[asterisk-commits] bebuild: tag 12.3.0-rc1 r414455 - in /tags/12.3.0-rc1: ./ contrib/realtime/my...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu May 22 11:31:19 CDT 2014


Author: bebuild
Date: Thu May 22 11:31:14 2014
New Revision: 414455

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=414455
Log:
Importing files for 12.3.0-rc1 release.

Added:
    tags/12.3.0-rc1/.lastclean   (with props)
    tags/12.3.0-rc1/.version   (with props)
    tags/12.3.0-rc1/ChangeLog   (with props)
    tags/12.3.0-rc1/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/12.3.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

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Added: tags/12.3.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.3.0-rc1/ChangeLog?view=auto&rev=414455
==============================================================================
--- tags/12.3.0-rc1/ChangeLog (added)
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+2014-05-22  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 12.3.0-rc1 Released.
+
+2014-05-22 16:08 +0000 [r414405]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/stasis_channels.c, res/res_stasis.c,
+	  main/manager_channels.c, main/stasis_endpoints.c,
+	  rest-api/api-docs/events.json, res/stasis/app.c,
+	  res/ari/resource_events.c, include/asterisk/stasis_app.h,
+	  include/asterisk/stasis.h, apps/app_userevent.c,
+	  res/ari/resource_events.h, res/ari/ari_model_validators.c,
+	  CHANGES, main/stasis.c, res/ari/ari_model_validators.h,
+	  include/asterisk/stasis_channels.h, res/res_ari_events.c: ARI:
+	  Add ability to raise arbitrary User Events User events can now be
+	  generated from ARI. Events can be signalled with arbitrary json
+	  variables, and include one or more of channel, bridge, or
+	  endpoint snapshots. An application must be specified which will
+	  receive the event message (other applications can subscribe to
+	  it). The message will also be delivered via AMI provided a
+	  channel is attached. Dialplan generated user event messages are
+	  still transmitted via the channel, and will only be received by a
+	  stasis application they are attached to or if the channel is
+	  subscribed to. This change also introduces the multi object blob
+	  mechanism used to send multiple snapshot types in a single
+	  message. The dialplan app UserEvent was also changed to use multi
+	  object blob, and a new stasis message type created to handle
+	  them. ASTERISK-22697 #close Review:
+	  https://reviewboard.asterisk.org/r/3494/
+
+2014-05-22 16:00 +0000 [r414404]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, apps/app_meetme.c: app_meetme: Don't interrupt MOH for
+	  waitmarked users. Occasionally, when the last marked user leaves
+	  the conference, waitmarked users don't get MOH if MOH is supposed
+	  to be played while a waitmarked user is waiting for another
+	  marked user. * Made not interrupt MOH when the user is a
+	  waitmarked user. The waitmarked user doesn't need to hear any
+	  leave announcements from the conference as the user would have
+	  already heard different leave announcements if they were enabled.
+	  Apparently DAHDI occasionally sends unending non-silent streams
+	  to these users or a normal user still in the conference has
+	  continuous high background noise. These non-silent streams cause
+	  MOH to be suspended while the never ending "announcement" is
+	  played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
+	  by: Tyler Stewart Review:
+	  https://reviewboard.asterisk.org/r/3543/ ........ Merged
+	  revisions 414401 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 414402 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-22 15:44 +0000 [r414400]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c, main/parking.c, main/bridge.c,
+	  main/bridge_basic.c, res/parking/parking_applications.c,
+	  include/asterisk/parking.h, include/asterisk/bridge.h,
+	  res/parking/parking_bridge_features.c, channels/chan_mgcp.c,
+	  res/res_pjsip_refer.c, channels/chan_dahdi.c,
+	  channels/sig_analog.c: res_pjsip_refer: Fix bugs involving
+	  Parking/PJSIP/transfers PJSIP would never send the final 200
+	  Notify for a blind transfer when transferring to parking. This
+	  patch fixes that. In addition, it fixes a reference leak when
+	  performing blind transfers to non-bridging extensions. Review:
+	  https://reviewboard.asterisk.org/r/3485/
+
+2014-05-22 14:01 +0000 [r414330-414347]  Matthew Jordan <mjordan at digium.com>
+
+	* /, UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag ........
+	  Merged revisions 414345 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 414346 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/event.c, main/stasis.c, include/asterisk/devicestate.h,
+	  include/asterisk/event.h, main/stasis_message.c,
+	  include/asterisk/event_defs.h, res/res_corosync.c,
+	  include/asterisk/stasis.h, main/app.c, main/devicestate.c:
+	  res_corosync: Update module to work with Stasis (and compile)
+	  This patch fixes res_corosync such that it works with Asterisk
+	  12. This restores the functionality that was present in previous
+	  versions of Asterisk, and ensures compatibility with those
+	  versions by restoring the binary message format needed to pass
+	  information from/to them. The following changes were made in the
+	  core to support this: * The event system has been partially
+	  restored. All event definition and event types in this patch were
+	  pulled from Asterisk 11. Previously, we had hoped that this
+	  information would live in res_corosync; however, the approach in
+	  this patch seems to be better for a few reasons: (1)
+	  Theoretically, ast_events can be used by any module as a binary
+	  representation of a Stasis message. Given the structure of an
+	  ast_event object, that information has to live in the core to be
+	  used universally. For example, defining the payload of a device
+	  state ast_event in res_corosync could result in an incompatible
+	  device state representation in another module. (2) Much of this
+	  representation already lived in the core, and was not easily
+	  extensible. (3) The code already existed. :-) * Stasis message
+	  types now have a message formatter that converts their payload to
+	  an ast_event object. * Stasis message forwarders now handle
+	  forwarding to themselves. Previously this would result in an
+	  infinite recursive call. Now, this simply creates a new
+	  forwarding object with no forwards set up (as it is the thing it
+	  is forwarding to). This is advantageous for res_corosync, as
+	  returning NULL would also imply an unrecoverable error. Returning
+	  a subscription in this case allows for easier handling of message
+	  types that are published directly to an aggregate topic that has
+	  forwarders. Review: https://reviewboard.asterisk.org/r/3486/
+	  ASTERISK-22912 #close ASTERISK-22372 #close
+
+2014-05-21 22:17 +0000 [r414272]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/core_unreal.c: core_unreal: Only block media frames when
+	  a generator is on both ends of an unreal channel. The fix for
+	  ASTERISK-12292 was a bit too aggressive. You could have
+	  generators pointed at each other on local channels but need to
+	  get other kinds of frames such as DTMF or CONNECTED_LINE frames
+	  accross. ........ Merged revisions 414269 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 414270 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-21 19:07 +0000 [r414216]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* /, funcs/func_strings.c: pbx.c: prevent potential crash from
+	  recursive replace() Recurisve usage of replace() resulted in
+	  corruption of the temporary string storage and potential crash.
+	  By changing the string to be allocated separtely per instance,
+	  this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
+	  Meer ASTEIRSK-23650 #close Review:
+	  https://reviewboard.asterisk.org/r/3539/ ........ Merged
+	  revisions 414214 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 414215 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-19 19:50 +0000 [r414195]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* res/res_stasis_answer.c: Replace __ast_answer with ast_raw_answer
+	  in app_control_answer While load testing an ARI application, I
+	  noticed asterisk was returning HTTP 500 internal server errors on
+	  channels/:id/answer. After talking to #asterisk-dev, the issue
+	  appeared to be a lack of media flowing after __ast_answer() was
+	  called. So now, we call ast_raw_answer instead and no longer wait
+	  for media. ASTERISK-23758 #close Review:
+	  https://reviewboard.asterisk.org/r/3549/
+
+2014-05-19 13:46 +0000 [r414154]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c, /: chan_ooh323: fix h323_log full path name
+	  * fix to use astlogdir option for h323_log file instead of
+	  hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky
+	  Patches: ooh323_logger_patch.diff ........ Merged revisions
+	  414152 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 414153 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-19 01:09 +0000 [r414122-414137]  Matthew Jordan <mjordan at digium.com>
+
+	* main/framehook.c, include/asterisk/channel.h,
+	  bridges/bridge_native_rtp.c, main/bridge_channel.c,
+	  res/res_pjsip_refer.c, res/res_pjsip_session.c, main/channel.c:
+	  Undo r414122 The Test Suite caught a few problems, undoing until
+	  those are resolved
+
+	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
+	  main/bridge_channel.c, res/res_pjsip_session.c, main/channel.c,
+	  main/framehook.c: bridge_native_rtp/bridge_channel: Fix direct
+	  media issues due to frame hook This patch fixes issues with
+	  direct media bridges that occur after a blind transfer. These
+	  issues were caught by the (currently failing)
+	  pjsip/transfers/blind_transfer/caller_direct_media test. The test
+	  currently fails primarily for two reasons: (1) When Bob and
+	  Charlie (the transfer target and the transfer destination) enter
+	  a bridge together, the framehook remains on the transfer target
+	  channel until both channels are in the bridge. As it consumes
+	  voice frames, the initial bridge type is a simple bridge. The
+	  framehook is removed when both channels are in the bridge;
+	  however, this does not currently cause the bridging framework to
+	  re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE
+	  poke to the transfer target channel when a framehook is removed
+	  so the bridge can re-evaluate itself. (2) When a channel leaves a
+	  native RTP bridge, it may be leaving due to being hung up.
+	  Sending a re-INVITE to a channel that is about to be hung up is
+	  not nice - in fact, there's a good chance we'll send the BYE
+	  request before the channel has had a chance to send back a 200
+	  OK. To be somewhat nicer, this patch adds a function to channel.h
+	  that allows the bridging framework to query for exactly why a
+	  channel is leaving a bridge via the channel's soft hangup flags.
+	  This allows it to only send the re-INVITE if there's a chance the
+	  channel will survive the native bridging experience. Review:
+	  https://reviewboard.asterisk.org/r/3535/
+
+2014-05-16 20:05 +0000 [r413993-414069]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone
+	  detection. * Check if waitingfordt (waitfordialtone) is enabled
+	  in dahdi_read() to allow the DSP to operate early enough to
+	  detect dialtone. * Made use the correct variable in
+	  my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
+	  Davies Patches: dialtone_detect_fix (license #5012) patch
+	  uploaded by Steve Davies Review:
+	  https://reviewboard.asterisk.org/r/3534/ ........ Merged
+	  revisions 414067 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 414068 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, channels/sig_pri.c: sig_pri.c: Pull the pri_dchannel()
+	  PRI_EVENT_RING case into its own function. * Populate the
+	  CALLERID(ani2) value (and the special CALLINGANI2 channel
+	  variable) with the ANI2 value in addition to the PRI specific
+	  ANI2 channel variable. * Made complete snapshot staging with the
+	  channel lock held. All channel snapshots need to be done while
+	  the channel lock is held. ........ Merged revisions 414050 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI
+	  conference data structure. Starting a conference recording using
+	  the admin menu overwrites the DAHDI conference data structure
+	  used to modify the admin user's conference mute mode. * Made no
+	  longer pass the user's DAHDI conference data structure into the
+	  menu functions. The menu now uses its own DAHDI conference data
+	  structure to start the recording channel. * Moved the unlock
+	  conf->playlock to before playing the conf-full message. No sense
+	  keeping the lock while that prompt is playing. The user is never
+	  going to get into the conference at that point. ........ Merged
+	  revisions 413991 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413992 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-14 15:39 +0000 [r413896]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* res/res_musiconhold.c, /: res_musiconhold: Minor cleanup. Fix a
+	  few free()'s that should be ast_free()'s. Reverted an old
+	  workaround that isn't necessary. Reorder a tiny bit of code.
+	  Remove a bit of commented-out code. Review:
+	  https://reviewboard.asterisk.org/r/3536/ ........ Merged
+	  revisions 413894 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413895 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-13 18:01 +0000 [r413877]  Jonathan Rose <jrose at digium.com>
+
+	* main/netsock2.c, /, channels/chan_sip.c,
+	  include/asterisk/netsock2.h: chan_sip: Add TLS and SRTP status to
+	  CLI command 'sip show channel' ASTERISK-23564 #close Reported by:
+	  Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/
+	  ........ Merged revisions 413876 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-13 13:52 +0000 [r413789-413792]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* res/res_format_attr_h264.c, /: h264: Fix H264 SDP payload format.
+	  https://tools.ietf.org/html/rfc3984#section-8.1 says
+	  profile-level-id takes 3 bytes in base16 (6 hex digits). This
+	  fixes video setup in certain cases. ASTERISK-23664 #close
+	  ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume
+	  Maudoux. Review: https://reviewboard.asterisk.org/r/3530/
+	  ........ Merged revisions 413791 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/rtp_engine.c, /: rtp: Fix case typo in H263+ mime.
+	  http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
+	  canonical mime subtype is "H263-1998", not "h263-1998". Original
+	  code was added in r183101 on 2009-03-19 02:26:50 +0100. This
+	  fixes issues with Polycom phones. ASTERISK-23665 #close
+	  ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
+	  Maudoux, backported by me. Review:
+	  https://reviewboard.asterisk.org/r/3529/ ........ Merged
+	  revisions 413787 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413788 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-13 00:25 +0000 [r413766-413771]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
+	  PROGRESS events when overlap dialing is enabled. When overlap
+	  dialing is enabled, the lack of inband audio available
+	  information in the SETUP_ACKNOWLEDGE events causes an
+	  interoperability problem with SIP. sig_pri doesn't know if there
+	  is dialtone present when a SETUP_ACKNOWLEDGE is received so it
+	  assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
+	  SIP channel driver then sends out a 183 Session Progress and
+	  blocks the desired 180 Ringing message when the ALERTING message
+	  comes in. * Made the configure script detect if the installed
+	  version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
+	  Using the new API, made generate an AST_CONTROL_PROGRESS frame on
+	  an incoming SETUP_ACKNOWLEDGE message when the message indicates
+	  inband audio is present instead of assuming that dialtone is
+	  present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
+	  inband audio available indication only if dialtone is expected.
+	  The change also makes the fallback behaviour of sending the
+	  PROGRESS message better by sending it only if dialtone is
+	  expected. * Changed receiving a PROCEEDING message to not
+	  generate an AST_CONTROL_PROGRESS frame if the progress indication
+	  ie indicates non-end-to-end-ISDN. This helps interoperability
+	  with SIP. * Changed sending a PROCEEDING message in response to
+	  an AST_CONTROL_PROCEEDING frame to not indicate inband audio
+	  available. It was silly to do so anyway because the channel
+	  driver doesn't know if inband audio is even available. This helps
+	  interoperability with SIP. This patch and a corresponding change
+	  in libpri work together to allow Asterisk to control the inband
+	  audio available progress indication ie on the SETUP_ACKNOWLEDGE
+	  message when dialtone is present. AST-1338 #close Reported by:
+	  Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
+	  ........ Merged revisions 413714 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413765 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup.
+
+2014-05-12 22:23 +0000 [r413712]  Jonathan Rose <jrose at digium.com>
+
+	* /, apps/app_chanspy.c: app_chanspy: Fix a test that was failing
+	  on account of r413551 ASTERISK-23381 #close ASTERISK-23381
+	  #comment Reported by: Robert Moss Review:
+	  https://reviewboard.asterisk.org/r/3505/ ........ Merged
+	  revisions 413710 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-11 02:05 +0000 [r413650-413681]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
+	  include/asterisk/framehook.h, main/channel.c, main/framehook.c,
+	  main/bridge_basic.c: framehooks: Add callback for determining if
+	  a hook is consuming frames of a specific type. In the past
+	  framehooks have had no capability to determine what frame types a
+	  hook is actually interested in consuming. This has meant that
+	  code has had to assume they want all frames, thus preventing
+	  native bridging. This change adds a callback which allows a
+	  framehook to be queried for whether it is consuming a frame of a
+	  specific type. The native RTP bridging module has also been
+	  updated to take advantange of this, allowing native bridging to
+	  occur when previously it would not. ASTERISK-23497 #comment
+	  Reported by: Etienne Lessard ASTERISK-23497 #close Review:
+	  https://reviewboard.asterisk.org/r/3522/
+
+	* main/framehook.c, main/bridge_basic.c,
+	  include/asterisk/channel.h, bridges/bridge_native_rtp.c,
+	  include/asterisk/framehook.h, main/channel.c: Undoing framehook
+	  support. Issues were uncovered by Bamboo.
+
+	* include/asterisk/channel.h, bridges/bridge_native_rtp.c,
+	  include/asterisk/framehook.h, main/channel.c, main/framehook.c,
+	  main/bridge_basic.c: framehooks: Add callback for determining if
+	  a hook is consuming frames of a specific type. In the past
+	  framehooks have had no capability to determine what frame types a
+	  hook is actually interested in consuming. This has meant that
+	  code has had to assume they want all frames, thus preventing
+	  native bridging. This change adds a callback which allows a
+	  framehook to be queried for whether it is consuming a frame of a
+	  specific type. The native RTP bridging module has also been
+	  updated to take advantange of this, allowing native bridging to
+	  occur when previously it would not. ASTERISK-23497 #comment
+	  Reported by: Etienne Lessard ASTERISK-23497 #close Review:
+	  https://reviewboard.asterisk.org/r/3522/
+
+2014-05-09 23:13 +0000 [r413588-413597]  Kinsey Moore <kmoore at digium.com>
+
+	* /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged
+	  revisions 413592 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413595 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/slinfactory.c, main/core_unreal.c, main/acl.c,
+	  res/res_pjsip_t38.c, channels/sig_pri.c, channels/chan_jingle.c,
+	  channels/chan_dahdi.c, channels/sig_analog.c,
+	  include/asterisk/astobj.h, res/res_corosync.c,
+	  res/res_stun_monitor.c, apps/app_sms.c, main/audiohook.c,
+	  pbx/pbx_config.c, channels/iax2/firmware.c, apps/app_adsiprog.c,
+	  channels/chan_sip.c, funcs/func_sysinfo.c, main/utils.c,
+	  res/res_format_attr_h263.c, res/res_jabber.c,
+	  res/res_http_websocket.c, res/res_pktccops.c, res/res_monitor.c,
+	  main/file.c, res/res_pjsip/pjsip_configuration.c, main/adsi.c,
+	  channels/sip/include/sip.h, cel/cel_pgsql.c, main/pbx.c,
+	  res/res_calendar_icalendar.c, res/res_crypto.c, main/aoc.c,
+	  channels/chan_gtalk.c, main/netsock.c, res/res_ari_model.c,
+	  res/res_config_odbc.c, res/res_pjsip_outbound_registration.c,
+	  main/event.c, funcs/func_iconv.c, apps/app_stack.c,
+	  res/res_calendar.c, res/res_sorcery_config.c, main/frame.c,
+	  main/parking.c, res/res_format_attr_h264.c, channels/chan_iax2.c,
+	  apps/confbridge/conf_config_parser.c, funcs/func_hangupcause.c,
+	  main/manager.c, formats/format_pcm.c, funcs/func_srv.c,
+	  res/res_format_attr_silk.c, main/asterisk.c, main/xmldoc.c,
+	  res/res_rtp_asterisk.c, main/format.c, main/ccss.c,
+	  res/res_calendar_caldav.c, main/enum.c, main/config.c,
+	  res/res_srtp.c, main/loader.c,
+	  channels/pjsip/dialplan_functions.c, funcs/func_channel.c,
+	  main/bucket.c, main/abstract_jb.c, res/res_stasis_recording.c,
+	  apps/app_verbose.c, main/dsp.c, apps/app_voicemail.c,
+	  main/stun.c, main/security_events.c, apps/app_festival.c,
+	  res/res_timing_dahdi.c, main/devicestate.c, res/res_xmpp.c,
+	  apps/app_getcpeid.c, main/cli.c, res/res_format_attr_celt.c,
+	  main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c,
+	  funcs/func_frame_trace.c, main/callerid.c, pbx/pbx_dundi.c,
+	  res/res_pjsip_pubsub.c, res/res_fax_spandsp.c,
+	  channels/chan_mgcp.c, res/res_stasis_playback.c, /,
+	  main/translate.c, cdr/cdr_adaptive_odbc.c, res/res_musiconhold.c,
+	  pbx/dundi-parser.c, apps/app_queue.c, res/res_calendar_ews.c,
+	  channels/iax2/parser.c, main/io.c, channels/chan_phone.c,
+	  res/res_agi.c, channels/chan_motif.c, apps/app_minivm.c,
+	  apps/app_dumpchan.c, main/logger.c, apps/app_confbridge.c,
+	  channels/sip/config_parser.c, res/res_odbc.c,
+	  main/manager_channels.c, main/udptl.c, apps/app_dial.c,
+	  res/res_fax.c, funcs/func_env.c, bridges/bridge_softmix.c,
+	  main/taskprocessor.c, res/res_stasis_snoop.c,
+	  res/res_format_attr_opus.c, res/ael/pval.c, main/channel.c,
+	  main/cdr.c, main/data.c, res/res_pjsip/location.c,
+	  main/config_options.c, main/app.c, channels/chan_alsa.c,
+	  main/stdtime/localtime.c, main/bridge_channel.c,
+	  res/res_pjsip_registrar.c, main/sched.c, channels/chan_unistim.c,
+	  main/rtp_engine.c: Allow Asterisk to compile under GCC 4.10 This
+	  resolves a large number of compiler warnings from GCC 4.10 which
+	  cause the build to fail under dev mode. The vast majority are
+	  signed/unsigned mismatches in printf-style format strings.
+	  ........ Merged revisions 413586 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413587 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-09 16:35 +0000 [r413556]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_chanspy.c, /: app_chanspy: Fix a bug where Barge mode
+	  could fail If the barge audiohook was attached prior to the spyee
+	  and its peer actually being bridged, the audiohook would not be
+	  applied and the connected peer would not be able to hear audio
+	  from the spy when the spy is in barge mode. (closes issue
+	  ASTERISK-23381) Reported by: Robert Moss Review:
+	  https://reviewboard.asterisk.org/r/3505/ ........ Merged
+	  revisions 413551 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-08 00:35 +0000 [r413487]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_queue.c, main/manager.c, /: app_queue: Extend
+	  documentation for various Manager actions and events. ........
+	  Merged revisions 413485 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413486 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-07 20:58 +0000 [r413452-413454]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_confbridge.c: app_confbridge: Fixed "CBAnn" channels not
+	  going away. Fixed a ref leak in conf_handle_talker_cb() everytime
+	  the conference bridge was found to report a channel's talker
+	  status change. The resulting leak caused the "CBAnn" channels and
+	  the conference bridge to never be destroyed. Thanks to Richard
+	  Kenner on the asterisk-user's list for locating the problem.
+	  Reported by: Richard Kenner
+
+	* /, apps/app_confbridge.c: app_confbridge: Fix ref leak in CLI
+	  "confbridge kick" command. Fixed ref leak in the CLI "confbridge
+	  kick" command when the channel to be kicked was not in the
+	  conference. ........ Merged revisions 413451 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-07 17:50 +0000 [r413306-413398]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_config_odbc.c, /: Fix encoding of custom prepare extra
+	  data. Patches: res_config_odbc-take2.patch by John Hardin
+	  (License #6512) ........ Merged revisions 413396 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413397 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* res/res_pjsip/presence_xml.c,
+	  res/res_pjsip_pidf_digium_body_supplement.c: Improve XML
+	  sanitization in NOTIFYs, especially for presence subtypes and
+	  messages. Embedded carriage return line feed combinations may
+	  appear in presence subtypes and messages since they may be
+	  derived from user input in an instant messenger client. As such,
+	  they need to be properly escaped so that XML parsers do not vomit
+	  when the messages are received.
+
+	* res/res_pjsip_registrar.c: Check for an act on failures to update
+	  contacts during registration. There was an underlying issue in a
+	  realtime backend where database updates would fail. Since we were
+	  not checking for failure, we would end up in a strange state
+	  where the old database entry was still present but Asterisk
+	  thought that it had been updated. Now when an entry fails to
+	  update, we print a warning and delete the old contact from
+	  sorcery so there is no mismatch between foreground and backend
+	  state. Patches: res_pjsip_registrar.patch by John Hardin (License
+	  #6512)
+
+	* res/res_config_odbc.c, /: Ensure that all parts of SQL UPDATEs
+	  and DELETEs are encoded. Patches: res_config_odbc.patch by John
+	  Hardin (License #6512) ........ Merged revisions 413304 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413305 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-02 20:35 +0000 [r413226-413282]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_config_odbc.c: Correct variable traversal logic in
+	  res_config_odbc's update_odbc function. Closes issue
+	  ASTERISK-23675 Reported by Leando Dardini Patches:
+	  asterisk-23675-odbc-linkedlist-traversal_12.diff uploaded by
+	  Michael L. Young (license #5026)
+
+	* res/res_config_odbc.c, /: Prevent crashes in res_config_odbc due
+	  to uninitialized string fields. Patches: odbc-crash.patch by John
+	  Hardin (License #6512) ........ Merged revisions 413241 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413251 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, res/res_config_pgsql.c: Return the number of rows affected by
+	  a SQL insert, rather than an object ID. The realtime API
+	  specifies that the store callback is supposed to return the
+	  number of rows affected. res_config_pgsql was instead returning
+	  an Oid cast as an int, which during any nominal execution would
+	  be cast to 0. Returning 0 when more than 0 rows were inserted
+	  causes problems to the function's callers. To give an idea of how
+	  strange code can be, this is the necessary code change to fix a
+	  device state issue reported against chan_pjsip in Asterisk 12+.
+	  The issue was that the registrar would attempt to insert contacts
+	  into the database. Because of the 0 return from res_config_pgsql,
+	  the registrar would think that the contact was not successfully
+	  inserted, even though it actually was. As such, even though the
+	  contact was query-able and it was possible to call the endpoint,
+	  Asterisk would "think" the endpoint was unregistered, meaning it
+	  would report the device state as UNAVAILABLE instead of
+	  NOT_INUSE. The necessary fix applies to all versions of Asterisk,
+	  so even though the bug reported only applies to Asterisk 12+, the
+	  code correction is being inserted into 1.8+. Closes issue
+	  ASTERISK-23707 Reported by Mark Michelson ........ Merged
+	  revisions 413224 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413225 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-02 16:33 +0000 [r413210]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c, UPGRADE.txt, res/res_pjsip_refer.c:
+	  res_pjsip_refer: Add Referred-By header on INVITE for blind
+	  transfers. Per rfc3892, the Referred-By header in a REFER must be
+	  copied into the referenced request (IE. The outgoing INVITE to
+	  the transfer target). * Automatically put the Referred-By header
+	  in the outgoing INVITE message if the SIPREFERREDBYHDR channel
+	  variable is defined with a value. * Made
+	  chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance
+	  so chan_pjsip has a better chance to interoperate. * Fixed
+	  refer_blind_callback() and refer_incoming_refer_request() to not
+	  modify the data in the pointer returned by
+	  pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data
+	  since the calling routine doesn't own the buffer. ASTERISK-23501
+	  #close Reported by: John Bigelow Review:
+	  https://reviewboard.asterisk.org/r/3514/
+
+2014-05-02 15:58 +0000 [r413196]  Jonathan Rose <jrose at digium.com>
+
+	* CHANGES, res/parking/parking_bridge_features.c,
+	  res/parking/parking_manager.c, res/parking/res_parking.h:
+	  Parking: Add 'AnnounceChannel' argument to manager action 'Park'
+	  (closes ASTERISK-23397) Reported by: Denis Review:
+	  https://reviewboard.asterisk.org/r/3446/
+
+2014-05-01 15:41 +0000 [r413173]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_pjsip_exten_state.c: Remove unnecessary repetition checks
+	  from res_pjsip_exten_state The PBX core already takes care of
+	  ensuring that repeated state changes are not communicated to
+	  exten state consumers. Because the check in res_pjsip_exten_state
+	  was incomplete, it was causing valid presence state changes not
+	  to be sent out. For instance, if the presence state did not
+	  change but the message or subtype did, then no presence-related
+	  NOTIFY request would be sent out. closes issue ASTERISK-23672
+	  Reported by Mark Michelson
+
+2014-05-01 12:30 +0000 [r413159]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip/config_transport.c: res_pjsip: Add the ability to
+	  configure ciphers based on name. Previously this code would only
+	  accept the OpenSSL identifier instead of the documented name.
+	  ASTERISK-23498 #close ASTERISK-23498 #comment Reported by:
+	  Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/
+
+2014-04-30 20:47 +0000 [r413142]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/message.c, /, channels/chan_sip.c,
+	  include/asterisk/message.h, res/res_pjsip_messaging.c:
+	  chan_sip.c: Fixed off-nominal message iterator ref count and
+	  alloc fail issues. * Fixed early exit in sip_msg_send() not
+	  destroying the message iterator. * Made
+	  ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
+	  tolerant of a NULL iter parameter in case
+	  ast_msg_var_iterator_init() fails. * Made
+	  ast_msg_var_iterator_destroy() clean up any current message data
+	  ref. * Made struct ast_msg_var_iterator,
+	  ast_msg_var_iterator_init(), ast_msg_var_iterator_next(),
+	  ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy()
+	  use iter instead of i. * Eliminated RAII_VAR usage in
+	  res_pjsip_messaging.c:vars_to_headers(). ........ Merged
+	  revisions 413139 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-04-30 20:38 +0000 [r413140]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_pjsip.c: chan_pjsip: Fix deadlock when retrieving
+	  call-id of channel. If a task was in-flight which required the
+	  channel or bridge lock it was possible for the synchronous task
+	  retrieving the call-id to deadlock as it holds those locks. After
+	  discussing with Mark Michelson the synchronous task was removed
+	  and the call-id accessed directly. This should be safe as each
+	  object involved is guaranteed to exist and the call-id will never
+	  change.
+
+2014-04-30 13:06 +0000 [r413124]  Kinsey Moore <kmoore at digium.com>
+
+	* /, res/res_http_websocket.c: Websocket: Add session locking and
+	  delay close This resolves a race condition where data could be
+	  written to a NULL FILE pointer causing a crash as a websocket
+	  connection was in the process of shutting down by adding locking
+	  to websocket session writes and by deferring session teardown
+	  until session destruction. (closes issue ASTERISK-23605) Review:
+	  https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan
+	  ........ Merged revisions 413123 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-04-30 12:41 +0000 [r413117-413121]  Joshua Colp <jcolp at digium.com>
+
+	* res/stasis/control.c: res_stasis: Add progress indications to
+	  operations which perform media. This change fixes operations
+	  which did not account for the fact that they may be executed on
+	  channels which have not been answered. These operations will now
+	  indicate progress when invoked. ASTERISK-23560 #close
+	  ASTERISk-23560 #comment Reported by: Jan Svoboda Review:
+	  https://reviewboard.asterisk.org/r/3495/
+
+	* res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix issue where
+	  sending a hold SDP twice could cause an unhold. This change fixes
+	  a bug where if an SDP with media address and sendonly was
+	  received twice the underlying call would go off hold, instead of
+	  remaining on hold. This occured because the code did not properly
+	  take into account that the SDP may contain both a valid media
+	  address and the sendonly attribute. The code now examines the
+	  sendonly attribute and media address first, so if the SDP is
+	  received again no change will occur. ASTERISK-23558 #comment
+	  Reported by: John Bigelow Review:
+	  https://reviewboard.asterisk.org/r/3472/
+
+	* channels/chan_pjsip.c, res/res_pjsip_session.c: chan_pjsip: Add
+	  support for picking up calls in the configured pickup group.
+	  AST-1363 Review: https://reviewboard.asterisk.org/r/3478/
+
+2014-04-29 15:09 +0000 [r413102]  George Joseph <george.joseph at fairview5.com>
+
+	* include/asterisk/spinlock.h: Add "destroy" implementation for
+	  spinlock. The original commit for spinlock was missing "destroy"
+	  implementations. Most of them are no-ops but phtread_spin and
+	  pthread_mutex do need their locks destroyed.
+
+2014-04-29 11:19 +0000 [r413088]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_pjsip.c: chan_pjsip: Implement core ability to get
+	  Call-ID of a channel. This changes implement the
+	  "get_pvt_uniqueid" which is used to return the technology
+	  specific unique identifier. In the case of SIP this is the
+	  Call-ID of the dialog. Review:
+	  https://reviewboard.asterisk.org/r/3480/
+
+2014-04-28 20:01 +0000 [r413073]  Kinsey Moore <kmoore at digium.com>
+
+	* main/bridge.c, main/bridge_basic.c: Bridging: Don't lock NULL
+	  bridges When bridge locking was added for bridge snapshot
+	  creation, some locations where bridge locking was added were not
+	  guaranteed to actually have a bridge and locking NULL AO2 objects
+	  tends to cause segfaults. This ensures that NULL bridges aren't
+	  locked.
+
+2014-04-25 17:48 +0000 [r413009]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Add support for DTLS
+	  handshake retransmissions On congested networks, it is possible
+	  for the DTLS handshake messages to get lost. This patch adds a
+	  timer to res_rtp_asterisk that will periodically check to see if
+	  the handshake has succeeded. If not, it will retransmit the DTLS
+	  handshake. Review: https://reviewboard.asterisk.org/r/3337
+	  ASTERISK-23649 #close Reported by: Nitesh Bansal patches:
+	  dtls_retransmission.patch uploaded by Nitesh Bansal (License
+	  6418) ........ Merged revisions 413008 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-04-24 14:37 +0000 [r412992]  Kevin Harwell <kharwell at digium.com>
+

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