[svn-commits] bebuild: tag 12.0.0-beta2 r403153 - in /tags/12.0.0-beta2: ./ contrib/realtim...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 25 15:39:37 CST 2013
Author: bebuild
Date: Mon Nov 25 15:39:35 2013
New Revision: 403153
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=403153
Log:
Importing files for 12.0.0-beta2 release.
Added:
tags/12.0.0-beta2/.lastclean (with props)
tags/12.0.0-beta2/.version (with props)
tags/12.0.0-beta2/ChangeLog (with props)
tags/12.0.0-beta2/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.0.0-beta2/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.0.0-beta2/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.0.0-beta2/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.0.0-beta2/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.0.0-beta2/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.0.0-beta2/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.0.0-beta2/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.0.0-beta2/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.0.0-beta2/.lastclean?view=auto&rev=403153
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URL: http://svnview.digium.com/svn/asterisk/tags/12.0.0-beta2/ChangeLog?view=auto&rev=403153
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--- tags/12.0.0-beta2/ChangeLog (added)
+++ tags/12.0.0-beta2/ChangeLog Mon Nov 25 15:39:35 2013
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+2013-11-25 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.0.0-beta2 Released.
+
+2013-11-23 17:38 +0000 [r403131-403134] Kevin Harwell <kharwell at digium.com>
+
+ * include/asterisk/stasis_app.h, main/devicestate.c,
+ res/stasis/app.h, rest-api/resources.json,
+ res/res_stasis_device_state.c (added),
+ res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
+ res/ari/resource_device_states.c (added),
+ rest-api/api-docs/deviceStates.json (added),
+ rest-api-templates/ari.make.mustache, res/ari.make,
+ rest-api/api-docs/applications.json,
+ include/asterisk/stasis_app_device_state.h (added),
+ res/ari/resource_device_states.h (added),
+ res/ari/resource_applications.h, res/res_stasis.c,
+ include/asterisk/devicestate.h,
+ res/res_stasis_device_state.exports.in (added),
+ rest-api/api-docs/events.json, res/res_ari_device_states.c
+ (added), res/stasis/app.c: ARI: Implement device state API
+ Created a data model and implemented functionality for an ARI
+ device state resource. The following operations have been added
+ that allow a user to manipulate an ARI controlled device:
+ Create/Change the state of an ARI controlled device PUT
+ /deviceStates/{deviceName}&{deviceState} Retrieve all ARI
+ controlled devices GET /deviceStates Retrieve the current state
+ of a device GET /deviceStates/{deviceName} Destroy a device-state
+ controlled by ARI DELETE /deviceStates/{deviceName} The ARI
+ controlled device must begin with 'Stasis:'. An example
+ controlled device name would be Stasis:Example. A
+ 'DeviceStateChanged' event has also been added so that an
+ application can subscribe and receive device change events. Any
+ device state, ARI controlled or not, can be subscribed to. While
+ adding the event, the underlying subscription control mechanism
+ was refactored so that all current and future resource
+ subscriptions would be the same. Each event resource must now
+ register itself in order to be able to properly handle
+ [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/3025/
+
+ * res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_mwi.c, include/asterisk/sorcery.h,
+ res/res_pjsip/pjsip_configuration.c, include/asterisk/strings.h,
+ res/res_pjsip_pubsub.c,
+ res/res_pjsip/include/res_pjsip_private.h,
+ res/res_pjsip/config_transport.c, res/res_pjsip_registrar.c,
+ main/sorcery.c, include/asterisk/res_pjsip.h,
+ include/asterisk/acl.h, res/res_pjsip/config_auth.c,
+ include/asterisk/utils.h, res/res_pjsip.exports.in,
+ res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, main/utils.c,
+ res/res_pjsip.c, res/res_pjsip_exten_state.c,
+ include/asterisk/res_pjsip_pubsub.h: res_pjsip: AMI commands and
+ events. Created the following AMI commands and corresponding
+ events for res_pjsip: PJSIPShowEndpoints - Provides a listing of
+ all pjsip endpoints and a few select attributes on each. Events:
+ EndpointList - for each endpoint a few attributes.
+ EndpointlistComplete - after all endpoints have been listed.
+ PJSIPShowEndpoint - Provides a detail list of attributes for a
+ specified endpoint. Events: EndpointDetail - attributes on an
+ endpoint. AorDetail - raised for each AOR on an endpoint.
+ AuthDetail - raised for each associated inbound and outbound auth
+ TransportDetail - transport attributes. IdentifyDetail -
+ attributes for the identify object associated with the endpoint.
+ EndpointDetailComplete - last event raised after all detail
+ events. PJSIPShowRegistrationsInbound - Provides a detail listing
+ of all inbound registrations. Events: InboundRegistrationDetail -
+ inbound registration attributes for each registration.
+ InboundRegistrationDetailComplete - raised after all detail
+ records have been listed. PJSIPShowRegistrationsOutbound -
+ Provides a detail listing of all outbound registrations. Events:
+ OutboundRegistrationDetail - outbound registration attributes for
+ each registration. OutboundRegistrationDetailComplete - raised
+ after all detail records have been listed.
+ PJSIPShowSubscriptionsInbound - A detail listing of all inbound
+ subscriptions and their attributes. Events: SubscriptionDetail -
+ on each subscription detailed attributes
+ SubscriptionDetailComplete - raised after all detail records have
+ been listed. PJSIPShowSubscriptionsOutbound - A detail listing of
+ all outboundbound subscriptions and their attributes. Events:
+ SubscriptionDetail - on each subscription detailed attributes
+ SubscriptionDetailComplete - raised after all detail records have
+ been listed. (issue ASTERISK-22609) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2959/
+
+2013-11-23 12:51 +0000 [r403117-403119] Joshua Colp <jcolp at digium.com>
+
+ * res/res_stasis_recording.c, res/ari/ari_model_validators.c,
+ rest-api/api-docs/recordings.json,
+ res/ari/ari_model_validators.h, res/res_stasis_playback.c,
+ rest-api/api-docs/events.json: ari: Add events for playback and
+ recording. While there were events defined for playback and
+ recording these were not actually sent. This change implements
+ the to_json handlers which produces them. (closes issue
+ ASTERISK-22710) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/3026/
+
+ * res/res_ari_channels.c, res/ari/resource_channels.h,
+ res/res_stasis_snoop.exports.in (added),
+ include/asterisk/stasis_app_snoop.h (added),
+ rest-api/api-docs/channels.json, main/audiohook.c,
+ res/ari/resource_channels.c, res/res_stasis_snoop.c (added): ari:
+ Add Snoop operation for spying/whispering on channels. The Snoop
+ operation can be invoked on a channel to spy or whisper on it. It
+ returns a channel that any channel operations can then be invoked
+ on (such as record to do monitoring). (closes issue
+ ASTERISK-22780) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3003/
+
+2013-11-22 23:44 +0000 [r403094] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_stasis.c, tests/test_stasis_channels.c: Make sure unit
+ tests compile This fixes the unit tests that were broken by
+ r403069 and several functions requiring a new parameter for
+ sanitization of JSON messages generated from object snapshots.
+
+2013-11-22 22:24 +0000 [r403082] Kevin Harwell <kharwell at digium.com>
+
+ * contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
+ res/res_pjsip/pjsip_configuration.c: res_pjsip: convert
+ configuration settings names to snake case some more Updated the
+ alembic script for pjsip. Also, the dtls config parsing stuff was
+ expecting strings with no underscores, so removed the underscores
+ from the option name before passing it to the parser.
+
+2013-11-22 20:01 +0000 [r403069] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_stasis.c, main/stasis_endpoints.c,
+ res/ari/resource_endpoints.c, main/rtp_engine.c,
+ res/stasis/app.c, include/asterisk/stasis_bridges.h,
+ include/asterisk/stasis_app.h, include/asterisk/stasis.h,
+ main/stasis_bridges.c, res/ari/resource_bridges.c, main/json.c,
+ main/stasis_message.c, include/asterisk/stasis_channels.h,
+ main/stasis_channels.c, res/ari/resource_channels.c,
+ include/asterisk/stasis_endpoints.h: ARI: Don't leak
+ implementation details This change prevents channels used as
+ implementation details from leaking out to ARI. It does this by
+ preventing creation of JSON blobs of channel snapshots created
+ from those channels and sanitizing JSON blobs of bridge snapshots
+ as they are created. This introduces a framework for excluding
+ information from output targeted at Stasis applications on a
+ consumer-by-consumer basis using channel sanitization callbacks
+ which could be extended to bridges or endpoints if necessary.
+ This prevents unhelpful error messages from being generated by
+ ast_json_pack. This also corrects a bug where BridgeCreated
+ events would not be created. (closes issue ASTERISK-22744)
+ Review: https://reviewboard.asterisk.org/r/2987/ Reported by:
+ David M. Lee
+
+2013-11-22 17:19 +0000 [r403022] Kevin Harwell <kharwell at digium.com>
+
+ * configs/pjsip.conf.sample, res/res_pjsip/config_system.c,
+ contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
+ res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c,
+ res/res_pjsip.c, res/res_pjsip/config_transport.c,
+ res/res_pjsip/config_global.c: res_pjsip: convert configuration
+ settings names to snake case Renamed, where appropriate, the
+ configuration options for chan/res_pjsip to use snake case
+ (compound words separated by an underscore). For example,
+ faxdetect will become fax_detect, recordofffeature will become
+ record_off_feature, etc... Review:
+ https://reviewboard.asterisk.org/r/3002/
+
+2013-11-22 17:11 +0000 [r403016] Joshua Colp <jcolp at digium.com>
+
+ * /, main/translate.c: translate: Move freeing of frame to after it
+ is used. When translating from one format to another it is
+ possible to inform the translation function that the source frame
+ should be freed. This was previously done immediately but shortly
+ afterwards the frame that was freed was accessed and used again.
+ This change moves code around a bit so that the frame is now
+ freed after it has been completely used. (closes issue
+ ASTERISK-22788) Reported by: Corey Farrell Patches:
+ translate-access-after-free-11up.patch uploaded by coreyfarrell
+ (license 5909) translate-access-after-free-1.8.patch uploaded by
+ coreyfarrell (license 5909) ........ Merged revisions 403014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 403015 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-21 22:35 +0000 [r402981-402993] David M. Lee <dlee at digium.com>
+
+ * rest-api-templates/res_ari_resource.c.mustache,
+ rest-api-templates/ari_resource.c.mustache: ari: Fix #include to
+ match generated headers for snakeCase resource files
+
+ * rest-api-templates/make_ari_stubs.py: ari: Fix generators for
+ resources with camelCase names. For the new deviceState resource,
+ we need to properly generate device_state.[ch] files.
+
+2013-11-21 19:21 +0000 [r402968] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_session.c: res_pjsip_session: Fix memory leak of
+ direct media format capabilities The direct media format
+ capabilities are always allocated in ast_sip_session_alloc and
+ were not freed in the session destructor. Whoops. (This being the
+ third whoops caught by Scott and Nitesh's valgrind work for the
+ Asterisk Test Suite. Nifty!)
+
+2013-11-21 19:08 +0000 [r402944-402956] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/app.h: voicemail: Fixup some doxygen comments.
+
+ * main/bucket.c: bucket: Fix scheme ref leak in
+ __ast_bucket_scheme_register().
+
+2013-11-21 17:52 +0000 [r402940-402941] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix use of
+ uninitialized value in PJSIP In PJMEDIA,
+ pjmedia_sdp_rtpmap_to_attr will attempt to use the string
+ rtpmap.param regardless of its length value. Simply setting the
+ length to 0 does not prevent the garbage on the stack in
+ rtpmap.param.ptr from being formatted in a sprintf call. This
+ patch initializes the string to NULL so that at the very least,
+ something is provided to the function that is predictable.
+
+ * res/res_pjsip_mwi.c: res_pjsip_mwi: Fix memory leak of MWI
+ subscriptions container This patch fixes a reference counting
+ memory leak on the ao2_container created as part of
+ create_mwi_subscriptions. When we create the container in this
+ routine, the intent is to hand lifetime ownership over to the
+ global container unsolicited_mwi. When
+ ao2_global_obj_replace_unref is called, the reference count on
+ mwi_subscriptions (the container) will be bumped by 1; however,
+ the function does not decrement the reference count on
+ mwi_subscriptions when this occurs. This will prevent the
+ container from being fully disposed of when Asterisk exits (or on
+ any subsequent call to this operation, such as during a reload).
+
+2013-11-21 15:55 +0000 [r402926] David M. Lee <dlee at digium.com>
+
+ * include/asterisk/stasis_app.h, rest-api/api-docs/channels.json,
+ res/ari/resource_channels.c, res/res_ari_channels.c,
+ res/ari/resource_channels.h, res/stasis/control.c: ari: Add
+ silence generator controls This patch adds the ability to start a
+ silence generator on a channel via ARI. This generator will play
+ silence on the channel (avoiding audio timeouts on the peer)
+ until it is stopped, or some other media operation is started
+ (like playing media, starting music on hold, etc.). (closes issue
+ ASTERISK-22514) Review: https://reviewboard.asterisk.org/r/3019/
+
+2013-11-19 23:17 +0000 [r402891] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_caller_id.c: res_pjsip_caller_id: Don't overwrite
+ user portion of the From header when fromuser is set. The
+ fromuser option is used to explicitly set the user within the
+ From header. The res_pjsip_caller_id module did not take this
+ setting into account when determining if the From header could be
+ modified or not. (closes issue ASTERISK-22866) Reported by:
+ Anthony Messina
+
+2013-11-16 13:44 +0000 [r402864] Joshua Colp <jcolp at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_pjsip/pjsip_distributor.c: res_pjsip: Add support for
+ building against pjproject with SIP transaction group lock
+ support. SIP transaction group lock support has been backported
+ into our pjproject. Since the code now internally uses a group
+ lock the code is now changed to unlock it if present. Note that
+ the act of finding the transaction is what actually returns it
+ locked. For further information about group locks check out the
+ wiki page at: http://trac.pjsip.org/repos/wiki/Group_Lock (issue
+ ASTERISK-22818) Reported by: Matt Jordan
+
+2013-11-15 14:35 +0000 [r402838] Kinsey Moore <kmoore at digium.com>
+
+ * main/cel.c: CEL: Fix crash when using CELGenUserEvent This fixes
+ a crash when CELGenUserEvent is called from the dialplan while
+ CEL is disabled. Currently, CEL does not create its topics and
+ forwards if it is not enabled and external entities may depend on
+ these topics blindly since they should always be available. This
+ patch breaks up route creation and topic/forward creation such
+ that the CEL topics and forwards will always exist while the
+ router and its associated routes will be torn down and recreated
+ as necessary. (closes issue ASTERISK-22799) Review:
+ https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan
+
+2013-11-14 15:01 +0000 [r402817] David M. Lee <dlee at digium.com>
+
+ * res/res_stasis.c: stasis: Fixed scoping problem with bridge
+ tracking.
+
+2013-11-13 23:09 +0000 [r402804] Joshua Colp <jcolp at digium.com>
+
+ * res/res_ari_channels.c, res/ari/resource_channels.h,
+ res/stasis/control.c, include/asterisk/stasis_app.h,
+ rest-api/api-docs/channels.json, res/ari/resource_channels.c:
+ res_ari_channels: Add the ability to stop locally generated
+ ringing on a channel. Using the 'ring' operation it is possible
+ to start locally generated ringback if the channel is answered.
+ This change adds the ability to stop it by using DELETE.
+
+2013-11-12 23:16 +0000 [r402787-402793] Kevin Harwell <kharwell at digium.com>
+
+ * res/ari/resource_endpoints.c: ari endpoints: GET
+ /ari/endpoints/{invalid-tech} should return a 404 Was returning a
+ 404 on a valid technology with an empty list of endpoints. Now
+ checking against the channel tech to make sure the tech itself is
+ valid and not just an empty list of endpoints. (issue
+ ASTERISK-22803) Reported by: David M. Lee
+
+ * rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c,
+ res/res_ari_endpoints.c: ari endpoints: GET
+ /ari/endpoints/{invalid-tech} should return a 404 Implementation
+ listing endpoints by technology returned an empty array if no
+ matching endpoints were found. Fixed so a "404 Not Found" will be
+ returned instead. (closes issue ASTERISK-22803) Reported by:
+ David M. Lee
+
+2013-11-12 19:11 +0000 [r402767-402769] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Switch to a scoped lock to avoid missing unlocks
+ in failure returns.
+
+ * main/channel.c: Move a NULL check to a place that makes more
+ sense. Two variables were being checked for NULLity immediately
+ after being declared NULL. I moved the NULL check until after the
+ variables are allocated. This allows for the "channelvars" option
+ in manager.conf to work as intended again.
+
+2013-11-12 16:45 +0000 [r402757] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_pjsip_header_funcs.c, res/res_pjsip_messaging.c:
+ pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer
+ dereferences Both res_pjsip_messaging and res_pjsip_header_funcs
+ were causing asterisk to crash because they were trying to
+ dereference a NULL pointer. In the case of res_pjsip_messaging it
+ was attempting to "print" a contact header that did not exist. In
+ fact contact headers should not be part of a SIP MESSAGE, so the
+ offending code was simply removed. In the case of
+ res_pjsip_header_funcs a null private channel tech was being
+ passed to the function and then later dereferenced. Added null
+ checks (and error logging) to the read/write function handlers to
+ guard against crashing. (closes issue ASTERISK-22821) Reported
+ by: Anthony Messina
+
+2013-11-12 16:33 +0000 [r402755] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_celgenuserevent.c: CELGenUserEvent: Fix error message
+ from ast_json_pack This prevents NULL from being passed into an
+ ast_json_pack call when no extra information is passed to the
+ application which prevents an error message about NULL arguments
+ from being generated.
+
+2013-11-12 15:26 +0000 [r402738] David M. Lee <dlee at digium.com>
+
+ * res/ari/ari_model_validators.h, rest-api/api-docs/events.json:
+ Fixed a typ.
+
+2013-11-12 15:02 +0000 [r402710] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID
+ read Asterisk will sometimes core dump during caller id read on
+ analog channels due to a negative return value from the read() in
+ my_get_callerid that slips through as a negative length argument
+ to callerid_feed() if the errno returned by DAHDI is ELAST. This
+ change ensures that the negative return is treated properly even
+ when it is ELAST. (closes issue ASTERISK-22746) Reported by:
+ Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
+ uploaded by Michael Walton (License 6502) ........ Merged
+ revisions 402708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 402709 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-11 19:26 +0000 [r402687] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_confbridge.c: Get rid of some inaccurate comments.
+ I'm doing some unrelated work in app_confbridge and finding these
+ "invalid pin" comments to be annoying. Get out! ........ Merged
+ revisions 402686 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-11 15:36 +0000 [r402647] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_queue.c, /: app_queue: Honor penalty limits of 0 In the
+ current app_queue code from 1.8 up to trunk the upper and lower
+ penalties can be set to 0 but the value is interpreted to be
+ disabled instead of actually setting limits. This is especially
+ evident if min and max limits are set to 0 and members with
+ penalties of 0 and 1 are in the queue since the member with
+ penalty 1 will still receive calls. This patch adjusts the
+ special disabled value to be INT_MAX instead of 0. (closes issue
+ ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
+ Reported by: Schmooze Com ........ Merged revisions 402645 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 402646 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-08 23:04 +0000 [r402606] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ keep same local (from) tag for outgoing register requests For
+ outbound register requests the tag on the From line was updated
+ every 20 seconds prior to a successful registration and also once
+ for each registration renewal. That behavior can possibly cause
+ the registration to be denied because of the different tag, and
+ is not aligned with the intention of RFC 3261 8.1.3.5 "...
+ request constitutes a new transaction and SHOULD have the same
+ value of the Call-ID, To, and From of the previous request...".
+ This updates chan_sip to have a field to keep the local tag in
+ the registration structure and use that tag for registration
+ requests where the callid is also unchanged. (closes issue
+ ASTERISK-12117) Reported by: Pawel Pierscionek Review:
+ https://reviewboard.asterisk.org/r/2988/ ........ Merged
+ revisions 402604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 402605 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-08 20:20 +0000 [r402593] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_stasis.c: res_stasis.c: Fix locking issues with the
+ app_bridge_moh container. * Fix unlinking from the
+ app_bridges_moh container in remove_bridge_moh() without a lock
+ under normal circumstances. * Made check
+ ast_bridge_set_after_callback() return value in
+ bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK()
+ locking over too much scope in stasis_app_bridge_moh_channel()
+ and stasis_app_bridge_moh_stop(). * Fixed unusual usage of
+ ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge
+ from off nominal path in stasis_app_bridge_create(). * Fixed
+ strange construct in stasis_app_unsubscribe(). From a bad merge?
+ * Made load_module() cleanup on failure. Review:
+ https://reviewboard.asterisk.org/r/2962/
+
+2013-11-08 19:28 +0000 [r402584] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, main/security_events.c,
+ configs/manager.conf.sample, CHANGES, include/asterisk/manager.h:
+ security_events: Push out security events over AMI events
+ Security Events will now be written to any listener of the new
+ 'security' class Review: https://reviewboard.asterisk.org/r/2998/
+
+2013-11-08 19:22 +0000 [r402582] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip.c: Clarify an ambiguous error message.
+
+2013-11-08 18:48 +0000 [r402561-402570] David M. Lee <dlee at digium.com>
+
+ * res/res_pjsip/config_system.c: res_pjsip: Print a helpful error
+ message if sorcery registration fails
+
+ * res/ari/resource_playbacks.h: Changes from make ari-stubs after
+ r402560
+
+2013-11-08 17:39 +0000 [r402560] Kevin Harwell <kharwell at digium.com>
+
+ * res/ari.make, rest-api/api-docs/playback.json (removed),
+ res/ari/resource_playback.c (removed), res/res_ari_playback.c
+ (removed), rest-api/api-docs/playbacks.json (added),
+ res/ari/resource_playbacks.c (added), rest-api/resources.json,
+ res/ari/resource_playback.h (removed), res/res_ari_playbacks.c
+ (added), res/ari/resource_playbacks.h (added): ARI playback:
+ Rename ARI Playback to Playbacks Before playback was the only non
+ plural resource. It has been renamed to playbacks for
+ consistency. (closes issue ASTERISK-22737) Reported by: Paul
+ Belanger
+
+2013-11-08 17:28 +0000 [r402555] David M. Lee <dlee at digium.com>
+
+ * res/res_ari.c, main/manager.c, main/http.c: ari: Add
+ application/x-www-form-urlencoded parameter support ARI POST
+ calls only accept parameters via the URL's query string. While
+ this works, it's atypical for HTTP API's in general, and
+ specifically frowned upon with RESTful API's. This patch adds
+ parsing for application/x-www-form-urlencoded request bodies if
+ they are sent in with the request. Any variables parsed this way
+ are prepended to the variable list supplied by the query string.
+ (closes issue ASTERISK-22743) Review:
+ https://reviewboard.asterisk.org/r/2986/
+
+2013-11-07 23:16 +0000 [r402537] Jonathan Rose <jrose at digium.com>
+
+ * res/res_pjsip_authenticator_digest.c: PJSIP: Improve error
+ handling in digest authenticator Previously, regardless of
+ whether failure to authenticate was due to lacking any
+ authentication or actually failing authentication, the Digest
+ Authenticator would simply return that a challenge was still
+ needed. It will continue to do that when no authentication
+ information is in the received SIP digest, but when
+ authentication information is present and does not pass
+ authentication, that will be treated as an authentication error.
+ This is to ensure that PJSIP will issue security events indicated
+ failed auths.
+
+2013-11-07 21:09 +0000 [r402528] David M. Lee <dlee at digium.com>
+
+ * res/ari/resource_sounds.h, rest-api/api-docs/asterisk.json,
+ rest-api/api-docs/applications.json, res/res_ari_endpoints.c,
+ res/ari/resource_asterisk.c, rest-api/api-docs/playback.json,
+ rest-api-templates/swagger_model.py, res/ari/resource_asterisk.h,
+ rest-api-templates/ari_resource.c.mustache,
+ rest-api-templates/asterisk_processor.py, res/res_ari_bridges.c,
+ rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c,
+ res/ari/resource_endpoints.h, res/res_ari_applications.c,
+ res/res_ari_playback.c, res/res_ari_channels.c,
+ rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
+ res/res_ari_recordings.c, res/ari/resource_bridges.h,
+ res/res_ari_events.c, res/ari/resource_applications.c,
+ res/ari/resource_playback.c, rest-api/api-docs/channels.json,
+ res/ari/resource_applications.h, res/ari/resource_channels.c,
+ res/ari/resource_playback.h, res/res_ari_sounds.c,
+ rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
+ res/ari/resource_channels.h,
+ rest-api-templates/ari_resource.h.mustache,
+ res/ari/resource_events.c, res/ari/resource_recordings.h,
+ rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c,
+ rest-api-templates/res_ari_resource.c.mustache,
+ res/ari/resource_events.h, rest-api/api-docs/sounds.json,
+ res/ari/resource_sounds.c: ari: User better nicknames for ARI
+ operations While working on building client libraries from the
+ Swagger API, I noticed a problem with the nicknames.
+ channel.deleteChannel() channel.answerChannel()
+ channel.muteChannel() Etc. We put the object name in the nickname
+ (since we were generating C code), but it makes OO generators
+ redundant. This patch makes the nicknames more OO friendly. This
+ resulted in a lot of name changing within the res_ari_*.so
+ modules, but not much else. There were a couple of other fixed I
+ made in the process. * When reversible operations (POST /hold,
+ POST /unhold) were made more RESTful (POST /hold, DELETE
+ /unhold), the path for the second operation was left in the API
+ declaration. This worked, but really the two operations should
+ have been on the same API. * The POST /unmute operation had still
+ not been REST-ified. Review:
+ https://reviewboard.asterisk.org/r/2940/
+
+2013-11-06 21:57 +0000 [r402517] Kevin Harwell <kharwell at digium.com>
+
+ * apps/app_queue.c: app_queue: crash if first agent is "busy" If
+ the first agent/member (via CLI "queue show") in a queue is
+ "busy" (dnd, circuit busy, etc...) and no agents answered then
+ app_queue would crash. This occurred because while the calling of
+ agent(s) remained valid the channel on "busy" agent would be set
+ to NULL and then later dereferenced upon a second "rna" function
+ call. The original intention of the code is to have only valid
+ "call attempt" objects (channels != NULL) checked while
+ attempting to call agent(s). It does this by building a
+ "call_next" list of valid "call attempt" objects. In the case of
+ the "busy" agent subsequent builds of the valid "call attempt"
+ list would sometimes include (the case mentioned above) an
+ invalid "call attempt" object. The fix was to make sure the "call
+ attempt" list was appropriately built on every iteration. A NULL
+ sanity check was also added at the original offending spot of the
+ crash just in case another one slipped by somehow. (closes issue
+ ASTERISK-22644) Reported by: Marco Signorini Review:
+ https://reviewboard.asterisk.org/r/2983/
+
+2013-11-05 21:16 +0000 [r402501-402507] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Use AST_AF* defined constant when
+ calling ast_get_ip While the structure passed to ast_get_ip
+ should be set memset to 0, thus initializing the ss_family member
+ to 0, explicitly setting it to AST_AF_UNSPEC is more portable.
+
+ * channels/chan_iax2.c: chan_iax2: Fix incorrect usage of
+ ast_get_ip involving uninitialized struct This started off as a
+ fix for the failing IAX2 acl_call test in the Asterisk Test
+ Suite. When inspecting why that test was failing, it became clear
+ that all attempts to bind to any local loopback address was
+ failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding
+ IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787]
+ netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28]
+ DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2
+ 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1",
+ "(null)", ...): ai_family not supported [Nov 2 15:56:28]
+ WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's
+ conceivably other ways for getaddrino to return EAI_FAMILY, the
+ most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not
+ provided as the desired family. The culprit was the call to
+ ast_get_ip, defined in acl.h. This function uses the family from
+ the passed in addr object (which it will also populate when it
+ returns!) when it eventually calls getaddrinfo. This patch fixes
+ the use of ast_get_ip that were not specifying the family in
+ chan_iax2. This prevents uninitialized use of the structure, so
+ that the addresses resolve correctly. Review:
+ https://reviewboard.asterisk.org/r/2991
+
+ * include/asterisk/netsock2.h, include/asterisk/acl.h: netsock2:
+ Define AST_AF_* enum constants to their AF_* equivalents This
+ patch explicitly defines AST_AF_* enum constants to their
+ sys/socket.h defined equivalents. It is certainly unclear why
+ these constants actually have to exist, given that netsock2.h
+ includes sys/socket.h; however, since the code base is already
+ liberally sprinkled with the usage of AST_AF_* (as well as with
+ direct calls to AF_*), this will at least keep the semantics
+ consistent between their usage across systems.
+
+ * main/stasis_channels.c: stasis_channels: Don't give preference to
+ ANI info in channel snapshots When publishing channel snapshots,
+ we currently compute the caller ID name and number by giving
+ preference first to ani.{name|number}, then to id.{name|number}.
+ However, when a channel driver (such as chan_sip) updates the
+ caller ID, it typically only updates the caller ID stored in
+ id.{name|number}. This means that we are currently giving
+ preference to stale information. When looking at the rest of the
+ code base, the only other place where we appear to use this same
+ logic is in app_amd. Everywhere else, we treat the party
+ information in ani as being separate to the party information in
+ id. This patch publishes only the caller ID name and number in
+ the snapshot field for caller_name and caller_num. Note that the
+ information in ANI is still available in caller_ani. Review:
+ https://reviewboard.asterisk.org/r/2992/
+
+2013-11-04 20:56 +0000 [r402452] Kevin Harwell <kharwell at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: notify dialog info ignores
+ presentation indicator in callerid The presentation indicator in
+ a callerid (e.g. set by dialplan function
+ Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+ Info Notifies are generated during extension monitoring. Added a
+ check to make sure the name and/or number presentations on the
+ callee (remote identity) are set to allow. If they are restricted
+ then "anonymous" is used instead. (closes issue AST-1175)
+ Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2976/ ........ Merged
+ revisions 402450 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-02 04:30 +0000 [r402398-402438] Richard Mudgett <rmudgett at digium.com>
+
+ * main/stasis.c, main/stasis_message_router.c,
+ include/asterisk/vector.h: vector: Uppercase API to follow C
+ convention. C does not support templates like C++.
+
+ * main/stasis.c, main/stasis_message_router.c,
+ include/asterisk/vector.h, include/asterisk/lock.h: vector:
+ Update API to be more flexible. Made the vector macro API be more
+ like linked lists. 1) Added a name parameter to ast_vector() to
+ name the vector struct. 2) Made the API take a pointer to the
+ vector struct instead of the struct itself. 3) Added an element
+ cleanup macro/function parameter when removing an element from
+ the vector for ast_vector_remove_cmp_unordered() and
+ ast_vector_remove_elem_unordered(). 4) Added
+ ast_vector_get_addr() in case the vector element is not a simple
+ pointer. * Converted an inline vector usage in
+ stasis_message_router to use the vector API. It needed the API
+ improvements so it could be converted. * Fixed topic reference
+ leak in router_dtor() when the stasis_message_router is
+ destroyed. * Fixed deadlock potential in stasis_forward_all() and
+ stasis_forward_cancel(). Locking two topics at the same time
+ requires deadlock avoidance. * Made internal_stasis_subscribe()
+ tolerant of a NULL topic. * Made stasis_message_router_add(),
+ stasis_message_router_add_cache_update(),
+ stasis_message_router_remove(), and
+ stasis_message_router_remove_cache_update() tolerant of a NULL
+ message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as
+ intended in dispatch_message(). Review:
+ https://reviewboard.asterisk.org/r/2903/
+
+ * apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
+ apps/confbridge/conf_state_multi_marked.c,
+ apps/confbridge/conf_state.c,
+ apps/confbridge/conf_state_single.c,
+ apps/confbridge/conf_state_inactive.c,
+ apps/confbridge/conf_state_single_marked.c, /,
+ apps/confbridge/include/confbridge.h: confbridge: Separate user
+ muting from system muting overrides. The system overrides the
+ user muting requests when MOH is playing or a waitmarked user is
+ waiting for a marked user to join. System muting overrides
+ interfere with what the user may wish the muting to be when the
+ system override ends. * User muting requests are now independent
+ of the system muting overrides. The effective muting is now the
+ logical or of the user request and system override. * Added a
+ Muted flag to the CLI "confbridge list <conference>" command. *
+ Added a Muted header to the AMI ConfbridgeList action
+ ConfbridgeList event. (closes issue AST-1102) Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........
+ Merged revisions 402425 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/config.c, apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample, /: config: Allow ConfBridge DTMF
+ menus to have '#' as the first digit. ConfBridge allows custom
+ DTMF menus to be created in the confbridge.conf file by assigning
+ a DTMF key sequence to a sequence of actions as follows:
+ DTMF-sequence = action,action... Unfortunately, the normal config
+ file processing code interprets an initial '#' character as
+ starting a directive such as #include. * Add the ability to
+ escape the first non-blank character in a config line so the '#'
+ character can be used without triggering the directive processing
+ code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported
+ by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch
+ (license #5621) patch uploaded by rmudgett (modified) Review:
+ https://reviewboard.asterisk.org/r/2969/ ........ Merged
+ revisions 402407 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk/app.h, main/app.c: voicemail: Simplify callback
[... 22648 lines stripped ...]
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