[asterisk-commits] bebuild: tag 10.12.0-digiumphones-rc2 r378742 - /tags/10.12.0-digiumphones-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jan 9 13:53:05 CST 2013


Author: bebuild
Date: Wed Jan  9 13:53:00 2013
New Revision: 378742

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=378742
Log:
Importing files for 10.12.0-digiumphones-rc2 release.

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    tags/10.12.0-digiumphones-rc2/.version   (with props)
    tags/10.12.0-digiumphones-rc2/ChangeLog   (with props)

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+2013-01-09  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.12.0-digiumphones-rc2 Released.
+
+2013-01-09 00:23 +0000 [r378686-378705]  Automerge script <automerge at asterisk.org>
+
+	* apps/app_queue.c, /: Merged revisions 378689 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r378689 | rmudgett | 2013-01-08 17:55:57 -0600 (Tue, 08 Jan 2013)
+	  | 4 lines app_queue: Fix incorrect assertion. (issue
+	  ASTERISK-16115) ........
+
+	* configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+	  apps/app_queue.c, /: Merged revisions 378683 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r378683 | rmudgett | 2013-01-08 16:55:43 -0600
+	  (Tue, 08 Jan 2013) | 29 lines app_queue: Fix multiple calls to a
+	  queue member that is in only one queue. When ringinuse=no queue
+	  members can receive more than one call if these calls happen at
+	  nearly the same time. * Fix so a queue member does not receive
+	  more than one call from a queue. NOTE: This fix does not prevent
+	  multiple calls to a member if the member is in more than one
+	  queue. * Did some refactoring to eliminate some code redundancy.
+	  (issue ASTERISK-16115) Reported by: nik600 Patches:
+	  jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+	  uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+	  changes adding the queues.conf check_state_unknown option as it
+	  was added in an attempt to fix this problem. The fix did not need
+	  to be optional. The fix should not have tried to explicitly set
+	  the device state. Setting the device state by something other
+	  than the device introduces a race condition. I also could not see
+	  how the change would be effective other than delaying the
+	  app_queue code long enough for the device state to propagate to
+	  app_queue. ........ Merged revisions 378663 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2013-01-08 19:13 +0000 [r378658-378662]  Jason Parker <jparker at digium.com>
+
+	* /: Re-enable automerge.
+
+	* include/asterisk/channel.h, main/aoc.c, apps/app_queue.c,
+	  main/cel.c, apps/confbridge/conf_state.c, main/loader.c,
+	  channels/chan_dahdi.c, include/asterisk/event_defs.h,
+	  main/features.c, main/http.c, main/event.c,
+	  apps/app_confbridge.c, apps/confbridge/conf_state_empty.c,
+	  res/res_calendar.c, main/udptl.c, main/stun.c,
+	  channels/chan_sip.c, channels/chan_agent.c, main/devicestate.c,
+	  main/taskprocessor.c, res/res_jabber.c, channels/chan_iax2.c,
+	  apps/confbridge/conf_state_multi_marked.c, main/channel.c,
+	  main/data.c, channels/chan_skinny.c,
+	  apps/confbridge/include/confbridge.h,
+	  include/asterisk/bridging.h, main/file.c, main/image.c,
+	  sounds/Makefile, funcs/func_devstate.c,
+	  channels/sip/include/sip.h, main/timing.c, res/res_clialiases.c,
+	  include/asterisk/devicestate.h, channels/chan_local.c,
+	  main/ccss.c, /, apps/app_meetme.c: Multiple revisions
+	  377838,377842,377848,377882,377923,377947,377992,378037,378089,378093,378120,378218,378286,378320
+	  ........ r377838 | rmudgett | 2012-12-11 14:42:59 -0600 (Tue, 11
+	  Dec 2012) | 16 lines Cleanup taskprocessor on exit. * Cleanup CLI
+	  commands on exit. * v10 only: Merged v1.8 -r374177 change to
+	  taskprocessor.c missed in v10 -r374178. (issue ASTERISK-20649)
+	  Reported by: Corey Farrell Patches:
+	  taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell taskprocessor-cleanup-10-only.patch
+	  (license #5909) patch uploaded by Corey Farrell Modified ........
+	  Merged revisions 377837 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377842 | mmichelson | 2012-12-11 14:48:16 -0600 (Tue, 11 Dec
+	  2012) | 13 lines Fix crash that can occur if CLI registration
+	  fails for an aliased command. A recent memory leak fix in
+	  main/cli.c causes an ast_cli_entry's command field to be freed
+	  and NULLed if ast_cli_register() fails. res_clialiases was
+	  ignoring the return value of ast_cli_register() and was then
+	  passing the NULL command off to a a hash function. This resulted
+	  in a crash. The fix is not to ignore the erroneous return value.
+	  If ast_cli_register() fails, then we do not continue trying to
+	  process the current alias. ........ Merged revisions 377840 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377848 | rmudgett | 2012-12-11 15:07:47 -0600 (Tue, 11 Dec 2012)
+	  | 14 lines Cleanup udptl on exit. * Cleanup CLI commands on exit.
+	  (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+	  Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell Modified ........ Merged revisions
+	  377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ r377882 | rmudgett | 2012-12-11 15:57:44 -0600 (Tue, 11
+	  Dec 2012) | 10 lines Cleanup CLI commands on exit for several
+	  files. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  unregister-cli-multiple-all.patch (license #5909) patch uploaded
+	  by Corey Farrell ........ Merged revisions 377881 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377923 | newtonr | 2012-12-12 16:41:24 -0600 (Wed, 12 Dec 2012)
+	  | 12 lines Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to
+	  1.4.12 for new Extra Sounds releases See CHANGES-* files in
+	  English extra 1.4.12 tarballs for new sound prompts added.
+	  (closes ASTERISK-20328) Reported by: Matt Jordan (closes AST-755)
+	  Reported by: John Bigelow ........ Merged revisions 377922 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377947 | kmoore | 2012-12-13 07:48:32 -0600 (Thu, 13 Dec 2012) |
+	  17 lines Ensure Min-SE is included in outbound INVITEs Asterisk
+	  now includes Min-SE in outbound INVITEs when the value is not 90
+	  (the default) and session timers are not disabled. This has the
+	  effect of Asterisk following RFC4028 more closely with regard to
+	  422 responses and preventing situations in which Asterisk would
+	  be forced to temporarily accept a call to tear it down based on a
+	  Session-Expires below the locally configured Min-SE. (issue
+	  SWP-5051) Review: https://reviewboard.asterisk.org/r/2222/
+	  Reported-by: Kinsey Moore Patch-by: Kinsey Moore ........ Merged
+	  revisions 377946 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377992 | rmudgett | 2012-12-13 14:52:26 -0600 (Thu, 13 Dec 2012)
+	  | 29 lines confbridge: Fix MOH on simultaneous user entry to a
+	  new conference. When two users entered a new conference
+	  simultaneously, one of the callers hears MOH. This happened if
+	  two unmarked users entered simultaneously and also if a
+	  waitmarked and a marked user entered simultaneously. * Created a
+	  confbridge internal MOH API to eliminate the inlined MOH handling
+	  code. Note that the conference mixing bridge needs to be locked
+	  when actually starting/stopping MOH because there is a small
+	  window between the conference join unsuspend MOH and actually
+	  joining the mixing bridge. * Created the concept of suspended MOH
+	  so it can be interrupted while conference join announcements to
+	  the user and DTMF features can operate. * Suspend any MOH until
+	  the user is about to actually join the mixing bridge of the
+	  conference. This way any pre-join file playback does not need to
+	  worry about MOH. * Made post-join actions only play deferred
+	  entry announcement files. Changing the user/conference state
+	  during that time is not protected or controlled by the state
+	  machine. (closes issue ASTERISK-20606) Reported by: Eugenia
+	  Belova Tested by: rmudgett Review:
+	  https://reviewboard.asterisk.org/r/2232/ ........ r378037 |
+	  rmudgett | 2012-12-14 15:30:46 -0600 (Fri, 14 Dec 2012) | 20
+	  lines app_queue: Revert bad ringinuse=no patch. With the option
+	  ringinuse=no set, the patch committed 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
+	  state is considered in use or unknown if the channel exists or
+	  not respectively. (closes issue ASTERISK-20801) Reported by:
+	  rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch
+	  (license #5621) patch uploaded by rmudgett ........ Merged
+	  revisions 378036 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r378089 | rmudgett | 2012-12-17 16:57:10 -0600 (Mon, 17 Dec 2012)
+	  | 16 lines Make chan_local module references tied to local_pvt
+	  lifetime. The chan_local module references were manually tied to
+	  the existence of the ;1 and ;2 channel links. * Made chan_local
+	  module references tied to the existence of the local_pvt
+	  structure as well as automatically take care of the module
+	  references. * Tweaked the wording of the local_fixup() failure
+	  warning message to make sense. Review:
+	  https://reviewboard.asterisk.org/r/2181/ ........ Merged
+	  revisions 378088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r378093 | rmudgett | 2012-12-17 17:08:40 -0600 (Mon, 17 Dec 2012)
+	  | 5 lines Fix potential double free when unloading a module.
+	  ........ Merged revisions 378092 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r378120 | kmoore | 2012-12-18 11:38:22 -0600 (Tue, 18 Dec 2012) |
+	  11 lines Add test events for time limit-related hangups This
+	  patch adds hangup-related test events in order to support testing
+	  of time-limited bridges. This aids in testing the S() and L()
+	  bridge options. (issue SWP-4713) ........ Merged revisions 378119
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r378218 | kmoore | 2012-12-31 08:43:26 -0600 (Mon, 31 Dec 2012) |
+	  12 lines Ensure chan_sip rejects encrypted streams without crypto
+	  info This ensures that Asterisk rejects encrypted media streams
+	  (RTP/SAVP audio and video) that are missing cryptographic keys
+	  and ensures that the incoming SDP is consistent with RFC4568 as
+	  far as having a crypto attribute present for any SAVP streams.
+	  Review: https://reviewboard.asterisk.org/r/2204/ ........ Merged
+	  revisions 378217 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r378286 | mjordan | 2013-01-02 09:23:57 -0600 (Wed, 02 Jan 2013)
+	  | 30 lines 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 (in res_jabber), the allocation has been eliminated
+	  since it was unnecesary. Note that the HTTP portion of this issue
+	  was independently found by Brandon Edwards of Exodus
+	  Intelligence. (issue ASTERISK-20658) Reported by: wdoekes,
+	  Brandon Edwards Tested by: mmichelson, wdoekes patches:
+	  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
+	  5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
+	  wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
+	  uploaded by wdoekes (license 5674) ........ Merged revisions
+	  378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ r378320 | mjordan | 2013-01-02 11:40:28 -0600 (Wed, 02
+	  Jan 2013) | 27 lines 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 calls 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. (issue ASTERISK-20175)
+	  Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by:
+	  kmoore patches: event-cachability-3.diff uploaded by jcolp
+	  (license 5000) ........ Merged revisions 378303 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions
+	  377838,377842,377848,377882,377923,377947,377992,378037,378089,378093,378120,378218,378286,378320
+	  from http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
+	  exit. * Unreference hints and statecbs containers on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
+	  Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
+	  Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
+	  uploaded by Corey Farrell Modified ........ Merged revisions
+	  377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 377807 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* include/asterisk/_private.h, main/indications.c, res/res_srtp.c,
+	  apps/confbridge/conf_config_parser.c, main/logger.c,
+	  main/event.c, main/astmm.c, apps/app_confbridge.c,
+	  addons/cdr_mysql.c, main/stdtime/localtime.c,
+	  codecs/codec_dahdi.c, main/asterisk.c, main/xmldoc.c,
+	  main/format.c, channels/chan_unistim.c,
+	  contrib/realtime/mysql/sippeers.sql, main/dnsmgr.c,
+	  channels/chan_sip.c, /, res/res_fax.c: Multiple revisions
+	  377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772
+	  ........ r377136 | rmudgett | 2012-12-03 14:33:08 -0600 (Mon, 03
+	  Dec 2012) | 17 lines 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 ........
+	  r377166 | rmudgett | 2012-12-03 16:53:58 -0600 (Mon, 03 Dec 2012)
+	  | 15 lines 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 ........
+	  r377212 | rmudgett | 2012-12-04 16:31:02 -0600 (Tue, 04 Dec 2012)
+	  | 1 line confbridge: Update online XML documentation. ........
+	  r377227 | rmudgett | 2012-12-04 18:49:53 -0600 (Tue, 04 Dec 2012)
+	  | 29 lines confbridge: Fix several small issues. * Made
+	  func_confbridge_helper() allow an empty value when setting
+	  options. You previously could not Set(CONFBRIDGE(user,pin)=) and
+	  clear the configured pin from the dialplan. * Made
+	  func_confbridge_helper() handle its datastore better if multiple
+	  threads attempt to set the first CONFBRIDGE option value on the
+	  channel. * Made the func_confbridge_helper() only output one
+	  diagnostic message concerning the option. * Made the bridge
+	  video_mode able to repeatedly change in the config file and
+	  CONFBRIDGE dialplan function. The video_mode option values are an
+	  enum and not independent of each other. * Made
+	  handle_cli_confbridge_show_bridge_profile() better handle the
+	  video_mode option. * Simplified datastore handling code in
+	  conf_find_user_profile() and conf_find_bridge_profile(). * Made
+	  parse_bridge(), parse_user(), and parse_menu() use var->file
+	  instead of CONFBRIDGE_CONFIG because the var could have been from
+	  an include file. (closes issue ASTERISK-20655) Reported by:
+	  Birger "WIMPy" Harzenetter ........ r377241 | rmudgett |
+	  2012-12-04 20:09:13 -0600 (Tue, 04 Dec 2012) | 4 lines * Fix
+	  registering core show codecs/codec CLI commands twice. * Fix
+	  registering atexit format_attr_shutdown() more than once.
+	  ........ r377258 | file | 2012-12-05 10:49:33 -0600 (Wed, 05 Dec
+	  2012) | 19 lines 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 ........
+	  r377261 | jrose | 2012-12-05 10:57:26 -0600 (Wed, 05 Dec 2012) |
+	  15 lines 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 ........
+	  r377354 | rmudgett | 2012-12-06 17:56:45 -0600 (Thu, 06 Dec 2012)
+	  | 24 lines 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. ........ r377382
+	  | kmoore | 2012-12-07 15:58:21 -0600 (Fri, 07 Dec 2012) | 17
+	  lines 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 ........ r377399 | rmudgett | 2012-12-07 17:42:03
+	  -0600 (Fri, 07 Dec 2012) | 5 lines MALLOC_DEBUG: Only wait if we
+	  want atexit allocation dumps. ........ Merged revisions 377398
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377432 | rmudgett | 2012-12-07 18:29:23 -0600 (Fri, 07 Dec 2012)
+	  | 14 lines 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 ........
+	  r377504 | tilghman | 2012-12-09 19:24:41 -0600 (Sun, 09 Dec 2012)
+	  | 5 lines 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 ........
+	  r377510 | tilghman | 2012-12-09 19:39:58 -0600 (Sun, 09 Dec 2012)
+	  | 16 lines 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
+	  ........ r377558 | igorg | 2012-12-09 23:04:36 -0600 (Sun, 09 Dec
+	  2012) | 8 lines Fix crash on transfer initiated from insreeen
+	  menu on Unistim phones. Removed CDR-related code that moved to
+	  do_masquarade before. (closes issue ASTERISK-20417) Reported by:
+	  Rudolf Migalin ........ Merged revisions 377557 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377592 | igorg | 2012-12-10 00:41:47 -0600 (Mon, 10 Dec 2012) |
+	  9 lines Fix codec mismatch Fix code to send in both rx and tx
+	  open stream messages correct codecs. Found that on phase 0/1
+	  phones wrong codecs cause to no audio in some situations. (issue
+	  ASTERISK-20183) ........ Merged revisions 377591 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377624 | kmoore | 2012-12-10 08:40:26 -0600 (Mon, 10 Dec 2012) |
+	  14 lines Handle Session-Expires less than local Min-SE in 200 OK
+	  Ensure that a call is immediately torn down if a Session-Expires
+	  value received in a 200 OK is less than the local Min-SE. This
+	  also prevents Asterisk from allowing calls with Session-Expires
+	  below the RFC4028-mandated minimum (90s). (closes issue
+	  ASTERISK-20653) Review: https://reviewboard.asterisk.org/r/2237/
+	  Patch-by: Kinsey Moore ........ Merged revisions 377623 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377656 | kmoore | 2012-12-10 10:53:16 -0600 (Mon, 10 Dec 2012) |
+	  14 lines Ensure ReceiveFax provides a CED tone via T.38 When
+	  using res_fax_digium, the T.38 CED tone was not being provided
+	  properly which would cause some incoming faxes to fail. This was
+	  not an issue with res_fax_spandsp since it does not strictly
+	  honor the send_ced flag and sends the CED tone whenever receiving
+	  a T.38 fax. (closes issue FAX-343) Reported-by: Benjamin Tietz
+	  Patch-by: Kinsey Moore ........ Merged revisions 377655 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377705 | rmudgett | 2012-12-10 18:32:40 -0600 (Mon, 10 Dec 2012)
+	  | 14 lines Cleanup dnsmgr on exit. * Cleanup dnsmgr thread and
+	  CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
+	  Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909) patch
+	  uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
+	  (license #5909) patch uploaded by Corey Farrell Modified ........
+	  Merged revisions 377704 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377709 | rmudgett | 2012-12-10 19:00:05 -0600 (Mon, 10 Dec 2012)
+	  | 15 lines Cleanup event on exit. * Cleanup CLI commands on exit.
+	  * v10 only: Merged v1.8 -r374177 change to event.c missed in v10
+	  -r374178. (issue ASTERISK-20649) Reported by: Corey Farrell
+	  Patches: event_shutdown-10-only.patch (license #5909) patch
+	  uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch
+	  (license #5909) patch uploaded by Corey Farrell ........ Merged
+	  revisions 377708 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377741 | rmudgett | 2012-12-10 20:11:29 -0600 (Mon, 10 Dec 2012)
+	  | 19 lines Cleanup indications on exit. * Made
+	  ast_unregister_indication_country() unlink the found tone zone
+	  before selecting a new default_tone_zone to make it impossible to
+	  select the tone zone being unregistered again. * Ringcadence is
+	  no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
+	  commands and destroy default_tone_zone on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  indications-cleanup-all.patch (license #5909) patch uploaded by
+	  Corey Farrell Modified ........ Merged revisions 377740 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r377772 | rmudgett | 2012-12-10 20:42:34 -0600 (Mon, 10 Dec 2012)
+	  | 13 lines Cleanup logger on exit. * Cleanup CLI commands,
+	  destroy verbosers and logchannels lists on exit. (issue
+	  ASTERISK-20649) Reported by: Corey Farrell Patches:
+	  logger-cleanup-all.patch (license #5909) patch uploaded by Corey
+	  Farrell Modified ........ Merged revisions 377771 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions
+	  377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772
+	  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
+
+2013-01-08 15:51 +0000 [r378655]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_mixmonitor.c: Fix incorrect pointer manipulation that
+	  caused mixmonitor recording to fail. (closes ASTERISK-20834)
+	  reported by Philippe Lindheimer Patches: ASTERISK-20834.patch
+	  uploaded by Mark Michelson (License #5049)
+
+2012-12-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.12.0-digiumphones-rc1 Released.
+
+2012-12-03 20:25 +0000 [r377068-377134]  Automerge script <automerge at asterisk.org>
+
+	* /: automerge cancel
+
+	* main/cli.c, main/cdr.c, /: Merged revisions 377070,377074 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r377070 | rmudgett | 2012-12-03 12:41:28 -0600
+	  (Mon, 03 Dec 2012) | 15 lines 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
+	  ................ r377074 | rmudgett | 2012-12-03 13:14:14 -0600
+	  (Mon, 03 Dec 2012) | 12 lines Cleanup CLI resources on exit and
+	  CLI command registration errors. (issue ASTERISK-20649) Reported
+	  by: Corey Farrell Patches: cli-leaks-1_8-10.patch (license #5909)
+	  patch uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+	  #5909) patch uploaded by Corey Farrell Modified ........ Merged
+	  revisions 377073 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, main/ccss.c: Merged revisions 377038 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r377038 | rmudgett | 2012-12-03 11:06:44 -0600
+	  (Mon, 03 Dec 2012) | 10 lines Fix CCSS CLI commands and logger
+	  level not unregistered. (issue ASTERISK-20649) Reported by: Corey
+	  Farrell Patches: ccss-cleanup-all.patch (license #5909) patch
+	  uploaded by Corey Farrell ........ Merged revisions 377037 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+2012-11-30 22:25 +0000 [r376656-376982]  Automerge script <automerge at asterisk.org>
+
+	* channels/misdn/isdn_lib.c, /: Merged revisions 376951 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376951 | rmudgett | 2012-11-30 15:33:38 -0600
+	  (Fri, 30 Nov 2012) | 18 lines 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
+	  ................
+
+	* /, channels/chan_sip.c, funcs/func_volume.c: Merged revisions
+	  376916,376920 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376916 | mmichelson | 2012-11-30 10:23:46 -0600
+	  (Fri, 30 Nov 2012) | 23 lines 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
+	  ................ r376920 | seanbright | 2012-11-30 11:06:21 -0600
+	  (Fri, 30 Nov 2012) | 5 lines Minor spelling fix to the VOLUME
+	  documentation. ........ Merged revisions 376919 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* channels/chan_local.c, /, channels/chan_sip.c: Merged revisions
+	  376865,376869 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376865 | rmudgett | 2012-11-29 16:30:26 -0600
+	  (Thu, 29 Nov 2012) | 7 lines Fix compile error. (issue
+	  ASTERISK-20724) ........ Merged revisions 376864 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................ r376869 | rmudgett | 2012-11-29 16:58:28 -0600
+	  (Thu, 29 Nov 2012) | 7 lines chan_local: Fix local_pvt ref leak
+	  in local_devicestate(). Regression introduced by ASTERISK-20390
+	  fix. ........ Merged revisions 376868 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 376835 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376835 | elguero | 2012-11-29 15:51:50 -0600
+	  (Thu, 29 Nov 2012) | 19 lines 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
+	  ................
+
+	* main/astmm.c, main/asterisk.c, /: Merged revisions 376789 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376789 | rmudgett | 2012-11-28 18:45:11 -0600
+	  (Wed, 28 Nov 2012) | 26 lines Add MALLOC_DEBUG atexit unreleased
+	  malloc memory summary. * Adds the following CLI commands to
+	  control MALLOC_DEBUG reporting of unreleased malloc memory when
+	  Asterisk is shut down. memory atexit list on memory atexit list
+	  off memory atexit summary byline memory atexit summary byfunc
+	  memory atexit summary byfile memory atexit summary off * Made
+	  check all remaining allocated region blocks atexit for fence
+	  violations. * Increased the allocated region hash table size by
+	  about three times. It still isn't large enough considering the
+	  number of malloced blocks Asterisk uses. * Made CLI "memory show
+	  allocations anomalies" use regions_check_all_fences(). Review:
+	  https://reviewboard.asterisk.org/r/2196/ ........ Merged
+	  revisions 376788 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* main/astmm.c, /: Merged revisions 376759 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376759 | rmudgett | 2012-11-28 18:05:25 -0600
+	  (Wed, 28 Nov 2012) | 19 lines 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
+	  ................
+
+	* main/manager.c, /: Merged revisions 376726 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376726 | jrose | 2012-11-28 10:30:27 -0600
+	  (Wed, 28 Nov 2012) | 16 lines 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
+	  ................
+
+	* UPGRADE.txt, main/pbx.c, /: Merged revisions 376689 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r376689 | rmudgett | 2012-11-27 17:58:23 -0600
+	  (Tue, 27 Nov 2012) | 33 lines 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

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