05 23:47 +0000 [r349782-349821]  Richard Mudgett <rmudgett@digium.com>

	* /, cel/cel_sqlite3_custom.c: Make not assume that the
	  cel_sqlite3_custom SQL table primary key is AcctId. If a table is
	  created by some other application and the primary key is not
	  named "AcctId", cel/cel_sqlite3_custom.c will always try to
	  create the table and fail because it already exists. * Change the
	  SQL table query to not require AcctId as the primary key. (closes
	  issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
	  (license #6337) patch uploaded by socketpair ........ Merged
	  revisions 349819 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349820 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* UPGRADE.txt, pbx/pbx_config.c: Make pbx_config.c use Gosub
	  instead of Macro call for stdexten. Users created by users.conf
	  with hasvoicemail=yes have been documented as using a Gosub to
	  stdexten since v1.6.0. However, the code still generates dialplan
	  to access stdexten as a Macro as documented in v1.4; which does
	  not work with the newer extensions.conf.sample file. * Make
	  generated dialplan access the stdexten dialplan with the
	  documented Gosub instead of the older Macro style. (closes issue
	  ASTERISK-18809) Reported by: Jay Allen Patches:
	  gosub_patch-pbx_config.patch (license #6323) patch uploaded by
	  Jay Allen (modified) Tested by: rmudgett

2012-01-05 22:11 +0000 [r349733]  Kinsey Moore <kmoore@digium.com>

	* /, main/file.c: Allow playback of formats that don't support
	  seeking ast_streamfile previously did unconditional seeking on
	  files that broke playback of formats that don't support that
	  functionality. This patch avoids the seek that was causing the
	  problem. This regression was introduced in r158062. (closes issue
	  ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions
	  349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 349732 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-05 22:02 +0000 [r349674-349730]  Jonathan Rose <jrose@digium.com>

	* /, main/dsp.c: Fix an issue where dsp.c would interpret multiple
	  dtmf events from a single key press. When receiving calls from a
	  mobile phone into a DISA system on a connection with significant
	  interference, the reporter's Asterisk system would interpret DTMF
	  incorrectly and replicate digits received. This patch resolves
	  that by increasing the number of frames a mismatch has to be
	  detected before assuming the DTMF is over by 1 frame and adjusts
	  dtmf_detect function to reset hits and misses only when an edge
	  is detected. (closes issue ASTERISK-17493) Reported by: Alec
	  Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
	  (license 5546) Review: https://reviewboard.asterisk.org/r/1130/
	  ........ Merged revisions 349728 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349729 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/asterisk.c: Ensures Asterisk closes when receiving
	  terminal signals in 'no fork' mode. When catching a signal, in no
	  fork mode the console thread is identical to the thread
	  responsible for catching the signal and closing Asterisk, which
	  requires it to first dispense with the console thread. Prior to
	  this patch, if these threads were identical, upon receiving a
	  killing signal, the thread will send an URG signal to itself,
	  which we also catch and then promptly do nothing with. Obviously
	  this isn't useful behavior. (closes issue ASTERISK-19127)
	  Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded
	  by Bryon Clark (license 6157) ........ Merged revisions 349672
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 349673 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-04 22:23 +0000 [r349609-349634]  Matthew Jordan <mjordan@digium.com>

	* /, apps/confbridge/conf_config_parser.c: Fix for ConfBridge
	  config parser unlocking channel mutex too many times When looking
	  up a ConfBridge profile, the config parser would, if it found a
	  channel datastore on the channel requesting the bridge profile,
	  unlock the channel mutex twice. Since that's a little aggressive,
	  it now only unlocks it once. (closes issue ASTERISK-19042)
	  Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042
	  uploaded by David Vossel (license 5628) ........ Merged revisions
	  349619 from http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_fax.c: Free successfully translated frame in
	  fax_gateway_framehook A frame that is translated via
	  ast_translate is also duplicated via ast_frdup. This will
	  allocate a new frame on the heap, which needs to be free'd at the
	  appropriate time. This issue reporter used valgrind to find that
	  this occurred in res_fax's fax_gateway_framehook; a quick search
	  through the code showed that only place this was currently not
	  handling the translatted frame properly. (closes issue
	  ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged
	  revisions 349608 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-04 20:55 +0000 [r349560]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for
	  CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
	  pointer checks in the following chan_dahdi channel callbacks:
	  dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
	  dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
	  Diego Aguirre Tested by: rmudgett ........ Merged revisions
	  349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 349559 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-04 20:24 +0000 [r349506-349535]  Kinsey Moore <kmoore@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: Make debian init script
	  conform to the LSB standard Previously, this init script would
	  return 1 if Asterisk was already running. This is incorrect
	  behavior according to the LSB standard and has been fixed by
	  returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
	  johnc ........ Merged revisions 349529 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349532 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, contrib/scripts/autosupport.8, contrib/scripts/autosupport:
	  Update autosupport script and man page Added information
	  collection from the output of the utilities: top, free, uptime,
	  ifconfig Added information collection from the output of the
	  Asterisk command 'dahdi show status' Added option / flag '-n,
	  --non-interactive' Updated man page to reflect new option / flag
	  '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
	  issue AST-749) ........ Merged revisions 349504 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349505 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2012-01-04 19:53 +0000 [r349452-349503]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Adds Subscription-State header to notify
	  with call completion. per RFC3265 (Closes issue ASTERISK-17953)
	  Reported by: George Konopacki Patches: 19400.patch uploaded by
	  mmichelson (license 5049) ........ Merged revisions 349482 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349502 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/pbx.c, /: Fix documentation for SayNumber to reflect the
	  fact that language is changed in CHANNEL() (closes issue
	  ASTERISK-18962) reported by: Nir Simionovich ........ Merged
	  revisions 349450 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349451 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-31 15:48 +0000 [r349409-349410]  Russell Bryant <russell@russellbryant.com>

	* channels/chan_sip.c: Fix some minor formatting issues based on
	  coding guidelines.

	* channels/sip/include/dialog.h, channels/chan_sip.c,
	  include/asterisk/astobj2.h, main/astobj2.c: Constify tag argument
	  in REF_DEBUG related code.

2011-12-29 15:16 +0000 [r349341]  Matthew Jordan <mjordan@digium.com>

	* main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames
	  in local bridge loop Failing to handle
	  AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
	  causes the loop to exit prematurely. This causes a variety of
	  negative side effects, depending on when the loop exits. This
	  patch handles the frame by essentially swallowing the frame in
	  the local loop, as the current channel drivers expect the RTP
	  bridge to handle the frame, and, in the case of the local bridge
	  loop, no additional action is necessary. (issue ASTERISK-19040)
	  (issue ASTERISK-19128) (issue ASTERISK-17725) (issue
	  ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan
	  Schmidt Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1640/ ........ Merged
	  revisions 349339 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349340 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-28 21:39 +0000 [r349291]  Sean Bright <sean@malleable.com>

	* /, main/audiohook.c: Use ast_audiohook_write_list_empty to
	  determine if our lists are empty instead of duplicating that
	  logic. ........ Merged revisions 349289 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349290 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-28 19:00 +0000 [r349249-349251]  Kevin P. Fleming <kpfleming@digium.com>

	* utils, /: Tell Subversion to gnore the 'astdb2bdb' binary file if
	  it exists. ........ Merged revisions 349250 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_fax.c, include/asterisk/dsp.h,
	  include/asterisk/res_fax.h, res/res_fax_spandsp.c, main/dsp.c:
	  Improve T.38 gateway V.21 preamble detection. This commit removes
	  the V.21 preamble detection code previously added to the generic
	  DSP implementation in Asterisk, and instead enhances the res_fax
	  module to be able to utilize V.21 preamble detection
	  functionality made available by FAX technology modules. This
	  commit also adds such support to res_fax_spandsp, which uses the
	  Spandsp modem tone detection code to do the V.21 preamble
	  detection. There should be no functional change here, other than
	  much more reliable V.21 preamble detection (and thus T.38 gateway
	  initiation). ........ Merged revisions 349248 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-27 20:55 +0000 [r349196]  Matthew Jordan <mjordan@digium.com>

	* /, res/res_timing_pthread.c, include/asterisk/module.h,
	  res/res_timing_dahdi.c, res/res_timing_timerfd.c,
	  res/res_musiconhold.c: Fix timing source dependency issues with
	  MOH Prior to this patch, res_musiconhold existed at the same
	  module priority level as the timing sources that it depends on.
	  This would cause a problem when music on hold was reloaded, as
	  the timing source could be changed after res_musiconhold was
	  processed. This patch adds a new module priority level,
	  AST_MODPRI_TIMING, that the various timing modules are now loaded
	  at. This now occurs before loading other resource modules, such
	  that the timing source is guaranteed to be set prior to resolving
	  the timing source dependencies. (closes issue ASTERISK-17474)
	  Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
	  Wes Van Tlghem, elguero, Thomas Arimont Patches:
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
	  uploaded by elguero (License #5026)
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
	  uploaded by elguero (License #5026)
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
	  elguero (License #5026) Review:
	  https://reviewboard.asterisk.org/r/1578/ ........ Merged
	  revisions 349194 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349195 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-27 17:17 +0000 [r349146]  Sean Bright <sean@malleable.com>

	* /, main/audiohook.c: Once an audiohook is attached to a channel,
	  we continue to transcode all of the frames, even after all of the
	  hooks are detached. This patch short-cicuits us out before we
	  transcode unnecessarily. ........ Merged revisions 349144 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 349145 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-23 21:19 +0000 [r349106]  Matthew Jordan <mjordan@digium.com>

	* contrib/realtime/mysql/voicemail.sql,
	  configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Allow overriding of IMAP server settings on a user by user basis
	  This patch allows the imapserver, imapport, and imapflags
	  settings to be overridden for any voicemail user. It also
	  documents the settings in the sample voicemail.conf file, and
	  updates the voicemail schema to allow storage of those columns.
	  (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/

2011-12-23 20:42 +0000 [r349097-349098]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c, main/features.c, configs/sip.conf.sample,
	  channels/sip/include/sip.h: INFO/Record request configurable to
	  use dynamic features Adds two new options to SIP peers allowing
	  them to specify features (dynamic or builtin) to use when sending
	  INFO/record requests. Recordonfeature activates whatever feature
	  is specified when recieving a record: on request while
	  recordofffeature activates whatever feature is specified when
	  receiving a record: off request. Both of these features can be
	  disabled by setting the feature to an empty string. (closes issue
	  ASTERISK-16507) Reported by: Jon Bright Review:
	  https://reviewboard.asterisk.org/r/1634/

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: chan_sip autocreatepeer=persist
	  option for auto-created peers to survive reload This patch moves
	  destruction of sip peers to immediately after the general section
	  of sip.conf is read so that autocreatepeer setting can be read
	  before deletion of peers. If autocreatepeer=persist at reload,
	  then peers created by the autocreatepeer setting will be skipped
	  when purging the current SIP peer list. (closes ASTERISK-16508)
	  Reported by: Kirill Katsnelson Patches:
	  017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill
	  Katsnelson (license 5845)

2011-12-23 17:36 +0000 [r349046]  Sean Bright <sean@malleable.com>

	* /, apps/app_chanspy.c: Merged revisions 349045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r349045 | seanbright | 2011-12-23 12:32:33 -0500
	  (Fri, 23 Dec 2011) | 25 lines Merged revisions 349044 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec
	  2011) | 18 lines In ChanSpy, don't create audiohooks that will
	  never be used. When ChanSpy is initialized it creates and
	  attaches 3 audiohooks: 1) Read audio off of the channel that we
	  are spying on 2) Write audio to the channel that we are spying on
	  3) Write audio to the channel that is bridged to the channel that
	  we are spying on. The first is always necessary, but the others
	  are used only when specific options are passed to the ChanSpy
	  application (B, d, w, and W to be specific). When those flags are
	  not passed, neither of those audiohooks are ever sent frames, but
	  we still try to process the hooks for each voice frame that we
	  recieve on the channel. So in short - only create and attach
	  audiohooks that we actually need. ........ ................

2011-12-23 15:26 +0000 [r348994]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c, /: Fix missing doc tags found while fixing
	  ASTERISK-18689 Add missing <variable></variable> tags in app_dial
	  documentation. ........ Merged revisions 348992 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348993 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-23 02:35 +0000 [r348953]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix
	  extension state callback references in chan_sip. Chan_sip gives a
	  dialog reference to the extension state callback and assumes that
	  when ast_extension_state_del() returns, the callback cannot
	  happen anymore. Chan_sip then reduces the dialog reference count
	  associated with the callback. Recent changes (ASTERISK-17760)
	  have resulted in the potential for the callback to happen after
	  ast_extension_state_del() has returned. For chan_sip, this could
	  be very bad because the dialog pointer could have already been
	  destroyed. * Added ast_extension_state_add_destroy() so chan_sip
	  can account for the sip_pvt reference given to the extension
	  state callback when the extension state callback is deleted. *
	  Fix pbx.c awkward statecbs handling in
	  ast_extension_state_add_destroy() and handle_statechange() now
	  that the struct ast_state_cb has a destructor to call. * Ensure
	  that ast_extension_state_add_destroy() will never return -1 or 0
	  for a successful registration. * Fixed pbx.c statecbs_cmp() to
	  compare the correct information. The passed in value to compare
	  is a change_cb function pointer not an object pointer. * Make
	  pbx.c ast_merge_contexts_and_delete() not perform callbacks with
	  AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
	  deadlocking when those locks are held during the callback. *
	  Removed unused lock declaration for the pbx.c store_hints list.
	  (closes issue ASTERISK-18844) Reported by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1635/ ........ Merged
	  revisions 348940 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348952 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-22 22:39 +0000 [r348890]  Matthew Jordan <mjordan@digium.com>

	* cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql
	  There were a number of issues in cel_pgsql's pgsql_log method: *
	  If either sql or sql2 could not be allocated, the method would
	  return while the pgsql_lock was still locked * If the execution
	  of the log statement succeeded, the sql and sql2 structs were
	  never free'd * Reconnection successes were logged as ERRORs. In
	  general, the severity of several logging statements was reduced
	  (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
	  ........ Merged revisions 348888 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348889 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-22 21:12 +0000 [r348849]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Fix segfault on answer. Only
	  update/change RTP source if RTP has already been started and
	  connected to the subchannel.

2011-12-22 20:44 +0000 [r348848]  Matthew Jordan <mjordan@digium.com>

	* /, main/say.c, main/file.c, main/app.c, apps/app_confbridge.c,
	  main/bridging.c: Add Asterisk TestSuite event hooks to support
	  ConfBridge testing This patch adds initial testsuite event hooks
	  so that ConfBridge tests can be executed in the Asterisk
	  TestSuite. (issue ASTERISK-19059) ........ Merged revisions
	  348846 from http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-22 20:39 +0000 [r348847]  Terry Wilson <twilson@digium.com>

	* /, include/asterisk/format_pref.h: Allow packetization vaules >
	  127 According to the RTP packetization documentation, and the
	  maximum values listed in AST_FORMAT_LIST, we should support
	  values > that the signed char array that ast_codec_pref makes
	  available to store the value. All places in the code treat the
	  framing field as though it were an int array instaead of a char
	  array anyway, so this just fixes the type of the array. (closes
	  issue ASTERISK-18876) Review:
	  https://reviewboard.asterisk.org/r/1639/ ........ Merged
	  revisions 348833 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348845 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-21 20:13 +0000 [r348737-348794]  Richard Mudgett <rmudgett@digium.com>

	* /, codecs/speex: Make codecs/speex ignore *.i files also.
	  ........ Merged revisions 348793 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/confbridge: Make apps/confbridge ignore *.i files also.
	  ........ Merged revisions 348790 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS
	  number if it is blank. Some ISDN switches complain or block the
	  call if the RDNIS number is empty. * Made chan_iax2 not save a
	  RDNIS number into the ast_channel if the string is blank. This is
	  what other channel drivers do. (closes issue ASTERISK-17152)
	  Reported by: rmudgett ........ Merged revisions 348735 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348736 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-20 20:06 +0000 [r348698]  Matthew Nicholson <mnicholson@digium.com>

	* contrib/scripts/safe_asterisk: This adds support for setting
	  several safe_asterisk parameters using environment variables and
	  also enables a custom run directory for asterisk (instead of
	  defaulting to /tmp). Patch by: Byron Clark (byronclark) (closes
	  ASTERISK-17810)

2011-12-19 21:43 +0000 [r348649]  Richard Mudgett <rmudgett@digium.com>

	* /, configure, configure.ac: Fix crashes on other platforms caused
	  by interference from Darwin weak symbol support. Support weak
	  symbols on a platform specific basis. The Mac OS X (Darwin)
	  support must be isolated from the other platforms because it has
	  caused other platforms to crash. Several other platforms
	  including Linux have GCC versions that define the weak attribute.
	  However, this attribute is only setup for use in the code by
	  Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
	  Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged
	  revisions 348647 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348648 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-19 19:55 +0000 [r348606]  Leif Madsen <leif@leifmadsen.com>

	* /, main/message.c: Update documentation for MESSAGE_SEND_STATUS
	  variable. (Closes issue ASTERISK-19056) Reported by: Yuri
	  Patches: 348360.diff uploaded by Yuri (license #5242) ........
	  Merged revisions 348605 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-19 01:36 +0000 [r348567]  Terry Wilson <twilson@digium.com>

	* /, res/res_srtp.c: Add a separate buffer for SRTCP packets The
	  function ast_srtp_protect used a common buffer for both SRTP and
	  SRTCP packets. Since this function can be called from multiple
	  threads for the same SRTP session (scheduler for SRTCP and
	  channel for SRTP) it was possible for the packets to become
	  corrupted as the buffer was used by both threads simultaneously.
	  This patch adds a separate buffer for SRTCP packets to avoid the
	  problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
	  Collins) ........ Merged revisions 347995 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347996 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-18 18:29 +0000 [r348518]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample
	  related to AST-2011-013. * The sample file listed *two* values
	  for the 'nat' option as being the default. Only 'force_rport' is
	  the default. * The warning about having differing 'nat' settings
	  confusingly referred to both peers and users. ........ Merged
	  revisions 348515 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
	  Merged revisions 348516 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348517 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-16 23:58 +0000 [r348466]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /, main/features.c: Clean-up on isle five for
	  __ast_request_and_dial() and ast_call_forward(). * Add locking
	  when a channel inherits variables and datastores in
	  __ast_request_and_dial() and ast_call_forward(). Note: The
	  involved channels are not active so there was minimal potential
	  for problems. * Remove calls to ast_set_callerid() in
	  __ast_request_and_dial() and ast_call_forward() because the set
	  information is for the wrong direction. * Don't use C++ keywords
	  for variable names in ast_call_forward(). * Run the redirecting
	  interception macro if defined when forwarding a call in
	  ast_call_forward(). Note: Currently will never execute because
	  the only callers that supply a calling channel supply a hungup or
	  zombie channel. * Make feature_request_and_dial() put the
	  transferee into autoservice when it calls ast_call_forward() in
	  case a redirection interception macro is run. Note: Currently
	  will never happen because the caller channel (Party B) is always
	  hungup at this time. * Make feature_request_and_dial() ignore the
	  AST_CONTROL_PROCEEDING frame to silence a log message. ........
	  Merged revisions 348464 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348465 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-16 22:00 +0000 [r348416]  Jonathan Rose <jrose@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Voicemail with the saycid option will now play a caller's name
	  based on cid if available. In order to check the availability of
	  the caller's name, app_voicemail will check for an audio file in
	  <astspooldir>/recordings/callerids/ This change sets a precedent
	  for where to put recordings of names. Currently the idea is that
	  recordings here could also be used for applications like
	  confbridge and meetme to find recorded names in this folder from
	  callerid (when another recording isn't available) (closes issue
	  ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by
	  Russel Brown (license 6182)

2011-12-16 21:30 +0000 [r348312-348408]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /: Fix cut and past error in ast_call_forward().
	  (issue ASTERISK-18836) ........ Merged revisions 348401 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348405 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/channel.c, main/pbx.c, /, apps/app_authenticate.c,
	  funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
	  apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix
	  crash during CDR update. The ast_cdr_setcid() and
	  ast_cdr_update() were shown in ASTERISK-18836 to be called by
	  different threads for the same channel. The channel driver thread
	  and the PBX thread running dialplan. * Add lock protection around
	  CDR API calls that access an ast_channel pointer. (closes issue
	  ASTERISK-18836) Reported by: gpluser Review:
	  https://reviewboard.asterisk.org/r/1628/ ........ Merged
	  revisions 348362 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348363 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
	  CallerID to the announcing channel. ParkAndAnnounce tried to pass
	  the CallerID to the announcing channel but the ID was wiped out
	  by the channel masquerade done when parking the call. * Save the
	  CallerID before parking the channel to pass it to the announcing
	  channel. * Fixed a minor memory leak in ParkAndAnnounce. *
	  Updated some ParkAndAnnounce log messages. ........ Merged
	  revisions 348310 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348311 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-14 22:36 +0000 [r348215-348266]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_originate.c: Added support for all slin formats to
	  app_originate Previously, app_originate could not originate a
	  call into a non-8kHz conference bridge as the formats for
	  non-8kHz slin codecs were not applied to the created channel.
	  This patch adds all of the formats by default, such that if a
	  created channel has a codec that supports a higher sampling rate,
	  a translation path can be built between it and other channels.
	  ........ Merged revisions 348265 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Fixed Asterisk crash when function
	  QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has
	  two required parameters (queuename, option). It was only checking
	  for the presence of queuename. The patch checks for the existence
	  of the option parameter and provides better error logging when
	  invalid values are provided for the option parameter as well.
	  ........ Merged revisions 348211 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-14 22:05 +0000 [r348214]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or
	  receiving. The user may set that variable. ASTERISK-18921
	  ........ Merged revisions 348212 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348213 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-14 21:08 +0000 [r348161]  Jonathan Rose <jrose@digium.com>

	* main/features.c, configs/features.conf.sample: Add and document
	  PARKEDCALL variable set during timeout PARKEDCALL variable tracks
	  which parking lot the call was last parked in. This can be used
	  afterwards for flow control when returntoorigin is set to off. I
	  went ahead and documented both this and the existing variable set
	  during timeout (PARKINGSLOT) in the sample features.conf since
	  there was no prior mention of variables being set during timeout.
	  (closes issue ASTERISK-16239) Reported By: Clod Patry Patches:
	  M17503.diff uploaded by Clod Patry (license 5138)

2011-12-14 20:51 +0000 [r348160]  Matthew Jordan <mjordan@digium.com>

	* apps/app_confbridge.c: Improve error message in CONFBRIDGE_INFO
	  Provided a more descriptive error message when a value supplied
	  for the parameter type is not one of the acceptable values.
	  (closes issue ASTERISK-18717) Reported by: Paul Belanger Patches:
	  __20111103-better-confbridge_info-error-msg.txt (License #4999)

2011-12-14 20:37 +0000 [r348156-348159]  Jonathan Rose <jrose@digium.com>

	* /, configs/features.conf.sample: Fix accidental use of tabs
	  instead of spaces from previous features.conf.sample change
	  ........ Merged revisions 348157 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348158 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, configs/features.conf.sample: Document PARKINGSLOT variable in
	  features.conf.sample (issue ASTERISK-16239) ........ Merged
	  revisions 348154 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348155 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-13 23:10 +0000 [r348103]  Richard Mudgett <rmudgett@digium.com>

	* /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix
	  FollowMe CallerID on outgoing calls. The addition of the
	  Connected Line support changed how CallerID is passed to outgoing
	  calls. The FollowMe application was not updated to pass CallerID
	  to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
	  * Restructured findmeexec() to fix several memory leaks and
	  eliminate some duplicated code. * Made check the return value of
	  create_followme_number(). Putting a NULL into the numbers list is
	  bad if create_followme_number() fails. * Fixed a couple uses of
	  ast_strdupa() inside loops. * The changes to
	  bridge_builtin_features.c fix a similar CallerID issue with the
	  bridging API attended and blind transfers. (Not used at this
	  time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
	  Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1612/ ........ Merged
	  revisions 348101 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348102 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-13 15:22 +0000 [r348061]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Fix possible misshandling of an incoming SIP
	  response as a peer poke response. Also make sure peer has even
	  qualify enabled when handle a peer poke response. (closes issue
	  ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
	  UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
	  by: David Vossel ........ Merged revisions 348048 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 348056 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-12 19:35 +0000 [r347997]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/logger.h, utils/refcounter.c, main/logger.c,
	  utils/hashtest.c, UPGRADE.txt, utils/ael_main.c,
	  utils/hashtest2.c, CHANGES, main/asterisk.c, main/config.c,
	  configs/logger.conf.sample, main/loader.c, main/cli.c: Backed out
	  core changes from r346391 During testing, it was discovered that
	  there were a number of side effects introduced by r346391 and
	  subsequent check-ins related to it (r346429, r346617, and
	  r346655). This included the /main/stdtime/ test 'hanging', as
	  well as the remote console option failing to receive the
	  appropriate output after a period of time. I only backed out the
	  changes to main/ and utils/, as this was adequate to reverse the
	  behavior experienced. (issue ASTERISK-18974)

2011-12-12 17:34 +0000 [r347954]  Richard Mudgett <rmudgett@digium.com>

	* configs/iax.conf.sample, configs/chan_dahdi.conf.sample, /,
	  configs/chan_ooh323.conf.sample, configs/vpb.conf.sample,
	  configs/extensions.lua.sample, configs/sip.conf.sample,
	  configs/extensions.conf.sample: Update sample configs to put
	  incoming calls into context public. * Add warning about the SIP
	  allowguest option in context public. (closes issue
	  ASTERISK-14122) Reported by: Alec Davis Review:
	  https://reviewboard.asterisk.org/r/719/ ........ Merged revisions
	  347953 from http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-09 21:47 +0000 [r347866-347903]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c: Adds MixMonitor and StopMixMonitor AMI
	  commands to the manager These commands work much like the
	  dialplan applications that would otherwise invoke them. A nice
	  benefit of these is that they can be invoked on a call remotely
	  and at any time during a call. They work much like the Monitor
	  and StopMonitor ami commands. (closes issue ASTERISK-17726)
	  Reported by: Sergio González Martín Patches:
	  mixmonitor_actions.diff uploaded by Sergio González Martín
	  (license 5644) Review: https://reviewboard.asterisk.org/r/1193/

	* include/asterisk/file.h, apps/app_sayunixtime.c, CHANGES: Remove
	  autojump extensions from SayUnixTime, make an option to perform
	  automatic jumps. When a caller sends DTMF while the SayUnixTime
	  application is saying the time, The call would jump to the next
	  extension much like it does during Background(). This patch adds
	  option 'j' to SayUnixTime which when used employs the old
	  behavior. Also, this patch allows arguments to sayunixtime to not
	  be used as empty strings in the case of something like
	  'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue
	  ASTERISK-16675) Reported by: jlpedrosa Patches:
	  patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license
	  5959) Review: https://reviewboard.asterisk.org/r/956/

2011-12-09 01:33 +0000 [r347813]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /: Fix some parsing issues in
	  add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
	  potential sign extension issue. * Fix infinite loop in
	  add_exten_to_pattern_tree() handling of character set escape
	  handling. * Added buffer overflow checks in
	  add_exten_to_pattern_tree() character set collection. * Made
	  ignore empty character sets. * Added escape character handling to
	  end-of-range character in character sets. This has a slight
	  change in behavior if the end-of-range character is an escape
	  character. You must now escape it. * Fix potential sign extension
	  issue when expanding character set ranges. * Made remove
	  duplicated characters from character sets. The duplicate
	  characters lower extension matching priority and prevent
	  duplicate extension detection. * Fix escape character handling
	  when the escape character is trying to escape the end-of-string.
	  We could have continued processing characters after the end of
	  the exten string. We could have added the previous character to
	  the pattern matching tree incorrectly. (closes issue
	  ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions
	  347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 347812 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-08 21:32 +0000 [r347735]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: Fix regression when using tcpenable=no
	  and tlsenable=yes. The tlsenable settings are tucked away in
	  main/tcptls.c, so I missed them when resolving ASTERISK-18837.
	  This should resolve the test suite breakage of the sip tls tests.
	  Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
	  Jordan ........ Merged revisions 347718 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347727 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-08 20:55 +0000 [r347658]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_queue.c: Fix regressed behavior of queue set penalty
	  to work without specifying 'in <queuename>' r325483 caused a
	  regression in Asterisk 10+ that would make Asterisk segfault when
	  attempting to set penalty on an interface without specifying a
	  queue in the queue set penalty CLI command. In addition, no
	  attempt would be made whatsoever to perform the penalty setting
	  on all the queues in the core list with either the cli command or
	  the non-segfaulting ami equivalent. This patch fixes that and
	  also makes an attempt to document and rename some functions
	  required by this command to better represent what they actually
	  do. Oh yeah, and the use of this command without specifying a
	  specific queue actually works now. Review:
	  https://reviewboard.asterisk.org/r/1609/ ........ Merged
	  revisions 347656 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-08 17:55 +0000 [r347601]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Mark channel running the h exten with the
	  soft-hangup flag. When a bridge is broken, ast_bridge_call()
	  might execute the h exten on the calling channel. However, that
	  channel may not have been the channel that broke the bridge by
	  hanging up. The channel executing the h exten must be in a hung
	  up state so things like AGI run in the correct mode. * Make sure
	  ast_bridge_call() marks the channel it is executing the h exten
	  on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
	  to match the pbx.c main dialplan execution loop when it executes
	  the h exten.) (closes issue ASTERISK-18811) Reported by: David
	  Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: David Hajek, rmudgett
	  ........ Merged revisions 347595 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347600 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-08 16:24 +0000 [r347533]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't crash on INFO automon request with
	  no channel AST-2011-014. When automon was enabled in
	  features.conf, it was possible to crash Asterisk by sending an
	  INFO request if no channel had been created yet. (closes issue
	  ASTERISK-18805) ........ Merged revisions 347530 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
	  Merged revisions 347531 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347532 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-08 06:59 +0000 [r347490]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Fix segfault on answer. Fix a segfault if
	  an attempt to answer a call is made between when the inbound call
	  gives up (and the channel is removed) and when the device is
	  notified and removes the call from the device.

2011-12-07 21:42 +0000 [r347440]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /: Update AMI Getvar and Setvar documentation
	  about supplying a channel name. (closes issue ASTERISK-18958)
	  Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license
	  #5621) patch uploaded by rmudgett ........ Merged revisions
	  347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 347439 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-07 20:34 +0000 [r347395]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: Fix: Meetme recording variables from
	  realtime DB use null entries over channel variables Meetme would
	  attempt to substitute the realtime values of RECORDING_FILE and
	  RECORDING_FORMAT from the meetme db entry instead of using the
	  channel variable set for those variables in spite of those
	  database entries being NULL or even lacking a column to represent
	  them. (closes issue ASTERISK-18873) Reported by: Byron Clark
	  Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
	  6157) ........ Merged revisions 347369 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347383 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-07 20:15 +0000 [r347345]  Terry Wilson <twilson@digium.com>

	* Makefile, include/asterisk/paths.h, /,
	  configs/asterisk.conf.sample, build_tools/make_defaults_h,
	  main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of
	  configurable paths This patch also makes astdb2sqlite3 and
	  astcanary use the configured directory instead of relying on
	  $PATH. (closes issue ASTERISK-18959) Review:
	  https://reviewboard.asterisk.org/r/1613/ ........ Merged
	  revisions 347344 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-06 23:58 +0000 [r347294]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
	  signals case insensitive. (closes issue ASTERISK-18924) Reported
	  by: Kevin Taylor ........ Merged revisions 347292 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347293 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-06 22:01 +0000 [r347241]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over
	  m([x]) in waitExten If waitExten specifies a music class to use
	  with its music on hold option, it will use CHANNEL(musicclass)
	  instead if that channel variable has been set on the initiating
	  channel. This documents that behavior in the waitExten app so
	  that this can be known without checking the documentation of the
	  code in function local_ast_moh_start. (closes issue
	  ASTERISK-18804) ........ Merged revisions 347239 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347240 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-06 20:23 +0000 [r347157-347192]  Walter Doekes <walter+asterisk@wjd.nu>

	* UPGRADE.txt, CHANGES, apps/app_voicemail.c: Add VM_INFO()
	  dialplan function to gather information about a mailbox.
	  Deprecates MAILBOX_EXISTS. Provides count, email, exists,
	  fullname, language, locale, pager, password, tz. (closes issue
	  ASTERISK-18634) Patch by: Kris Shaw Review:
	  https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter
	  Doekes

	* /, channels/chan_sip.c: Don't allow transport=tcp when
	  tcpenable=no. When tcpenable=no, sending to transport=tcp hosts
	  was still allowed. Resolving the source address wasn't possible
	  and yielded the string "(null)" in SIP messages. Fixed that and a
	  couple of not-so-correct log messages. (closes issue
	  ASTERISK-18837) Reported by: Andreas Topp Review:
	  https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
	  ........ Merged revisions 347166 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347167 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_voicemail.c: Add regression tests for issue
	  ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
	  Reviewed by: Matt Jordan ........ Merged revisions 347131 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347146 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_voicemail.c: The voicemail [general] zonetag and
	  locale variables weren't loaded until after the mailboxes were
	  initialized. This caused the settings to be unset for those
	  mailboxes until a reload was performed. (closes issue
	  ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
	  Reviewed by: Matt Jordan ........ Merged revisions 347111 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347124 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-06 19:09 +0000 [r347110]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/dlinkedlists.h, tests/test_linkedlists.c: Doubly
	  linked lists unit test and update to implementation. Update the
	  doubly linked list implementation. Now safe traversing can insert
	  before and after the current node when traversing in either
	  direction. Updated the linked lists unit test test_linkedlist to
	  also test doubly linked lists. The old test_dlinkedlist requires
	  a manual check of results and probably should be removed. Review:
	  https://reviewboard.asterisk.org/r/1569/

2011-12-06 17:34 +0000 [r347069]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Fixed crash from orphaned MWI
	  subscriptions in chan_sip This patch resolves the issue where MWI
	  subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
	  When a peer is removed, either by pruning realtime SIP peers or
	  by unloading / loading chan_sip, the MWI subscriptions that were
	  orphaned would still be on the event engine list of valid
	  subscriptions but have a pointer to a peer that no longer was
	  valid. When an MWI event would occur, this would cause a seg
	  fault. (closes issue ASTERISK-18663) Reported by: Ross Beer
	  Tested by: Ross Beer, Matt Jordan Patches:
	  blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
	  Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged
	  revisions 347058 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347068 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-05 17:44 +0000 [r347008]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/sig_analog.h: Restore call progress code for analog
	  ports. Extracting sig_analog from chan_dahdi lost call progress
	  detection functionality. * Fix analog ports from considering a
	  call answered immediately after dialing has completed if the
	  callprogress option is enabled. (closes issue ASTERISK-18841)
	  Reported by: Richard Miller Patches: chan_dahdi.diff (license
	  #5685) patch uploaded by Richard Miller (Modified by me)
	  sig_analog.c.diff (license #5685) patch uploaded by Richard
	  Miller (Modified by me) sig_analog.h.diff (license #5685) patch
	  uploaded by Richard Miller ........ Merged revisions 347006 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 347007 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-05 15:04 +0000 [r346956]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c, /: Resolve duplicate label used in multiple
	  priorities for the same extension. Prior to this patch, if labels
	  with the same name were used for different priorities in the same
	  extension, the new label would be accepted, but it would be
	  unusable since attempts to reach that label would just go to the
	  first one. Now pbx.c detects this, generates a warning in logs,
	  and culls the label before adding it to the dialplan. (closes
	  issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
	  pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........
	  Merged revisions 346954 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346955 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-05 14:47 +0000 [r346953]  Kinsey Moore <kmoore@digium.com>

	* res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load
	  regression introduced in r346087 Add missing symbol exports for
	  ast_aji_client_destroy and ast_aji_buddy_destroy for usage
	  outside res_jabber. Testing of these changes focused on
	  res_jabber itself, so this problem was missed. Reported-by:
	  Michael Spiceland ........ Merged revisions 346951 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346952 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-04 10:08 +0000 [r346901]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
	  domain ACL bypass. The code that allowed admins to create users
	  with domain-only uri's had stopped to work in 1.8 because of the
	  reqresp parser rewrites. This is fixed now: if you have a
	  [mydomain.com] sip user, you can register with useraddr
	  sip:mydomain.com. Note that in that case -- if you're using
	  domain ACLs (a configured domain list) -- mydomain.com must be in
	  the allow list as well. Reviewboard r1606 shows a list of
	  registration combinations and which SIP response codes are
	  returned. Review: https://reviewboard.asterisk.org/r/1533/
	  Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
	  issue ASTERISK-18741) ........ Merged revisions 346899 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346900 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-02 23:30 +0000 [r346857]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Update SIP MESSAGE To parsing to
	  correctly handle URI The previous patch (r346040) incorrectly
	  parsed the URI in the presence of a port, e.g.,
	  user@hostname:port would fail as the port would be double
	  appended to the SIP message. This patch uses the parse_uri
	  function to correctly parse the URI into its username and
	  hostname parts, and places them in the correct fields in the
	  sip_pvt structure. (issue ASTERISK-18903) Review:
	  https://reviewboard.asterisk.org/r/1597/ ........ Merged
	  revisions 346856 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-02 19:40 +0000 [r346777-346816]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: implement nat option for rtp channels with
	  ooh323

	* addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions
	  346763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri,
	  02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7
	  lines process null frame pointer returned by
	  ast_rtp_instance_read correctly (closes issue ASTERISK-16697)
	  Reported by: under Patches: segfault.diff (License #5871) patch
	  uploaded by under ........ ................

2011-12-01 21:19 +0000 [r346709]  Richard Mudgett <rmudgett@digium.com>

	* main/stun.c, /, res/res_stun_monitor.c,
	  configs/res_stun_monitor.conf.sample, include/asterisk/stun.h:
	  Re-resolve the STUN address if a STUN poll fails for
	  res_stun_monitor. The STUN socket must remain open between polls
	  or the external address seen by the STUN server is likely to
	  change. However, if the STUN request poll fails then the STUN
	  server address needs to be re-resolved and the STUN socket needs
	  to be closed and reopened. * Re-resolve the STUN server address
	  and create a new socket if the STUN request poll fails. * Fix
	  ast_stun_request() return value consistency. * Fix
	  ast_stun_request() to check the received packet for expected
	  message type and transaction ID. * Fix ast_stun_request() to read
	  packets until timeout or an associated response packet is found.
	  The stun_purge_socket() hack is no longer required. * Reduce
	  ast_stun_request() error messages to debug output. * No longer
	  pass in the destination address to ast_stun_request() if the
	  socket is already bound or connected to the destination. (closes
	  issue ASTERISK-18327) Reported by: Wolfram Joost Tested by:
	  rmudgett Review: https://reviewboard.asterisk.org/r/1595/
	  ........ Merged revisions 346700 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346701 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-12-01 20:46 +0000 [r346699]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
	  ringing. 183 Ringing isn't even a thing. 183 is actually a
	  session progress message. (closes issue ASTERISK-18925) Reported
	  by: Sebastian Denz Tested by: jrose Patches:
	  asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
	  Denz (License #6139) ........ Merged revisions 346697 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346698 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-30 23:38 +0000 [r346617-346655]  Tilghman Lesher <tilghman@meg.abyt.es>

	* channels/chan_unistim.c, main/tcptls.c, channels/chan_sip.c,
	  main/config.c, main/loader.c: Remove the few places where we try
	  to ast_verbose() without a newline.

	* main/asterisk.c: Fix edge case for overflow buffer.

2011-11-30 22:03 +0000 [r346525-346566]  Jonathan Rose <jrose@digium.com>

	* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
	  r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
	  18 lines Cleaning up chan_sip/tcptls file descriptor closing.
	  This patch attempts to eliminate various possible instances of
	  undefined behavior caused by invoking close/fclose in situations
	  where fclose may have already been issued on a
	  tcptls_session_instance and/or closing file descriptors that
	  don't have a valid index for fd (-1). Thanks for more than a
	  little help from wdoekes. (closes issue ASTERISK-18700) Reported
	  by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
	  Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
	  Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged
	  revisions 346564 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346565 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
	  Reverting 346525 due to accidental patch against trunk instead of
	  1.8

	* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
	  Cleaning up chan_sip/tcptls file descriptor closing. This patch
	  attempts to eliminate various possible instances of undefined
	  behavior caused by invoking close/fclose in situations where
	  fclose may have already been issued on a tcptls_session_instance
	  and/or closing file descriptors that don't have a valid index for
	  fd (-1). Thanks for more than a little help from wdoekes. (closes
	  issue ASTERISK-18700) Reported by: Erik Wallin (issue
	  ASTERISK-18345) Reported by: Stephane Cazelas (issue
	  ASTERISK-18342) Reported by: Stephane Chazelas Review:
	  https://reviewboard.asterisk.org/r/1576/

2011-11-30 19:37 +0000 [r346474]  Leif Madsen <leif@leifmadsen.com>

	* configs/queues.conf.sample: Update queues.conf.sample
	  documentation. Update the documentation surrounding the use of
	  MONITOR_EXEC to make it more clear that it can be used for both
	  Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
	  Reported by: David Woolley Patches:
	  issue18817_mixmonitor_queues_doc.diff by Michael L. Young
	  (License #5026) ........ Merged revisions 346472 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346473 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-29 20:32 +0000 [r346391-346429]  Tilghman Lesher <tilghman@meg.abyt.es>

	* utils/refcounter.c, utils/hashtest.c, utils/ael_main.c,
	  utils/hashtest2.c: Fix compilation of utilities (caught by
	  Bamboo).

	* addons/chan_ooh323.c, channels/chan_sip.c, main/say.c,
	  res/res_fax.c, UPGRADE.txt, res/res_musiconhold.c,
	  res/res_jabber.c, CHANGES, configs/logger.conf.sample,
	  main/cli.c, channels/chan_usbradio.c, include/asterisk/logger.h,
	  main/dial.c, channels/chan_skinny.c, main/logger.c,
	  codecs/codec_dahdi.c, apps/app_rpt.c, apps/app_verbose.c,
	  main/asterisk.c, main/bridging.c, res/res_clialiases.c,
	  addons/res_config_mysql.c, apps/app_voicemail.c: Allow each
	  logging destination and console to have its own notion of the
	  verbosity level. Review: https://reviewboard.asterisk.org/r/1599

2011-11-29 00:03 +0000 [r346350]  David Vossel <dvossel@digium.com>

	* /, include/asterisk/message.h, main/message.c: Merged revisions
	  346349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011)
	  | 10 lines Fixes memory leak in message API. The ast_msg_get_var
	  function did not properly decrement the ref count of the var it
	  retrieves. The way this is implemented is a bit tricky, as we
	  must decrement the var and then return the var's value. As long
	  as the documentation for the function is followed, this will not
	  result in a dangling pointer as the ast_msg structure owns its
	  own reference to the var while it exists in the var container.
	  ........

2011-11-28 14:34 +0000 [r346294]  Stefan Schmidt <sst@sil.at>

	* res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set
	  debup ip' only works when also a port was specified. (closes
	  issue ASTERISK-18693) Reported by: Davide Dal Fra Review:
	  https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
	  Doekes ........ Merged revisions 346292 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346293 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 23:03 +0000 [r346241]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/acl.h, /, channels/chan_skinny.c,
	  channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls
	  to ast_get_ip() not initializing the address family. ........
	  Merged revisions 346239 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346240 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 20:48 +0000 [r346146-346199]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
	  function. In r116240, get_msg_text() got an extra parameter to
	  fix the unwanted addition of trailing newlines to SIP MESSAGE
	  bodies. This caused all linefeeds to be trimmed, which isn't
	  right either. This is a stop-gap; the right fix is to return the
	  original SIP request body. Review:
	  https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
	  ........ Merged revisions 346147 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346198 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, include/asterisk/strings.h: Fix ast_str_truncate signedness
	  warning and documentation. Review:
	  https://reviewboard.asterisk.org/r/1594 ........ Merged revisions
	  346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 346145 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 17:16 +0000 [r346088]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_jingle.c, /, include/asterisk/jabber.h,
	  channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource
	  leaks This should fix almost all resource leaks in res_jabber
	  that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous
	  situation where ast_aji_get_client would sometimes bump an
	  object's refcount and sometimes not. Review:
	  https://reviewboard.asterisk.org/r/1553 ........ Merged revisions
	  346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 346087 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 16:23 +0000 [r346053]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Fixed SendMessage stripping extension
	  from To: header in SIP MESSAGE When using the MessageSend
	  application to send a SIP MESSAGE to a non-peer, chan_sip
	  attempted to validate the hostname or IP Address. In the process,
	  it stripped off the extension and failed to add it back to the
	  sip_pvt structure before transmitting. This patch adds the full
	  URI passed in from the message core to the sip_pvt structure.
	  (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/
	  ........ Merged revisions 346040 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 16:12 +0000 [r346033]  Terry Wilson <twilson@digium.com>

	* /, res/res_musiconhold.c: Resume playing existing hold music for
	  cached realtime MOH As a result of the fix for ASTERISK-18039,
	  realtime caching MOH no longer properly resumes playing back a
	  file between different holds in the same call. This is because
	  scanning for new files causes the existing file array to be
	  emptied and we were just comparing that the saved pointer to the
	  filename matched the pointer to the filename in a particular
	  position in the array. An easy fix is to save the filename
	  instead of a pointer to it and then do a strcmp instead of
	  comparing the addresses. (closes issue ASTERISK-18912) Review:
	  https://reviewboard.asterisk.org/r/1596/ ........ Merged
	  revisions 346030 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 346031 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-23 16:10 +0000 [r346032]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Added
	  support level for new modules ........ Merged revisions 346029
	  from http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-22 23:06 +0000 [r345978]  Richard Mudgett <rmudgett@digium.com>

	* main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries
	  to ask for the same address family each time. The dnsmgr refresh
	  would always get the first address found regardless of the
	  original address family requested. So if you asked for only IPv4
	  addresses originally, you might get an IPv6 address on refresh. *
	  Saved the original address family requested by
	  ast_dnsmgr_lookup() to be used when the address is refreshed.
	  ........ Merged revisions 345976 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345977 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-22 20:32 +0000 [r345925]  Walter Doekes <walter+asterisk@wjd.nu>

	* include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros
	  exist next to the LOG_* macros. (issue ASTERISK-17973) ........
	  Merged revisions 345923 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345924 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-22 16:41 +0000 [r345883]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/confbridge/conf_config_parser.c: Add missing
	  sound_only_one config variable (closes issue ASTERISK-18895)
	  Reported by: zvision Patches: conf_config_parser.diff (license
	  #5755) patch uploaded by zvision ........ Merged revisions 345882
	  from http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-21 21:09 +0000 [r345831]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default
	  to nat=yes; warn when nat in general and peer differ It is
	  possible to enumerate SIP usernames when the general and
	  user/peer nat settings differ in whether to respond to the port a
	  request is sent from or the port listed for responses in the Via
	  header. In 1.4 and 1.6.2, this would mean if one setting was
	  nat=yes or nat=route and the other was either nat=no or
	  nat=never. In 1.8 and 10, this would mean when one was
	  nat=force_rport and the other was nat=no. In order to address
	  this problem, it was decided to switch the default behavior to
	  nat=yes/force_rport as it is the most commonly used option and to
	  strongly discourage setting nat per-peer/user when at all
	  possible. For more discussion of the issue, please see:
	  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
	  (closes issue ASTERISK-18862) Review:
	  https://reviewboard.asterisk.org/r/1591/ ........ Merged
	  revisions 345776 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
	  revisions 345800 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
	  Merged revisions 345828 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345830 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-21 16:40 +0000 [r345735]  Paul Belanger <paul.belanger@polybeacon.com>

	* CHANGES, main/config.c: Add #tryinclude statement This provides
	  the same functionality as #include however an asterisk module
	  will still load if the filename does not exist. Review:
	  https://reviewboard.asterisk.org/r/1476/

2011-11-19 15:11 +0000 [r345643-345684]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, main/db.c: Update the documentation to better clarify how the
	  existing commands work. Review:
	  https://reviewboard.asterisk.org/r/1593/ ........ Merged
	  revisions 345682 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345683 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/db.c: Fix a change in behavior in 'database show' from
	  1.8. In 1.8 and previous versions, one could use any fullword
	  portion of the key name, including the full key, to obtain the
	  record. Until this patch, this did not work for the full key.
	  Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson
	  (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged
	  revisions 345640 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-17 19:47 +0000 [r345560-345601]  Matthew Jordan <mjordan@digium.com>

	* contrib/realtime/mysql/sipfriends.sql (removed): Accidentally
	  readded sipfriends.sql in r345560. This was removed in r342871

	* configs/confbridge.conf.sample,
	  apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
	  CHANGES, contrib/realtime/mysql/sipfriends.sql (added),
	  apps/confbridge/conf_config_parser.c: Add admin toggle mute all
	  and participant count menu options to app_confbridge This patch
	  adds two new menu features to app_confbridge, admin_toggle_menu_
	  participants and participant_count. The admin action will
	  globally mute / unmute all non-admin participants on a
	  converence, while the participant count simply exposes the
	  existing participant count function to the conference bridge
	  menu. This also adds configuration options to change the sound
	  played when the conference is globally muted / unmuted, as well
	  as the necessary config hooks to place these functions in the
	  DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin
	  Reeves Tested by: Matt Jordan Patches:
	  app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
	  confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
	  Review: https://reviewboard.asterisk.org/r/1518/

2011-11-17 17:31 +0000 [r345559]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Remove dead code since pri_grab() can
	  never fail. Dead code makes programmers sick. I am sick of
	  looking at it. ........ Merged revisions 345546 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345558 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-16 14:56 +0000 [r345489]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_voicemail.c: Guarantee messages go into the right
	  folders with multiple recipients Before, using the U flag in
	  Voicemail with multiple recipients would put urgent messages in
	  the INBOX folder for all users past the first thanks to a bug
	  with the message copying function. This would also cause messages
	  to fail to be sent if the INBOX directory hadn't been created for
	  that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
	  Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged
	  revisions 345487 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345488 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-15 20:11 +0000 [r345221-345433]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug
	  output. * Change from using send() to ast_agi_send() so the
	  HANGUP shows up in the AGI debug output. (closes issue
	  ASTERISK-18723) Reported by: James Van Vleet Patches:
	  jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by
	  rmudgett ........ Merged revisions 345431 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345432 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure
	  name. It is fortunate that the typo does not alter generated code
	  since the e->restart.channel and e->ring.channel members are in
	  the same position. (closes issue ASTERISK-18868) Reported by:
	  zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
	  zvision ........ Merged revisions 345370 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345371 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is
	  paused for AMI and realtime. * Add parameter to queue log
	  ADDMEMBER to indicate if the member is paused. (closes issue
	  ASTERISK-18645) Reported by: garlew Patches: paused.diff (License
	  #5337) patch uploaded by garlew Tested by: rmudgett, garlew
	  Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged
	  revisions 345285 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345290 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt,
	  channels/sip/include/sip.h: Restore SIP DTMF overlap dialing
	  method. The recent fix for ASTERISK-17288 to get RFC3578 SIP
	  overlap support working correctly removed a long standing ability
	  to do overlap dialing using DTMF in the early media phase of a
	  call. See ASTERISK-18702 it has a very good description of the
	  issue. I started with Pavel Troller's chan_sip.diff patch on
	  issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
	  allowoverlap config option. The new option value causes the
	  Incomplte application to not send anything with chan_sip so the
	  caller can supply more digits via DTMF. * Renames
	  SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
	  since that is what it really means. * Fixed get_destination()
	  inconsistency with the pickup extension matching. * Fixed
	  initialization of PAGE3 of global_flags in reload_config().
	  (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
	  https://reviewboard.asterisk.org/r/1517/ Review:
	  https://reviewboard.asterisk.org/r/1582/ ........ Merged
	  revisions 345273 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345275 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes
	  issue ASTERISK-18857) Reported by: David M Patches:
	  mainpbx-trivial.patch (License #6326) patch uploaded by David M
	  ........ Merged revisions 345219 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345220 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-14 19:12 +0000 [r345165]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /: Don't read past end of input when calling
	  write() int blah = 1; ... write(chan->alertpipe[1], &blah,
	  new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is
	  only valid when new_frames == 1. Otherwise we start reading into
	  adjacent variables declared on the stack. The read end discards
	  what is read, so the values don't matter but it's not a good idea
	  to read past where we want even though new_frames is almost
	  always 1 and should never be large. This patch is basically taken
	  out of kpfleming's eventfd branch, as he mentioned that he
	  remembered fixing it there when I talked to him about this issue.
	  Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged
	  revisions 345163 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345164 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-14 19:03 +0000 [r345162]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/sip/include/reqresp_parser.h: Update reqresp_parser
	  parse_uri doxygen comments. The issue mentioned in the bug report
	  had been fixed recently by twilson. The reporter included this
	  documentation fix. (closes issue ASTERISK-18572) Reported by:
	  Richard Miller Patch by: Richard Miller (modified) ........
	  Merged revisions 345160 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345161 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-14 16:21 +0000 [r345120]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_voicemail.c: Moves voicemail setup password entry to
	  the end of the setup process. This change was made because
	  forcegreeting and forcename settings in voicemail could be
	  circumvented by hanging up after entering a password, because the
	  only way voicemail currently observes whether a mailbox is new or
	  not is by checking to see if the password is the same as the
	  mailbox number or not. (closes issue ASTERISK-18282) Reported by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
	  ........ Merged revisions 345062 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345117 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-14 15:11 +0000 [r345065]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Ensure that a null vmexten does not cause
	  a segfault When sip_send_mwi_to_peer was modified recently to
	  avoid deadlocks, vmexten was not expected to be null. This change
	  handles that situation to avoid a segfault. ........ Merged
	  revisions 345063 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 345064 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-14 01:25 +0000 [r345023]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Increased max number of destinations.

2011-11-12 16:32 +0000 [r344979]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_misdn.c, /: mISDN Round Robin break when no channel
	  is available Prevent channels been parsed repetitively. ........
	  Merged revisions 344965 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344966 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-12 00:36 +0000 [r344901]  Terry Wilson <twilson@digium.com>

	* /, res/res_musiconhold.c: Don't forget to rescan MOH files for
	  cached realtime classes Realtime MOH class caching was
	  implemented because without it, you would build a completely new
	  MOH class and would start the music over at the beginning each
	  time hold was pressed in a conversation. Unfortunately, this
	  broke re-scanning for file changes for realtime MOH classes. This
	  patch corrects that issue. (closes issue ASTERISK-18039) Review:
	  https://reviewboard.asterisk.org/r/1579/ ........ Merged
	  revisions 344899 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344900 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 22:00 +0000 [r344846]  Walter Doekes <walter+asterisk@wjd.nu>

	* include/asterisk/utils.h, /, main/utils.c,
	  include/asterisk/stringfields.h: Use __alignof__ instead of
	  sizeof for stringfield length storage. Kevin P Fleming suggested
	  that r343157 should use __alignof__ instead of sizeof. For most
	  systems this won't be an issue, but better fix it now while it's
	  still fresh. Review: https://reviewboard.asterisk.org/r/1573
	  ........ Merged revisions 344843 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344845 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 21:57 +0000 [r344844]  Matthew Jordan <mjordan@digium.com>

	* /, main/file.c: Video format was treated as audio when removed
	  from the file playback scheduler This patch fixes the format type
	  check in ast_closestream and filestream_destructor. Previously a
	  comparison operator was used, but since audio formats are no
	  longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
	  that have a value greater than the video formats), a bitwise AND
	  operation is used instead. Duplicated code was also moved to
	  filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
	  Bedrij Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1580/ ........ Merged
	  revisions 344823 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344842 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 21:37 +0000 [r344838-344840]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/sip/reqresp_parser.c: Remove unneeded if(params)
	  checks in reqresp_parser. Nick Lewis added them in
	  https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
	  reason. There is no way that params could become NULL in that
	  piece of code, so I removed these excess checks again. ........
	  Merged revisions 344837 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344839 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* main/manager.c, /: Fix bad quoting of multiline mxml opaque_data
	  that caused invalid xml. The opaque_data was added and enclosed
	  in single quotes, assuming it would be only a single line. The
	  rest of the lines were appended after the closing quote. (closes
	  issue ASTERISK-18852) Reported by: peep_ on IRC Review:
	  https://reviewboard.asterisk.org/r/1577 ........ Merged revisions
	  344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 344836 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 20:15 +0000 [r344771]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Fix regression introduced by SDP fixups
	  If capability is adjusted when switching to UDPTL during fax
	  transmission, fax teardown fails. Make sure capability is only
	  touched if RTP is active. This regression was introduced in
	  R344385. ........ Merged revisions 344769 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344770 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 18:37 +0000 [r344663-344717]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Check sip.conf maxforwards parameter for
	  range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions
	  344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 344716 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, main/cli.c: Make CLI "core show channel" not hold the channel
	  lock during console output. Holding the channel lock while the
	  CLI "core show channel" command is executing can slow down the
	  system. It could block the system if the console output is halted
	  or paused. * Made capture the CLI "core show channel" output into
	  a buffer to be output after the channel is unlocked. * Removed
	  use of C++ keyword as a variable name. out renamed to obuf. *
	  Checked allocation of obuf for failure so will not crash. (closes
	  issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
	  rmudgett ........ Merged revisions 344661 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344662 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-11 15:47 +0000 [r344610]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c, /: Fix a segmentation fault when using an extension
	  with CID matching and no CID. Attempting to call an extension
	  which used Caller ID matching with a channel that has an empty
	  caller id string would result in a segmentation fault. (closes
	  issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged
	  revisions 344608 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344609 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-10 23:21 +0000 [r344538-344560]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_macro.c: Fix app_macro.c MODULEINFO section
	  termination. (closes issue ASTERISK-18848) Reported by: Tony
	  Mountifield ........ Merged revisions 344557 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Fix potential deadlock calling ast_call()
	  with channel locks held. Fixed app_queue.c:ring_entry() calling
	  ast_call() with the channel locks held. Chan_local attempts to do
	  deadlock avoidance in its ast_call() callback and could deadlock
	  if a channel lock is already held. ........ Merged revisions
	  344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 344540 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Make AMI event AgentCalled get
	  CallerID/ConnectedLine info from the incoming channel. It was
	  strange that the AgentCalled AMI event would get most of its
	  information from the incoming channel but then get the CallerID
	  information from the outgoing channel. Before connected line
	  support was added, this information was always the same at this
	  point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
	  Tested by: rmudgett ........ Merged revisions 344536 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344537 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-10 21:56 +0000 [r344494]  David Vossel <dvossel@digium.com>

	* /, main/bridging.c: Merged revisions 344493 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011)
	  | 12 lines Fixes issue with ConfBridge participants hanging up
	  during DTMF feature menu usage getting stuck in conference
	  forever. When a conference user enters the DTMF menu they are
	  suspended from the bridge while the channel is handed off to the
	  DTMF feature code. If a user entered this state and hungup, there
	  existed a race condition where the channel could not exit the
	  conference because it was waiting on a signal that would never
	  arrive. This patch fixes that, because it would stupid for me to
	  talk about the problem and commit a patch for something else.
	  (closes issue ASTERISK-18829) Reported by: zvision ........

2011-11-10 21:15 +0000 [r344387-344441]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_meetme.c: Fix another incorrect case with meetme's
	  PIN logic and add documentation This fixes an issue where a user
	  of a dynamic conference was asked for a PIN twice. This also adds
	  documentation to assist in future modifications to the piece of
	  code responsible for PIN checking. (closes issue AST-670)
	  ........ Merged revisions 344439 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344440 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several
	  bugs with SDP parsing and well-formedness of responses Fix bug
	  ASTERISK-16558 which dealt with the order of responses to
	  incoming streams defined by SDP. Fix unreported bug where
	  offering multiple same-type streams would cause Asterisk to reply
	  with an incorrect SDP response missing one or more streams
	  without a proper declination. Fix bugs related to a single
	  non-audio stream being offered with responses requesting codecs
	  that were not offered in the initial invite along with an
	  additional audio stream that was not in the initial invite.
	  Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged
	  revisions 344385 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344386 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-10 16:29 +0000 [r344335]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_rtp_asterisk.c, /: only attempt to do stun handling on
	  ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny
	  (modified) ASTERISK-18490 ........ Merged revisions 344330 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344334 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-09 20:55 +0000 [r344272]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix deadlock during dialplan reload.
	  Another deadlock between the conlock/hints and channels/channel
	  locking orders. * Don't hold the channel and private lock in
	  sip_new() when calling ast_exists_extension(). (closes issue
	  ASTERISK-18740) Reported by: Byron Clark Patches:
	  sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
	  Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
	  uploaded by Byron Clark Tested by: Byron Clark ........ Merged
	  revisions 344268 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344271 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-09 20:10 +0000 [r344214-344217]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, channels/sip/reqresp_parser.c,
	  channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h: Don't treat a host:port
	  string as a domain The domain matching code prior to 1.8 used to
	  manually remove the port from the host:port string when
	  determining if an incoming request matched the list of domains.
	  When switching to the new parsing functions, the documentation
	  implied that the "domain" was being returned by these functions,
	  when instead it was returning the "hostport" as defined by RFC
	  3261. This led to confusion and resulted in 1.8+ rejecting an
	  incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set
	  in sip.conf. This patch renames the "domain" variables in the
	  parsing functions to "hostport" to more accurately describe what
	  it is that they are returning and also properly truncates the
	  resulting hostport strings when dealing with domain matching.
	  Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged
	  revisions 344215 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344216 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, tests/test_netsock2.c: Add a unit test for
	  ast_sockaddr_split_hostport Review:
	  https://reviewboard.asterisk.org/r/1575/ ........ Merged
	  revisions 344157 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344175 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-09 19:08 +0000 [r344161]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c,
	  addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h,
	  addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c:
	  Generate response to Status Enquiry message with Status q.931
	  message. Some PBXes require this for call status checking (closes
	  issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
	  ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
	  Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 344159 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-09 17:15 +0000 [r344104]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_meetme.c: Fix pin parameter behavior regression in
	  MeetMe The last time this code was touched (by me), a subtlety
	  was missed based on the difference between needing to check a
	  pin's validity and the need to prompt for a pin. (closes issue
	  ASTERISK-18488) ........ Merged revisions 344102 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344103 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-09 15:28 +0000 [r344050]  Matthew Nicholson <mnicholson@digium.com>

	* /, formats/format_wav.c: don't call ltohl() twice on the same
	  value ASTERISK-18739 Patch by: pawel (modified) ........ Merged
	  revisions 344048 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 344049 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-08 22:14 +0000 [r344005]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Residual changes for Asterisk v10 branch
	  from ASTERISK-18747. Residual changes for Asterisk v10 branch
	  from ASTERISK-18747 after
	  https://reviewboard.asterisk.org/r/1564/ commit and associated
	  dialogs callid hash key change fix. * Make check_rtp_timeout()
	  return CMP_MATCH if need to delete dialog from dialogs_rtpcheck.
	  This is an optimization to avoid an unneeded lock/unlock and
	  object search when using ao2_unlink. * Prevent crash in
	  check_rtp_timeout() if dialog->rtp is NULL. Review:
	  https://reviewboard.asterisk.org/r/1557/ ........ Merged
	  revisions 344004 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-08 19:29 +0000 [r343951]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, pbx/pbx_config.c: Fix crash when dialplan remove include is
	  called with too few arguments. "dialplan remove include x from y"
	  crashed when the amount of arguments was less than 6. (closes
	  issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
	  Andrey Solovyev ........ Merged revisions 343936 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343944 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-08 18:35 +0000 [r343905]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 343900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011)
	  | 11 lines Fixes regression caused by r343635 There was a missing
	  unlock for a function return that is only present in Asterisk 10
	  and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by:
	  Michael L. Young Patches:
	  asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch
	  uploaded by Michael L. Young ........

2011-11-08 18:02 +0000 [r343853]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
	  variable if unknown host configured crash. * Fixed a LOG_ERROR
	  message referencing the config variable list v that had
	  previously been processed and became NULL. * Added error return
	  value set that was missing in an ast_append_ha() error return
	  path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
	  issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
	  (license #5674) patch uploaded by Walter Doekes Tested by:
	  Michele ........ Merged revisions 343851 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343852 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-08 13:23 +0000 [r343790]  Leif Madsen <leif@leifmadsen.com>

	* /, build_tools/prep_tarball: Fix boo-boo in prep_tarball script.
	  A hardcoded a branch number was in the prep_tarball which could
	  not work. Changed it to the variable. ........ Merged revisions
	  343789 from http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 22:37 +0000 [r343744]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Make "sip show settings" CLI command get
	  RPID flags from the right global page The "Trust RPID" and "Send
	  RPID" entries in the "sip show settings" CLI command pulled the
	  flags from the incorrect global flags page. These are now read
	  from sip global flags page 0. (closes issue AST-711) ........
	  Merged revisions 343743 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 21:58 +0000 [r343693]  Leif Madsen <leif@leifmadsen.com>

	* configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Allow built
	  in variables to be used with dynamic weights. You can now use the
	  built in variables , , and within a dynamic weight. For example,
	  this could be useful when you want to pass requested lookup
	  number to the SHELL() function which could be used to execute a
	  script to dynamically set the weight of the result. (Closes issue
	  ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen,
	  Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded
	  by Joel Vandal (License #5374)

2011-11-07 21:44 +0000 [r343692]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: respect case changes in peer names on sip
	  reload ASTERISK-18669 ........ Merged revisions 343690 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343691 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 21:29 +0000 [r343684]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
	  changing dialogs hash key callid. Changing an object value used
	  as a container key requires removing the object from the
	  container and reinserting it. * Created change_callid_pvt() to
	  call instead of build_callid_pvt(). The change_callid_pvt() will
	  correctly change the dialog callid so the ao2 conainter can
	  explicitly unlink it. ........ Merged revisions 343637 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343677 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 20:35 +0000 [r343636]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Prevent BLF subscriptions from causing
	  deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
	  was causing deadlocks. This function now requires that both the
	  peer and associated pvt be unlocked before it is called for cases
	  where peer and peer->mwipvt form a circular reference. (closes
	  issue ASTERISK-18663) Review:
	  https://reviewboard.asterisk.org/r/1563/ ........ Merged
	  revisions 343621 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343635 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 19:58 +0000 [r343581]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/udptl.c, /, UPGRADE.txt: Correct the default udptl port
	  range. The udptl port range was defined as 4000-4999 in the
	  udptl.conf.sample, as 4500-4599 if you didn't have a config and
	  4500-4999 if your config was broken. Default is now 4000-4999.
	  (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher
	  Review: https://reviewboard.asterisk.org/r/1565 ........ Merged
	  revisions 343580 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 19:54 +0000 [r343579]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix deadlock if peer is destroyed while
	  sending MWI notice. A dialog cannot be destroyed by the
	  ao2_callback dialog_needdestroy because of a deadlock between the
	  dialogs container lock and the RWLOCK of the events subscription
	  list. * Create dialogs_to_destroy container to hold dialogs that
	  will be destroyed. * Ensure that the event subscription callback
	  will never happen with an invalid peer pointer by making the
	  event callback removal the first thing in the peer destructor
	  callback. NOTE: This particular deadlock will not happen with
	  Asterisk 10, but some of the changes still apply. (closes issue
	  ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review:
	  https://reviewboard.asterisk.org/r/1564/ ........ Merged
	  revisions 343577 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343578 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-07 18:42 +0000 [r343534]  Matthew Nicholson <mnicholson@digium.com>

	* main/format.c, /: list all of the codecs associated with a
	  particular format id for CLI command "core show codec" AST-699
	  ........ Merged revisions 343533 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-06 09:51 +0000 [r343492]  Olle Johansson <oej@edvina.net>

	* main/tcptls.c, include/asterisk/tcptls.h: Formatting and doxygen
	  improvements

2011-11-04 19:50 +0000 [r343448]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h,
	  addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c:
	  Final fix memleaks in GkClient codes, same for Timer codes.
	  (these memleaks stop development of gk codes, now i can continue)
	  Fix printHandler 'Unbalanced Structure' issues with locking
	  printHandler data for single thread. ........ Merged revisions
	  343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 343445 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-03 20:37 +0000 [r343394]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, res/res_config_sqlite.c: Fix sqlite config driver segfault and
	  broken queries The sqlite realtime handler assumed you had a
	  static config configured as well. The realtime multientry handler
	  assumed that you weren't using dynamic realtime. (closes issue
	  ASTERISK-18354) (closes issue ASTERISK-18355) Review:
	  https://reviewboard.asterisk.org/r/1561 ........ Merged revisions
	  343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 343393 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-03 19:57 +0000 [r343338]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_dialgroup.c: Remove invalid flag given to iterator
	  in func_dialgroup.c ........ Merged revisions 343336 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343337 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-03 15:40 +0000 [r343222-343278]  Terry Wilson <twilson@digium.com>

	* /, channels/sip/include/sip.h: Make room for the fax detect flags
	  The original REGISTERTRYING flag, in addition to being impossible
	  to check, also encroached on the space for the flag above it.
	  This patch moves the flags that were below REGISTERTRYING back to
	  where they were as though we had just removed the REGISTERTRYING
	  option. ........ Merged revisions 343276 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343277 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
	  channels/sip/include/sip.h: Remove registertrying option in
	  chan_sip This option is not only useless, but has been broken
	  since inception since the flag was never copied from the peer
	  where it is set to the pvt where it was checked. RFC 3261
	  specificially states that you should not send a provisional
	  response to a non-INVITE request, and if we did fix the code so
	  that it worked, it would cause the same kind of user enumeration
	  vulnerability that we've discussed with the nat= setting. This
	  patch removes registertrying option and any code that would have
	  sent a 100 response to a register. Review:
	  https://reviewboard.asterisk.org/r/1562/ ........ Merged
	  revisions 343220 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343221 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-02 22:46 +0000 [r343163-343219]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c: Fix improper warning introduced by
	  r342927 and more tweaks Changeset r342927 introduced a warning
	  which was only supposed to be emitted when a found realtime peer
	  had an empty (or no) name. It turned out that there were some
	  inconsistencies left. Now found peers with an empty name are
	  explicitly ignored like before r342927 but better. Reviewed by:
	  Stefan Schmidts, Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/1560 ........ Merged revisions
	  343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 343192 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* include/asterisk/utils.h, /, main/utils.c,
	  include/asterisk/stringfields.h: Ensure that string field lengths
	  are properly aligned Integers should always be aligned. For some
	  platforms (ARM, SPARC) this is more important than for others.
	  This changeset ensures that the string field string lengths are
	  aligned on *all* platforms, not just on the SPARC for which there
	  was a workaround. It also fixes that the length integer can be
	  resized to 32 bits without problems if needed. (closes issue
	  ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
	  Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
	  https://reviewboard.asterisk.org/r/1549 ........ Merged revisions
	  343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 343158 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-02 19:33 +0000 [r343049-343104]  Leif Madsen <leif@leifmadsen.com>

	* apps/app_authenticate.c: Add note about how Authenticate()
	  application with option 'd' works. (closes issue ASTERISK-17422)
	  Reported by: Leif Madsen ........ Merged revisions 343102 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343103 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* configs/queues.conf.sample: Update documentation for leastrecent
	  strategy. In queues.conf.sample the leastrecent strategy was
	  incorrectly described. Now updated to reflect how the strategy
	  actually checks peers. (closes issue ASTERISK-17854) Reported by:
	  Sebastian Denz Patches: queues.conf-doc_issue.patch (License
	  #6139) ........ Merged revisions 343047 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 343048 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-02 13:46 +0000 [r342992]  Kevin P. Fleming <kpfleming@digium.com>

	* /, apps/app_meetme.c: Modify comments in MeetMe application
	  documentation about DAHDI. The MeetMe application documentation
	  has some comments about usage of DAHDI, and they were a bit
	  outdated relative to modern DAHDI releases. This patch changes
	  the comment to just tell the user that a functional DAHDI timing
	  source is required, and no longer mention 'dahdi_dummy', since
	  that module does not exist in current DAHDI releases. ........
	  Merged revisions 342990 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342991 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-11-01 21:02 +0000 [r342871-342930]  Walter Doekes <walter+asterisk@wjd.nu>

	* /, channels/chan_sip.c, configs/extconfig.conf.sample,
	  include/asterisk/config.h, main/config.c: Several fixes to the
	  chan_sip dynamic realtime peer/user lookup There were several
	  problems with the dynamic realtime peer/user lookup code. The
	  lookup logic had become rather hard to read due to lots of
	  incremental changes to the realtime_peer function. And, during
	  the addition of the sipregs functionality, several possibilities
	  for memory leaks had been introduced. The insecure=port matching
	  has always been broken for anyone using the sipregs family. And,
	  related, the broken implementation forced those using sipregs to
	  *still* have an ipaddr column on their sippeers table. Thanks
	  Terry Wilson for comprehensive testing and finding and fixing
	  unexpected behaviour from the multientry realtime call which
	  caused the realtime_peer to have a completely unused code path.
	  This changeset fixes the leaks, the lookup inconsistenties and
	  that you won't need an ipaddr column on your sippeers table
	  anymore (when you're using sipregs). Beware that when you're
	  using sipregs, peers with insecure=port will now start matching!
	  (closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
	  Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
	  Wilson Review: https://reviewboard.asterisk.org/r/1395 ........
	  Merged revisions 342927 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342929 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* contrib/realtime/mysql/sippeers.sql (added),
	  configs/res_config_mysql.conf.sample, /,
	  configs/extconfig.conf.sample, configs/res_ldap.conf.sample,
	  res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample,
	  main/config.c, contrib/realtime/mysql/sipfriends.sql (removed):
	  Cleanup references to sipusers and sipfriends dynamic realtime
	  families Somewhere between 1.4 and 1.8 the sipusers family has
	  become completely unused. Before that, the sipfriends family had
	  been obsoleted in favor of separate sipusers and sippeers
	  families. Apparently, they have been merged back again into a
	  single family which is now called "sippeers". Reviewed by:
	  irroot, oej, pabelanger Review:
	  https://reviewboard.asterisk.org/r/1523 ........ Merged revisions
	  342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ........ Merged revisions 342870 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-31 17:51 +0000 [r342825]  Richard Mudgett <rmudgett@digium.com>

	* main/format.c, /, main/format_cap.c: Misc format capability
	  fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the
	  wrong variable. * Fix potential race between checking if an
	  interface exists and adding it to the container in
	  format.c:ast_format_attr_reg_interface(). * Fixed double rwlock
	  destroy in format.c:ast_format_attr_init() error exit path. *
	  Simplified format.c:find_interface() and
	  format.c:has_interface(). ........ Merged revisions 342824 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-31 16:10 +0000 [r342771]  Matthew Jordan <mjordan@digium.com>

	* main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access
	  when adding extension to pattern match tree When an extension is
	  removed from a context, its entry in the pattern match tree is
	  not deleted. Instead, the extension is marked as deleted. When an
	  extension is removed and re-added, if that extension is also a
	  prefix of another extension, several log messages would report an
	  error and did not check whether or not the extension was deleted
	  before accessing the memory. Additionally, if the extension was
	  already in the tree but previously deleted, and the pattern was
	  at the end of a match, the findonly flag was not honored and the
	  extension would be erroneously undeleted. Additionaly, it was
	  discovered that an IAX2 peer could be unregistered via the CLI,
	  while at the same time it could be scheduled for unregistration
	  by Asterisk. The unregistration method now checks to see if the
	  peer was already unregistered before continuing with an
	  unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
	  Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1526 ........ Merged
	  revisions 342769 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342770 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-30 02:31 +0000 [r342716]  Terry Wilson <twilson@digium.com>

	* /, res/res_calendar.c: Don't crash on empty notify channel
	  ........ Merged revisions 342715 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-29 04:41 +0000 [r342663-342664]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/linkedlists.h: Whitespace and some better macro
	  variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev
	  to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to
	  __extracted.

	* /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix
	  AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
	  AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
	  iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
	  the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
	  list if AST_LIST_INSERT_BEFORE_CURRENT() or
	  AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
	  cut and paste error using the wrong variable in
	  AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
	  for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
	  AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 342662 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-27 20:11 +0000 [r342606]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/dsp.c: tweak the v21 detector to detect an additional
	  pattern of hits and misses ........ Merged revisions 342605 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-27 19:48 +0000 [r342557-342604]  Jonathan Rose <jrose@digium.com>

	* res/res_rtp_multicast.c, /: Fix sequence number overflow over 16
	  bits causing codec change in RTP packets. Sequence number was
	  handled as an unsigned integer (usually 32 bits I think, more
	  depending on the architecture) and was put into the rtp packet
	  which is basically just a bunch of bits using an or operation.
	  Sequence number only has 16 bits allocated to it in an RTP packet
	  anyway, so it would add to the next field which just happened to
	  be the codec. This makes sure the sequence number is set to be a
	  16 bit integer regardless of architecture (hopefully) and also
	  makes it so the incrementing of the sequence number does bitwise
	  or at the peak of a 16 bit number so that the value will be set
	  back to 0 when going beyond 65535 anyway. (closes issue
	  ASTERISK-18291) Reported by: Will Schick Review:
	  https://reviewboard.asterisk.org/r/1542/ ........ Merged
	  revisions 342602 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342603 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_jabber.c: Cleanup reference leaks in res_jabber
	  res_jabber.c had a number of places where astobjs would be
	  referenced and have their reference counts bumped without having
	  a dereference made before the object lost scope. This patch adds
	  a number of ASTOBJ_UNREFs to resolve that. Review:
	  https://reviewboard.asterisk.org/r/1478/ ........ Merged
	  revisions 342545 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342546 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-25 22:06 +0000 [r342486-342489]  Richard Mudgett <rmudgett@digium.com>

	* /, main/astobj2.c: Check fopen return value for ao2 reference
	  debug output. Reported by: wdoekes Patched by: wdoekes Review:
	  https://reviewboard.asterisk.org/r/1539/ ........ Merged
	  revisions 342487 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342488 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, channels/sig_pri.c: Change D-channel warning to be less
	  confusing on non-NFAS setups. The "No D-channels available! Using
	  Primary channel as D-channel anyway!" WARNING message has been
	  confusing on non-NFAS setups. The message refers to things that
	  are NFAS specific. * Changed the warning to several different
	  warnings to be more accurate for the situation and less confusing
	  as a result: "No D-channels up! Switching selected D-channel from
	  X to Y.", "No D-channels up!", and "D-channel is down!". ........
	  Merged revisions 342484 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342485 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-25 21:11 +0000 [r342382-342437]  Terry Wilson <twilson@digium.com>

	* /, apps/app_queue.c: Use int for storing ao2_container_count
	  instad of size_t AST-676 ........ Merged revisions 342435 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342436 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Simplify queue membercount code Despite an
	  ominous sounding comment stating that membercount was for "logged
	  in" members only and thus we couldn't use ao2_container_count(),
	  I could not find a single place in the code where that seemed to
	  be accurate. The only time we decremented membercount was when we
	  were marking something dead or actually removing it. The only
	  places we incremented it were either after ao2_link(), or trying
	  to correct for having set it to 0 during a reload. In every case
	  where we were correcting the value, it seemed that we were trying
	  to make the count actually match what ao2_container_count() would
	  return. The only place I could find where we made a determination
	  about something being "logged in" or not, we didn't trust the
	  membercount, but instead looked at devicestate, paused, etc. This
	  patch removes membercount, replaces its use with
	  ao2_container_count, and manually adds the results of
	  ao2_container_count to a "membercount" field for ast_data queue
	  query results. This patch also would fix AST-676, but as it is
	  slightly riskier than the previously committed fix, the two
	  commits have been made separately. Reivew:
	  https://reviewboard.asterisk.org/r/1541/ ........ Merged
	  revisions 342383 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342384 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Properly update membercount for reloaded
	  members Since q->membercount is set to 0 before reloading, it is
	  important to increment it again for reloaded members as well as
	  added. (closes issue AST-676) Review:
	  https://reviewboard.asterisk.org/r/1541/ ........ Merged
	  revisions 342380 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342381 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-25 19:09 +0000 [r342278-342330]  Kinsey Moore <kmoore@digium.com>

	* pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for
	  pbx_spool.c One of the changes in the recent spool handling of
	  hardlinks patch was just outside a HAVE_INOTIFY block and caused
	  compilation to fail in some build environments. This has been
	  corrected. ........ Merged revisions 342328 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342329 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* pbx/pbx_spool.c, /: Merged revisions 342277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r342277 | kmoore | 2011-10-25 11:08:04 -0500
	  (Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) |
	  18 lines Fix spool handling to allow call files to be hardlinked
	  into place This fixes the inotify code to handle call files being
	  hardlinked into the spool directory. The smsq utility does this,
	  instead of rename(), to ensure that it cannot accidentally
	  overwrite an existing spool file. A rename() might do that, but
	  link() will definitely not. The inotify code had broken this,
	  because it would wait for an IN_CLOSE_WRITE event on the file...
	  which was never forthcoming, since it was never opened. Now we
	  look for IN_OPEN events following the IN_CREATE event, and only
	  wait for an IN_CLOSE_WRITE if the file was actually opened.
	  Patch-by: dwmw2 (closes issue ASTERISK-18331) Review:
	  https://reviewboard.asterisk.org/r/1391/ ........
	  ................

2011-10-25 01:29 +0000 [r342225]  Terry Wilson <twilson@digium.com>

	* /, include/asterisk/config.h, main/config.c: Return NULL when no
	  results returned for realtime_multientry It was not documented
	  what the return value should be when no entries were returned
	  with the multientry realtime callback. This change forces
	  consistent behavior even if the backends return an empty
	  ast_config. Review: https://reviewboard.asterisk.org/r/1521/
	  ........ Merged revisions 342223 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 342224 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-24 22:37 +0000 [r342184]  Richard Mudgett <rmudgett@digium.com>

	* /, include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add
	  missing link/unlink nolock debug defines. ........ Merged
	  revisions 342183 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-24 22:09 +0000 [r342148]  Jonathan Rose <jrose@digium.com>

	* main/features.c: Fixes a segfault caused by referencing null
	  frames introduced in r338623

2011-10-24 21:01 +0000 [r342112]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Fix use of OBJ_KEY in Queue application. To use
	  the new OBJ_KEY flag, the container hash and compare callback
	  functions must be updated to support OBJ_KEY. Otherwise, bad
	  things happen. (issue ASTERISK-14769)

2011-10-24 20:01 +0000 [r342063]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now
	  include fromuser of related peer. This behavior matches up more
	  closely with the way invite/register/etc are handled. This patch
	  also modifies some adjacent code for code style compliance.
	  Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
	  Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
	  by Jeremy Kister (license #6232) ........ Merged revisions 342061
	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
	  Merged revisions 342062 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-24 07:40 +0000 [r341923-342018]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, apps/app_queue.c: queues container needs locking when using
	  the OBJ_NOLOCK flag ........ Merged revisions 342017 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_queue.c: Remove some ref leaks and a return without
	  unlock. There some resource leaks introduced in asterisk 10 make
	  sure that locks are not held on return and we release ref's held.
	  ........ Merged revisions 341972 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
	  patch is related to work on RB1538

2011-10-22 12:03 +0000 [r341869]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, /: Merged revisions 341313 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed,
	  19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3
	  lines fix issue on channel numbering (calls could have same
	  channel number on heavy loaded system) ........ ................

2011-10-21 16:42 +0000 [r341808-341811]  Matthew Nicholson <mnicholson@digium.com>

	* /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395
	  ........ Merged revisions 341809 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341810 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, pbx/pbx_lua.c: don't limit the length of app and function
	  arguments ASTERISK-18395 ........ Merged revisions 341806 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341807 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-21 09:16 +0000 [r341769]  Gregory Nietsky <gregory@distrotech.co.za>

	* res/res_fax.c: White space fixes in res_fax

2011-10-20 22:03 +0000 [r341719]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c, res/res_agi.c, include/asterisk/features.h:
	  Fix AGI exec Park to honor the Park application parameters. The
	  fix for ASTERISK-12715 and ASTERISK-12685 added a check for the
	  Park application because the channel needed to be masqueraded to
	  prevent a crash. Since the Park application now always
	  masquerades the channel into the parking lot, the special check
	  is no longer needed. The fix also resulted in AGI exec Park
	  attempting to double park the call and not honor the Park
	  application parameters. * Removed no longer necessary call to
	  ast_masq_park_call() by AGI exec for the Park application.
	  (Reverts -r146923) * Fix Park application to only return 0 or -1.
	  The AGI exec Park was causing broken pipe error messages because
	  the Park application returned 1 on successful park. (closes issue
	  ASTERISK-18737) ........ Merged revisions 341717 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341718 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-20 21:28 +0000 [r341666-341713]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, funcs/func_callerid.c: Fixed typo from previous commit
	  ........ Merged revisions 341704 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341707 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, funcs/func_callerid.c: Updated documentation for the optional
	  CID parameter with CALLERID ........ Merged revisions 341664 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341665 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-20 18:27 +0000 [r341583-341624]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, configs/queues.conf.sample: Merged revisions 341599 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20
	  Oct 2011) | 8 lines add documentation for check_state_unknown in
	  configs/queues.conf.sample app_queue allows calls to members in a
	  "Unknown" state to be treated as available setting
	  check_state_unknown = yes will cause app_queue to query the
	  channel driver to better determine the state this only applies to
	  queues with ringinuse or ignorebusy set appropriately. ........

	* /, CHANGES, apps/app_queue.c: Merged revisions 341580 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20
	  Oct 2011) | 15 lines Add option to check state when state is
	  unknown r341486 reverts r325483 this is a rework of the patch.
	  optimize to minimize load. add option check_state_unknown to
	  control whether a member with unknown device state is checked
	  there is a small % chance that calls will be sent to the member
	  when they on a call. app_queue will see a device with unknown
	  state as available and does not try verify the state without this
	  option enabled. Review: https://reviewboard.asterisk.org/r/1535/
	  ........

2011-10-20 15:17 +0000 [r341533]  Terry Wilson <twilson@digium.com>

	* /, include/asterisk/strings.h: Clean up ast_check_digits The code
	  was originally copied from the is_int() function in the AEL code.
	  wdoekes pointed out that the function should take a const char*
	  and that their was an unneeded variable. This is now fixed.
	  ........ Merged revisions 341529 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341530 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-19 21:24 +0000 [r341487]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 341486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct
	  2011) | 18 lines Fix a performance regression introduced in
	  r325483. The regression was caused by a call to
	  ast_parse_device_state() in app_queue's ring_entry() function.
	  The ast_parse_device_state() function eventually calls
	  ast_channel_get_full() with a channel name prefix which causes it
	  to walk the channel list causing massive lock contention and slow
	  downs. This patch fixes the regression by removing the call to
	  ast_parase_device_state() which should be unnecessary. Queue
	  member device state should be maintained by device state events.
	  Some users have seen instances where busy agents were called when
	  they shouldn't have, which is the reason the call to
	  ast_parse_device_state() was added. That change appears to have
	  resolved that issue but also causes this performance regression.
	  There may still be issues with queue member status, and if so,
	  alternative methods should be investigated to resolve them.
	  AST-695 ........

2011-10-19 19:02 +0000 [r341437]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google
	  has recently make some changes (again) to their protocol. Rather
	  then patching asterisk to flip between the two different methods,
	  we now allow both. Lets hope this keeps Google Voice happy for a
	  while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
	  Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
	  6311) ........ Merged revisions 341435 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341436 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-19 07:45 +0000 [r341381]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, include/asterisk/strings.h: Don't use
	  is_int() since it doesn't link well on all platforms Just create
	  an normal API function in strings.h that does the same thing just
	  to be safe. ASTERISK-17146 ........ Merged revisions 341379 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341380 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-19 07:27 +0000 [r341378]  Stefan Schmidt <sst@sil.at>

	* /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
	  when Asterisk has not yet received a Contact URI from a UAS
	  ........ Merged revisions 341366 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341377 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-18 23:45 +0000 [r341316]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't resolve numeric hosts or contact
	  unresolved hosts If a SIP dial string contains a numeric hostname
	  that is not a peer name, don't try to resolve it as it is
	  unlikely that someone really means Dial(SIP/0.0.4.26) when
	  Dial(SIP/1050) is called. Also, make sure that create_addr
	  returns -1 if an address isn't resolved so that we don't attempt
	  to send SIP requests to an address that doesn't resolve. (closes
	  issue ASTERISK-17146, ASTERISK-17716) Review:
	  https://reviewboard.asterisk.org/r/1532/ ........ Merged
	  revisions 341314 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341315 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-18 21:15 +0000 [r341256]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/chan_sip.c, main/features.c, channels/chan_iax2.c,
	  channels/sip/include/sip.h, channels/chan_mgcp.c,
	  include/asterisk/features.h: More parking issues. * Fix potential
	  deadlocks in SIP and IAX blind transfer to parking. * Fix SIP,
	  IAX, DAHDI analog, and MGCP channel drivers to respect the
	  parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
	  parameter). Created ast_park_call_exten() and
	  ast_masq_park_call_exten() to maintian API compatibility. * Made
	  masq_park_call() handle a failed ast_channel_masquerade() setup.
	  * Reduced excessive struct parkeduser.peername[] size. ........
	  Merged revisions 341254 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341255 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-17 17:58 +0000 [r341198]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
	  include of asterisk/md5.h in pbx_realtime.c . A commit needed to
	  test the commit message. Merged-From:
	  http://svn.asterisk.org/svn/asterisk/branches/1.8@341074
	  Merged-From:
	  http://svn.asterisk.org/svn/asterisk/branches/10@341148

2011-10-17 17:38 +0000 [r341191]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Initialize variables before calling
	  parse_uri If parse_uri was called with an empty URI, some
	  pointers would be modified and an invalid read could result. This
	  patch avoids calling parse_uri with an empty contact uri when
	  parsing REGISTER requests. AST-2011-012 (closes issue
	  ASTERISK-18668) ........ Merged revisions 341189 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341190 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-17 16:39 +0000 [r341126-341147]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, tests/test_format_api.c: Set 'core' support level for
	  test_format_api.c ........ Merged revisions 341146 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_voicemail.c: Multiple revisions 341108,341112
	  ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon,
	  17 Oct 2011) | 2 lines Voicemail compiler flags are 'core'
	  support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400
	  (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged
	  revisions 341108,341112 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341122 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-17 16:18 +0000 [r341096]  Jason Parker <jparker@digium.com>

	* /, CHANGES: Add information about limitations of new codec
	  support in channel drivers. (issue ASTERISK-18680) ........
	  Merged revisions 341094 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-17 15:45 +0000 [r341090]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't try to remove peers without IPs
	  from peers_by_ip (closes issue ASTERISK-18696) ........ Merged
	  revisions 341088 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341089 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-14 21:37 +0000 [r341024]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change
	  the internal name of the menuselect options that are used to
	  control whether modules are embedded or not; using just the bare
	  category name led to accidentally enabling these options when
	  users used the wrong "--enable" operation on the menuselect
	  command line. Now the internal option names are prefixed with
	  "EMBED_", so they won't be the same as the name of the category
	  containing the modules they control the embedding of. ........
	  Merged revisions 341022 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 341023 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-14 21:15 +0000 [r340973]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Fix simple switch to not progress a call
	  when call already progressed. If a simple switch was started on a
	  device and then a specific call made (such as redial or speed
	  dial), on timeout of the simple switch the call would be
	  attempted again. This patch only allows the simple switch to make
	  a call if the substate is still in the collecting digits mode.
	  Also added small debug message to dialAndAactivate sub. Tested by
	  snuff and myself.

2011-10-14 20:51 +0000 [r340972]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
	  340971 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500
	  (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) |
	  8 lines Quiet RTCP Receiver Reports during fax transmission RTCP
	  is now disabled for "inactive" RTP audio streams during SIP T.38
	  sessions. The ability to disable RTCP streams in res_rtp_asterisk
	  was missing, so this code was added to support the bug fix.
	  (closes issue ASTERISK-18400) ........ ................

2011-10-14 18:38 +0000 [r340932]  Jonathan Rose <jrose@digium.com>

	* utils/utils.xml, /, funcs/func_jitterbuffer.c: Some additional
	  module documentation changes for 10 for the menuselect change.
	  (issue ASTERISK-18268) ........ Merged revisions 340931 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-14 16:45 +0000 [r340880]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /: Avoid unnecessary WARNING message Add
	  AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
	  displaying a WARNING message. (closes issue ASTERISK-18610) Patch
	  by: Kristijan_Vrban ........ Merged revisions 340878 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340879 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-13 23:08 +0000 [r340811-340813]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Fix DTMF blind transfer continuing to execute
	  dialplan after transfer. Party A calls Party B. Party A DTMF
	  blind transfers Party B to Party C. Party A channel continues to
	  execute dialplan. * Fixed the return value of
	  builtin_blindtransfer() to return the correct value after a
	  transfer so the dialplan will not keep executing. * Removed
	  unnecessary connected line update that did not really do
	  anything. * Made access to GOTO_ON_BLINDXFR thread safe in
	  check_goto_on_transfer(). * Fixed leak of xferchan for failure
	  cases in check_goto_on_transfer(). * Updated debug messages in
	  builtin_blindtransfer() and check_goto_on_transfer(). (closes
	  issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
	  ........ Merged revisions 340809 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340810 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /: Update 10 merged property.

	* /: Restore branch 10 merge properties.

2011-10-13 08:53 +0000 [r340771]  Gregory Nietsky <gregory@distrotech.co.za>

	* /: Merged revisions 339463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
	  9 lines Only change the capabilities on the gateway when the
	  session is been destroyed there is still a race condition that
	  ends in a segfault. if the caps are changed the logic in
	  res_fax_spandsp will run T30 code not gateway code to end the
	  session. this has been experienced on a "slower" under spec
	  system. ........

2011-10-13 07:05 +0000 [r340720]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Merged revisions 340718 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340718 | schmidts | 2011-10-13 06:59:50 +0000
	  (Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13
	  Oct 2011) | 3 lines storing the route-set also on a 181 response
	  not only on 180,182 or 183. ........ ................

2011-10-13 07:02 +0000 [r340665-340719]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Initialize ast_sockaddr before calling
	  ast_sockaddr_resolve Avoid possible jump based on unitialized
	  value ........ Merged revisions 340715 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340716 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, res/res_config_sqlite.c: Don't skip the query field on a
	  realtime multi query There is no documented reason to not add the
	  query field to the varlist returned by a realtime multi query,
	  despite the config category being set to its value. Of course,
	  there is no documentation that the category should be set to the
	  value either. There is lots of no documentation when it comes to
	  realtime. But, other engines do not skip this field so I am
	  forcing this backend to follow the convention, because not doing
	  so is very silly. ........ Merged revisions 340662 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340663 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-12 21:28 +0000 [r340626]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Merged revisions 340577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340577 | schmidts | 2011-10-12 20:33:37 +0000
	  (Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12
	  Okt 2011) | 3 lines Store route-set from provisional SIP
	  responses so early-dialog requests can be routed properly
	  ........ ................

2011-10-12 21:02 +0000 [r340579]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 340578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340578 | twilson | 2011-10-12 13:57:19 -0700
	  (Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011)
	  | 9 lines Update SIP realtime fullcontact regardless of caching
	  We should update the fullcontact field in the realtime table
	  whether or not rtcachefriends is set. There is no reason to treat
	  a non-cached realtime entity differently than a cached in this
	  regard. (closes issue ASTERISK-18446) Reported by: wdoekes
	  ........ ................

2011-10-12 20:09 +0000 [r340472-340524]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Initialize the PRI channel alarms
	  properly on startup. The PRI channel alarms were initialized with
	  an inverted sense. (closes issue ASTERISK-18710) Reported by:
	  Tzafrir Cohen ........ Merged revisions 340522 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340523 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_meetme.c: Update MeetMe p and X option documentation
	  when interacting with the s option. ASTERISK-12175 changed the p
	  and X options to not interfere with the s option when they are
	  used together. It makes more sense for the s option to have
	  priority for the DTMF '*' key since it cannot change its
	  activation code. Otherwise, you could not use option s with the p
	  or X options. JIRA AST-671 ........ Merged revisions 340470 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340471 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-12 16:29 +0000 [r340420]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was
	  added (closes issue ASTERISK-18612) Reported by: Tim Osman
	  ........ Merged revisions 340418 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340419 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-11 21:06 +0000 [r340318-340367]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
	  Add protection for SS7 channel allocation and better glare
	  handling. * Added a CLI "ss7 show channels" command that might
	  prove useful for future debugging. * Made the incoming SS7
	  channel event check and gripe message uniform. * Made sure that
	  the DNID string for an incoming call is always initialized.
	  (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
	  Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
	  patch uploaded by rmudgett ........ Merged revisions 340365 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340366 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some
	  potential deadlocks pointed out by helgrind. * Fixed deadlock
	  potential calling dialog_unlink_all() in __sip_autodestruct().
	  Found by helgrind. * Fixed deadlock potential in
	  handle_request_invite() after calling sip_new(). Found by
	  helgrind. * The sip_new() function now returns with the created
	  channel already locked. * Removed the dead code that starts a PBX
	  in in sip_new(). No sip_new() callers caused that code to be
	  executed and it was a bad thing to do anyway. * Removed unused
	  parameters and return value from dialog_unlink_all(). * Made
	  dialog_unlink_all() and __sip_autodestruct() safely obtain the
	  owner and private channel locks without a deadlock avoidance
	  loop. ........ Merged revisions 340284 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340310 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-11 19:06 +0000 [r340283]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update
	  SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from
	  which the code was originally taken. It has a slightly better
	  code, and a better phrased license (simple 3-clause BSD). *
	  main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
	  * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
	  6234. * Removed unused include of asterisk/sha1.h from
	  main/channels.c Review: https://reviewboard.asterisk.org/r/1503/
	  Merge-From:
	  http://svn.asterisk.org/svn/asterisk/branches/1.8@340263
	  Merge-From:
	  http://svn.asterisk.org/svn/asterisk/branches/10@340280

2011-10-11 18:57 +0000 [r340282]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, /, include/asterisk/manager.h: Convert registered
	  AMI actions to ao2 objects. * Fixed race between calling an AMI
	  action callback and unregistering that action. Refixes
	  ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
	  memory leak if an AMI action failed to get registered because is
	  already was registered. Part of the ao2 conversion. * Fixed AMI
	  ListCommands action not walking the actions list with a lock
	  held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
	  stack usage. * Fix AMI Originate action Variable header requiring
	  a space after the header colon. Reported by Yaroslav Panych on
	  the asterisk-dev list. * Increased the number of listed variables
	  allowed per AMI Originate action Variable header to 64. * Fixed
	  AMI GetConfigJSON action output format. * Fixed usage of res
	  contents outside of scope in append_channel_vars(). * Fixed
	  inconsistency of config file channelvars option. The values no
	  longer accumulate with every channelvars option in the config
	  file. Only the last value is kept to be consistent with the CLI
	  "manager show settings" command. (closes issue ASTERISK-18479)
	  Reported by: Jaco Kroon ........ Merged revisions 340279 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 340281 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-10-10 23:10 +0000 [r340221-340224]  Terry Wilson <twilson@digium.com>

	* UPGRADE.txt, main/db.c: Return error when no rows are deleted for
	  AMI DBDelTree (closes issue AST-654)

	* /, main/db.c: Merged revisions 340222 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011)
	  | 8 lines On astdb conversion, also warn about permissions
	  requirements The user running Asterisk must have permission to
	  the directory the Asterisk database resides in since SQLite 3
	  needs to be able to create a journal file. (closes issue
	  ASTERISK-18174) ........

	* utils/Makefile, utils/utils.xml, /, UPGRADE.txt,
	  utils/astdb2bdb.c (added): Merged revisions 340219-340220 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10
	  Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to
	  SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10
	  they can use the astdb2bdb utility to convert the database back
	  to the Berkeley format that Asterisk 1.8 uses. Review:
	  https://reviewboard.asterisk.org/r/1502/ ........ r340220 |
	  twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
	  Add a missing file for the astdb2bdb conversion utility ........

2011-10-10 20:39 +0000 [r340166]  Matthew Jordan <mjordan@digium.com>

	* /, channels/chan_sip.c: Merged revisions 340165 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340165 | mjordan | 2011-10-10 15:30:18 -0500
	  (Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011)
	  | 13 lines Updated chan_sip to place calls on hold if SDP address
	  in INVITE is ANY This patch fixes the case where an INVITE is
	  received with c=0.0.0.0 or ::. In this case, the call should be
	  placed on hold. Previously, we checked for the address being
	  null; this patch keeps that behavior but also checks for the ANY
	  IP addresses. Review: https://reviewboard.asterisk.org/r/1504/
	  (closes issue ASTERISK-18086) Reported by: James Bottomley Tested
	  by: Matt Jordan ........ ................

2011-10-10 14:16 +0000 [r340110]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c,
	  include/asterisk/module.h, res/res_agi.c,
	  include/asterisk/xmldoc.h, doc/appdocsxml.dtd, main/loader.c,
	  main/xmldoc.c: Merged revisions 340109 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500
	  (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct
	  2011) | 11 lines Load the proper XML documentation when multiple
	  modules document the same application. This patch adds an
	  optional "module" attribute to the XML documentation spec that
	  allows the documentation processor to match apps with identical
	  names from different modules to their documentation. This patch
	  also fixes a number of bugs with the documentation processor and
	  should make it a little more efficient. Support for multiple
	  languages has also been properly implemented. ASTERISK-18130
	  Review: https://reviewboard.asterisk.org/r/1485/ ........
	  ................

2011-10-10 00:57 +0000 [r339993-340071]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Add skinny version 17 protocol support.
	  Added some data to skinny packet structures to make compatible
	  with v17. Added protocolversion to device, set on registration
	  based on the version provided by device. v17 includes some
	  increased ip space for ip6. This patch increases ip space in the
	  packets but still only uses ip4. Some packet structures
	  duplicated (ip4 and ip6 types). ip4 type used unless version is
	  greater or equal to 17. Tested by snuff and myself on 7961 with
	  recent 8.5 firmware. Also tested compatible with old 7960 and
	  older 30VIPs.

	* channels/chan_skinny.c: Increase SKINNY_MAX_PACKET and add some
	  logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some
	  messages in v17 that are greater than the old 1000 bytes. Also
	  add some useful logging regarding packet and session handling. A
	  device (with protocol v17) was sending a packet with length
	  greater than 1000 which resulted in the TCP session being
	  destroyed and registration being retryed.

	* /, channels/chan_skinny.c: Merged revisions 340031 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011)
	  | 8 lines Return -1 to skinny_session if register rejected. If
	  device registration is rejected, return -1 so that the session is
	  destroyed immediately. Previously, a segfault would occur on a
	  graceful shutdown if a register is rejected and the
	  skinny_session has not yet timed out. ........

	* /, channels/chan_skinny.c: Merged revisions 339992 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011)
	  | 9 lines Remove log message on traverse session list. On
	  destroying a session, a list of sessions is traversed to find the
	  matching session. For each session not matching, skinny
	  erroneously logged that the session was not matched. While
	  technically correct the message was misleading, and tended to
	  indicate errors that were not there. ........

2011-10-09 01:19 +0000 [r339832-339947]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c, /: Merged revisions 339942 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339942 | igorg | 2011-10-09 08:18:02 +0700
	  (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) |
	  6 lines Fix compilation issue, caused by missed session structure
	  (closes issue ASTERISK-18694) Reported by: alex70 ........
	  ................

	* channels/chan_unistim.c, /: Merged revisions 339885 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339885 | igorg | 2011-10-08 22:46:27 +0700
	  (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) |
	  7 lines Fix segfault in Unistim channel (closes issue
	  ASTERISK-18638) Reported by: jonnt ........ ................

	* channels/chan_unistim.c, /: Merged revisions 339831 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339831 | igorg | 2011-10-08 22:01:35 +0700
	  (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) |
	  8 lines Fix char array cast as short array in send_client()
	  function (for ARM platform) (closes issue ASTERISK-17314)
	  Reported by: jjoshua ........ ................

2011-10-07 19:37 +0000 [r339721-339778]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_url.c: Merged revisions 339777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500
	  (Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011)
	  | 5 lines Initialize option flags for SendURL application.
	  (closes issue ASTERISK-18574) Reported by: marcelloceschia
	  ........ ................

	* /: Recorded merge of revisions 339681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011)
	  | 10 lines Fixed segfault on core stop gracefully. There was an
	  issue that the cap and confcap pointers for each line and device
	  were being memcpy'd so they all pointed to the same
	  ast_format_cap. On destroying, a segfault occured on the second
	  call to the same struct. skinny reload now works again as well.
	  Tested by snuff (in trunk) and myself. ........

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  autoconf/ast_ext_lib.m4: Merged revisions 339720 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500
	  (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011)
	  | 20 lines Fix regression in configure script for libpri
	  capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE
	  option to fix BRI PTMP TE layer 2 persistence issues with some
	  telcos. ASTERISK-18535 attempted to fix the unexpected
	  requirement that libpri *must* have that feature to work with
	  Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
	  optional features required. Unfortunately, I thought
	  AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri
	  and deleted those lines for libpri. The result was the
	  HAVE_PRI_xxx defines that control the ability to use optional
	  libpri features were also deleted. * Created
	  AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
	  features in a library that the source code could take advantage
	  of if the code supports the feature. (closes issue
	  ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........
	  ................

2011-10-06 20:18 +0000 [r339680]  Damien Wedhorn <voip@facts.com.au>

	* channels/chan_skinny.c: Fixed segfault on core stop gracefully.
	  There was an issue that the cap and confcap pointers for each
	  line and device were being memcpy'd so they all pointed to the
	  same ast_format_cap. On destroying, a segfault occured on the
	  second call to the same struct. skinny reload now works again as
	  well. Tested by snuff and myself.

2011-10-06 17:54 +0000 [r339627]  Richard Mudgett <rmudgett@digium.com>

	* main/udptl.c, /, channels/chan_sip.c: Merged revisions 339626 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500
	  (Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011)
	  | 18 lines Fix debugging messages generated by 'udptl debug'. *
	  Makes chan_sip set the tag to the channel name. * Fixes received
	  debug message sequence number. * Removed tx/rx debug message type
	  since it was hard coded to 0. * Made udptl.c logged message
	  header consistent if possible: "UDPTL (%s): ". * Removed unused
	  rx_expected_seq_no from struct ast_udptl. (closes issue
	  ASTERISK-18401) Reported by: Kevin P. Fleming Patches:
	  jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Matthew Nicholson ........ ................

2011-10-06 13:43 +0000 [r339587]  Leif Madsen <leif@leifmadsen.com>

	* build_tools/prep_tarball: Merged revisions 339586 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500
	  (Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011)
	  | 8 lines Update prep_tarball script to download pre-exported
	  documentation. I've updated the prep_tarball script to now
	  download the pre-exported documentation from the Asterisk wiki.
	  This will give us more control over what is being included in the
	  tarball releases, and will make both the PDF and HTML exported
	  documentation look much better (especially when viewing from a
	  console). (Closes issue ASTERISK-18677) ........ ................

2011-10-05 17:02 +0000 [r339510-339513]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, /: Merged revisions 339512 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500
	  (Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05
	  Oct 2011) | 1 line Fix Dial F option notes formatting. ........
	  ................

	* main/manager.c, /: Merged revisions 339508 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500
	  (Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011)
	  | 7 lines Add missing documentation of required AMI action
	  Challenge AuthType header. (closes issue ASTERISK-18554) Reported
	  by: Vlad Povorozniuc Patches:
	  __20110919-manager-challenge-docs.patch.txt (license #4999) patch
	  uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05
	  11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI
	  action Challenge. ........ ................

2011-10-05 16:35 +0000 [r339509]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 339507 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500
	  (Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct
	  2011) | 3 lines The app name in the documentation must match what
	  we register the application as. ........ ................

2011-10-05 06:50 +0000 [r339464-339465]  Gregory Nietsky <gregory@distrotech.co.za>

	* res/res_fax.c, include/asterisk/res_fax.h, CHANGES: Add generic
	  faxdetect framehook to res_fax Added func
	  FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan
	  faxdetect allowing more flexibility. as soon as a fax tone is
	  detected the framehook is removed. there is a penalty involved in
	  running this framehook on non G711 channels as they will be
	  transcoded. CNG tone is suppresed using the SQUELCH flag to allow
	  WaitForNoise to be run on the channel to detect Voice. (Closes
	  issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew
	  Nicholson, Kevin Fleming Review:
	  https://reviewboard.asterisk.org/r/1116/

	* /, res/res_fax.c: Merged revisions 339463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
	  9 lines Only change the capabilities on the gateway when the
	  session is been destroyed there is still a race condition that
	  ends in a segfault. if the caps are changed the logic in
	  res_fax_spandsp will run T30 code not gateway code to end the
	  session. this has been experienced on a "slower" under spec
	  system. ........

2011-10-04 22:59 +0000 [r339408]  Richard Mudgett <rmudgett@digium.com>

	* Makefile, /: Merged revisions 339407 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500
	  (Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011)
	  | 8 lines Make always create the MOH directory
	  (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
	  by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
	  #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
	  Keuter ........ ................

2011-10-04 19:51 +0000 [r339315-339354]  Jonathan Rose <jrose@digium.com>

	* /, main/say.c: Merged revisions 339353 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339353 | jrose | 2011-10-04 14:44:02 -0500
	  (Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) |
	  12 lines Removes improper use of sound 'and' in German language
	  mode from application saynumber Asterisk would say 'Five hundert
	  und sechs und zwanzig' instead of 'Five hundert sechs und
	  zwanzig'... which is both weird sounding and wrong. This patch
	  makes sure Asterisk will only say the 'and' word between the
	  single digit and double digit places. (closes issue
	  ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
	  upstream_germand_no_and.diff (License #5402) uploaded by Lionel
	  Elie Mamane ........ ................

	* /, res/res_jabber.c: Merged revisions 339298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339298 | jrose | 2011-10-04 09:09:50 -0500
	  (Tue, 04 Oct 2011) | 19 lines Merged revisions 339297 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) |
	  13 lines Reverting revision 333265 due to component connection
	  problems it introduces. I'm going to attempt some generic
	  res_jabber cleanup and come up with a new fix for this problem,
	  but first it seems prudent to remove this rather broad attempt to
	  fix it and instead approach this problem either from the same
	  angle but looking only at canceling (or possibly rescheduling)
	  the send when we absolutely know it will cause a segfault or, if
	  that can't be easily accomplished, strictly from the devstate
	  side of things. Also, I'm pretty sure a lot of the code in
	  res_jabber isn't thread safe. (issue ASTERISK-18626) (issue
	  ASTERISK-18078) ........ ................

2011-10-04 12:27 +0000 [r339262]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/memheap.c: Merged revisions 339245 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339245 | may | 2011-10-04 15:49:49 +0400 (Tue,
	  04 Oct 2011) | 9 lines Merged revisions 339244 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2
	  lines fix forget declaration in previous change ........
	  ................

2011-10-04 09:43 +0000 [r339206]  Olle Johansson <oej@edvina.net>

	* main/manager.c, CHANGES: Generate error message when AMI action
	  originate extension doesn't exist Review:
	  https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new
	  feature? No responses on Asterisk-dev so I'm committing to trunk
	  only.

2011-10-03 20:13 +0000 [r339146-339149]  Leif Madsen <leif@leifmadsen.com>

	* channels/chan_sip.c: Merged revisions 339148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339148 | lmadsen | 2011-10-03 15:13:16 -0500
	  (Mon, 03 Oct 2011) | 14 lines Merged revisions 339147 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011)
	  | 6 lines Remove duplicated Maxforwards line in AMI output.
	  (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny
	  Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by
	  Jacek Konieczny ........ ................

	* apps/app_dial.c: Merged revisions 339145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339145 | lmadsen | 2011-10-03 14:55:15 -0500
	  (Mon, 03 Oct 2011) | 13 lines Merged revisions 339144 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011)
	  | 6 lines Make documentation for Dial() options 'F' and 'F()'
	  more clear. (Closes issue ASTERISK-18646) Reported by: Physis
	  Heckman Tested by: Richard Mudgett ........ ................

2011-10-03 19:16 +0000 [r339091]  Alexandr Anikin <may@telecom-service.ru>

	* /, addons/ooh323c/src/memheap.c: Merged revisions 339089 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon,
	  03 Oct 2011) | 10 lines Merged revisions 339087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4
	  lines destroy memheap mutex properly before memheap deleted (fix
	  memory leak occured after r304950 changes with DEBUG_THREAD
	  compile option) ........ ................

2011-10-03 18:58 +0000 [r339090]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, main/file.c: Merged revisions 339088 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r339088 | twilson | 2011-10-03 11:44:27 -0700
	  (Mon, 03 Oct 2011) | 17 lines Merged revisions 339086 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011)
	  | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more
	  places After the change in r336294, the new
	  AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite
	  happens. If we receive a re-invite from a device the
	  waitstream_core was not aware of the new control frame and would
	  drop the call. (closes issue ASTERISK-18610) Reported by:
	  Kristijan_Vrban ........ ................

2011-10-03 15:55 +0000 [r339021-339046]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 339045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct
	  2011) | 4 lines Ported ast_fax_caps_to_str() to 10, not sure why
	  it wasn't already here. This function prints a list of caps
	  instead of a hex bitfield. ........

	* /, res/res_fax.c: Merged revisions 339043 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct
	  2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right
	  after we set it. ........

	* /, res/res_fax.c: Merged revisions 339011 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct
	  2011) | 2 lines properly remove the AST_FAX_TECH_GATEWAY flag
	  (instead of setting all of the other flags) ........

2011-10-03 14:40 +0000 [r338905-338998]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, CHANGES: Merged revisions 338997 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) |
	  1 line Documentation noting the extension of CHANNEL() for
	  chan_ooh323 ........

	* addons/chan_ooh323.c, /, funcs/func_channel.c: Merged revisions
	  338995 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) |
	  6 lines Remove the channel function OOH323() and place its
	  options into CHANNEL() channel drivers should not have there own
	  dialplan functions. ........

	* /, res/res_fax.c: Merged revisions 338950 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) |
	  14 lines Fixup a race condition in res_fax.c where
	  FAXOPT(gateway)=no will turn off the gateway but the framehook is
	  not destroyed. this problem happens when a gateway is attempted
	  in the dialplan and the device is not available i may want to do
	  fax to mail in the server it will not be allowed. instead of
	  checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts
	  338904 Fix some white space. ........

	* /, res/res_fax.c: Merged revisions 338904 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) |
	  8 lines Remove T38 Gateway capability when detaching framehook.
	  SET(FAXOPT(gateway)=no) does not remove the capability when
	  detaching the framehook. small patch to fix this problem.
	  ........

2011-10-01 01:56 +0000 [r338855]  TransNexus OSP Development <support@transnexus.com>

	* configure: Update "configure" based on r338139.

2011-09-30 22:08 +0000 [r338802]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 338801 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500
	  (Fri, 30 Sep 2011) | 19 lines Merged revisions 338800 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011)
	  | 12 lines Fix segfault in analog_ss_thread() not checking
	  ast_read() for NULL. NOTE: The problem was reported against
	  v1.6.2. It is unlikely to ever happen on v1.8 and above since
	  chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
	  version in sig_analog.c has largely replaced it. (closes issue
	  ASTERISK-18648) Reported by: Stephan Bosch Patches:
	  jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Stephan Bosch ........ ................

2011-09-30 19:25 +0000 [r338755]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting changes only --Denna och
	  nedanstående rader kommer inte med i loggmeddelandet-- M
	  channels/chan_sip.c

2011-09-30 18:59 +0000 [r338720]  Jonathan Rose <jrose@digium.com>

	* /, configs/queues.conf.sample: Merged revisions 338719 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338719 | jrose | 2011-09-30 13:55:27 -0500
	  (Fri, 30 Sep 2011) | 9 lines Merged revisions 338718 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep
	  2011) | 1 line Adds documentation for QueueMemberStatus event
	  generation ........ ................

2011-09-30 16:40 +0000 [r338665]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/chan_sip.c: Fix formatting of AMI header for SIP show
	  peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
	  issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
	  asterisk-sipshowpeer_response_end.patch (license #6298) patch
	  uploaded by Jacek Konieczny ........ Merged revisions 338663 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
	  revisions 338664 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

2011-09-30 13:21 +0000 [r338623]  Olle Johansson <oej@edvina.net>

	* main/features.c: Preserve DTMF length in main/features.c Review:
	  https://reviewboard.asterisk.org/r/1463/ A small part of much
	  larger work with DTMF duration in Asterisk, funded by IPvision AS
	  in Denmark. Thanks to irroot for the review!

2011-09-29 21:16 +0000 [r338557]  Paul Belanger <paul.belanger@polybeacon.com>

	* tests/test_security_events.c, /, tests/test_locale.c,
	  tests/test_logger.c, tests/test_dlinklists.c,
	  tests/test_linkedlists.c, tests/test_amihooks.c: Merged revisions
	  338556 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400
	  (Thu, 29 Sep 2011) | 9 lines Merged revisions 338555 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu,
	  29 Sep 2011) | 2 lines Test modules should depend on the
	  TEST_FRAMEWORK flag ........ ................

2011-09-29 20:55 +0000 [r338553]  Jason Parker <jparker@digium.com>

	* /, tests/test_db.c, tests/test_netsock2.c: Merged revisions
	  338552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338552 | qwell | 2011-09-29 15:54:55 -0500
	  (Thu, 29 Sep 2011) | 9 lines Merged revisions 338551 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep
	  2011) | 1 line Test modules have a support level of core.
	  ........ ................

2011-09-29 12:22 +0000 [r338435]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
	  revisions 338417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338417 | irroot | 2011-09-29 14:16:42 +0200
	  (Thu, 29 Sep 2011) | 19 lines Merged revisions 338416 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) |
	  12 lines The rtptimeout setting is ignored on a per peer basis.
	  Not only is the rtptimeout ignored in some cases but rtpkeepalive
	  and rtpholdtimeout is affected. this commit also removes
	  rtptimeout/rtpholdtimeout on text rtp. (closes issue
	  ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
	  ........ ................

2011-09-29 12:03 +0000 [r338377-338415]  Olle Johansson <oej@edvina.net>

	* cdr/cdr_pgsql.c, CHANGES: Add CLI command "cdr show pgsql status"
	  based on "cdr mysql status" Review:
	  https://reviewboard.asterisk.org/r/923/ Thanks all for the code
	  reviews and feedback.

	* res/res_agi.c: Just formatting.

2011-09-28 22:38 +0000 [r338284-338324]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 338323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500
	  (Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011)
	  | 5 lines Make duplicate call ptr warning message more helpful. *
	  Adds the value of the call ptr to the duplicate call ptr message
	  to help trace why there is a duplicate call ptr. ........
	  ................

	* include/asterisk/logger.h, /: Merged revisions 338253 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500
	  (Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011)
	  | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE
	  declaration. (closes issue ASTERISK-17973) Reported by: Luke H
	  Patches: logger_h.patch (license #6278) patch uploaded by Luke H
	  ........ ................

2011-09-28 20:55 +0000 [r338229]  Jason Parker <jparker@digium.com>

	* build_tools/cflags.xml, channels/chan_usbradio.c,
	  build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /,
	  build_tools/embed_modules.xml, tests/test_db.c,
	  tests/test_netsock2.c: Merged revisions 338228 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338228 | qwell | 2011-09-28 15:54:35 -0500
	  (Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep
	  2011) | 1 line Add support levels to non-module sections of
	  menuselect (cflags, utils, etc). ........ ................

2011-09-28 20:28 +0000 [r338226]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 338225 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500
	  (Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011)
	  | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7
	  not present. (closes issue ASTERISK-18357) Reported by: Matthew
	  Nicholson ........ ................

2011-09-28 17:00 +0000 [r338187-338188]  Terry Wilson <twilson@digium.com>

	* CHANGES: Update CHANGES to reflect autopausebusy not being in
	  Asterisk 10

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add
	  autopausebusy and autopauseunavail queue options Make it possible
	  to autopause on a busy or unavailable response from a device.
	  (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches:
	  autopausebusy.txt by twilson Review:
	  https://reviewboard.asterisk.org/r/1399/

2011-09-28 07:30 +0000 [r338136-338139]  TransNexus OSP Development <support@transnexus.com>

	* configure.ac: Updated for checking OSP Toolkit version 4.0.0.

	* apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.

2011-09-27 20:15 +0000 [r338086]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/app_macro.c: Merged revisions 338085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400
	  (Tue, 27 Sep 2011) | 9 lines Merged revisions 338084 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue,
	  27 Sep 2011) | 2 lines Upgrade app_macro to core ........
	  ................

2011-09-27 12:45 +0000 [r338042]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Whitespace (red blobs) fixes

2011-09-26 19:40 +0000 [r337975]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
	  include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
	  include/asterisk/channel.h, main/cel.c, main/manager.c,
	  funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
	  main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_custom.c,
	  cdr/cdr_manager.c, apps/app_voicemail.c: Merged revisions 337974
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500
	  (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
	  | 30 lines Fix deadlock when using dummy channels. Dummy channels
	  created by ast_dummy_channel_alloc() should be destoyed by
	  ast_channel_unref(). Using ast_channel_release() needlessly grabs
	  the channel container lock and can cause a deadlock as a result.
	  * Analyzed use of ast_dummy_channel_alloc() and made use
	  ast_channel_unref() when done with the dummy channel. (Primary
	  reason for the reported deadlock.) * Made
	  app_dial.c:dial_exec_full() not call ast_call() holding any
	  channel locks. Chan_local could not perform deadlock avoidance
	  correctly. (Potential deadlock exposed by this issue. Secondary
	  reason for the reported deadlock since the held lock was part of
	  the deadlock chain.) * Fixed some uses of
	  ast_dummy_channel_alloc() not checking the returned channel
	  pointer for failure. * Fixed some potential chan=NULL pointer
	  usage in func_odbc.c. Protected by testing the bogus_chan value.
	  * Fixed needlessly clearing a 1024 char auto array when setting
	  the first char to zero is enough in manager.c:action_getvar().
	  (closes issue ASTERISK-18613) Reported by: Thomas Arimont
	  Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Tested by: Thomas Arimont ........
	  ................

2011-09-23 19:20 +0000 [r337855-337910]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337902
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337902 | irroot | 2011-09-23 21:18:14 +0200
	  (Fri, 23 Sep 2011) | 10 lines Merged revisions 337898 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) |
	  4 lines Spelling fix ........ ................

	* /, apps/app_queue.c: Merged revisions 337840 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337840 | irroot | 2011-09-23 10:39:22 +0200
	  (Fri, 23 Sep 2011) | 17 lines Merged revisions 337839 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) |
	  11 lines Make sure a CDR is on the stack for call in the Queue.
	  Only let update_cdr act on the last CDR in the stack. In some
	  circumstances [Attended transfer to queue] a CDR record is not
	  inserted for this call where it should. (closes issue
	  ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
	  ........ ................

2011-09-23 00:47 +0000 [r337776]  Russell Bryant <russell@russellbryant.com>

	* /, configs/res_pktccops.conf.sample: Merged revisions 337775 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337775 | russell | 2011-09-22 19:45:35 -0500
	  (Thu, 22 Sep 2011) | 18 lines Merged revisions 337774 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011)
	  | 11 lines Comment out entries in sample res_pktccops.conf. With
	  these options enabled, they can cause Asterisk to freak out by
	  SYN flooding a network and eating the CPU. Obviously it would be
	  good to fix the code so that this can't happen, but we can at
	  least change the default configuration so it doesn't happen. This
	  was reported downstream to the Fedora issue tracker:
	  https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........
	  ................

2011-09-22 21:42 +0000 [r337722]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 337721 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500
	  (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011)
	  | 18 lines Made ISDN not add numbering plan prefix strings to
	  empty numbers. When the Caller-ID is restricted, the expected
	  behavior is for the Caller-ID to be blank. In chan_dahdi, the
	  national prefix is placed onto the Caller-ID number even if it is
	  restricted (empty) causing the Caller-ID to be the national
	  prefix rather than blank. This behavior was lost when sig_pri was
	  extracted from chan_dahdi. * Made not add prefix strings to empty
	  connected line, calling, and ANI number strings. (closes issue
	  ASTERISK-18577) Reported by: Kris Shaw Patches:
	  jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Kris Shaw ........ ................

2011-09-22 16:35 +0000 [r337600]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c, include/asterisk/event_defs.h,
	  main/security_events.c, channels/sip/security_events.c (added),
	  main/event.c, CHANGES, channels/sip/include/security_events.h
	  (added), channels/sip/include/sip.h,
	  include/asterisk/security_events_defs.h,
	  configs/logger.conf.sample: Merged revisions 337595,337597 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep
	  2011) | 12 lines Generate Security events in chan_sip using new
	  Security Events Framework Security Events Framework was added in
	  1.8 and support was added for AMI to generate events at that
	  time. This patch adds support for chan_sip to generate security
	  events. (closes issue ASTERISK-18264) Reported by: Michael L.
	  Young Patches: security_events_chan_sip_v4.patch (license #5026)
	  by Michael L. Young Review:
	  https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose
	  | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot
	  to svn add new files to r337595 Part of Generating security
	  events for chan_sip (issue ASTERISK-18264) Reported by: Michael
	  L. Young Patches: security_events_chan_sip_v4.patch (License
	  #5026) by Michael L. Young Reviewboard:
	  https://reviewboard.asterisk.org/r/1362/ ........

2011-09-22 11:46 +0000 [r337432-337543]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, res/res_srtp.c: Merged revisions 337542 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337542 | irroot | 2011-09-22 13:44:22 +0200
	  (Thu, 22 Sep 2011) | 14 lines Merged revisions 337541 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) |
	  8 lines Add warned to ast_srtp to prevent errors on each frame
	  from libsrtp The first 9 frames are not reported as some devices
	  dont use srtp from first frame these are suppresed. the warning
	  is then output only once every 100 frames. ........
	  ................

	* /, channels/chan_h323.c: Merged revisions 337487 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337487 | irroot | 2011-09-22 11:26:26 +0200
	  (Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) |
	  10 lines If IP address is used in chan_h323 host parameter of
	  peer configuration. module tries to resolve IP address to IP
	  address and fails. Simple fix to set family of socket this is a
	  hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue
	  ASTERISK-17278) (issue ASTERISK-17500) ........ ................

	* main/channel.c, /: Merged revisions 337431 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337431 | irroot | 2011-09-22 08:29:09 +0200
	  (Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) |
	  19 lines Its possible to loose audio on ast_write when the
	  channel is not transcoded correctly. in the case of DAHDI the
	  channel is hungup. This patch tries to "fix" the problem and make
	  the channel compatiable and warn the user of this problem. Please
	  note there is a underlying problem with codec negotion this does
	  not fix the problem it does try to rectify it and prevent loss of
	  service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
	  issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
	  ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
	  (issue ASTERISK-18422) ........ ................

2011-09-21 21:26 +0000 [r337343-337385]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, apps/app_voicemail.c: More silly spacing changes ..... Merged
	  revisions 337353 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged
	  revisions 337380 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* /, apps/app_voicemail.c: ................ ........ Dumb little
	  spacing fix. ........ Merged revisions 337344 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.8
	  ................ Merged revisions 337345 from
	  http://svn.asterisk.org/svn/asterisk/branches/10

	* funcs/func_curl.c, /: ................ ........ Escape commas in
	  keys and values, when keys and values are enumerated by commas.
	  Review: https://reviewboard.asterisk.org/r/1433 ........ Merged
	  revisions 337325 from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ................ Merged revisions 337342 from
	  https://origsvn.digium.com/svn/asterisk/branches/10

2011-09-21 11:21 +0000 [r337262-337283]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, configs/sip.conf.sample: Merged revisions 337263 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) |
	  1 line Whitespace fixup from SRTP patch ........

	* /, apps/app_originate.c, CHANGES: Merged revisions 337261 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21
	  Sep 2011) | 10 lines Adds a timeout argument to app_originate the
	  default is 30s this will be used if the timout supplied is
	  invalid or no timeout is supplied. Contributed by: jacco (thank
	  you for the work) Review:
	  https://reviewboard.asterisk.org/r/1310/ ........

2011-09-21 09:39 +0000 [r337179-337220]  Olle Johansson <oej@edvina.net>

	* main/pbx.c, /, CHANGES, configs/extensions.conf.sample: Merged
	  revisions 337219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13
	  lines Make ast_pbx_run() not default to s@default if extension is
	  not found Review: https://reviewboard.asterisk.org/r/1446/ This
	  is a bug - or architecture mistake - that has been in Asterisk
	  for a very long time. It was exposed by the AMI originate action
	  and possibly some other applications. Most channel drivers checks
	  if an extension exists BEFORE starting a pbx on an inbound call,
	  so most calls will not depend on this issue. Thanks everyone
	  involved in the review and on IRC and the mailing list for a
	  quick review and all the feedback. (closes issue ASTERISK-18578)
	  ........

	* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES:
	  Merged revisions 337178 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14
	  lines Change strictrtp option to default to yes in the RTP module
	  Suggested by Kapejod on Facebook Review:
	  https://reviewboard.asterisk.org/r/1448/ (closes issue
	  ASTERISK-18587) Thanks for quick feedback to kpfleming and
	  Tilghman --Denna och nedanstående rader kommer inte med i
	  loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M
	  res/res_rtp_asterisk.c ........

2011-09-20 23:02 +0000 [r337124]  Matthew Jordan <mjordan@digium.com>

	* apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
	  apps/app_minivm.c, main/app.c, apps/app_confbridge.c,
	  apps/app_followme.c, apps/app_voicemail.c: Merged revisions
	  337120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500
	  (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011)
	  | 21 lines Fix for incorrect voicemail duration in external
	  notifications This patch fixes an issue where the voicemail
	  duration was being reported with a duration significantly less
	  than the actual sound file duration. Voicemails that contained
	  mostly silence were reporting the duration of only the sound in
	  the file, as opposed to the duration of the file with the
	  silence. This patch fixes this by having two durations reported
	  in the __ast_play_and_record family of functions - the
	  sound_duration and the actual duration of the file. The
	  sound_duration, which is optional, now reports the duration of
	  the sound in the file, while the actual full duration of the file
	  is reported in the duration parameter. This allows the voicemail
	  applications to use the sound_duration for minimum duration
	  checking, while reporting the full duration to external parties
	  if the voicemail is kept. (issue ASTERISK-2234) (closes issue
	  ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
	  House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1443 ........ ................

2011-09-20 22:54 +0000 [r337121-337123]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_strings.c: Merged revisions 337119 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011)
	  | 16 lines Fix crash with STRREPLACE function. The
	  ast_func_read() function calls the .read2 callback with the len
	  parameter set to zero indicating no size restrictions on the
	  supplied ast_str buffer. The value was used to dimension a local
	  starts[] array with the array subsequently used. * Reworked the
	  strreplace() function to perform the string replacement in a
	  straight forward manner. Eliminated the need for the starts[]
	  array. (closes issue ASTERISK-18545) Reported by: Federico Alves
	  Patches: jira_asterisk_18545_v10.patch (license #5621) patch
	  uploaded by rmudgett Tested by: rmudgett, Federico Alves ........

	* /: Updated 10 merge property.

	* /: Restore branch-10 merge properties.

2011-09-20 22:29 +0000 [r337117]  Leif Madsen <leif@leifmadsen.com>

	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011)
	  | 7 lines Update RedHat Init script to work with Heartbeat. The
	  current RedHat init script was not LSB compatible. This change
	  will make it LSB compatible so that it can work correctly with
	  Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa
	  ........

2011-09-20 21:05 +0000 [r337063]  Kinsey Moore <kmoore@digium.com>

	* main/pbx.c, /, tests/test_pbx.c: Merged revisions 337062 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337062 | kmoore | 2011-09-20 16:05:01 -0500
	  (Tue, 20 Sep 2011) | 18 lines Merged revisions 337061 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) |
	  11 lines Make CANMATCH with the new pattern match engine behave
	  more like the old one When checking an extension for E_CANMATCH
	  using the new extension matching algorithm, an exact match was
	  not returned as a possible match resulting in the queue failing
	  to allow a caller to exit on DTMF. This removes the requirement
	  that an extension be longer than acquired digits for an
	  E_CANMATCH operation to succeed. (closes issue ASTERISK-18044)
	  Review: https://reviewboard.asterisk.org/r/1367/ ........
	  ................

2011-09-20 19:13 +0000 [r336988-337009]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_ss7.c: Merged revisions 337008 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500
	  (Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011)
	  | 15 lines Check if a channel was created before using the
	  pointer in sig_ss7_new_ast_channel(). Fixes the crash in
	  ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
	  libss7 access lock protection. * Prevent cancelling the
	  ss7_linkset() thread at inoportune times just like the
	  pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
	  Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
	  patch uploaded by rmudgett (attached to related ASTERISK-17966)
	  ........ ................

	* /, channels/sig_ss7.c: Merged revisions 336978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500
	  (Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011)
	  | 21 lines Fix deadlock from not releasing SS7 linkset lock.
	  sig_ss7_hangup() failed to release the SS7 linkset lock if the
	  call had the alreadyhungup flag set. * Made unlock the SS7
	  linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
	  set. * Made ss7_start_call() not hold any locks while creating
	  the channel for an incoming call to prevent deadlock. * Made
	  ss7_grab() a void function, since it could never fail, to
	  simplify calling code. * Made obtain the channel lock to do
	  softhangup in some places. Patches: jira_ast_668_v1.8.patch
	  (license #5621) patch uploaded by rmudgett JIRA AST-668 ........
	  ................

2011-09-20 16:56 +0000 [r336937]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/sip/sdp_crypto.c, /, channels/chan_sip.c,
	  channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h,
	  configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h:
	  Merged revisions 336936 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) |
	  14 lines Allow Setting Auth Tag Bit length Based on invite or
	  config option Update the SIP SRTP API to allow use of 32 or 80
	  bit taglen. Curently only 80 bit is supported. The outgoing
	  invite will use the taglen of the incoming invite preventing
	  one-way audio. (Closes issue ASTERISK-17895) Review:
	  https://reviewboard.asterisk.org/r/1173/ ........

2011-09-20 01:11 +0000 [r336879]  Russell Bryant <russell@russellbryant.com>

	* res/res_rtp_asterisk.c, /: Merged revisions 336878 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336878 | russell | 2011-09-19 20:03:55 -0500
	  (Mon, 19 Sep 2011) | 43 lines Merged revisions 336877 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011)
	  | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses
	  crashes related to RTCP handling. The backtraces just show a
	  crash in ast_rtcp_write() where it appears that the RTP instance
	  is no longer valid. There is a race condition with scheduled RTCP
	  transmissions and the destruction of the RTP instance. This patch
	  utilizes the fact that ast_rtp_instance is a reference counted
	  object and ensures that it will not get destroyed while a
	  reference is still around due to scheduled RTCP transmissions.
	  RTCP transmissions are scheduled and executed from the chan_sip
	  scheduler context. This scheduler context is processed in the SIP
	  monitor thread. The destruction of an RTP instance occurs when
	  the associated sip_pvt gets destroyed (which happens when the
	  sip_pvt reference count reaches 0). However, the SIP monitor
	  thread is not the only thread that can cause a sip_pvt to get
	  destroyed. The sip_hangup function, executed from a channel
	  thread, also decrements the reference count on a sip_pvt and
	  could cause it to get destroyed. While this is being changed
	  anyway, the patch also removes calling ast_sched_del() from
	  within the RTCP scheduler callback. It's not helpful. Simply
	  returning 0 prevents the callback from being rescheduled. (closes
	  issue ASTERISK-18570) Related issues that look like they are the
	  same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
	  (issue ASTERISK-15257) (issue ASTERISK-13334) (issue
	  ASTERISK-9977) (issue ASTERISK-9716) Review:
	  https://reviewboard.asterisk.org/r/1444/ ........
	  ................

2011-09-19 22:28 +0000 [r336837]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 336792 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336792 | twilson | 2011-09-19 17:13:34 -0500
	  (Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19
	  Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is
	  an update to the fix for ASTERISK-18340 and ASTERISK-17725
	  ........ ................

2011-09-19 21:42 +0000 [r336735-336790]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, funcs/func_strings.c: Merged revisions 336789 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011)
	  | 2 lines Ensure substring will not be found in the previous
	  match. ........

	* Makefile, /, configure, include/asterisk/autoconfig.h.in,
	  main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules,
	  include/asterisk/optional_api.h: Merged revisions 336734 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336734 | tilghman | 2011-09-19 15:29:40 -0500
	  (Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011)
	  | 11 lines Various changes to allow 1.8 to compile on Mac OS X
	  Lion (10.7) * Makefile workaround for 10.6 extended to work on
	  10.7 and later. * Now uses the 'weak' symbol for Lion systems,
	  which no longer support 'weak_import' Closes ASTERISK-17612.
	  Closes ASTERISK-18213. Tested by: tilghman, oej. ........
	  ................

2011-09-19 20:23 +0000 [r336732]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
	  apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c,
	  apps/app_mixmonitor.c: Merged revisions 336717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336717 | jrose | 2011-09-19 15:16:23 -0500
	  (Mon, 19 Sep 2011) | 14 lines Merged revisions 336716 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) |
	  7 lines Document applications that play audio and do not answer
	  unanswered calls. This patch is part of an effort to document
	  early media and its usage. If you are interested in contributing
	  to this documentation effort, there are probably other
	  applications worth documenting as well as an Asterisk wiki
	  article at
	  https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
	  ........ ................

2011-09-19 19:03 +0000 [r336660-336662]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336659 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336659 | rmudgett | 2011-09-19 13:51:19 -0500
	  (Mon, 19 Sep 2011) | 38 lines Merged revisions 336658 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011)
	  | 31 lines Made Dial d and H options no longer immediately
	  auto-answer the calling leg. The Dial d and H options break DTMF
	  attended transfer atxferdropcall option. 1) Party A calls party
	  B. 2) Party B does a DTMF attended transfer to Party C. If the
	  dialplan uses the Dial d or H options to call Party C then the
	  Dial application answers the call immediately before initiating
	  the call leg to Party C. The premature answer causes the transfer
	  code to not invoke the atxferdropcall=no behavior for a blonde
	  transfer since Party C has "answered". The transfer code thinks
	  that Party B has "consulted" with Party C when Party B hangs up
	  and completes the transfer to Party A. Party A now hears ringback
	  until Party C actually answers. ASTERISK-13294 Dial d option.
	  ASTERISK-11067 Dial H option to disconnect before answer. The
	  referenced issues made Dial answer with the d and H options
	  because many SIP and ISDN phones cannot send DTMF before the call
	  is connected. * Made require the dialplan to control when or if
	  the call needs to be answered to use the Dial application d and H
	  options. (The call is no longer surprise answered when using the
	  Dial d or H options.) Review:
	  https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
	  AST-666 ........ ................

	* /: Update merge 10 branch merge propterty.

	* /: Restore 10 branch merge properties.

2011-09-19 16:22 +0000 [r336600]  Jason Parker <jparker@digium.com>

	* cel/cel_odbc.c, configs/cel_odbc.conf.sample, sounds/Makefile:
	  Remove weird mergeinfo props that make merges annoying sometimes.

2011-09-19 15:48 +0000 [r336574]  Leif Madsen <leif@leifmadsen.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011)
	  | 7 lines Update get_ilbc_source.sh script to work again.
	  Recently iLBC support in Asterisk has changed after the
	  acquisition of GIPS by Google. More information about how this
	  may affect you is available in a blog post at:
	  http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
	  ........

2011-09-19 15:36 +0000 [r336571]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 336570 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500
	  (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011)
	  | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA
	  AST-675 ........ ................

2011-09-19 13:57 +0000 [r336505]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 336502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån,
	  19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5
	  lines Add diversion header to a 302 redirect response if we have
	  diversion data (closes issue ASTERISK-18143) patch by oej
	  ........ ................

2011-09-19 13:41 +0000 [r336503]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, channels/chan_h323.c: Merged revisions 336500 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336500 | irroot | 2011-09-19 15:31:50 +0200
	  (Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) |
	  13 lines A long time ago in a galaxy far far away a IPv6 update
	  was made, chan_h323 was not updated causeing all to flee to
	  chan_ooh323. the brave Jedi [asterisk developers] pondered this
	  miscarrige of justice and restored order to the force for the
	  sake of closing out 2 old issues. (closes issue ASTERISK-17278)
	  (closes issue ASTERISK-17500) Reported by: dread, sybasesql
	  Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........
	  ................

2011-09-19 12:20 +0000 [r336382-336453]  Olle Johansson <oej@edvina.net>

	* main/manager.c, /: Merged revisions 336441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån,
	  19 Sep 2011) | 9 lines Merged revisions 336440 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2
	  lines Make sure manager_debug option is reset at reload ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 336381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån,
	  19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9
	  lines Add missing unlock at MWI message sending time (closes
	  issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041)
	  by Gregory Hinton Nietsky Thanks to irrot for the reminder, to
	  Gregory for the patch! ........ ................

2011-09-16 22:12 +0000 [r336315-336317]  Terry Wilson <twilson@digium.com>

	* /, funcs/func_frame_trace.c: Merged revisions 336316 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336316 | twilson | 2011-09-16 17:11:39 -0500
	  (Fri, 16 Sep 2011) | 9 lines Merged revisions 336314 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16
	  Sep 2011) | 2 lines Whitespace fix ........ ................

	* /, funcs/func_frame_trace.c: Merged revisions 336313 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336313 | twilson | 2011-09-16 17:07:00 -0500
	  (Fri, 16 Sep 2011) | 12 lines Merged revisions 336312 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011)
	  | 5 lines Add missing frame types to func_frame_trace Also casts
	  control frames to the proper enum so that the compile will catch
	  new additions. ........ ................

2011-09-16 21:20 +0000 [r336311]  Jonathan Rose <jrose@digium.com>

	* main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c,
	  include/asterisk/frame.h: Merged revisions 336307 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336307 | jrose | 2011-09-16 16:09:20 -0500
	  (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) |
	  13 lines Fix bad RTP media bridges in directmedia calls on peers
	  separated by multiple Asterisk nodes. In a situation involving
	  devices on separate Asterisk trunks, the remote RTP bridge would
	  break when starting a call with directmedia. This patch queues a
	  new type of control frame so that our RTP bridge loop can
	  properly detect when these situations occur and check to see if
	  peers need to be updated in order to send their media to the
	  proper location. (Closes issue ASTERISK-18340) Reported by:
	  Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk
	  Tested by: twilson, jrose ........ ................

2011-09-16 19:11 +0000 [r336236]  Sean Bright <sean@malleable.com>

	* /, UPGRADE-1.8.txt: Merged revisions 336235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336235 | seanbright | 2011-09-16 15:10:39 -0400
	  (Fri, 16 Sep 2011) | 9 lines Merged revisions 336234 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri,
	  16 Sep 2011) | 2 lines Make a note that inotify won't work with
	  an NFS mounted spooler directory. ........ ................

2011-09-16 10:16 +0000 [r336095-336168]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_misdn.c, /: Merged revisions 336167 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336167 | irroot | 2011-09-16 12:12:03 +0200
	  (Fri, 16 Sep 2011) | 22 lines Merged revisions 336166 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) |
	  16 lines The round robin routing routine in chan_misdn.c is
	  broken. it rotates between ports but never checks the channels in
	  the ports. i have extensivly tested it and verified it works on 1
	  upto 4 ports. before the patch only 1 out of each port was used
	  now all are used as expected. (closes issue ASTERISK-18413)
	  Reported by: irroot Tested by: irroot Reviewed by: irroot Review:
	  https://reviewboard.asterisk.org/r/1410/ ........
	  ................

	* /, apps/app_queue.c: Merged revisions 336094 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r336094 | irroot | 2011-09-15 17:54:46 +0200
	  (Thu, 15 Sep 2011) | 26 lines Merged revisions 336093 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) |
	  20 lines Locking order in app_queue.c causes deadlocks. a channel
	  lock must never be held with the queues container lock held. the
	  deadlock occured on masquerade. the queues container lock is a
	  relic of the past the old queue module lock. with ao2 there is no
	  need to hold this lock when dealing with members this patch
	  removes unneeded locks. (closes issue ASTERISK-18101) (closes
	  issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault
	  Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew
	  Nicholson Review: https://reviewboard.asterisk.org/r/1402/
	  ........ ................

2011-09-15 15:19 +0000 [r336092]  David Vossel <dvossel@digium.com>

	* /, main/format_cap.c: Merged revisions 336091 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011)
	  | 2 lines Removes some no-op code found in format_cap.c. ........

2011-09-15 12:50 +0000 [r336043]  Olle Johansson <oej@edvina.net>

	* CREDITS, /, apps/app_meetme.c, CHANGES: Merged revisions 336042
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12
	  lines Meetme: Introducing a new option "k" to kill a conference
	  if there's only a single member left. When using Meetme as a
	  modular call bridge from third party applications, it's handy to
	  make it behave like a normal call bridge. When the second to last
	  person exists, the last person will be kicked out of the
	  conference when this option is enabled. (closes issue
	  ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/
	  Patch by oej, sponsored by ClearIT, Solna, Sweden ........

2011-09-15 08:40 +0000 [r335993]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, channels/chan_agent.c: Merged revisions 335991 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335991 | irroot | 2011-09-15 10:29:12 +0200
	  (Thu, 15 Sep 2011) | 17 lines Merged revisions 335978 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) |
	  11 lines lock the channel before calling ast_bridged_channel() to
	  prevent a seg fault. AMI agents list called on shutdown causes a
	  segfault, introducing proper locking will prevent this. (closes
	  issue ASTERISK-18092) Reported by: agustina Patches:
	  chan_agent.patch (License #5041) patch uploaded by irroot
	  ........ ................

2011-09-14 18:38 +0000 [r335853-335913]  Richard Mudgett <rmudgett@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 335912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500
	  (Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011)
	  | 13 lines Remove unnecessary libpri dependency checks in the
	  configure script. Using the --with-pri option with the configure
	  script generated an error about not having PRI_L2_PERSISTENCE if
	  you did not have the absolute latest libpri SVN checkout
	  installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the
	  configure.ac script seems to be for libraries that are dependent
	  upon other libraries and not necessarily for optional/added
	  features within a library. (closes issue ASTERISK-18535) Reported
	  by: Michael Keuter ........ ................

	* channels/chan_dahdi.c, /: Merged revisions 335852 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335852 | rmudgett | 2011-09-14 11:00:37 -0500
	  (Wed, 14 Sep 2011) | 18 lines Merged revisions 335851 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011)
	  | 11 lines Fixed cut-n-paste regression using the wrong variable.
	  Fixes the missing DAHDI channels when using the newer
	  chan_dahdi.conf sections for channel configuration. (closes issue
	  ASTERISK-18496) Reported by: Sean Darcy Patches:
	  jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Sean Darcy, rmudgett ........
	  ................

2011-09-14 13:29 +0000 [r335792]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c, /: Merged revisions 335791 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335791 | mnicholson | 2011-09-14 08:28:50 -0500
	  (Wed, 14 Sep 2011) | 11 lines Merged revisions 335790 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep
	  2011) | 4 lines The tech and data members of
	  fast_originate_helper are not string fields. ASTERISK-17709
	  ........ ................

2011-09-13 22:11 +0000 [r335722]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_directed_pickup.c: Merged revisions 335721 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335721 | rmudgett | 2011-09-13 17:10:44 -0500
	  (Tue, 13 Sep 2011) | 9 lines Merged revisions 335720 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13
	  Sep 2011) | 1 line Remove obsolete todo comment about
	  PICKUPRESULT. ........ ................

2011-09-13 21:52 +0000 [r335719]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/dnsmgr.c: Additional updates for parsing dnsmgr.conf Review:
	  https://reviewboard.asterisk.org/r/1432/

2011-09-13 21:40 +0000 [r335718]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
	  parse the option "defaultlanguage" from the [options] section of
	  asterisk.conf, as in the sample config file. Otherwise the
	  build-time default language (normally "en") is always the default
	  one. Review: https://reviewboard.asterisk.org/r/1342/
	  Signed-off-by: Tzafrir Cohen (License #5035)
	  <tzafrir.cohen@xorcom.com> Original-Commit:
	  http://svn.digium.com/svn/asterisk/branches/1.8@335716
	  Original-Commit:
	  http://svn.digium.com/svn/asterisk/branches/10@335717

2011-09-13 18:56 +0000 [r335657]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, configure, configure.ac: Merged revisions 335656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335656 | tilghman | 2011-09-13 13:55:33 -0500
	  (Tue, 13 Sep 2011) | 11 lines Merged revisions 335655 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011)
	  | 4 lines Move mandatory checks closer to the beginning of the
	  file. If these are going to fail, they should fail as quickly as
	  possible. ........ ................

2011-09-13 18:49 +0000 [r335654]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, main/manager.c, /: Merged revisions 335653 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335653 | mnicholson | 2011-09-13 13:47:57 -0500
	  (Tue, 13 Sep 2011) | 12 lines Merged revisions 335618 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep
	  2011) | 5 lines Don't limit the size of appdata for manager
	  originate actions. ASTERISK-17709 Patch by: tilghman (with
	  modifications) ........ ................

2011-09-13 18:11 +0000 [r335555-335603]  Paul Belanger <paul.belanger@polybeacon.com>

	* UPGRADE.txt, main/dsp.c: Clean up dsp.conf parsing Review:
	  https://reviewboard.asterisk.org/r/1434/

	* UPGRADE.txt, cdr/cdr_csv.c: Clean up cdr.conf parsing for [csv]
	  section Review: https://reviewboard.asterisk.org/r/1427/

	* main/dnsmgr.c, UPGRADE.txt: Clean up dnsmgr.conf parsing Review:
	  https://reviewboard.asterisk.org/r/1432/

2011-09-13 07:35 +0000 [r335511]  Russell Bryant <russell@russellbryant.com>

	* include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged
	  revisions 335510 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335510 | russell | 2011-09-13 02:24:34 -0500
	  (Tue, 13 Sep 2011) | 22 lines Merged revisions 335497 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011)
	  | 15 lines Fix a crash in res_ais. This patch resolves a crash
	  observed in a load testing environment that involved the use of
	  the res_ais module. I observed some crashes where the event
	  delivery callback would get called, but the length parameter
	  incidcating how much data there was to read was 0. The code
	  assumed (with good reason I would think) that if this callback
	  got called, there was an event available to read. However, if the
	  rare case that there's nothing there, catch it and return instead
	  of blowing up. More specifically, the change always ensure that
	  the size of the received event in the cluster is always big
	  enough to be a real ast_event. Review:
	  https://reviewboard.asterisk.org/r/1423/ ........
	  ................

2011-09-12 15:56 +0000 [r335435]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, /: Merged revisions 335434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500
	  (Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep
	  2011) | 6 lines Properly set caller_warning and callee_warning
	  before we try to use them. ASTERISK-18199 Patch by: elguero
	  Testing by: rtang ........ ................

2011-09-12 14:33 +0000 [r335385]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Documentation updates

2011-09-12 14:24 +0000 [r335354]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c, /: Merged revisions 335346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335346 | kmoore | 2011-09-12 09:22:15 -0500
	  (Mon, 12 Sep 2011) | 17 lines Merged revisions 335341 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) |
	  10 lines Ensure frames are not written to dialed channel if
	  ringback is requested When a single channel was dialed and there
	  was media to be forwarded to the calling channel, the media was
	  written without regard for ringback causing silence to be heard
	  in some circumstances. This regression was introduced when the
	  meaning of "single" changed to mean only the number of channels
	  dialed. (closes issue ASTERISK-18083) ........ ................

2011-09-12 14:22 +0000 [r335324-335349]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Small documentation updates

	* CREDITS, channels/chan_sip.c, include/asterisk/indications.h,
	  UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
	  New sip.conf option for setting default tonezone for channel or
	  individual devices Review:
	  https://reviewboard.asterisk.org/r/1429/ (closes issue
	  ASTERISK-18497) Thanks to russellb for peer review.

	* /, channels/chan_sip.c: Merged revisions 335323 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån,
	  12 Sep 2011) | 19 lines Merged revisions 335319 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12
	  lines Lock the peer->mvipvt to avoid crashes with SIP history
	  enabled After the launch of 1.6 event-based MWI we have two
	  threads handling the peer->mwipvt, which cause issues with SIP
	  history additions in combination with the max limit for number of
	  history entries. Review: https://reviewboard.asterisk.org/r/1373/
	  (closes issue ASTERISK-18288) Thanks to irrot for peer review.
	  Work with this bug funded by IPvision AS ........
	  ................

2011-09-12 13:27 +0000 [r335322]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 335321 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335321 | kmoore | 2011-09-12 08:27:04 -0500
	  (Mon, 12 Sep 2011) | 16 lines Merged revisions 335320 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) |
	  9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2
	  does not support IPv6 and getting such addresses from DNS can
	  cause error messages on the remote end involving bad IPv4 address
	  casts in the presence of IPv6/IPv4 tunnels. This patch ensures
	  that IAX2 will not encounter IPv6 addresses via DNS queries.
	  (closes issue ASTERISK-18090) ........ ................

2011-09-12 11:15 +0000 [r335261]  Stefan Schmidt <sst@sil.at>

	* /, channels/chan_sip.c: Merged revisions 335260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335260 | schmidts | 2011-09-12 11:11:45 +0000
	  (Mon, 12 Sep 2011) | 12 lines Merged revisions 335259 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011)
	  | 6 lines build_peer doesnt unlink a peer object from peers_by_ip
	  container which leads to a wrong refcounter value. adding an
	  ao2_unlink from the peers_by_ip container fix it. Review:
	  https://reviewboard.asterisk.org/r/1428/ ........
	  ................

2011-09-12 03:10 +0000 [r335170-335212]  Paul Belanger <paul.belanger@polybeacon.com>

	* UPGRADE.txt: Be more specific on which section has changed.

	* main/cdr.c, UPGRADE.txt: Iterate though cdr.conf setting Review:
	  https://reviewboard.asterisk.org/r/1426/

2011-09-11 17:09 +0000 [r335129]  Terry Wilson <twilson@digium.com>

	* configs/res_config_sqlite3.conf.sample (added),
	  res/res_config_sqlite3.c (added): Add SQLite 3 realtime support

2011-09-09 16:28 +0000 [r335079]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
	  addons/chan_ooh323.c, channels/chan_sip.c,
	  channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
	  main/channel.c, channels/chan_usbradio.c, main/dial.c,
	  channels/chan_dahdi.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, funcs/func_frame_trace.c,
	  main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
	  include/asterisk/frame.h, channels/sig_ss7.c,
	  channels/chan_mgcp.c: Merged revisions 335078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500
	  (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011)
	  | 23 lines Updated SIP 484 handling; added Incomplete control
	  frame When a SIP phone uses the dial application and receives a
	  484 Address Incomplete response, if overlapped dialing is enabled
	  for SIP, then the 484 Address Incomplete is forwarded back to the
	  SIP phone and the HANGUPCAUSE channel variable is set to 28.
	  Previously, the Incomplete application dialplan logic was
	  automatically triggered; now, explicit dialplan usage of the
	  application is required. Additionally, this patch adds a new
	  AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel
	  driver receives this control frame, it is an indication that the
	  dialplan expects more digits back from the device. If the device
	  supports overlap dialing it should attempt to notify the device
	  that the dialplan is waiting for more digits; otherwise, it can
	  handle the frame in a manner appropriate to the channel driver.
	  (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested
	  by: Matthew Jordan Review:
	  https://reviewboard.asterisk.org/r/1416/ ........
	  ................

2011-09-09 07:28 +0000 [r335015]  Gregory Nietsky <gregory@distrotech.co.za>

	* funcs/func_dialplan.c, /, apps/app_readexten.c, CHANGES: Merged
	  revisions 335014 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) |
	  9 lines Move code for VALID_EXTEN from app_readexten to
	  func_dialplan Mark VALID_EXTEN deprecated. Review:
	  https://reviewboard.asterisk.org/r/1396/ ........

2011-09-08 22:30 +0000 [r334955]  Richard Mudgett <rmudgett@digium.com>

	* /, main/logger.c: Merged revisions 334954 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334954 | rmudgett | 2011-09-08 17:28:56 -0500
	  (Thu, 08 Sep 2011) | 17 lines Merged revisions 334953 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011)
	  | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core
	  stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
	  enabled when res_fax tries to unregister its logger level. * Make
	  ast_logger_unregister_level() use ast_free() instead of free().
	  When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
	  a call to free(). Therefore, if you allocated memory with a form
	  of ast_malloc you must free it with ast_free. ........
	  ................

2011-09-08 13:36 +0000 [r334907]  Jonathan Rose <jrose@digium.com>

	* main/cdr.c, main/pbx.c: Removes colorful verb statements
	  erroneously commited with r332760

2011-09-07 19:38 +0000 [r334845]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_iax2.c: Merged revisions 334844 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334844 | pabelanger | 2011-09-07 15:37:24 -0400
	  (Wed, 07 Sep 2011) | 11 lines Merged revisions 334843 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep
	  2011) | 4 lines Cleanup chan_iax2.c log messages Review:
	  https://code.asterisk.org/code/cru/CR-AST-11 ........
	  ................

2011-09-07 19:35 +0000 [r334842]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 334841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500
	  (Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011)
	  | 10 lines Fix AMI action Park crash. * Made AMI action Park not
	  say anything to the parker channel (AMI header Channel2) since
	  the AMI action is a third party parking the call. (This is a
	  change in behavior that cannot be preserved without a lot of
	  effort.) * Made not play pbx-parkingfailed if the Park 's' option
	  is used. JIRA AST-660 ........ ................

2011-09-07 15:37 +0000 [r334683-334792]  Stefan Schmidt <sst@sil.at>

	* /, main/features.c: Merged revisions 334747 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334747 | schmidts | 2011-09-07 15:10:37 +0000
	  (Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07
	  Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in
	  a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
	  before doing a masquerade in the pickup function. ........
	  ................

	* main/features.c: clean up wrong merged stuff

	* /, main/features.c: Merged revisions 334682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011)
	  | 3 lines Adding the Feature to sent a Reason Header in a SIP
	  Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before
	  doing a masquerade in the pickup function. ........

	* main/features.c: Adding the Feature to sent a Reason Header in a
	  SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
	  before doing a masquerade in the pickup function.

2011-09-07 08:17 +0000 [r334618-334623]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, CHANGES, apps/app_queue.c: Merged revisions 334621 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334621 | alecdavis | 2011-09-07 20:14:50 +1200
	  (Wed, 07 Sep 2011) | 9 lines Merged revisions 334620 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07
	  Sep 2011) | 2 lines peroid typo ........ ................

	* main/logger.c: log Asterisk Version number, Build etc into each
	  log file Allow tracking of previous versions through log file
	  records to be tracked. Each time log file is created or opened,
	  log Asterisk Version, Buildinfo. etc. alecdavis (license 585)
	  Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1409/

	* main/pbx.c, /: Merged revisions 334617 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334617 | alecdavis | 2011-09-07 19:45:00 +1200
	  (Wed, 07 Sep 2011) | 17 lines Merged revisions 334616 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep
	  2011) | 10 lines Prevent segfault if call arrives before Asterisk
	  is fully booted. Prevent ast_pbx_start and ast_run_start from
	  starting a new thread unless asterisk is fully booted. alecdavis
	  (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1407/ ........
	  ................

2011-09-07 00:54 +0000 [r334574]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/frame.c, contrib/realtime/mysql/iaxfriends.sql,
	  contrib/realtime/postgresql/realtime.sql,
	  configs/sip.conf.sample, CHANGES,
	  contrib/realtime/mysql/sipfriends.sql: Implement the '!' negation
	  element to negate codecs directly in the allow keyword. This
	  permits the list of codecs to be specified in one configuration
	  line, instead of two or more, generally with the aim of either
	  allowing all codecs with the exception of a few or disallowing
	  most but permitting a few. Review:
	  https://reviewboard.asterisk.org/r/1411/

2011-09-06 16:15 +0000 [r334519]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334455 | irroot | 2011-09-06 15:58:56 +0200
	  (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
	  13 lines Make SQL query in app_voicemail.c portable LIMIT is not
	  portable. Regression from r312212 (closes issue ASTERISK-18255)
	  Reported by: Leif Madsen Tested by: Leif Madsen Review:
	  https://reviewboard.asterisk.org/r/1415/ ........
	  ................

2011-09-06 16:08 +0000 [r334517]  Paul Belanger <paul.belanger@polybeacon.com>

	* configs/iax.conf.sample, /, CHANGES, channels/chan_iax2.c: Merged
	  revisions 334514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep
	  2011) | 6 lines authdebug is now disabled by default To enable
	  this functionaility again set authdebug = yes in iax.conf Review:
	  https://reviewboard.asterisk.org/r/1414/ ........

2011-09-06 16:04 +0000 [r334472-334515]  Gregory Nietsky <gregory@distrotech.co.za>

	* /, apps/app_voicemail.c: Revert r334472 due to properties going
	  missing

	* /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334455 | irroot | 2011-09-06 15:58:56 +0200
	  (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
	  13 lines Make SQL query in app_voicemail.c portable LIMIT is not
	  portable. Regression from r312212 (closes issue ASTERISK-18255)
	  Reported by: Leif Madsen Tested by: Leif Madsen Review:
	  https://reviewboard.asterisk.org/r/1415/ ........
	  ................

2011-09-02 21:09 +0000 [r334304-334358]  Richard Mudgett <rmudgett@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 334357 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334357 | rmudgett | 2011-09-02 16:08:16 -0500
	  (Fri, 02 Sep 2011) | 26 lines Merged revisions 334355 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011)
	  | 19 lines MusicOnHold has extra unref which may lead to memory
	  corruption and crash. The problem happens when a call is
	  disconnected and you had started a MOH class that does not use
	  the files mode. If you define REF_DEBUG and recreate the problem,
	  it will announce itself with the following warning: Attempt to
	  unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
	  class is still in a container! * Fixed moh_alloc() and
	  moh_release() functions not handling the state->class reference
	  consistently. (closes issue ASTERISK-18346) Reported by: Mark
	  Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
	  Review: https://reviewboard.asterisk.org/r/1404/ ........
	  ................

	* /, include/asterisk/config.h, main/config.c: Merged revisions
	  334297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334297 | rmudgett | 2011-09-02 12:15:08 -0500
	  (Fri, 02 Sep 2011) | 46 lines Merged revisions 334296 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011)
	  | 39 lines Fix potential memory allocation failure crashes in
	  config.c. * Added required checks to the returned memory
	  allocation pointers to prevent crashes. * Made
	  ast_include_rename() create a replacement ast_variable list node
	  if the new filename is longer than the available space. Fixes
	  potential crash and memory leak. * Factored out
	  ast_variable_move() from ast_variable_update() so
	  ast_include_rename() can also use it when creating a replacement
	  ast_variable list node. * Made the filename stuffed at the end of
	  the struct a minimum allocated size in ast_variable_new() in case
	  ast_include_rename() changes the stored filename. * Constify
	  struct char pointers pointing to strings stuffed at the end of
	  the struct for: ast_variable, cache_file_mtime, and
	  ast_config_map. * Factored out cfmtime_new() to remove inlined
	  code and allow some struct pointers to become const. * Removed
	  the list lock from struct cache_file_mtime that was never used. *
	  Added doxygen comments to several structure elements and better
	  documented what strings are stuffed at the struct end char array.
	  * Reworked ast_config_text_file_save() and set_fn() to handle
	  allocation failure of the include file scratch pad object
	  tracking blank lines. * Made ast_config_text_file_save() fn[]
	  declared with PATH_MAX to ensure it is long enough for any
	  filename with path. Also reduced the number of container fileset
	  buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
	  https://reviewboard.asterisk.org/r/1378/ ........
	  ................

2011-09-01 17:41 +0000 [r334231-334236]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/pbx.c, /: Merged revisions 334235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334235 | tilghman | 2011-09-01 12:39:32 -0500
	  (Thu, 01 Sep 2011) | 9 lines Merged revisions 334234 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01
	  Sep 2011) | 2 lines Remove 1.6 compatibility documentation from
	  1.8, as it no longer applies. ........ ................

	* res/res_config_odbc.c, /: Merged revisions 334230 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334230 | tilghman | 2011-09-01 12:30:19 -0500
	  (Thu, 01 Sep 2011) | 25 lines Merged revisions 334229 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011)
	  | 18 lines Create a local alias for ast_odbc_clear_cache. As a
	  function pointer, the reference has to be resolved at load time
	  irrespective of the RTLD_LAZY flag. Creating a local alias solves
	  this problem, because the structure is initialized with that
	  local function pointer, while the actual function can remain
	  lazily linked until runtime. The reason why this is important is
	  because we lazily load function references during the module
	  loading process, in order to obtain priority values for each
	  module, ensuring that modules are loaded in the correct order.
	  Previous to this change, when this module was initially loaded,
	  the module loader would emit a symbol resolution error, because
	  of the above requirement. Closes ASTERISK-18399 (reported by
	  Mikael Carlsson, fix suggested by Walter Doekes, patch by me)
	  ........ ................

2011-08-31 18:54 +0000 [r334158]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 334157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334157 | mnicholson | 2011-08-31 13:53:40 -0500
	  (Wed, 31 Aug 2011) | 11 lines Merged revisions 334156 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug
	  2011) | 4 lines Disable T.38 when we get a invite with image
	  media port set to 0 ASTERISK-17678 ........ ................

2011-08-31 18:11 +0000 [r334115]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Optimize chan_sip.c check_rtp_timeout()
	  function. * Make check_rtp_timeout() remember the values returned
	  by ast_rtp_instance_get_timeout(),
	  ast_rtp_instance_get_hold_timeout(), and
	  ast_rtp_instance_get_keepalive() instead of repeatedly calling
	  them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon
	  Patches: issue-18319-trunk-r333066.diff (License #6159) patch
	  uploaded by Rob Gagnon Review:
	  https://reviewboard.asterisk.org/r/1377/

2011-08-31 16:31 +0000 [r334067]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 334064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug
	  2011) | 4 lines only alter the gateway_timeout when attching the
	  gateway to a channel ASTERISK-18219 ........

2011-08-31 16:02 +0000 [r334011-334014]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 334013 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334013 | rmudgett | 2011-08-31 11:00:49 -0500
	  (Wed, 31 Aug 2011) | 30 lines Merged revisions 334012 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011)
	  | 23 lines No DAHDI channel available for conference, user
	  introduction disabled. The following error will consistently
	  occur when trying to dial into a MeetMe conference when the
	  server does not have DAHDI hardware installed: app_meetme.c: No
	  DAHDI channel available for conference, user introduction
	  disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
	  correctly during compilation and install of Asterisk/Dahdi,
	  including associated modules, etc., a chan_dahdi.conf
	  configuration file in /etc/asterisk is not created by FreePBX if
	  hardware does not exist, causing MeetMe to be unable to open a
	  DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
	  channel when there is no chan_dahdi.conf file to load. (closes
	  issue ASTERISK-17398) Reported by: Preston Edwards Patches:
	  jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett ........ ................

	* main/channel.c, /, channels/chan_agent.c: Merged revisions 334010
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500
	  (Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011)
	  | 43 lines Call pickup race leaves orphaned channels or crashes.
	  Multiple users attempting to pickup a call that has been forked
	  to multiple extensions either crashes or fails a masquerade with
	  a "bad things may happen" message. This is the scenario that is
	  causing all the grief: 1) Pickup target is selected 2) target is
	  marked as being picked up in ast_do_pickup() 3) target is
	  unlocked by ast_do_pickup() 4) app dial or queue gets a chance to
	  hang up losing calls and calls ast_hangup() on target 5) SINCE A
	  MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
	  ast_channel_masquerade(), ast_hangup() completes successfully and
	  the channel is no longer in the channels container. 6)
	  ast_do_pickup() then calls ast_channel_masquerade() to schedule
	  the masquerade on the dead channel. 7) ast_do_pickup() then calls
	  ast_do_masquerade() on the dead channel 8) bad things happen
	  while doing the masquerade and in the process ast_do_masquerade()
	  puts the dead channel back into the channels container 9) The
	  "orphaned" channel is visible in the channels list if a crash
	  does not happen. This patch does the following: * Made
	  ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up
	  channel and not release the channel lock until that has happened.
	  * Made __ast_channel_masquerade() not setup a masquerade if
	  either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse
	  of AST_FLAG_ZOMBIE since it would no longer work. (closes issue
	  ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec
	  Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273)
	  Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis,
	  irroot, Karsten Wemheuer Review:
	  https://reviewboard.asterisk.org/r/1400/ ........
	  ................

2011-08-31 15:20 +0000 [r334008]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Merged revisions 334007 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r334007 | kmoore | 2011-08-31 10:19:30 -0500
	  (Wed, 31 Aug 2011) | 14 lines Merged revisions 334006 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) |
	  7 lines Correct an AMI protocol violation with SIPshowpeer The
	  response of SIPshowpeer ends with "\r\n\r\n". Since other
	  commands are ended by using \r\n this confuses any interfacing
	  script. (closes issue ASTERISK-17486) ........ ................

2011-08-30 22:16 +0000 [r333963]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /,
	  addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: Merged
	  revisions 333961-333962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333961 | may | 2011-08-31 01:21:53 +0400 (Wed,
	  31 Aug 2011) | 11 lines Merged revisions 333947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5
	  lines cleanups in ACF/ARJ GK replies processing fixed long (24
	  sec) pause if acf/arj proccessed before ast_cond_wait called to
	  wait this ........ ................ r333962 | may | 2011-08-31
	  01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines security fix. really
	  drop call if signalling addr is not same as socket addr
	  ................

2011-08-30 14:03 +0000 [r333896]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 333895 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug
	  2011) | 6 lines Replaced FAXOPT(gwtimeout) with a second
	  parameter to FAXOPT(gateway). Patch by: irroot Review:
	  https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........

2011-08-29 21:43 +0000 [r333838]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 333837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333837 | twilson | 2011-08-29 16:41:13 -0500
	  (Mon, 29 Aug 2011) | 22 lines Merged revisions 333836 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011)
	  | 15 lines Refresh peer address if DNS unavailable at peer
	  creation If Asterisk starts and no DNS is available, outbound
	  registrations will fail indefinitely. This patch copies the
	  address from the sip_registry struct, which will be updated, to
	  the peer->addr when necessary. If dnsmgr is enabled, the
	  registration fails without the patch because even though the
	  address on the registry is updated via dnsmgr, the address is
	  just copied on the first try. Since we use ast_sockaddr_copy,
	  dnsmgr can't update the address that is copied to the sip_pvt or
	  peers. Closes issue ASTERISK-18000 Review:
	  https://reviewboard.asterisk.org/r/1335/ ........
	  ................

2011-08-29 21:17 +0000 [r333789]  Richard Mudgett <rmudgett@digium.com>

	* /, include/asterisk/channel.h, addons/chan_mobile.c: Merged
	  revisions 333786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333786 | rmudgett | 2011-08-29 16:12:29 -0500
	  (Mon, 29 Aug 2011) | 13 lines Merged revisions 333784-333785 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011)
	  | 2 lines Fix deadlock potential of
	  chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett |
	  2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do
	  not hold locks notes to channel.h ........ ................

2011-08-29 18:28 +0000 [r333736]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax_spandsp.c: Merged revisions 333716 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug
	  2011) | 5 lines It is possible for the gateway to be attached
	  when the channel is still negotiating T.38. This change handles
	  that case. ASTERISK-18329 ........

2011-08-29 17:31 +0000 [r333689]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /, CHANGES: Merged revisions 333681 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011)
	  | 7 lines Use realtime text when it is negotiated This patch make
	  use of wirte_text() realtime text instead of send_text() if T.140
	  is in native formats. ASTERISK-17937 Review:
	  https://reviewboard.asterisk.org/r/1356/ ........

2011-08-29 17:14 +0000 [r333632]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Merged revisions 333631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333631 | mjordan | 2011-08-29 12:12:55 -0500
	  (Mon, 29 Aug 2011) | 9 lines Merged revisions 333630 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29
	  Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI
	  message in app_voicemail ........ ................

2011-08-29 15:58 +0000 [r333571]  Jonathan Rose <jrose@digium.com>

	* /, res/res_jabber.c: Merged revisions 333570 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333570 | jrose | 2011-08-29 10:56:56 -0500
	  (Mon, 29 Aug 2011) | 11 lines Merged revisions 333569 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) |
	  4 lines Accidental use of variable client->status instead of
	  client->state in from ASTERISK-18078 (issue ASTERISK-18078)
	  ........ ................

2011-08-28 09:57 +0000 [r333509]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
	  GCC 4.6 detects variables that get assined to, but never used
	  later. Also removes some remmed-out lines that become invalid.
	  (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
	  (License #5035) <tzafrir.cohen@xorcom.com>,

2011-08-26 16:38 +0000 [r333428]  Jonathan Rose <jrose@digium.com>

	* /, res/res_jabber.c: Merged revisions 333410 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333410 | jrose | 2011-08-26 11:28:03 -0500
	  (Fri, 26 Aug 2011) | 19 lines Merged revisions 333378 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) |
	  13 lines [patch] Buddies are always auto-registered when
	  processing the roster Reporter said autoregister flag was ignored
	  for registering 'buddies' which had a subscription to us.
	  Verified that this was the case and observed how the patch
	  addressed this and made sure it didn't break anything. (closes
	  issue ASTERISK-14233) Reported by: Simon Arlott Patches:
	  asterisk-0015229.patch (license #5756) patch uploaded by Simon
	  Arlott Tested by: Jonathan Rose ........ ................

2011-08-26 16:12 +0000 [r333371]  Matthew Jordan <mjordan@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 333370 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333370 | mjordan | 2011-08-26 10:58:37 -0500
	  (Fri, 26 Aug 2011) | 26 lines Merged revisions 333339 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011)
	  | 20 lines Bug fixes for voicemail user emailsubject / emailbody.
	  This code change fixes a few issues with the voicemail user
	  override of emailbody and emailsubject, including escaping the
	  strings, potential memory leaks, and not overriding the voicemail
	  defaults. Revision 325877 fixed this for ASTERISK-16795, but did
	  not fix it for ASTERISK-16781. A subsequent check-in prevented
	  325877 from being applied to 10. This check-in resolves both
	  issues, and applies the changes to 1.8, 10, and trunk. (closes
	  issue ASTERISK-16781) Reported by: Sebastien Couture Tested by:
	  mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested
	  by: mjordan Review: https://reviewboard.asterisk.org/r/1374
	  ........ ................

2011-08-25 19:13 +0000 [r333276]  Jonathan Rose <jrose@digium.com>

	* /, res/res_jabber.c: Merged revisions 333266 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333266 | jrose | 2011-08-25 14:00:05 -0500
	  (Thu, 25 Aug 2011) | 20 lines Merged revisions 333265 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) |
	  14 lines Segfault when publishing device states via XMPP and not
	  connected When using publishing device state with res_jabber,
	  Asterisk will attempt to send a device state using the
	  unconnected client using iks_send_raw and crash. This patch
	  checks the validity of the connection before attempting to send
	  the device state. (closes issue ASTERISK-18078) Reported by:
	  Michael L. Young Patches:
	  res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
	  patch uploaded by Michael L. Young Tested by: Jonathan Rose
	  ........ ................

2011-08-25 19:01 +0000 [r333159-333269]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 333268 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333268 | qwell | 2011-08-25 14:01:18 -0500
	  (Thu, 25 Aug 2011) | 9 lines Merged revisions 333267 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug
	  2011) | 2 lines Fix for DESTDIR spaces patch. ........
	  ................

	* Makefile, build_tools/mkpkgconfig, /, configure, configure.ac,
	  makeopts.in, sounds/Makefile: Merged revisions 333203 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333203 | qwell | 2011-08-25 10:29:56 -0500
	  (Thu, 25 Aug 2011) | 15 lines Merged revisions 333201 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) |
	  8 lines Fix installation into directories containing spaces. This
	  also vastly simplifies the logic in sounds/Makefile (Closes issue
	  ASTERISK-18290) Reported by: Paul Belanger Review:
	  https://reviewboard.asterisk.org/r/1379/ ........
	  ................

	* channels/chan_local.c: Fix typo from r333070

2011-08-24 16:52 +0000 [r333117]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 333115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug
	  2011) | 4 lines Changed the "timeout" option to "gwtimeout".
	  ASTERISK-18219 ........

2011-08-24 09:17 +0000 [r333070-333075]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c: Formatting changes - Removing some red
	  white space and adding some curly brackets.

	* CHANGES: Add documentation for new manager event in chan_local
	  AST-17623

	* channels/chan_local.c: Add manager event for local channel
	  semi-bridge (issue AST-17623) Review:
	  https://reviewboard.asterisk.org/r/1154

2011-08-23 18:17 +0000 [r332881-333014]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_queue.c: Merged revisions 333011 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r333011 | rmudgett | 2011-08-23 13:15:49 -0500
	  (Tue, 23 Aug 2011) | 19 lines Merged revisions 333010 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011)
	  | 12 lines Memory Leak in app_queue The patch that was committed
	  in the 1.6.x versions of Asterisk for ASTERISK-15862 actually
	  fixed two issues. One was not applicable to 1.8 but the other is.
	  queue_leak.patch fixes the portion applicable to 1.8. (closes
	  issue ASTERISK-18265) Reported by: Fred Schroeder Patches:
	  queue_leak.patch (license #5049) patch uploaded by mmichelson
	  Tested by: Thomas Arimont ........ ................

	* /, main/config.c: Merged revisions 332940 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332940 | rmudgett | 2011-08-22 16:23:40 -0500
	  (Mon, 22 Aug 2011) | 14 lines Merged revisions 332939 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011)
	  | 7 lines Minor code optimizations. * Simplify
	  ast_category_browse() logic for easier understanding. * Remove
	  dead code in ast_variable_delete() and simplify some of its
	  logic. ........ ................

	* /, apps/app_queue.c: Merged revisions 332875,332878 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332875 | rmudgett | 2011-08-22 14:41:03 -0500
	  (Mon, 22 Aug 2011) | 1 line Fix merge property. ................
	  r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011)
	  | 25 lines Merged revisions 332874 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011)
	  | 18 lines Reference leaks in app_queue. * Fixed
	  load_realtime_queue() leaking a queue reference when it
	  overwrites q when processing a realtime queue. (issue
	  ASTERISK-18265) * Make join_queue() unreference the queue
	  returned by load_realtime_queue() when it is done with the
	  pointer. The load_realtime_queue() returns a reference to the
	  just loaded realtime queue. * Fixed queues container reference
	  leak in queues_data_provider_get(). * queue_unref() should not
	  return q that was just unreferenced. * Made logic in
	  __queues_show() and queues_data_provider_get() when calling
	  load_realtime_queue() easier to understand. ........
	  ................

2011-08-22 19:56 +0000 [r332880]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_gtalk.c: Merged revisions 332877 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332877 | pabelanger | 2011-08-22 15:43:33 -0400
	  (Mon, 22 Aug 2011) | 13 lines Merged revisions 332876 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug
	  2011) | 6 lines Revert previous commit It seems google is still
	  making changes to the protocol. (issue ASTERISK-18301) ........
	  ................

2011-08-22 19:52 +0000 [r332879]  Richard Mudgett <rmudgett@digium.com>

	* /: Fix merge 10 branch merge properties.

2011-08-22 19:19 +0000 [r332844]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/test.h, main/manager.c, /, main/file.c,
	  main/test.c, main/app.c, configs/manager.conf.sample,
	  include/asterisk/manager.h, apps/app_voicemail.c: Merged
	  revisions 332817 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011)
	  | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This
	  update adds a new AMI event, TestEvent, which is enabled when the
	  TEST_FRAMEWORK compiler flag is defined. It also adds initial
	  usage of this event to app_voicemail. The TestEvent AMI event is
	  used extensively by the voicemail tests in the Asterisk Test
	  Suite. ........

2011-08-22 18:33 +0000 [r332762-332831]  Richard Mudgett <rmudgett@digium.com>

	* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
	  revisions 332830 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500
	  (Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011)
	  | 8 lines Memory leaks in realtime_multi_xxx() when database
	  access returns error. * Fix realtime_multi_pgsql() configuration
	  memory leak when the database access returns an error. * Fix
	  realtime_multi_odbc() configuration category use after free when
	  the database access returns an error. ........ ................

	* /, main/config.c: Merged revisions 332761 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500
	  (Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011)
	  | 15 lines Memory leak reading realtime database variable list.
	  Calling ast_load_realtime() can leak the last list node if the
	  read list only contains empty variable value items. * Fixed list
	  filter loop in ast_load_realtime() to delete the list node
	  immediately instead of the next time through the loop. The next
	  time through the loop may not happen if the node to delete is the
	  last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
	  Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
	  patch uploaded by rmudgett ........ ................

2011-08-22 17:05 +0000 [r332760]  Jonathan Rose <jrose@digium.com>

	* main/cdr.c, main/pbx.c, configs/cdr.conf.sample,
	  include/asterisk/cdr.h, CHANGES: Add option for logging congested
	  calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a
	  CDR option to cdr.conf that will allow CDR files to log calls
	  ending with congestion in a way that is unique from other
	  unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec
	  Davis Patches: cdr_congestion.diff.txt (License #5546) patch
	  uploaded by Alec Davis

2011-08-22 16:31 +0000 [r332757]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions
	  332756 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug
	  2011) | 4 lines add a way to disable and/or modify the gateway
	  timeout ASTERISK-18219 ........

2011-08-21 14:34 +0000 [r332701]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_gtalk.c: Merged revisions 332700 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400
	  (Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug
	  2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue
	  ASTERISK-18301) Reported by: az1324 ........ ................

2011-08-19 20:00 +0000 [r332655]  Kinsey Moore <kmoore@digium.com>

	* /, apps/app_confbridge.c: Merged revisions 332654 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) |
	  8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO
	  doesn't behave as well in edge cases as MEETME_INFO. With this
	  patch, CONFBRIDGE_INFO should behave in a much more reasonable
	  manner when presented with invalid conferences and keywords.
	  Review: https://reviewboard.asterisk.org/r/1359/ ........

2011-08-19 17:24 +0000 [r332615]  Richard Mudgett <rmudgett@digium.com>

	* res/res_config_ldap.c: Fix infinite loop releasing the same
	  memory in ldap_loadentry(). * Fixed memory leak of vars in
	  ldap_loadentry(). * Fixed potential NULL ptr dereference of vars
	  in ldap_loadentry().

2011-08-18 21:39 +0000 [r332561]  Terry Wilson <twilson@digium.com>

	* main/netsock2.c, /: Merged revisions 332560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332560 | twilson | 2011-08-18 16:34:04 -0500
	  (Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011)
	  | 5 lines Fix possible error on stringification of IPv4-mapped
	  addrs The FreeBSD netsock2 test has been failing for a while. We
	  were pasing sa->len to getnameinfo instead of sa_tmp->len.
	  ASTERISK-18289 ........ ................

2011-08-18 19:30 +0000 [r332505]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 332504 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332504 | kmoore | 2011-08-18 14:29:15 -0500
	  (Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) |
	  8 lines CRC4 in "dahdi show status" gives wrong impression to T1
	  users Change CRC4 to CRC in the output of "dahdi show status" so
	  that it can apply in more situations without confusing users,
	  especially since T1 lines use CRC6 instead of CRC4. (closes issue
	  AST-471) ........ ................

2011-08-18 14:49 +0000 [r332388-332448]  Tilghman Lesher <tilghman@meg.abyt.es>

	* build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged
	  revisions 332447 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332447 | tilghman | 2011-08-18 09:48:40 -0500
	  (Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18
	  Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development
	  mode, as it's useful when DEBUG_THREADS is enabled. ........
	  ................

	* Makefile, agi/Makefile, utils/Makefile, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged
	  revisions 332369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332369 | tilghman | 2011-08-17 14:24:59 -0500
	  (Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011)
	  | 10 lines Re-add support for spaces in pathnames, including now
	  spaces in DESTDIR. This was initially added to 1.8 prior to
	  release, primarily to support the standard paths on Mac OS X, but
	  was partially reverted recently in Subversion, due to the lack of
	  support for spaces in DESTDIR. This commit restores support for
	  the standard paths on Mac OS X, and also includes support for
	  spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by:
	  pabelanger Review: https://reviewboard.asterisk.org/r/1326/
	  ........ ................

2011-08-17 18:31 +0000 [r332337]  Terry Wilson <twilson@digium.com>

	* /, res/res_timing_timerfd.c: Merged revisions 332321 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332321 | twilson | 2011-08-17 13:09:49 -0500
	  (Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011)
	  | 10 lines Don't read from a disarmed or invalid timerfd Numerous
	  isues have been reported for deadlocks that are caused by a
	  blocking read in res_timing_timerfd on a file descriptor that
	  will never be written to. This patch adds some checks to make
	  sure that the timerfd is both valid and armed before calling
	  read(). Should fix: ASTERISK-18142, ASTERISK-18166,
	  ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others.
	  Review: https://reviewboard.asterisk.org/r/1361/ ........
	  ................

2011-08-17 16:18 +0000 [r332270]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  channels/sig_pri.c: Merged revisions 332265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500
	  (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011)
	  | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with
	  HA8, HB8, and B410P cards. France Telecom brings layer 2 and
	  layer 1 down on BRI lines when the line is idle. When layer 1
	  goes down Asterisk cannot make outgoing calls and the HA8 and HB8
	  cards also get IRQ misses. The inability to make outgoing calls
	  is because the line is in red alarm and Asterisk will not make
	  calls over a line it considers unavailable. The IRQ misses for
	  the HA8 and HB8 card are because the hardware is switching clock
	  sources from the line which just brought layer 1 down to internal
	  timing. There is a DAHDI option for the B410P card to not tell
	  Asterisk that layer 1 went down so Asterisk will allow outgoing
	  calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI
	  option for the HA8 and HB8 cards: "modprobe wctdm24xxp
	  bri_teignored=1". Unfortunately that will not clear up the IRQ
	  misses when the telco brings layer 1 down. * Add layer 2
	  persistence option to customize the layer 2 behavior on BRI PTMP
	  lines. The new option has three settings: 1) Use libpri default
	  layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when
	  the peer brings it down. 3) Leave layer 2 down when the peer
	  brings it down. Layer 2 will be brought up as needed for outgoing
	  calls. JIRA AST-598 ........ ................

2011-08-16 20:15 +0000 [r332178]  Paul Belanger <paul.belanger@polybeacon.com>

	* tests/test_substitution.c, tests/test_heap.c, /,
	  tests/test_expr.c, tests/test_ast_format_str_reduce.c,
	  tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
	  tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c,
	  tests/test_linkedlists.c, tests/test_sched.c,
	  tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
	  tests/test_func_file.c, tests/test_security_events.c,
	  tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
	  tests/test_acl.c, tests/test_locale.c, tests/test_utils.c,
	  tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c,
	  tests/test_poll.c, tests/test_amihooks.c: Merged revisions 332177
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400
	  (Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug
	  2011) | 4 lines Flag test modules as 'core' Review:
	  https://reviewboard.asterisk.org/r/1369/ ........
	  ................

2011-08-16 17:53 +0000 [r332120]  Jonathan Rose <jrose@digium.com>

	* /, channels/chan_sip.c: Merged revisions 332119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332119 | jrose | 2011-08-16 12:45:38 -0500
	  (Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) |
	  16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects
	  mailbox value, multiple MWI subs Before, having multiple
	  subscriptions to mailboxes on a sip peer set via the mailbox
	  setting in sip.conf would only result in updates being sent on
	  whichever mailbox triggered the mwi event. Now all of them get
	  counted regardless. Also fixes a bug involving parsing of the
	  mailbox option in sip.conf so that trailing and leading spaces
	  before/after commas are trimmed. (closes issue ASTERISK-18067)
	  Reported by: aragon (closes issue ASTERISK-15479) Reported by:
	  Ben Winslow Patches:
	  chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
	  patch uploaded by Ben Winslow ........ ................

2011-08-16 17:23 +0000 [r332117]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c, CHANGES, configs/features.conf.sample,
	  main/asterisk.c: Merged revisions 332101 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500
	  (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011)
	  | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183
	  Multi-parkinglot directs calls to wrong parkinglot. JIRA
	  ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
	  ParkedCall() with no extension should pickup first available call
	  and does not. JIRA AST-576 Issues with parking lots * Removed
	  searching for parking lots by extension. Parking lots can only be
	  found by the parking lot name since parking lot access extensions
	  and spaces are not guaranteed to be unique. * Added
	  parking_lot_name option to the Park and ParkedCall applications.
	  Updated documentation for Park and ParkedCall applications. * Add
	  parkext_exclusive configuration option to make parking entry
	  extensions specify which parking lot they access. (closes issue
	  ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
	  David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
	  Quezada (closes issue ASTERISK-17430) Reported by: Philippe
	  Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
	  AST-624 'next' setting for findslot does nothing * Reimplemented
	  since findslot feature option broken by -r114655. (closes issue
	  ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
	  JIRA ASTERISK-15792 Dialplan continues execution after transfer
	  to park. This happens for DTMF attended transfer, DTMF blind
	  transfer, and DTMF one-touch-parking if the party initiating
	  these features also initiated the call. * Fixed the return code
	  from the affected builtin features when parking a call. (closes
	  issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
	  rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
	  the expected call when picking up a parked call. This is mostly a
	  documentation problem. However, the option is not reset to the
	  default when features.conf is reloaded. * Updated
	  features.conf.sample documentation for courtesytone and
	  parkedplay options. * Reset the parkedplay option to default when
	  features.conf is reloaded. JIRA AST-615 AMI Park action followed
	  by features reload results in orphaned channels in parking lot. *
	  Reloading features.conf will not touch parking lots that have
	  calls still parked in them. Reload again at a later time. Misc
	  additional fixes: * Added unit test for parking lot dialplan
	  usage checking. * Made update connected line when a parked call
	  is retrieved from a parking lot. * Made retrieved parked call
	  stop ringing or MOH depending upon how the call was waiting in
	  the parking lot. * Made CLI "features show" indicate if the
	  parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
	  variable to allow dynamic parking lots to specify the parking lot
	  access extension. * Made AMI ParkedCalls action ParkedCall events
	  have a Parkinglot header. * Made AMI ParkedCalls action
	  ParkedCallsComplete event have a Total header. * Fixed potential
	  deadlock from AMI Park action holding channel locks while calling
	  masq_park_call(). * Fixed several places where ast_strdupa() were
	  used inside of loops. (Mostly fixed by refactoring the loop body
	  into its own function.) * Fixed copy_parkinglot() copying too
	  much from the source parking lot. Extracted the parking lot
	  configuration settings into struct parkinglot_cfg. * Refactored
	  courtesytone playing code to put the channel not playing the tone
	  in autoservice. * Fix when pbx-parkingfailed is played that the
	  other channel is put in autoservice if it exists. * Fixed
	  parkinglot reference leak in parked_call_exec() error paths. *
	  Fixed parkinglot_unref() use of parkinglot after it was unreffed.
	  * Made destroy the struct ast_parkinglot parkings lock when done.
	  * Refactored the features.conf parking lot configuration code to
	  eliminate redundancy. * Fixed feature reload to better protect
	  parking lots. * Fixed parking lot container reference leak in
	  handle_parkedcalls(). * Fixed the total count in
	  handle_parkedcalls(). Review:
	  https://reviewboard.asterisk.org/r/1358/ ........
	  ................

2011-08-16 15:21 +0000 [r332028-332044]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/sip/include/sip.h: Merged revisions 332042 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue,
	  16 Aug 2011) | 2 lines fix a code comment AST-580 ........

	* /, UPGRADE.txt, CHANGES: Merged revisions 332029 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug
	  2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt
	  from CHANGES AST-580 ........

	* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
	  revisions 332027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500
	  (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue,
	  16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the
	  default value for the 'storesipcause' option AST-580 ........
	  ................

2011-08-16 14:47 +0000 [r332024]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c: Formatting changes while working with
	  DTMF...

2011-08-16 14:41 +0000 [r332023]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged
	  revisions 332022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500
	  (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is
	  disabled by default. Merged revisions 332021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug
	  2011) | 7 lines Added the 'storesipcause' option to sip.conf to
	  allow the user to disable the setting of HASH(SIP_CAUSE,<chan
	  name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan
	  name>) on the channel carries a significant performance penalty
	  because of the usage of the MASTER_CHANNEL() dialplan function.
	  AST-580 ........ ................

2011-08-15 17:36 +0000 [r331957]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 331956 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500
	  (Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011)
	  | 13 lines Fix some minor chan_dahdi config load issues. *
	  Address chan_dahdi.conf dahdichan option todo item about needing
	  line number. * Make ignore_failed_channels option also apply to
	  dahdichan option. * Don't attempt to create a default pseudo
	  channel if the chan_dahdi.conf channel/channels option is not
	  allowed. * Add a similar check for dahdichan in normal
	  chan_dahdi.conf sections as is done in users.conf. ........
	  ................

2011-08-15 15:24 +0000 [r331903]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/rtp_engine.c, /: Merged revisions 331894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400
	  (Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug
	  2011) | 5 lines Fix noisy message when briding channels (closes
	  issue ASTERISK-18270) Reported by: Federico Alves ........
	  ................

2011-08-15 15:15 +0000 [r331869]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 331868 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331868 | dvossel | 2011-08-15 10:14:13 -0500
	  (Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011)
	  | 6 lines Fixes locking inversion issues present in the handling
	  of the sip REFER method. (closes issue ASTERISK-18082) Reported
	  by: James Van Vleet ........ ................

2011-08-15 13:27 +0000 [r331830]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting guideline fixes

2011-08-12 19:06 +0000 [r331776]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 331775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500
	  (Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug
	  2011) | 11 lines Unlock the channel before calling update_queue.
	  Holding the channel lock when calling update_queue which attempts
	  to lock the queue lock can cause a deadlock. This deadlock
	  involves the following chain: 1. hold chan lock -> wait queue
	  lock 2. hold queue lock -> wait agent list lock 3. hold agent
	  list lock -> wait chan list lock 4. hold chan list lock -> wait
	  chan lock ........ ................

2011-08-12 19:01 +0000 [r331773]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 331772 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500
	  (Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011)
	  | 8 lines Suppress warning message when using DAHDITransfer or
	  DAHDIHangup. * The fake event should only be processed by the
	  channel that currently owns the private and not the associated
	  call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616
	  ........ ................

2011-08-12 18:03 +0000 [r331717]  Jonathan Rose <jrose@digium.com>

	* apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331644
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331644 | jrose | 2011-08-12 11:18:57 -0500
	  (Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug
	  2011) | 1 line Fixes 32bit compilation warnings brought on by
	  331634 in app_dial and app_meetme ........ ................

2011-08-12 17:56 +0000 [r331716]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 331715 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500
	  (Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011)
	  | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no
	  effect. The AMI actions DAHDIHangup and DAHDITransfer have no
	  effect on a DAHDI channel. These two AMI actions are highly
	  specialized to analog channels and appear to make the channel
	  behave like a jack port for headsets. * Made the faked DAHDI
	  event get processed before a normal media stream read in
	  dahdi_read() instead of trying to trigger an exception read by
	  setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
	  long ago that changed how AST_FLAG_EXCEPTION is processed in the
	  core. Unfortunately, the faked DAHDI events no longer worked when
	  that happened. * Updated the DAHDI AMI action documentation for
	  the following actions: DAHDITransfer, DAHDIHangup,
	  DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
	  DAHDIRestart. * Made use sscanf() instead of atoi() for better
	  error checking of the DAHDIChannel header string. JIRA AST-620
	  JIRA SWP-3616 ........ ................

2011-08-12 16:32 +0000 [r331660]  Terry Wilson <twilson@digium.com>

	* /, tests/test_netsock2.c: Merged revisions 331659 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331659 | twilson | 2011-08-12 11:31:21 -0500
	  (Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011)
	  | 4 lines Fix netsock2 multiple zero-expansion test Remove
	  erroneous single bracket. ........ ................

2011-08-12 16:22 +0000 [r331657]  Kinsey Moore <kmoore@digium.com>

	* /, main/logger.c: Merged revisions 331654 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331654 | kmoore | 2011-08-12 11:21:37 -0500
	  (Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) |
	  12 lines Logger does not warn of failure to open logging channels
	  Currently, logger only prints an error message to stderr when it
	  fails to open a logger channel where many users will not see it
	  because the logger lock is held. The alternative provided by this
	  patch is to log the error to all attached consoles in the hopes
	  that it will be easier to see. Additionally, this patch prevents
	  the failed logger channel from being added to the list where it
	  would silently fail on each call to the Asterisk logger. (closes
	  issue ASTERISK-16231) Review:
	  https://reviewboard.asterisk.org/r/1338 ........ ................

2011-08-11 21:55 +0000 [r331580]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331579
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331579 | qwell | 2011-08-11 16:54:54 -0500
	  (Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) |
	  6 lines Use proper values for 64-bit option flags. Also, reusing
	  bits es no bueno, so change the value of a duplicate. (issue
	  ASTERISK-18239) ........ ................

2011-08-11 21:44 +0000 [r331577]  Richard Mudgett <rmudgett@digium.com>

	* /, funcs/func_shell.c: Merged revisions 331576 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500
	  (Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011)
	  | 9 lines Segfault in shell_helper in func_shell.c. The return
	  value of popen() was not checked for failure to open. (closes
	  issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles
	  Tested by: rmudgett ........ ................

2011-08-10 22:24 +0000 [r331519]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Merged revisions 331518 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331518 | kmoore | 2011-08-10 17:23:49 -0500
	  (Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) |
	  10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify
	  sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
	  the additional ref just before the invite and adding an unref
	  following it corrects the issue as seen via REF_DEBUG. The unref
	  existed in a distant revision and it appears as though the wrong
	  ref operation was removed. (closes issue ASTERISK-18091) Review:
	  https://reviewboard.asterisk.org/r/1332/ ........
	  ................

2011-08-10 20:51 +0000 [r331419-331463]  Richard Mudgett <rmudgett@digium.com>

	* /, main/logger.c: Merged revisions 331462 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500
	  (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011)
	  | 30 lines Output of queue log not started until logger reloaded.
	  ASTERISK-15863 caused a regression with queue logging. The output
	  of the queue log is not started until the logger configuration is
	  reloaded. * Queue log initialization is completely delayed until
	  the first message is posted to the queue log system. Including
	  the initial opening of the queue log file. * Fixed rotate_file()
	  ROTATE strategy to give the file just rotated out to the
	  configured exec function after rotate. Just like the other
	  strategies. * Fixed logger reload to always post the queue reload
	  entry instead of just if there is a queue log file. * Refactored
	  some code to eliminate some redundancy and to reduce stack
	  utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported
	  by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch
	  (license #5621) patch uploaded by rmudgett Tested by: rmudgett
	  (closes issue ASTERISK-18208) Reported by: Christian Pinedo
	  Review: https://reviewboard.asterisk.org/r/1333/ ........
	  ................

	* /, main/features.c: Merged revisions 331420 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011)
	  | 2 lines Make sure feature_request_and_dial() initializes
	  outstate if passed in. ........

	* /, main/features.c, CHANGES: Merged revisions 331418 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011)
	  | 6 lines Revert -r318141. It was a band-aid that only partially
	  fixed parking. A better fix is on reviewboard review 1358. (issue
	  ASTERISK-17374) ........

2011-08-10 15:45 +0000 [r331371]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c, CHANGES: SIP display-name needed to be empty
	  for Avaya IP500 In order to address a compatability issue with
	  certain features on certain devices which rely on display name
	  content to change behavior, initreqprep in chan_sip.c has been
	  changed to no longer substitute cid_number into the display name
	  when cid_name isn't present. Instead, it will send no display
	  name in that case. (closes issue ASTERISK-16198) Reported by:
	  Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/

2011-08-10 13:49 +0000 [r331317]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c, /: Merged revisions 331316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500
	  (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) |
	  8 lines AMI action ModuleReload returns Error if Module: missing
	  or empty An empty string was not being checked for properly
	  causing identification of the module to be reloaded to fail and
	  return an Error with message "No such module." (closes issue
	  AST-616) ........ ................

2011-08-09 23:17 +0000 [r331266]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /, channels/chan_sip.c, main/features.c,
	  channels/chan_iax2.c, apps/app_parkandannounce.c: Merged
	  revisions 331265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500
	  (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011)
	  | 15 lines Misc minor items found in code. * Add some reentrancy
	  protection in pbx.c when creating the contexts_table hash table.
	  * Fix inverted test in chan_sip.c conditional code. * Fix
	  uninitialized variable and use of the wrong variable in
	  chan_iax2.c. * Fix test of return value in app_parkandannounce.c.
	  Explicitly testing for -1 is bad if the function does not
	  actually return that value when it fails. * Fixup some comments
	  and add some curly braces in features.c. ........
	  ................

2011-08-09 17:12 +0000 [r331202]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /,
	  addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooq931.c:
	  Merged revisions 331147,331200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue,
	  09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4
	  lines move ast_cond_signal for admitted call after all data
	  filled/freed clear all log channels by pointed number not only
	  first free allocated callToken in ooh323_answer ........
	  ................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue,
	  09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode
	  Added additional check for IP semantics before parse destination
	  by ast_parse_args due to it can parse numeric as IP. (closes
	  issue ASTERISK-18218) Reported by: slesru Patch:
	  ASTERISK-18218.patch ................

2011-08-09 17:08 +0000 [r331201]  Kinsey Moore <kmoore@digium.com>

	* funcs/func_enum.c, UPGRADE.txt, main/enum.c: Allow ENUM query
	  functions to report lookup errors The ENUM dialplan functions do
	  not report DNS query errors properly. It is useful to
	  differentiate between failed query (e.g. non-existent domain) vs.
	  no data records of the appropriate type. This is required to make
	  overlapped dialing work. (closes issue ASTERISK-13769) Review:
	  https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras

2011-08-09 16:02 +0000 [r331140-331144]  Jason Parker <jparker@digium.com>

	* /, doc/asterisk.8: Merged revisions 331143 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331143 | qwell | 2011-08-09 10:59:54 -0500
	  (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug
	  2011) | 1 line Regenerate asterisk man page from sgml. ........
	  ................

	* /, doc/asterisk.8, configs/asterisk.conf.sample,
	  configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
	  revisions 331139 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331139 | qwell | 2011-08-09 10:50:07 -0500
	  (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011)
	  | 12 lines Documentation Updates Note default polling setting in
	  voicemail.conf Add missing config to asterisk.conf Update manpage
	  (issue #16505) Reported by: tzafrir Patches:
	  asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
	  Tested by: lathama, tzafrir ........ ................

	* doc/asterisk.8, configs/asterisk.conf.sample,
	  configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
	  revisions 331138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) |
	  1 line Revert merge of r306999, due to merge conflict. ........

2011-08-08 22:59 +0000 [r331042-331098]  Terry Wilson <twilson@digium.com>

	* /, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Merged
	  revisions 331097 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011)
	  | 5 lines Bump the AMI protocol version to 1.2 As a result of
	  converting Unlink events that were missed in the AMI 1.1 update
	  to Bridge events, the AMI protocol version is being incremented.
	  ........

	* main/channel.c, /, CHANGES: Merged revisions 331041 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011)
	  | 6 lines Replace AMI Unlink events with Bridge events A previous
	  update converted some of the Link and Unlink events to Bridge
	  events, but a couple of Unlink events were missed. This patch
	  rectifies the situation. (closes issues ASTERISK-17455) ........

2011-08-08 20:54 +0000 [r331000-331040]  Kinsey Moore <kmoore@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 331039 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r331039 | kmoore | 2011-08-08 15:53:30 -0500
	  (Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) |
	  11 lines In-queue MOH stops after a periodic announcement If the
	  seek value is past the end of file when resuming G.722 MOH, MOH
	  will cease to function for the duration of the MOH session
	  through all starts and stops until saved state is cleared.
	  Adjusting the code to guarantee a single valid read (which is
	  already assumed) fixes the bug. (closes issue ASTERISK-18077)
	  Review: https://reviewboard.asterisk.org/r/1328/ Tested-by:
	  Jonathan Rose <jrose@digium.com> ........ ................

	* configs/queues.conf.sample, apps/app_queue.c: Log queue member
	  name when state_interface is set for ADDMEMBER and REMOVEMEMBER
	  events app_queue logs the events ADDMEMBER and REMOVEMEMBER with
	  the agent field set to the interface value rather than the
	  membername value when a member is added with a state_interface
	  value set. However all other member related queue events are
	  logged with the membername when a state_interface is set. This
	  patch makes these fields optionally more consistent and correct.
	  (closes issue ASTERISK-14769) Review:
	  https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey
	  Tested-by: Kinsey Moore <kmoore@digium.com>

	* apps/app_queue.c: app_queue: Add StateInterface to output of
	  "queue show" and "QueueStatus" This patch adds the
	  state_interface of the queue member struct to the output of
	  "queue show" (CLI command) and "QueueStatus" (AMI action) when
	  displaying relevant queue member information. For the AMI event
	  message the variable StateInterface has been added. (closes issue
	  ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/
	  Patch-by: Jamuel Starkey

2011-08-05 15:57 +0000 [r330941]  David Vossel <dvossel@digium.com>

	* /, codecs/codec_resample.c: Merged revisions 330940 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011)
	  | 2 lines The slin resampler is no longer dependent on an
	  external library, but the dependency was not removed correctly.
	  ........

2011-08-05 08:47 +0000 [r330903]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c, /,
	  addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330899 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330899 | may | 2011-08-05 11:38:28 +0400 (Fri,
	  05 Aug 2011) | 11 lines Merged revisions 330827 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4
	  lines change gk client behaivour on rrq/grq failures to setup
	  timers and next tries after timeout instead of complete failure
	  in the ooh323 stack ........ ................

2011-08-04 20:53 +0000 [r330845]  Terry Wilson <twilson@digium.com>

	* /, configure, configure.ac: Merged revisions 330844 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330844 | twilson | 2011-08-04 15:51:23 -0500
	  (Thu, 04 Aug 2011) | 11 lines Merged revisions 330843 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011)
	  | 4 lines Make libsrtp instructions more explicit when linking
	  fails (closes issue ASTERISK-18139) ........ ................

2011-08-03 15:16 +0000 [r330707-330764]  Kinsey Moore <kmoore@digium.com>

	* /, main/Makefile: Merged revisions 330763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330763 | kmoore | 2011-08-03 10:15:26 -0500
	  (Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) |
	  9 lines editing files in main/editline does not ensure rebuild of
	  libedit.a When editing a source file in main/editline, the build
	  system does not rebuild libedit.a and uses the already existing
	  one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem.
	  (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........
	  ................

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  330706 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330706 | kmoore | 2011-08-03 08:39:06 -0500
	  (Wed, 03 Aug 2011) | 17 lines Merged revisions 330705 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) |
	  10 lines Call pickup broken for DAHDI channels when beginning
	  with # The call pickup feature did not work on DAHDI devices for
	  anything other than feature codes beginning with * since all
	  feature codes in chan_dahdi were originally hard-coded to begin
	  with *. This patch is also applied to chan_dahdi.c to fix this
	  bug with radio modes. (closes issue AST-621) Review:
	  https://reviewboard.asterisk.org/r/1336/ ........
	  ................

2011-08-02 20:54 +0000 [r330650]  Kevin P. Fleming <kpfleming@digium.com>

	* /, res/res_jabber.c: Merged revisions 330649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330649 | kpfleming | 2011-08-02 15:52:44 -0500
	  (Tue, 02 Aug 2011) | 9 lines Merged revisions 330648 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02
	  Aug 2011) | 2 lines Convert an error message to actually be
	  helpful. ........ ................

2011-08-02 16:19 +0000 [r330577-330593]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 330586 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330586 | dvossel | 2011-08-02 11:17:59 -0500
	  (Tue, 02 Aug 2011) | 15 lines Merged revisions 330581 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011)
	  | 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2
	  resulting from an edge case in the way control frames are queued
	  during calltoken negotiation is complete. (closes issue
	  ASTERISK-17610) Reported by: mgrobecker ........ ................

	* /, channels/chan_sip.c: Merged revisions 330579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330579 | dvossel | 2011-08-02 11:08:57 -0500
	  (Tue, 02 Aug 2011) | 9 lines Merged revisions 330578 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02
	  Aug 2011) | 2 lines Optimization to buffer initialization fix.
	  ........ ................

	* /, channels/chan_sip.c: Merged revisions 330576 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330576 | dvossel | 2011-08-02 10:55:36 -0500
	  (Tue, 02 Aug 2011) | 12 lines Merged revisions 330575 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011)
	  | 5 lines Fixes uninitialized string buffer in log message.
	  (closes issue ASTERISK-17200) Reported by: lmadsen ........
	  ................

2011-08-01 15:24 +0000 [r330435]  Kinsey Moore <kmoore@digium.com>

	* /, main/say.c: Merged revisions 330434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330434 | kmoore | 2011-08-01 10:23:29 -0500
	  (Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) |
	  9 lines Incorrect playback for Spanish in some circumstances When
	  you say the time in spanish and it is 01:00 - 01:59 or 13:00 -
	  13:59 you must use female pronunciation "1F". The function
	  "say_date_with_format_es" does not take this in account. (closes
	  ASTERISK-15016) Patch-by: Luis Jimenez ........ ................

2011-07-31 00:19 +0000 [r330370-330379]  Richard Mudgett <rmudgett@digium.com>

	* main/astobj2.c: Fixed compiler warning and a couple prototype
	  mismatches.

	* main/channel.c, /: Merged revisions 330369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500
	  (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011)
	  | 4 lines Remove some redundant locking code in
	  ast_do_masquerade(). Also updated some comments. ........
	  ................

2011-07-30 15:54 +0000 [r330313]  Gregory Nietsky <gregory@distrotech.co.za>

	* main/channel.c, /: Merged revisions 330312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330312 | irroot | 2011-07-30 17:34:41 +0200
	  (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) |
	  9 lines prevent double masqurading channels when one is been hung
	  up and deadlock avoidance is used. There is a race condition in
	  ast_do_masquerade / ast_hangup (at least) Reported by me signed
	  off by schmidts with input from David Vossel Review:
	  https://reviewboard.asterisk.org/r/1323/ ........
	  ................

2011-07-29 19:34 +0000 [r330273]  Russell Bryant <russell@russellbryant.com>

	* include/asterisk/astobj2.h, tests/test_astobj2.c,
	  channels/chan_iax2.c, main/astobj2.c: astobj2: Avoid using
	  temporary objects + ao2_find() with OBJ_POINTER. There is a
	  fairly common pattern making its way through the code base where
	  we put a temporary object on the stack so we can call ao2_find()
	  with OBJ_POINTER. The purpose is so that it can be passed into
	  the object hash function. However, this really seems like a hack
	  and potentially error prone. This patch is a first stab at
	  approach to avoid having to do that. It adds a new flag, OBJ_KEY,
	  which can be used instead of OBJ_POINTER in these situations.
	  Then, the hash function can know whether it was given an object
	  or some custom data to hash. The patch also changes some uses of
	  ao2_find() for iax2_user and iax2_peer objects to reflect how
	  OBJ_KEY would be used. So long, and thanks for all the fish.
	  Review: https://reviewboard.asterisk.org/r/1184/

2011-07-29 17:20 +0000 [r330205-330221]  Sean Bright <sean@malleable.com>

	* /, formats/format_wav.c: Merged revisions 330217 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330217 | seanbright | 2011-07-29 13:19:42 -0400
	  (Fri, 29 Jul 2011) | 9 lines Merged revisions 330213 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri,
	  29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........
	  ................

	* /, formats/format_wav.c: Merged revisions 330204 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330204 | seanbright | 2011-07-29 12:58:40 -0400
	  (Fri, 29 Jul 2011) | 9 lines Merged revisions 330203 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri,
	  29 Jul 2011) | 2 lines Only write to wav files that were opened
	  to be written to. ........ ................

2011-07-29 05:27 +0000 [r330163]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/app_confbridge.c: Merged revisions 330162 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul
	  2011) | 4 lines Fix typo pointed out on #asterisk Thanks notten
	  ........

2011-07-28 21:46 +0000 [r330109]  Terry Wilson <twilson@digium.com>

	* /, main/term.c: Merged revisions 330108 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330108 | twilson | 2011-07-28 16:44:31 -0500
	  (Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28
	  Jul 2011) | 2 lines Make console colors work for
	  TERM=xterm-256color ........ ................

2011-07-28 17:16 +0000 [r330052]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 330051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500
	  (Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500
	  (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
	  28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
	  outgoing call legs of a data call are using different formats:
	  a-law, u-law. When the call is bridged, the media stream is run
	  through translation to convert the media formats. The translation
	  is bad for data calls. * Make incoming call that does not
	  explicitly specify u-law or a-law use the DAHDI channel's default
	  law. The outgoing call always uses the default law from the DAHDI
	  channel. (closes issue ABE-2800) Patches:
	  jira_abe_2800_companding.patch (license #5621) patch uploaded by
	  rmudgett .......... ................ ................

2011-07-28 15:46 +0000 [r329996]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 329995 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329995 | qwell | 2011-07-28 10:45:49 -0500
	  (Thu, 28 Jul 2011) | 13 lines Merged revisions 329994 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) |
	  6 lines Fix a SIP transfer deadlock. The locking in this function
	  is very scary. There are like 6 structs involved. (closes issue
	  AST-470) ........ ................

2011-07-28 15:30 +0000 [r329993]  Matthew Nicholson <mnicholson@digium.com>

	* /, res/res_fax.c: Merged revisions 329992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329992 | mnicholson | 2011-07-28 10:28:21 -0500
	  (Thu, 28 Jul 2011) | 13 lines Merged revisions 329991 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul
	  2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading
	  the res_fax config file Patch by: tzafrir Reported by: tzafrir
	  (closes issue ASTERISK-18161) ........ ................

2011-07-28 13:04 +0000 [r329897-329953]  Sean Bright <sean@malleable.com>

	* configs/confbridge.conf.sample, /: Merged revisions 329952 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu,
	  28 Jul 2011) | 4 lines The default conf-usermenu says that '8'
	  can be used to leave the conference, so put that in the sample
	  user menu. '5' is supposed to extend the conference, but there
	  doesn't appear to be a concept of that in the menu actions.
	  ........

	* /, apps/app_confbridge.c: Merged revisions 329950 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul
	  2011) | 1 line Correct the spelling of 'conference.' ........

	* /, channels/chan_sip.c: Merged revisions 329896 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329896 | seanbright | 2011-07-28 07:35:27 -0400
	  (Thu, 28 Jul 2011) | 9 lines Merged revisions 329895 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu,
	  28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show
	  settings' more consistent. ........ ................

2011-07-27 21:22 +0000 [r329835-329856]  Jonathan Rose <jrose@digium.com>

	* main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES:
	  reverting 329840 due to failing tests. Going to change this
	  feature to be purely optional.

	* main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Adds cdr
	  logging of calls resulting in CONGESTION Applies a patch made a
	  long time ago by alecdavis which adds a CDR feature for logging
	  calls that failed due to congestion. (closes issue #15907)
	  Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded
	  by alecdavis (license #5546) Review:
	  https://reviewboard.asterisk.org/r/454/

2011-07-27 19:19 +0000 [r329775]  Sean Bright <sean@malleable.com>

	* /, Makefile.moddir_rules: Merged revisions 329771 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329771 | seanbright | 2011-07-27 15:18:47 -0400
	  (Wed, 27 Jul 2011) | 15 lines Merged revisions 329767 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul
	  2011) | 8 lines Explicitly sort the module list so that the
	  menuselect lists are sorted. (closes ASTERISK-18141) Reported by:
	  Richard Miller Patches: sort-order.diff uploaded by seanbright
	  (License #5060) Tested by: leifmadsen ........ ................

2011-07-27 18:12 +0000 [r329711]  Jonathan Rose <jrose@digium.com>

	* /, configs/indications.conf.sample: Merged revisions 329710 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329710 | jrose | 2011-07-27 13:11:07 -0500
	  (Wed, 27 Jul 2011) | 14 lines Merged revisions 329709 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) |
	  8 lines Fix New Zealand indications profile based on
	  http://www.telepermit.co.nz/TNA102.pdf (closes issue
	  ASTERISK-16263) Reported by: richardf Patches:
	  nz-indications.patch uploaded by richardf (License #6015)
	  ........ ................

2011-07-27 15:26 +0000 [r329671]  Sean Bright <sean@malleable.com>

	* /, main/loader.c: Merged revisions 329670 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul
	  2011) | 2 lines Sort the module list so that 'module show' is
	  alphabetical. ........

2011-07-27 04:27 +0000 [r329615]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, cdr/cdr_odbc.c: Merged revisions 329614 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329614 | tilghman | 2011-07-26 23:25:26 -0500
	  (Tue, 26 Jul 2011) | 13 lines Merged revisions 329613 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011)
	  | 6 lines Duration and billsec are swapped in high resolution
	  time. Closes ASTERISK-18024 Patches:
	  20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
	  ........ ................

2011-07-26 14:27 +0000 [r329530-329564]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 329538 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329538 | jrose | 2011-07-26 09:19:34 -0500
	  (Tue, 26 Jul 2011) | 11 lines Merged revisions 329529 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) |
	  5 lines Changes sound file for prepend "then-press-pound" to
	  "vm-then-pound" which is the same prompt, only it turned out
	  "then-press-pound" was part of extra sounds. Also, vm is more
	  appropriate anyway. ........ ................

	* include/asterisk/app.h, /, configs/voicemail.conf.sample,
	  main/app.c, apps/app_voicemail.c: Merged revisions 329528 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329528 | jrose | 2011-07-26 08:52:34 -0500
	  (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) |
	  17 lines Fixes some voicemail forwarding behavior based around
	  prepend mode. Formerly, prepend forwarding would have the user
	  record a message with no useful prompt and an expectation for the
	  user to push a button on the phone when finished recording. If a
	  length of silence was detected instead, the recording would be
	  canceled and the user would re-enter the voicemail forwarding
	  menu. Subsequent time-outs in prepend recording would also bug
	  out in the sense that they would write over the original message
	  and get sent to the recipient regardless of whether they timed
	  out or were accepted. This patch fixes this issue and adds a
	  prompt which will be played after a timeout informing the user
	  that they needed to press a button. Currently, the sound files
	  that we have are somewhat inadquate for this, so after the call
	  we simply have Allison say "Please try again. Then press pound."
	  which actually relies on two separate sound files. Just one would
	  be more appropriate. reporter: Vlad Povorozniuc Review:
	  https://reviewboard.asterisk.org/r/1327/ ........
	  ................

2011-07-25 19:57 +0000 [r329473]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, main/enum.c: Merged revisions 329472 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400
	  (Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
	  ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon,
	  25 Jul 2011) | 2 lines Decrease verbose messages to debug, to
	  help clean up CLI. ........ ................

2011-07-25 14:07 +0000 [r329391-329432]  Gregory Nietsky <gregory@distrotech.co.za>

	* include/asterisk/dsp.h, main/dsp.c: dsp_process was enhanced to
	  work with alaw and ulaw in addition to slin. noticed that some
	  functions could be refactored here it is. Reported by: irroot
	  Tested by: irroot, mnicholson Review:
	  https://reviewboard.asterisk.org/r/1304/

	* channels/chan_sip.c, channels/sip/include/sip.h: Remove
	  lastmsgssent from sip it has not been working since 1.6 Clean up
	  the return values to be consistant not currently used Add doxygen
	  returns MWI Event is sent on Register (closes issue
	  ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak
	  Review: https://reviewboard.asterisk.org/r/1172/

2011-07-22 21:15 +0000 [r329332-329335]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c, /: Merged revisions 329334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10 ........
	  r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011)
	  | 1 line Make use less redundant loop construct for iterating
	  over hints. ........

	* main/pbx.c, /: Merged revisions 329331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500
	  (Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011)
	  | 48 lines Deadlocks dealing with dialplan hints during reload.
	  There are two remaining different deadlocks reported dealing with
	  dialplan hints. The deadlock in ASTERISK-17666 is caused by
	  invalid locking order in ast_remove_hint(). The hints container
	  must be locked before the hint object. The deadlock in
	  ASTERISK-17760 is caused by a catch-22 situation in
	  handle_statechange(). The deadlock is caused by not having the
	  conlock before calling the watcher callbacks. Unfortunately,
	  having that lock causes a different deadlock as reported in
	  ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
	  handle_statechange() no longer call the watcher callbacks holding
	  any locks that matter. * Made hint ao2 destructor do the watcher
	  callbacks for extension deactivation to guarantee that they get
	  called. * Fixed hint reference leak in ast_add_hint() if the
	  callback container constructor failed. * Fixed hint reference
	  leak in complete_core_show_hint() for every hint it found for CLI
	  tab completion. * Adjusted locking in
	  ast_merge_contexts_and_delete() for safety. * Added
	  context_merge_lock to prevent ast_merge_contexts_and_delete() and
	  handle_statechange() from interfering with each other. * Fixed
	  ast_change_hint() not taking into account that the extension is
	  used for the hash key. (closes issue ASTERISK-17666) Reported by:
	  irroot Tested by: irroot JIRA SWP-3318 (closes issue
	  ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
	  SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
	  ........ ................

2011-07-21 20:26 +0000 [r329258]  Russell Bryant <russell@russellbryant.com>

	* channels/chan_dahdi.c, /, main/features.c,
	  include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c,
	  include/asterisk/rtp_engine.h: Merged revisions 329257 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21
	  Jul 2011) | 2 lines s/1.10/10.0/ ........

2011-07-21 18:06 +0000 [r329146-329205]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
	  revisions 329204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500
	  (Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011)
	  | 6 lines Document parkinglot in chan_dahdi.conf.sample. *
	  Document existing feature in chan_dahdi.conf.sample. * Remove
	  some dead code related to the parkinglot option. ........
	  ................

	* /, apps/app_directed_pickup.c: Merged revisions 329200 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500
	  (Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011)
	  | 17 lines Update PickupChan documentation. The PickupChan uses
	  the ampersand as the argument separator. Was documented as:
	  PickupChan(channel[,channel2[,...][,options]]) Fixed
	  documentation to:
	  PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
	  This is a continuation of ASTERISK-17494 for v1.8 and later.
	  (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
	  pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
	  by Erik Smith Tested by: Erik Smith ........ ................

	* /, main/features.c: Merged revisions 329145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/10
	  ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500
	  (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
	  r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011)
	  | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed
	  more times than we've locked! This appears to be a leftover from
	  when ast_channel was converted to ao2 objects. Simply removed the
	  extraneous unlock. (closes issue ASTERISK-17772) ........
	  ................

2011-07-21 16:22 +0000 [r329106-329130]  Jason Parker <jparker@digium.com>

	* UPGRADE-1.10.txt (removed), UPGRADE-10.txt (added), UPGRADE.txt:
	  Fix UPGRADE.txt files for Asterisk 10.

	* /: Remove another 2.0 property.

2011-07-21 16:05 +0000 [r329105]  Russell Bryant <russell@russellbryant.com>

	* /: Fix merge properties to reflect Asterisk 10 branch

