[svn-commits] bebuild: tag 12.1.0-rc1 r407608 - in /tags/12.1.0-rc1: ./ contrib/realtime/my...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Feb 6 15:36:08 CST 2014
Author: bebuild
Date: Thu Feb 6 15:35:58 2014
New Revision: 407608
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=407608
Log:
Importing files for 12.1.0-rc1 release.
Added:
tags/12.1.0-rc1/.lastclean (with props)
tags/12.1.0-rc1/.version (with props)
tags/12.1.0-rc1/ChangeLog (with props)
tags/12.1.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.1.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.1.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.1.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.1.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.1.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.1.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.1.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.1.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.1.0-rc1/.lastclean?view=auto&rev=407608
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Added: tags/12.1.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.1.0-rc1/ChangeLog?view=auto&rev=407608
==============================================================================
--- tags/12.1.0-rc1/ChangeLog (added)
+++ tags/12.1.0-rc1/ChangeLog Thu Feb 6 15:35:58 2014
@@ -1,0 +1,23829 @@
+2013-02-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.1.0-rc1 Released.
+
+2014-02-06 20:06 +0000 [r407589] Matthew Jordan <mjordan at digium.com>
+
+ * main/security_events.c, UPGRADE.txt, CHANGES: security_events:
+ Add AMI documentation; output optional fields This patch adds
+ documentation for the Security Events that are emited over AMI.
+ It also notes these events in the UPGRADE/CHANGES file.
+
+2014-02-06 19:57 +0000 [r407587] Rusty Newton <rnewton at digium.com>
+
+ * configs/pjsip.conf.sample: configs/pjsip.conf.sample:
+ Configuration section naming in pjsip.conf.sample needs a little
+ clarification There is a bit of nuance to how you name things in
+ pjsip.conf. This is a documentation patch to at least clear it up
+ a little for users. Review:
+ https://reviewboard.asterisk.org/r/3180/
+
+2014-02-06 17:54 +0000 [r407572] Kevin Harwell <kharwell at digium.com>
+
+ * contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py:
+ pjsip realtime: already created enum failure for postgresql If an
+ enum had been previously created the alembic script would attempt
+ to re-create it and an error would be generated while running
+ migrations for a postgresql server. The work around for this is
+ to use the ENUM object type for postgres as opposed to the
+ generic enum type used by sqlalchemy. Using this type in the
+ script seems to work properly for both postgres and mysql.
+
+2014-02-06 17:06 +0000 [r407568] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip_logger.c,
+ res/res_pjsip/include/res_pjsip_private.h,
+ res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
+ res/res_pjsip/config_auth.c, res/res_pjsip/location.c,
+ res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ include/asterisk/res_pjsip_cli.h, res/res_pjsip/pjsip_cli.c,
+ res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip/config_domain_aliases.c: res_pjsip: Updates and
+ adds more PJSIP CLI commands. * Adds identify, transport, and
+ registration support to the PJSIP CLI. * Creates three additional
+ callbacks, one for an iterator, one for a comparator, and one for
+ a container. This eliminates the link dependency from higher
+ level modules to lower level ones. * Eliminates duplicate sorting
+ in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. *
+ Pushes CLI command registration down to the implementing source
+ file. * Adds several ast_sip_destroy_sorcery functions to
+ complement existing ast_sip_sorcery_initialize functions. The
+ destroy functions unregister PJSIP CLI commands and PJSIP CLI
+ formatters. Reported by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3104/
+
+2014-02-06 16:53 +0000 [r407567] Mark Michelson <mmichelson at digium.com>
+
+ * contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py:
+ Fix alembic script to work properly in offline mode. When run in
+ offline mode, this would attempt to check the database for the
+ presence of a type it was going to try to create. I now check the
+ context to see if we're running in offline mode and change a
+ parameter accordingly.
+
+2014-02-05 23:03 +0000 [r407513] Rusty Newton <rnewton at digium.com>
+
+ * /, formats/format_wav.c: formats/format_wav: enhancing log
+ message "Not a wav file" to be clear on what is supported
+ Modifying the log message to be more specific as to what is
+ supported. Specifically it seems format_wav supports only PCM
+ encoded versions with a lower-case '.wav' extension. (closes
+ issues ASTERISK-22310) Reported by: Jim Credland Review:
+ https://reviewboard.asterisk.org/r/3188/ ........ Merged
+ revisions 407511 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407512 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-05 20:55 +0000 [r407461] Jonathan Rose <jrose at digium.com>
+
+ * CHANGES: CHANGES: Improved description of Name/Creator changes to
+ bridge ARI, adds AMI The changes log was written with language
+ that was a little too internal Asterisk specific, so it's been
+ changed to be more in the frame of reference of an ARI user.
+ Also, previously the AMI event changes were omitted from the
+ change log as well as the ability to include a bridge name in the
+ ARI post bridges command.
+
+2014-02-05 20:43 +0000 [r407458] Kinsey Moore <kmoore at digium.com>
+
+ * main/logger.c, /: Logger: Fix handling of absolute paths This
+ fixes path handling for log files so that an extra / is not
+ appended to the file path when the path is absolute (begins with
+ /). This would previously result in different but functionally
+ equivalent paths in the output of 'logger show channels'.
+ ........ Merged revisions 407455 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407456 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-05 19:41 +0000 [r407442] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_pjsip/config_global.c: res_pjsip: When no global type the
+ debug option defaults to "yes" If the global section was not
+ specified in pjsip.conf then the configuration object does not
+ exist in sorcery so when retrieving "debug" option it would
+ return NULL. Then the NULL result was passed to ast_false utils
+ function which would return false because it wasn't set to some
+ representation of false, thus enabling sip debug logging. Made it
+ so if the global config object does not exist then it will return
+ a default of "no" for sip debugging. (issue ASTERISK-23038)
+ Reported by: Rusty Newton
+
+2014-02-05 17:27 +0000 [r407423] Kinsey Moore <kmoore at digium.com>
+
+ * UPGRADE.txt: UPGRADE: Note change in behavior for device state
+ subscriptions
+
+2014-02-05 17:12 +0000 [r407419] Jonathan Rose <jrose at digium.com>
+
+ * CHANGES: CHANGES: Update changes log to include new bridge fields
+ added in r404042
+
+2014-02-05 14:22 +0000 [r407389-407402] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, rest-api/api-docs/channels.json,
+ rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES,
+ include/asterisk/manager.h, rest-api/api-docs/bridges.json,
+ rest-api/api-docs/recordings.json,
+ rest-api/api-docs/deviceStates.json,
+ rest-api/api-docs/endpoints.json,
+ rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
+ rest-api/api-docs/asterisk.json,
+ rest-api/api-docs/applications.json,
+ rest-api/api-docs/playbacks.json: ARI/AMI: Update versions;
+ update UPGRADE/CHANGES notes for 12.1.0 changes Due to backwards
+ compatible changes made to AMI/ARI, the version needs to be
+ bumped to 1.1.0/2.1.0, respectively.
+
+ * rest-api-templates/api.wiki.mustache,
+ rest-api-templates/swagger_model.py: api.wiki.mustache: Update
+ wiki template to support body parameters This patch updates the
+ api.wiki.mustache template and the swagger_model python script to
+ understand if an operation has a body parameter. If an operation
+ does have a body parameter, it will now be displayed in the
+ corresponding wiki entry.
+
+2014-02-04 20:08 +0000 [r407274-407339] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c:
+ devicestate: Make ast_devstate_changed_literal() return value and
+ doxygen consistent. Nothing actually cares about the value
+ anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose
+ ........ Merged revisions 407337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407338 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix assertion for
+ pjsip.conf authorization list options. (closes issue
+ ASTERISK-23168) Reported by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3143/
+
+ * configs/sip.conf.sample, main/tcptls.c, /: tcptls.c: Made TLS
+ handle a certificate chain file. Thanks to Guillaume Martres for
+ doing the necessary research to validate the change. (closes
+ issue ASTERISK-17727) Reported by: LN Patches:
+ use_certificate_chain.patch (license #5864) patch uploaded by st
+ documente_certificate_chain.patch (license #6576) patch uploaded
+ by Guillaume Martres ........ Merged revisions 407272 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407273 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-04 16:54 +0000 [r407259] Matthew Jordan <mjordan at digium.com>
+
+ * funcs/func_cdr.c: funcs/func_cdr: Fix non-epoch timestamps broken
+ by improper char array deref Thanks to snuffy for pointing this
+ issue out and fixing it. (closes issue ASTERISK-23250) Reported
+ by: snuffy patches: func_cdr-fix.diff uploaded by snuffy (License
+ 5024)
+
+2014-02-04 02:21 +0000 [r407213] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_clialiases.c: res_clialiases: Fix crash when reloading
+ and re-aliasing an alias that is in use. The code assumed that
+ unregistering the alias would always succeed while in practice
+ this is not actually true. A common case is the "reload" command
+ itself. If the cli_aliases.conf configuration file was changed
+ and reload executed the command would fail to unregister and
+ ultimately point to freed memory. The reload process now checks
+ whether unregistering succeeded or not and if not the old CLI
+ alias is retained. (closes issue ASTERISK-19773) Reported by:
+ Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth
+ Blades ........ Merged revisions 407205 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407210 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-04 02:04 +0000 [r407197] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Skinny - Fix deadlock when pickup of no
+ call. Locking issues in skinny when picking up a call that
+ doesn't exist. Cleaned up sub locking by fully removing and using
+ the chan lock instead. Also changed ast_call_pickup to check
+ whether chan was masq'd. (closes issue ASTERISK-23249) Reported
+ by: wedhorn Tested by: snuffy, myself Patches:
+ skinny-locking01.diff uploaded by wedhorn (license 5019)
+
+2014-02-03 01:14 +0000 [r407166] Matthew Jordan <mjordan at digium.com>
+
+ * main/cdr.c: cdrs: Check for applications to lock onto during dial
+ begin handling This patch brings CDR processing further in line
+ with r407085. During some dial operations, the application would
+ not be locked to the Dial application and would instead continue
+ to show the previously known application. In particular, this
+ would occur when a Parked call would time out. This was due to a
+ previous snapshot already locking the application to Park -
+ processing this in a Dial Begin allows the Dial application to
+ reassert its rightful place. (CDRs. Ugh.) But hooray for the
+ Parked Call tests for catching this in the Asterisk Test Suite.
+
+2014-02-01 16:23 +0000 [r407153] Joshua Colp <jcolp at digium.com>
+
+ * res/ari/ari_model_validators.c, res/res_stasis.c,
+ main/stasis_bridges.c, res/ari/ari_model_validators.h,
+ rest-api/api-docs/events.json, res/stasis/app.c: res_stasis:
+ Enable transfers and provide events when they occur. This change
+ enables transfers within ARI created bridges and adds events for
+ when they occur. Unlike other events these will be received if
+ *any* subscribed object is involved in the transfer. (closes
+ issue ASTERISK-22984) Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/3120/
+
+2014-02-01 00:24 +0000 [r407104] coreyfarrell <coreyfarrell at localhost>:
+
+ * /, apps/app_stack.c: app_stack: protect against missing
+ parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2
+ parameters and LOCAL_PEEK requires 1 parameter. This protects
+ against situations where those parameters are blank or missing by
+ logging an error and returning. (closes issue ASTERISK-23220)
+ Reported by: James Sharp ........ Merged revisions 407100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 407103 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-31 23:40 +0000 [r407082-407084] Matthew Jordan <mjordan at digium.com>
+
+ * main/manager_channels.c, apps/app_dial.c, main/cdr.c, main/pbx.c,
+ main/bridge_after.c, UPGRADE.txt: CDRs: fix a variety of dial
+ status problems, h/hangup handler creating CDRs This patch fixes
+ a number of small-ish problems that were noticed when witnessing
+ the records that the FreePBX dialplan produces: (1) Mid-call
+ events (as well as privacy options) have the ability to change
+ the overall state of the Dial operation after the called party
+ answers. This means that publishing the DialEnd event when the
+ called party is premature; we have to wait for the execution of
+ these subroutines to complete before we can signal the overall
+ status of the DialEnd. This patch moves that publication and adds
+ handlers for the mid-call events. (2) The AST_FLAG_OUTGOING
+ channel flag is cleared if an after bridge goto datastore is
+ detected. This flag was preventing CDRs from being recorded for
+ all outbound channels that had a 'continue' option enabled on
+ them by the Dial application. (3) The CDR engine now locks the
+ 'Dial' application as being the CDR application if it detects
+ that the current CDR has entered that app. This is similar to the
+ logic that is done for Parking. In general, if we entered into
+ Dial, then we want that CDR to record the application as such -
+ this prevents pre-dial handlers, mid-call handlers, and other
+ shenaniganry from changing the application value. (4) The CDR
+ engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more
+ places to determine if the channel is in hangup logic or dead. In
+ either case, we don't want to record changes in the channel. (5)
+ The default option for "endbeforehexten" has been changed to
+ "yes". In general, you don't want to see CDRs in the 'h' exten or
+ in hangup logic. Since the semantics of that option changed in
+ 12, it made sense to update the default value as well. (6)
+ Finally, because we now have the ability to synchronize on the
+ messages published to the CDR topic, on shutdown the CDR engine
+ will now synchronize to the messages currently in flight. This
+ helps to ensure that all in-flight CDRs are written before
+ shutting down. (closes issue ASTERISK-23164) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/3154
+
+ * apps/app_dial.c, /: app_dial: Allow macro/gosub pre-bridge
+ execution to occur on priorities The parsing for the destination
+ of the macro/gosub uses the '^' character to separate out
+ context, extension, and priority. However, the logic for the
+ macro/gosub execution was written such that it would only do the
+ actual macro/gosub jump if a '^' character existed. This doesn't
+ apply when the macro/gosub jump occurs in a priority/priority
+ label. This patch changes the logic so that the parsing still
+ occurs, but the jump will occur even for priorities/priority
+ labels. (issue ASTERISK-23164) Review:
+ https://reviewboard.asterisk.org/r/3154 ........ Merged revisions
+ 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 407074 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-31 23:14 +0000 [r407034-407036] Kevin Harwell <kharwell at digium.com>
+
+ * contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py
+ (added), configs/pjsip.conf.sample, UPGRADE.txt,
+ res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c:
+ res_pjsip: Config option to enable PJSIP logger at load time.
+ Added a "debug" configuration option for res_pjsip that when set
+ to "yes" enables SIP messages to be logged. It is specified under
+ the "system" type. Also added an alembic script to add the option
+ to realtime. (closes issue ASTERISK-23038) Reported by: Rusty
+ Newton Review: https://reviewboard.asterisk.org/r/3148/
+
+ * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Exporting
+ global symbols caused load order issues Removed the exportation
+ of global symbols from the module as it is no longer needed and
+ it could potentially cause load problems as on some systems it
+ would try to load before res_pjsip_pubsub
+
+2014-01-31 22:38 +0000 [r407031] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/res_pjsip_presence_xml.h (added): Add file that
+ apparently got missed in the merge.
+
+2014-01-31 22:17 +0000 [r407019] Kevin Harwell <kharwell at digium.com>
+
+ * UPGRADE.txt,
+ contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py,
+ contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py:
+ alembic: script modifications due to errors A couple of the
+ scripts had errors that would not allow a full migration to take
+ place. The extensions table needed to make its 'id' column a
+ primary key in order to work with mysql. The other script
+ ...add_endpoints... was missing tables that it was trying to add
+ columns to. Added the primary key on id for extensions and added
+ the tables in for the missing pjsip configuration options. While
+ it is not ideal to modify already released scripts this was a
+ case where it had to be done due to errors in the script and
+ lacking a better alternative. Review:
+ https://reviewboard.asterisk.org/r/3167/
+
+2014-01-31 22:11 +0000 [r407016] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_xpidf_body_generator.c (added),
+ res/res_pjsip_mwi_body_generator.c (added),
+ res/res_pjsip_pubsub.c, res/res_pjsip_pidf.c (removed),
+ res/res_pjsip_pidf_eyebeam_body_supplement.c (added),
+ res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c
+ (added), include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.c (added),
+ include/asterisk/res_pjsip_exten_state.h (removed),
+ res/res_pjsip_pubsub.exports.in,
+ include/asterisk/res_pjsip_body_generator_types.h (added),
+ res/res_pjsip_mwi.c: Decouple subscription handling from
+ NOTIFY/PUBLISH body generation. When the PJSIP pubsub framework
+ was created, subscription handlers were required to state what
+ event they handled along with what body types they knew how to
+ generate. While this serves well when implementing a base RFC, it
+ has problems when trying to extend the body to support
+ non-standard or proprietary body elements. The code also was
+ NOTIFY-specific, meaning that when the time comes that we start
+ writing code to send out PUBLISH requests with MWI or presence
+ bodies, we would likely find ourselves duplicating code that had
+ previously been written. This changeset introduces the concept of
+ body generators and body supplements. A body generator is
+ responsible for allocating a native structure for a given body
+ type, providing the primary body content, converting the native
+ structure to a string, and deallocating resources. A body
+ supplement takes the primary body content (the native structure,
+ not a string) generated by the body generator and adds
+ nonstandard elements to the body. With these elements living in
+ their own module, it becomes easy to extend our support for body
+ types and to re-use resources when sending a PUBLISH request.
+ Body generators and body supplements register themselves with the
+ pubsub core, similar to how subscription and publish handlers had
+ done. Now, subscription handlers do not need to know what type of
+ body content they generate, but they still need to inform the
+ pubsub core about what the default body type for a given event
+ package is. The pubsub core keeps track of what body generators
+ and body supplements have been registered. When a SUBSCRIBE
+ arrives, the pubsub core will check that there is a subscription
+ handler for the event in the SUBSCRIBE, then it will check that
+ there is a body generator that can provide the content specified
+ in the Accept header(s). Because of the nature of body generators
+ and supplements, it means res_pjsip_exten_state and res_pjsip_mwi
+ have been completely gutted. They no longer worry about body
+ types, instead calling ast_sip_pubsub_generate_body_content()
+ when they need to generate a NOTIFY body. Review:
+ https://reviewboard.asterisk.org/r/3150
+
+2014-01-31 22:05 +0000 [r407014] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_pjsip_mwi.c: res_pjsip_mwi: Subscribe fails when missing
+ aor name When subscribing to MWI (res_pjsip_mwi) and the sip uri
+ did not contain a name (ex: sip:<ip address>) then the
+ subscription would fail since it would be unable to locate an
+ associated aor. This patch makes it so that when a subscribe
+ comes with no aor name then it will subscribe to all aors on the
+ located endpoint. (closes issue ASTERISK-23072) Reported by: Bob
+ M Review: https://reviewboard.asterisk.org/r/3164/
+
+2014-01-31 15:01 +0000 [r407000] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_pjsip_nat.c: PJSIP: Fix address for ACK in NAT situations
+ In NAT scenarios where a call is placed to a Grandstream phone,
+ res_pjsip will sometimes send the ACK to a 200 OK to the private
+ address of the device behind the NAT instead of the address of
+ the NAT device. This corrects that behavior by rewriting the
+ address in the Contact header in the incoming 200 OK and the
+ dialog's target address if necessary (since it has already been
+ rewritten to the incorrect private address). (closes issue
+ ASTERISK-23106) Review: https://reviewboard.asterisk.org/r/3168/
+ Reported by: Matt Jordan
+
+2014-01-31 05:28 +0000 [r406987] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Skinny: fix up possible double unlock of
+ chan. Return before chan is possibly unlocked a second time when
+ hanging up a channel in SUBSTATE_OFFHOOK.
+
+2014-01-30 20:34 +0000 [r406935] coreyfarrell <coreyfarrell at localhost>:
+
+ * main/udptl.c, res/res_rtp_asterisk.c, /: res_rtp_asterisk &
+ udptl: fix port selection to work with SELinux restrictions
+ ast_bind to a port reserved for another program by SELinux causes
+ errno == EACCES. This caused random failures when binding rtp or
+ udptl sockets. Treat EACCES as a non-fatal error, try next port.
+ (closes issue ASTERISK-23134) Reported by: Corey Farrell ........
+ Merged revisions 406933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406934 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-30 17:33 +0000 [r406919] Sean Bright <sean at malleable.com>
+
+ * main/manager.c, /: Make a NOTICE about an invalid channel name
+ more useful. ........ Merged revisions 406918 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-29 00:42 +0000 [r406862] Russell Bryant <russell at russellbryant.com>
+
+ * /, configs/queues.conf.sample: queues.conf.sample Fix documented
+ default for persistentmembers Closes issue ASTERISK-22662
+ ........ Merged revisions 406860 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406861 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-28 23:35 +0000 [r406788-406847] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_pjsip_pubsub.c: res_pjsip_pubsub: potential crash on
+ timeout What seems to be happening is if a subscription has been
+ terminated and the subscription timeout/expires is less than the
+ time it takes for all pending transactions (currently on the
+ subscription) to end then the subscription timer will not have
+ been canceled yet and sub will be null. Since the subscription
+ has already been canceled nothing needs to be done so a null
+ check in the asterisk code is sufficient in working around this
+ problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins
+
+ * cdr/cdr_radius.c, cel/cel_radius.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: cdr_radius,
+ cel_radius: build agains libfreeradius-client Asterisk's RADIUS
+ module currently build against libradiusclient-ng, but this
+ project has been superseeded by libfreeradius-client. The API is
+ 99% compatible except that the header name has changed, the
+ library name has changed, and the configuration file location has
+ changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé
+ Patches: freeradius-client.patch uploaded by sharky (license
+ 6561) ........ Merged revisions 406801 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406802 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip/include/res_pjsip_private.h,
+ include/asterisk/compat.h: res_pjsip,compat: INFINITY and NAN
+ undefined On some systems the values for INFINITY and NAN are not
+ defined thus causing a build error on those systems. Added
+ definitions for those if they had not previously been defined.
+ (closes issue ASTERISK-23056) Reported by: capouch Patches:
+ inf-nan-patch.txt uploaded by capouch (license 6564)
+
+2014-01-28 19:13 +0000 [r406775] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_stasis_device_state.c: ARI: Make double subscribe respond
+ with success Currently, attempting to subscribe an application to
+ a device state that it has already subscribed to will generate a
+ 500 error response. This will now be treated as a subscription
+ refresh even though ARI subscriptions don't currently support
+ lifetimes and will respond with the normal response for a
+ successful subscription (200 OK). (closes issue ASTERISK-23143)
+ Reported by: Matt Jordan
+
+2014-01-28 16:41 +0000 [r406723] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/rtp_engine.c, /: rtp_engine: improved handling of
+ get_rtp_info failure In ast_rtp_instance_make_compatible(), after
+ a failure of channel tech call get_rtp_info() to return
+ peer_instance, the null pointer would be passed to ao2_ref,
+ producing an error that looked like a refernce counting problem
+ but is not. This patch corrects that and adds helpful LOG_ERROR
+ messages to indicate which failure path occurred. (issue
+ AST-1276) Review: https://reviewboard.asterisk.org/r/3156/
+ ........ Merged revisions 406721 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406722 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-28 00:11 +0000 [r406707] Richard Mudgett <rmudgett at digium.com>
+
+ * tests/test_cel.c, tests/test_cdr.c: test_cdr.c, test_cel.c:
+ Correctly destroy created bridges. * Fixed the
+ test_cel_attended_transfer_bridges_link unit test to also account
+ for the local channel link being destroyed now that the bridges
+ are actually destroyed. * Made CDR unit test use its own version
+ of do_sleep() from the CEL unit tests.
+
+2014-01-27 20:36 +0000 [r406574-406645] Russell Bryant <russell at russellbryant.com>
+
+ * /, main/config.c: Allow nested #includes in extconfig.conf
+ extconfig.conf was hard-coded to not allow nested includes for
+ some reason. The code has been this way since a patch was merged
+ for ASTERISK-3333 (revision 4889), which was a significant update
+ to this code ("Merge config updates"). I can't figure out any
+ good reason why this should be limited. This patch just removes
+ the limit and uses the default nesting depth limit. Closes issue
+ ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/
+ ........ Merged revisions 406643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406644 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/channel.c, /, main/file.c, include/asterisk/channel.h:
+ Protect ast_filestream object when on a channel The
+ ast_filestream object gets tacked on to a channel via
+ chan->timingdata. It's a reference counted object, but the
+ reference count isn't used when putting it on a channel. It's
+ theoretically possible for another thread to interfere with the
+ channel while it's unlocked and cause the filestream to get
+ destroyed. Use the astobj2 reference count to make sure that as
+ long as this code path is holding on the ast_filestream and
+ passing it into the file.c playback code, that it knows it's
+ valid. Bug reported by Leif Madsen. Review:
+ https://reviewboard.asterisk.org/r/3135/ ........ Merged
+ revisions 406566 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406567 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-26 23:03 +0000 [r406516] Richard Mudgett <rmudgett at digium.com>
+
+ * main/tcptls.c, /: tcptls.c: Add missing cleanup on off nominal
+ path. ........ Merged revisions 406514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406515 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-26 02:10 +0000 [r406489] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_session.c: res_pjsip_session: Be less strict with
+ core requested outgoing capabilities. The core may (depending on
+ circumstances) request a single codec on outgoing calls. Many
+ channel drivers ignore or treat this as a suggestion while still
+ including configured codecs. The res_pjsip_session logic treated
+ this as an explicit request, leaving out other configured codecs.
+ This change makes res_pjsip_session behave like other channel
+ driver and simply adds the requested codec to the list. (closes
+ issue ASTERISK-23082) Reported by: xrobau Review:
+ https://reviewboard.asterisk.org/r/3140/
+
+2014-01-24 23:29 +0000 [r406401-406465] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cel.c, /: CEL: Protect data structures during reload and
+ shutdown. The CEL data structures need to be protected during a
+ configuration reload and shutdown. Asterisk crashed during a
+ shutdown because CEL events were still in flight and the CEL data
+ structures were already destroyed. * Protected the cel_backends,
+ cel_dialstatus_store, and cel_linkedids ao2 containers with a
+ global ao2 object wrapper. * Added NULL checks before use of the
+ cel_backends, cel_dialstatus_store, and cel_linkedids ao2
+ containers in case the CEL module is already shutdown. * Fixed
+ overloading of the cel_linkedids held objects reference count.
+ During shutdown any held objects would be leaked. * Fixed memory
+ leak of cel_linkedids held objects if the LINKEDID_END is not
+ being tracked. The objects in the cel_linkedids container were
+ not removed if the LINKEDID_END event is not used. * Added access
+ protection to the cel_backends container during the CLI "cel show
+ status" command. * Made cel_backends, cel_dialstatus_store, and
+ cel_linkedids use the standard ao2 callback templates for the
+ hash and cmp functions. * Eliminated unnecessary uses of
+ RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated
+ resources on failure. (closes issue AST-1253) Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3128/ ........ Merged
+ revisions 406417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406418 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /: manager: Register atexit shutdown routine only
+ once. * Made register atexit shutdown routine only once in
+ __init_manager(). * Fixed some initial load failure conditions in
+ __init_manager(). * Made reset options to defaults on reload when
+ the reload will actually happen. * Removed unnecessary container
+ traversals of the white/black filters during manager_free_user().
+ * ast_free() does not need a NULL check before calling. ........
+ Merged revisions 406359 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406400 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-24 21:25 +0000 [r406389] Jonathan Rose <jrose at digium.com>
+
+ * res/res_config_pgsql.c, /: res_config_pgsql: Fix a memory leak
+ and use RAII_VAR for cleanup when practical Review:
+ https://reviewboard.asterisk.org/r/3141/ ........ Merged
+ revisions 406360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406361 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-24 18:04 +0000 [r406342] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /: manager: Protect data structures during
+ shutdown. Occasionally, the manager module would get an
+ "INTERNAL_OBJ: bad magic number" error on a "core restart
+ gracefully" command if an AMI connection is established. * Added
+ ao2_global_obj protection to the sessions global container. *
+ Fixed the order of unreferencing a session object in
+ session_destroy(). * Removed unnecessary container traversals of
+ the white/black filters during session_destructor(). (closes
+ issue AST-1242) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3144/ ........ Merged
+ revisions 406341 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-23 23:41 +0000 [r406327] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_pidf.c: Today is not my day for writing code that
+ compiles.
+
+2014-01-23 22:54 +0000 [r406311] Michael L. Young <elgueromexicano at gmail.com>
+
+ * addons/res_config_mysql.c: res_config_mysql: Fix Setting The
+ Column Name Incorrectly When support for a realtime sorcery
+ module was added in revision 386731, the wrong property was
+ accidentally used for setting the column name to be updated in
+ the database table. This patch fixes the typo. (closes issue
+ ASTERISK-23177) Reported by: Denis Tested by: Denis Patches:
+ asterisk-23177-use-field-name.diff by Michael L. Young (license
+ 5026)
+
+2014-01-23 21:09 +0000 [r406294-406295] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_pidf.c: Fix presence body errors found during
+ testing: * PIDF bodies were reporting an "open" state in many
+ cases where it should have been reporting "closed" * XPIDF bodies
+ had XML nodes placed incorrectly within the hierarchy. * SIP URIs
+ in XPIDF bodies did not go through XML sanitization * XML
+ sanitization had some errors: * Right angle bracket was being
+ replaced with "&rt;" instead of ">" * Double quote,
+ apostrophe, and ampersand were not being escaped.
+
+ * res/res_pjsip_pidf.c: Fix presence body errors found during
+ testing: * PIDF bodies were reporting an "open" state in many
+ cases where it should have been reporting "closed" * XPIDF bodies
+ had XML nodes placed incorrectly within the hierarchy. * SIP URIs
+ in XPIDF bodies did not go through XML sanitization * XML
+ sanitization had some errors: * Right angle bracket was being
+ replaced with "&rt;" instead of ">" * Double quote,
+ apostrophe, and ampersand were not being escaped.
+
+2014-01-22 22:23 +0000 [r406264] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * utils/extconf.c, main/pbx.c, /: pbx.c: Pre-initialize timezone to
+ avoid crash on destroy In ast_build_timing, initialize the
+ timezone value to NULL in order to avoid deferencing an
+ uninitialized value later when calling ast_destroy_timing. The
+ timezone value could be uninitialized if ast_build_timing were to
+ fail due to a zero length time string. (closes issue
+ ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review:
+ https://reviewboard.asterisk.org/r/3134/ Patches:
+ ast_build_timing-initialize-timezone.patch uploaded by
+ coreyfarrell (license 5909) ........ Merged revisions 406241 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 406245 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-22 19:34 +0000 [r406152-406223] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_confbridge.c, /: ConfBridge: Fix channel parameter
+ documentation Confbridge AMI and CLI commands for mute, unmute,
+ and setting the single video source can accept channel prefixes
+ in lieu of a full channel name, but documentation states only
+ that it is required and is a channel name. This corrects the
+ documentation. (closes issue PQ-1397) Reported by: Steve Pitts
+ ........ Merged revisions 406217 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
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