[svn-commits] bebuild: tag certified-13.1-cert1 r431519 - in /certified/tags/13.1-cert1: ./...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jan 30 15:55:24 CST 2015
Author: bebuild
Date: Fri Jan 30 15:55:22 2015
New Revision: 431519
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=431519
Log:
Importing files for 13.1-cert1 release.
Modified:
certified/tags/13.1-cert1/.version
certified/tags/13.1-cert1/ChangeLog
certified/tags/13.1-cert1/contrib/realtime/mysql/mysql_config.sql
certified/tags/13.1-cert1/contrib/realtime/oracle/oracle_config.sql
certified/tags/13.1-cert1/contrib/realtime/postgresql/postgresql_config.sql
certified/tags/13.1-cert1/contrib/realtime/sqlserver/mssql_config.sql
Modified: certified/tags/13.1-cert1/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1/.version?view=diff&rev=431519&r1=431518&r2=431519
==============================================================================
--- certified/tags/13.1-cert1/.version (original)
+++ certified/tags/13.1-cert1/.version Fri Jan 30 15:55:22 2015
@@ -1,1 +1,1 @@
-13.1.0
+13.1-cert1
Modified: certified/tags/13.1-cert1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/13.1-cert1/ChangeLog?view=diff&rev=431519&r1=431518&r2=431519
==============================================================================
--- certified/tags/13.1-cert1/ChangeLog (original)
+++ certified/tags/13.1-cert1/ChangeLog Fri Jan 30 15:55:22 2015
@@ -1,3 +1,809 @@
+2015-01-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 13.1-cert1 Released.
+
+2015-01-30 17:53 +0000 [r431494] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_agent_pool.c, /: app_agent_pool: Fix initial module load
+ agent device state reporting. When the app_agent_pool module
+ initially loads there is a race condition between the thread
+ loading agents.conf and the device state internal processing
+ thread. If the device state internal processing thread handles
+ the agent creation state updates before the thread that loaded
+ agents.conf registers the device state provider callback then the
+ cached agent state is "Invalid". When a consumer module like
+ app_queue asks for the agent state it gets the cached "Invalid"
+ state instead of the real state from the provider. * Moved
+ loading the agents.conf configuration to the last thing setup by
+ app_agent_pool in load_module(). Now the device state provider
+ callback is registered before the config is loaded so the agent
+ creation state updates are guaranteed to get the initial device
+ state. * Removed some now redundant config cleanup on error in
+ load_config(). * Added lock protection when accessing the device
+ state in agent_pvt_devstate_get() and eliminated the RAII_VAR()
+ usage. ASTERISK-24737 #close Reported by: Steve Pitts Review:
+ https://reviewboard.asterisk.org/r/4390/ ........ Merged
+ revisions 431492 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-30 16:50 +0000 [r431470] Mark Michelson <mmichelson at digium.com>
+
+ * main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c,
+ res/res_pjsip_refer.c, main/pbx.c, main/manager.c,
+ pbx/pbx_spool.c, /, main/bridge_after.c: Fix some memory leaks.
+ These memory leaks were found and fixed by John Hardin. I'm just
+ committing them for him. ASTERISK-24736 #close Reported by Mark
+ Michelson Review: https://reviewboard.asterisk.org/r/4389
+ ........ Merged revisions 431468 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-30 16:41 +0000 [r431467] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: Merge r431153 from asterisk/branches/13
+ r431153 | jrose | 2015-01-27 11:22:52 -0600 (Tue, 27 Jan 2015) |
+ 9 lines Manager: Fix Manager Action ModuleLoad to give correct
+ response when reloading Prior to this patch, ModuleLoad would
+ respond with an error indicating that the requested module wasn't
+ found in spite of finding and reloading the module. Review:
+ https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 #close
+
+2015-01-28 21:53 +0000 [r431326-431334] Mark Michelson <mmichelson at digium.com>
+
+ * funcs/func_curl.c, /: Multiple revisions 431297-431298 ........
+ r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan
+ 2015) | 17 lines Mitigate possible HTTP injection attacks using
+ CURL() function in Asterisk. CVE-2014-8150 disclosed a
+ vulnerability in libcURL where HTTP request injection can be
+ performed given properly-crafted URLs. Since Asterisk makes use
+ of libcURL, and it is possible that users of Asterisk may get
+ cURL URLs from user input or remote sources, we have made a patch
+ to Asterisk to prevent such HTTP injection attacks from
+ originating from Asterisk. ASTERISK-24676 #close Reported by Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/4364
+ AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49
+ -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from
+ previous patch. ........ Merged revisions 431297-431298 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 431299 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 431301 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_pjsip_t38.c, res/res_pjsip_session.c, /,
+ res/res_pjsip_sdp_rtp.c: Fix file descriptor leak in RTP code.
+ SIP requests that offered codecs incompatible with configured
+ values could result in the allocation of RTP and RTCP ports that
+ would not get reclaimed later. ASTERISK-24666 #close Reported by
+ Y Ateya Review: https://reviewboard.asterisk.org/r/4323
+ AST-2015-001 ........ Merged revisions 431300 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 431303 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-28 04:11 +0000 [r431244] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c,
+ main/sorcery.c: res_pjsip_outbound_registration: Fix reload race
+ condition. Performing a CLI "module reload" command when there
+ are new pjsip.conf registration objects defined frequently failed
+ to load them correctly. What happens is a race condition between
+ res_pjsip pushing its reload into an asynchronous task processor
+ task and the thread that does the rest of the reloads when it
+ gets to reloading the res_pjsip_outbound_registration module. A
+ similar race condition happens between a reload and the CLI/AMI
+ show registrations commands. The reload updates the
+ current_states container and the CLI/AMI commands call
+ get_registrations() which builds a new current_states container.
+ * Made res_pjsip.c reload_module() use
+ ast_sip_push_task_synchronous() instead of ast_sip_push_task() to
+ eliminate two threads processing config reloads at the same time.
+ * Made get_registrations() not replace the global current_states
+ container so the CLI/AMI show registrations command cannot
+ interfere with reloading. You could never add/remove objects in
+ the container without the possibility of the container being
+ replaced out from under you by get_registrations(). * Added a
+ registration loaded sorcery instance observer to purge any dead
+ registration objects since get_registrations() cannot do this job
+ anymore. The struct ast_sorcery_instance_observer callbacks must
+ be used because the callback happens inline with the load
+ process. The struct ast_sorcery_observer callbacks are pushed to
+ a different thread. * Added some global current_states NULL
+ pointer checks in case the container disappears because of
+ unload_module(). * Made sorcery's struct
+ ast_sorcery_instance_observer.object_type_loaded callbacks
+ guaranteed to be called before any struct
+ ast_sorcery_observer.loaded callbacks will be called. * Moved the
+ check for non-reloadable objects to before the sorcery instance
+ loading callbacks happen to short circuit unnecessary work.
+ Previously with non-reloadable objects, the sorcery instance
+ loading/loaded callbacks would always happen, the individual
+ wizard loading/loaded would be prevented, and the non-reloadable
+ type logging message would be logged for each associated wizard.
+ ASTERISK-24729 #close Review:
+ https://reviewboard.asterisk.org/r/4381/ ........ Merged
+ revisions 431243 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 23:02 +0000 [r431200-431221] Kevin Harwell <kharwell at digium.com>
+
+ * main/tcptls.c, /: tcptls: Bad file descriptor error when
+ reloading chan_sip While running through some scenarios using
+ chan_sip and tcp a problem would occur that resulted in a flood
+ of bad file descriptor messages on the cli: tcptls.c:712
+ ast_tcptls_server_root: Accept failed: Bad file descriptor The
+ message is received because the underlying socket has been
+ closed, so is valid. This is probably happening because unloading
+ of chan_sip is not atomic. That however is outside the scope of
+ this patch. This patch simply stops the logging of multiple
+ occurrences of that message. ASTERISK-24728 #close Reported by:
+ Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/
+ ........ Merged revisions 431218 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 431219 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, channels/chan_sip.c: chan_sip: stale nonce causes failure When
+ refreshing (with a small expiration) a registration that was sent
+ to chan_sip the nonce would be considered stale and reject the
+ registration. What was happening was that the initial
+ registration's "dialog" still existed in the dialogs container
+ and upon refresh the dialog match algorithm would choose that as
+ the "dialog" instead of the newly created one. This occurred
+ because the algorithm did not check to see if the from tag
+ matched if authentication info was available after the 401. So,
+ it ended up assuming the original "dialog" was a match and
+ stopped the search. The old "dialog" of course had an old nonce,
+ thus the stale nonce message. This fix attempts to leave the
+ original functionality alone except in the case of a REGISTER. If
+ a REGISTER is received if searches for an existing "dialog"
+ matching only on the callid. If the expires value is low enough
+ it will reuse dialog that is there, otherwise it will create a
+ new one. ASTERISK-24715 #close Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/4367/ ........ Merged
+ revisions 431187 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 431194 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:52 +0000 [r431162] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
+ app_confbridge: Repeatedly starting and stopping recording ref
+ leaks the recording channel. Starting and stopping conference
+ recording more than once causes the recording channels to be
+ leaked. For v13 the channels also show up in the CLI "core show
+ channels" output. * Reworked and simplified the recording channel
+ code to use ast_bridge_impart() instead of managing the recording
+ thread in the ConfBridge code. The recording channel's ref
+ handling easily falls into place and other off nominal code paths
+ get handled better as a result. ASTERISK-24719 #close Reported
+ by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/
+ Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged
+ revisions 431135 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 431160 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:35 +0000 [r431159] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c, main/bridge_channel.c, /: bridge /
+ res_pjsip_sdp_rtp: Fix issues with media not being reinvited
+ during direct media. This change fixes two issues: 1. During a
+ swap operation bridging added the new channel before having the
+ swap channel leave. This was not handled in bridge_native_rtp and
+ could result in a channel not getting reinvited back to Asterisk.
+ After this change the swap channel will leave first and the new
+ channel will then join. 2. If a re-invite was received after a
+ session had been established any upstream elements (such as
+ bridge_native_rtp) were not notified that they may want to
+ re-evaluate things. After this change an UPDATE_RTP_PEER control
+ frame is queued when this situation occurs and upstream can
+ react. AST-1524 #close Review:
+ https://reviewboard.asterisk.org/r/4378/ ........ Merged
+ revisions 431157 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:18 +0000 [r431140] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/confbridge/include/confbridge.h,
+ apps/confbridge/conf_config_parser.c: app_confbridge: Restore
+ user's menu name to CLI output of 'confbridge list' When issuing
+ a 'confbridge list XXXX' CLI command, the resulting output no
+ longer displays the menu associated with a ConfBridge
+ participant. The issue was caused by ASTERISK-22760. When that
+ patch was done, it removed the copying of the menu name
+ associated with the user from the actual user profile. This patch
+ fixes the issue by copying the menu name over to the user profile
+ when the menu hooks are applied to the user. Since that function
+ now does a little bit more than just apply the hooks, the name of
+ the function has been changed to cover the copying of the menu
+ name over as well. In addition, there is a disparity between the
+ menu name length as it is stored on the conf_menu structure and
+ the confbridge_user structure; this patch makes the lengths match
+ so that a strcpy can be used. Review:
+ https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close
+ Reported by: Steve Pitts ........ Merged revisions 431134 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 11:48 +0000 [r431116] Joshua Colp <jcolp at digium.com>
+
+ * res/parking/parking_manager.c, /: res_parking: Fix crash due to
+ race condition when unloading. There is currently a race
+ condition when unloading the res_parking module. Depending on the
+ will of the universe the subscription invocation may occur AFTER
+ the module is unloaded. This is because the module does NOT use
+ stasis_unsubscribe_and_join when terminating the subscription. It
+ merely uses stasis_unsubscribe. This change makes it use
+ stasis_unsubscribe_and_join which is documented for usage in this
+ exact scenario. AST-1520 #close Review:
+ https://reviewboard.asterisk.org/r/4375/ ........ Merged
+ revisions 431114 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-23 15:24 +0000 [r431016] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_ari_events.c, include/asterisk/stasis_app.h,
+ res/res_pjsip_mwi.c, res/parking/parking_applications.c,
+ channels/chan_iax2.c, res/res_pjsip/pjsip_global_headers.c,
+ res/res_pjsip_pubsub.c, res/res_ari_channels.c, res/res_stasis.c,
+ rest-api-templates/param_parsing.mustache, /,
+ res/res_ari_endpoints.c: Investigate and fix memory leaks in
+ Asterisk Fixed memory leaks that were found in Asterisk.
+ ASTERISK-24693 #close Reported by: Kevin Harwell Review:
+ https://reviewboard.asterisk.org/r/4347/ ........ Merged
+ revisions 430999 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-21 19:47 +0000 [r430898] Richard Mudgett <rmudgett at digium.com>
+
+ * CHANGES, /, res/res_pjsip_outbound_registration.c: Multiple
+ revisions 430223,430373,430395 ........ r430223 | gtjoseph |
+ 2015-01-06 11:35:21 -0600 (Tue, 06 Jan 2015) | 24 lines
+ outbound_registration: Add 'pjsip send register' and update 'send
+ unregister' The current behavior of 'pjsip send unregister' is to
+ send the unregister (REGISTER with 0 exp) but let the next
+ scheduled register proceed normally. I don't think that's a good
+ idea. If you unregister, it should stay unregistered until you
+ decide to start registrations again. So this patch just adds a
+ cancel_registration call to the current unregister_task to cancel
+ the timer. Of course, now you need a way to start registration
+ again so I've added a 'pjsip send register' command that
+ unregisters and cancels any existing registration (the same as
+ send unregister), then sends an immediate registration and starts
+ the timer back up again. Both changes also ripple to AMI. There's
+ a new PJSIPRegister command. There's no harm in calling either
+ command repeatedly. They don't care about the actual state.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4301/ ........ r430373 |
+ gtjoseph | 2015-01-08 11:48:29 -0600 (Thu, 08 Jan 2015) | 25
+ lines res_pjsip_outbound_registration: Fix several reload issues
+ There are 2 issues with reloading registrations... 1. The
+ 'can_reuse_registration' test wasn't considering the intervals or
+ expiration in its determination of whether a registration changed
+ or not so if you changed any of the intervals or the expiration
+ and reloaded, the object would get reloaded but the actual timers
+ wouldn't change. can_reuse_registration now does a sorcery diff
+ on the old and new objects instead of discretely testing certain
+ fields. Now if you change expiration for instance, and reload,
+ the timer is updated and re-registration will occur on the new
+ value. 2. If you mung up your password on an outbound
+ registration you get a permanent failure. If you fix the password
+ (on the outbound_auth object) and reload, nothing tells
+ outbound_registration to try again because the registration
+ itself didn't change. This patch adds an observer on the "auth"
+ object type and if any auth changes, existing registration states
+ are searched and those in a REJECTED_PERMANENT state are retried.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4304/ ........ r430395 |
+ gtjoseph | 2015-01-08 15:37:42 -0600 (Thu, 08 Jan 2015) | 14
+ lines res_pjsip_outbound_registration: Fix reference leak. Every
+ time a registration started,
+ sip_outbound_registration_response_cb bumps the ref count on
+ client_state then pushes a handle_registration_response task.
+ handle_registration_response never unreffed it though. So every
+ time a registration goes out, the ref count goes up by one. This
+ patch adds the unreffs to handle_registration_response.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4303/ ........ Merged
+ revisions 430223,430373,430395 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-21 13:36 +0000 [r430843-430865] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: channels/chan_sip: Fix registration leak
+ during reload When the SIP registrations were migrated to using
+ ao2 in what was then trunk, the explicit destruction of the
+ registrations on module reload was removed and not replaced with
+ an ao2 equivalent. Debugging done by Stefan Engström, the issue
+ reporter, on ASTERISK-24673 confirmed that the reference in the
+ registry_list container was being leaked. Since the purpose of
+ cleanup_all_regs is to prep a registration for destruction, this
+ function now calls an ao2_callback function callback with the
+ OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the
+ registrations. This cleans up each registration, and also removes
+ it from the registration container registry_list. Review:
+ https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close
+ Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan
+ Engström Tested by: Stefan Engström ........ Merged revisions
+ 430864 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * apps/app_dial.c, /: apps/app_dial: Don't publish DialEnd twice on
+ unexpected GoSub/Macro values The Dial application has some
+ interesting options with the mid-call Macro (M) and GoSub (U)
+ options. If the MACRO_RESULT/GOSUB_RESULT returns specific
+ values, the Dial application will take some action upon the
+ channels involved in the dial operation (such as hanging up a
+ particular party, etc.) The Dial application ensures that a
+ Stasis message is published in the event that
+ MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial
+ operation, so that there is a corresponding DialEnd event
+ published in AMI/ARI for the DialBegin event that preceeded it. A
+ bug exists where that same DialEnd event will be published on
+ Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is
+ not one that the Dial application cares about. This causes two
+ DialEnd events to be published - one with the
+ MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is
+ all sorts of wrong. This patch fixes the bug by ensuring that we
+ only publish a DialEnd message to Stasis if the Dial
+ application's mid-call Macro/GoSub returns something that Dial
+ cares about. Review: https://reviewboard.asterisk.org/r/4336
+ ASTERISK-24682 #close Reported by: Matt Jordan ........ Merged
+ revisions 430842 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-19 18:18 +0000 [r430782] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c, /: Call extension state callbacks at hint creation.
+ When a hint gets created, any subsequent device or presence state
+ changes result in extension status events getting sent out to
+ interested parties. However, at the time of hint creation, no
+ such event gets sent out, so watchers of extension state are
+ potentially left in the dark until the first state change after
+ hint creation. Patch contributed by John Hardin (License #6512)
+ ........ Merged revisions 430776 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-15 12:11 +0000 [r430666] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_pjsip_outbound_registration.c:
+ res_pjsip_outbound_registration: Fix race condition when
+ reloading and listing registrations. Due to the split of outbound
+ registration state from configuration it is possible during a
+ reload for a "pjsip show registrations" CLI command to be
+ executed which gets an older snapshot of the configuration. This
+ configuration may include outbound registrations which have been
+ removed due to a reload operation occurring at the same time. The
+ code for printing the outbound registration did not take this
+ into account but now it does. AST-1506 #close Review:
+ https://reviewboard.asterisk.org/r/4338/ ........ Merged
+ revisions 430664 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-07 03:29 +0000 [r430253-430293] Matthew Jordan <mjordan at digium.com>
+
+ * utils/conf2ael.c, apps/app_waitforring.c, formats/format_vox.c,
+ res/res_timing_pthread.c, pbx/pbx_ael.c,
+ cel/cel_sqlite3_custom.c, res/res_hep_rtcp.c,
+ formats/format_jpeg.c, apps/app_jack.c, apps/app_adsiprog.c,
+ cdr/cdr_sqlite3_custom.c, res/res_snmp.c, channels/chan_sip.c,
+ cel/cel_tds.c, apps/app_dictate.c, apps/app_festival.c,
+ agi/eagi-test.c, res/res_hep_pjsip.c, apps/app_alarmreceiver.c,
+ apps/app_image.c, channels/chan_console.c, apps/app_getcpeid.c,
+ apps/app_talkdetect.c, channels/chan_oss.c,
+ channels/chan_misdn.c, apps/app_mp3.c, channels/chan_alsa.c,
+ pbx/pbx_dundi.c, channels/chan_nbs.c, utils/extconf.c,
+ apps/app_zapateller.c, cel/cel_pgsql.c, res/res_config_pgsql.c,
+ utils/muted.c, apps/app_test.c, utils/smsq.c,
+ apps/app_morsecode.c, apps/app_ices.c, cdr/cdr_csv.c,
+ channels/chan_phone.c, funcs/func_pitchshift.c,
+ funcs/func_audiohookinherit.c,
+ res/res_pjsip_phoneprov_provider.c, apps/app_minivm.c,
+ res/res_statsd.c, apps/app_sms.c, res/res_config_ldap.c,
+ utils/streamplayer.c, utils/check_expr.c, cel/cel_radius.c,
+ apps/app_nbscat.c, res/res_hep.c, apps/app_waitforsilence.c,
+ apps/app_dahdiras.c, pbx/pbx_lua.c, res/res_ael_share.c,
+ cdr/cdr_radius.c, cdr/cdr_tds.c, utils/stereorize.c,
+ apps/app_osplookup.c, channels/chan_skinny.c,
+ funcs/func_frame_trace.c, apps/app_amd.c, pbx/pbx_realtime.c,
+ apps/app_url.c, apps/app_externalivr.c, cdr/cdr_odbc.c,
+ res/res_timing_kqueue.c, channels/chan_mgcp.c,
+ channels/chan_unistim.c, res/res_phoneprov.c, utils/astman.c,
+ cdr/cdr_pgsql.c, res/res_config_sqlite.c: Disable extended
+ support modules
+
+ * /,
+ contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py:
+ contrib/ast-db-manage: Correct down_revision path for
+ user_eq_phone When the user_eq_phone patch was backported to 13,
+ it referenced the downward revision that the PJSIP optimistic
+ encryption option also references. This creates a multi-path
+ upgrade Exception when generating the SQL files. This patch
+ corrects this in the 13 branch. Note that trunk, which already
+ contained both of these features, is unaffected by this problem.
+ ........ Merged revisions 430252 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-06 19:53 +0000 [r430245] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/bridge_basic.c, /: bridge: avoid leaking channel during
+ blond transfer pt2 A blond transfer to a failed destination, when
+ followed by a recall attempt, lead to a leak of the reference to
+ the destination channel. In addition to correcting the regression
+ on the previous attempt (r429826) this fixes the leak and two
+ additional reference leaks on failures of bridge_import.
+ ASTERISK-24513 #close Review:
+ https://reviewboard.asterisk.org/r/4302/ ........ Merged
+ revisions 430199 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 430200 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-24 15:27 +0000 [r430085-430094] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_agi.c, /: res/res_agi: Make Verbose message for 'stream
+ file' match other playbacks The Verbose message displayed when a
+ file is played back via 'stream file' was formatted differently
+ than other playbacks: * It didn't include the channel name * It
+ didn't include the channel language It does, however, include the
+ playback offset as well as any escape digits. That information
+ was kept; however, this patch updates the formatting to more
+ closely match the Verbose messages displayed when a file is
+ played back by 'control stream file', Playback, ControlPlayback,
+ or any other file playback operation. ........ Merged revisions
+ 429519 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
+ (added), /, res/res_pjsip.c: res_pjsip: Backport missing commits
+ for user_eq_phone This backports the following from trunk, which
+ were missed: r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04
+ Nov 2014) | 2 lines res_pjsip: Allow + at the beginning of a
+ phone number when user_eq_phone is enabled. r427259 | file |
+ 2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip:
+ Apply the 'user_eq_phone' setting to the To header as well. It
+ also adds the Alembic script for the option. ASTERISK-24643
+ ........ Merged revisions 430092 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, tests/test_stasis_channels.c: Stasis: Update unittest for
+ channel snapshots This adjusts the unit test for channel
+ snapshots to take the new language key into account. ........
+ Merged revisions 429352 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h,
+ res/res_pjsip_keepalive.c (added), res/res_pjsip/config_global.c,
+ /, configs/samples/pjsip.conf.sample: res_pjsip_keepalive: Add
+ runtime configurable keepalive module for connection-oriented
+ transports. Note that this is backport from trunk of r425825.
+ This change adds a module which is configurable using the
+ keep_alive_interval setting in the global section that will send
+ a CRLF keep alive to all active connection-oriented transports at
+ the provided interval. This is useful because it can help keep
+ connections open through NATs. This functionality also exists
+ within PJSIP but can not be controlled at runtime and requires
+ recompiling it. Review: https://reviewboard.asterisk.org/r/4084/
+ ASTERISK-24644 #close ........ Merged revisions 430084 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip_caller_id.c, CHANGES, res/res_pjsip.c,
+ include/asterisk/res_pjsip.h: res_pjsip: Add 'user_eq_phone'
+ option to add a 'user=phone' parameter when applicable. Note that
+ this is a backport of r425804 from trunk. This change adds a
+ configuration option which adds a 'user=phone' parameter if the
+ user portion of the request URI or the From URI is determined to
+ be a number. Review: https://reviewboard.asterisk.org/r/4073/
+ ASTERISK-24643 #close ........ Merged revisions 430083 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 21:22 +0000 [r430030-430046] Richard Mudgett <rmudgett at digium.com>
+
+ * main/bridge_basic.c, /: DTMF atxfer: Setup recall channels as if
+ the transferee initiated the call. After the initial DTMF atxfer
+ call attempt to the transfer target fails to answer during a
+ blonde transfer, the recall callback channels do not get setup
+ with information from the initial transferrer channel. As a
+ result, the recall callback to the transferrer does not have
+ callid, channel variables, datastores, accountcode, peeraccount,
+ COLP, and CLID setup. A similar situation happens with the recall
+ callback to the transfer target but it is less visible. The
+ recall callback to the transfer target does not have callid,
+ channel variables, datastores, accountcode, peeraccount, and COLP
+ setup. * Added missing information to the recall callback
+ channels before initiating the call. callid, channel variables,
+ datastores, accountcode, peeraccount, COLP, and CLID * Set callid
+ of the transferrer channel on the DTMF atxfer controller thread
+ attended_transfer_monitor_thread(). * Added missing channel
+ unlocks and props unref to off nominal paths in
+ attended_transfer_properties_alloc(). ASTERISK-23841 #close
+ Reported by: Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/4259/ ........ Merged
+ revisions 430034 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * include/asterisk/_private.h, main/asterisk.c, /, main/logger.c:
+ queue_log: Post QUEUESTART entry when Asterisk fully boots. The
+ QUEUESTART log entry has historically acted like a fully booted
+ event for the queue_log file. When the QUEUESTART entry was
+ posted to the log was broken by the change made by
+ ASTERISK-15863. * Made post the QUEUESTART queue_log entry when
+ Asterisk fully boots. This restores the intent of that log entry
+ and happens after realtime has had a chance to load. AST-1444
+ #close Reported by: Denis Martinez Review:
+ https://reviewboard.asterisk.org/r/4282/ ........ Merged
+ revisions 430009 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 430010 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 18:35 +0000 [r430007-430008] bebuild <bebuild at localhost>:
+
+ * /, res/res_pjsip/pjsip_options.c: Multiple revisions
+ 429128,429246 ........ r429128 | kmoore | 2014-12-09 08:00:50
+ -0600 (Tue, 09 Dec 2014) | 12 lines PJSIP: Stagger outbound
+ qualifies This change staggers initiation of outbound qualify
+ (OPTIONS) attempts to reduce instantaneous server load and
+ prevent network congestion. Review:
+ https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close
+ Reported by: Richard Mudgett ........ Merged revisions 429127
+ from http://svn.asterisk.org/svn/asterisk/branches/12 ........
+ r429246 | kmoore | 2014-12-10 07:14:56 -0600 (Wed, 10 Dec 2014) |
+ 8 lines PJSIP: Fix assert on initial mass qualify This fixes the
+ MWI test regressions caused by r429127 and ensures that contacts
+ have non-zero qualify_frequency before attempting scheduling.
+ ........ Merged revisions 429245 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429128,429246 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * main/manager.c, /: Prevent possible race condition on dual
+ redirect of channels in the same bridge. The
+ AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent
+ bridges from prematurely acting on orphaned channels in bridges.
+ The problem with the AMI redirect action was that it was setting
+ this flag on channels based on the presence of a PBX, not whether
+ the channel was in a bridge. Whether a channel has a PBX is
+ irrelevant, so the condition has been altered to check if the
+ channel is in a bridge. ASTERISK-24536 #close Reported by Niklas
+ Larsson Review: https://reviewboard.asterisk.org/r/4268 ........
+ Merged revisions 429741 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-19 21:52 +0000 [r429855-429892] bebuild <bebuild at localhost>:
+
+ * CHANGES, res/res_ari_channels.c, res/ari/resource_channels.h, /,
+ rest-api/api-docs/channels.json, res/ari/resource_channels.c:
+ ari: Add support for specifying an originator channel when
+ originating. If an originator channel is specified when
+ originating a channel the linked ID of it will be applied to the
+ newly originated outgoing channel. This allows an association to
+ be made between the two so it is known that the originator has
+ dialed the originated channel. ASTERISK-24552 #close Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/
+ ........ Merged revisions 429153 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/ari/ari_model_validators.c, main/manager_channels.c,
+ res/ari/ari_model_validators.h, /, main/stasis_channels.c,
+ rest-api/api-docs/channels.json: ARI/AMI: Include language in
+ standard channel snapshot output The channel "language" was
+ already part of a channel snapshot, however is was not sent out
+ over AMI or ARI. This patch makes it so the channel "language" is
+ included in the appropriate AMI or ARI events. ASTERISK-24553
+ #close Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4245/ ........ Merged
+ revisions 429204 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429206 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_pjsip_session.c, /: res_pjsip_session: Fix issue where a
+ declined media stream in a re-INVITE would fail SDP negotiation.
+ In the past the SDP negotiation within res_pjsip_session was made
+ more tolerant of certain situations. The only case where SDP
+ negotiation will fail is when a major error occurs during
+ negotiation. Receiving an already declined media stream is not
+ considered a major error. When producing the local SDP the logic
+ took this into account so on the initial INVITE the declined
+ media stream did not cause an SDP negotiation failure.
+ Unfortunately the logic for handling media streams with a handler
+ did not mirror this logic and considered an already declined
+ media stream an error and thus failed the SDP negotiation. This
+ change makes the logic between both situations match so only
+ under major errors will the SDP negotiation fail. ASTERISK-24607
+ #close Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4254/ ........ Merged
+ revisions 429407 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * include/asterisk/format.h, main/format.c, /, main/codec.c: media:
+ Fix crash when determining sample count of a frame during
+ shutdown. When shutting down Asterisk the codecs are cleaned up.
+ As a result anything attempting to get a codec based on ID or
+ details will find that no codec exists. This currently occurs
+ when determining the sample count of a frame. This code did not
+ take this situation into account. This change fixes this by
+ getting the codec directly from the format and eliminates the
+ lookup. This is both faster and also provides a guarantee that
+ the codec will exist and will be valid. ASTERISK-24604 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4260/ ........ Merged
+ revisions 429497 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_outbound_registration.c: Prevent potential
+ infinite outbound authentication loops in registration. Prior to
+ this patch, Asterisk would always respond to 401 responses to
+ registration attempts by trying to provide a registration with
+ authentication credentials. Even if subsequent attempts were
+ rejected with 401 responses, Asterisk would continue this
+ behavior. If authentication credentials were incorrect, this
+ could continue forever. With this patch, we keep track of whether
+ we have attempted authentication on an outbound registration
+ attempt. If we already have, we don not try again until the next
+ attempt. This prevents the infinite loop scenario. Review:
+ https://reviewboard.asterisk.org/r/4273 ........ Merged revisions
+ 429761 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * res/res_pjsip_outbound_publish.c, /: res_pjsip_outbound_publish:
+ stack overflow when using non-default sorcery wizard When using a
+ non-default sorcery wizard (in this instance realtime) for
+ outbound publishes Asterisk will crash after a stack overflow
+ occurs due to the code infinitely recursing. The fix entails
+ removing the outbound publish state dependency from the outbound
+ publish sorcery object and instead keeping an in memory container
+ that can be used to lookup the state when needed. ASTERISK-24514
+ #close Reported by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/4178/ ........ Merged
+ revisions 429175 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive'
+ streams for hold This allows use of the 'inactive' stream
+ direction identifier to be used for hold where 'sendonly' is
+ normally used. Some Seimens phones use 'inactive' and this change
+ allows music on hold to operate properly. Review:
+ https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts
+ ........ Merged revisions 429432 from
+ http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+ revisions 429433 from
+ http://svn.asterisk.org/svn/asterisk/branches/13
+
+ * channels/chan_pjsip.c, res/res_pjsip_session.c,
+ include/asterisk/res_pjsip_session.h, /,
+ res/res_pjsip_session.exports.in: res_pjsip_session: Delay
+ sending BYE if a re-INVITE transaction is in progress. Given the
+ scenario where a PJSIP channel is in a native RTP bridge with
+ direct media and the channel is then hung up the code will
+ currently re-INVITE the channel back to Asterisk and send a BYE
+ at the same time. Many SIP implementations dislike this greatly.
+ This change makes it so that if a re-INVITE transaction is in
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