[svn-commits] bebuild: tag 12.3.0-rc1 r414455 - in /tags/12.3.0-rc1: ./ contrib/realtime/my...
SVN commits to the Digium repositories
svn-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)
Added: tags/12.3.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.3.0-rc1/.lastclean?view=auto&rev=414455
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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)
+++ tags/12.3.0-rc1/ChangeLog Thu May 22 11:31:14 2014
@@ -1,0 +1,27423 @@
+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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