[asterisk-commits] bebuild: tag 12.5.0-rc1 r420813 - in /tags/12.5.0-rc1: ./ contrib/realtime/my...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Aug 11 13:54:51 CDT 2014
Author: bebuild
Date: Mon Aug 11 13:54:48 2014
New Revision: 420813
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=420813
Log:
Importing files for 12.5.0-rc1 release.
Added:
tags/12.5.0-rc1/.lastclean (with props)
tags/12.5.0-rc1/.version (with props)
tags/12.5.0-rc1/ChangeLog (with props)
tags/12.5.0-rc1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/12.5.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.5.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.5.0-rc1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/12.5.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.5.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.5.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/12.5.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.5.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.5.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/12.5.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.5.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.5.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.5.0-rc1/.lastclean?view=auto&rev=420813
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--- tags/12.5.0-rc1/ChangeLog (added)
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+2014-08-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.5.0-rc1 Released.
+
+2014-08-11 18:48 +0000 [r420805] Matthew Jordan <mjordan at digium.com>
+
+ * rest-api/api-docs/playbacks.json, UPGRADE.txt,
+ rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+ rest-api/resources.json, 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: AMI/ARI: Update version to
+ 2.5.0/1.5.0 respectively This is to support the backwards
+ compatible changes made in the next version of Asterisk.
+
+2014-08-11 18:45 +0000 [r420795-420802] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_stasis.c: Stasis: Use the correct return value Return the
+ correct value instead of always returning 0 when setting internal
+ status on unreal channels. Reported by: Richard Mudgett
+
+ * res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h,
+ res/res_stasis.c, res/ari/resource_bridges.c: Stasis: Allow
+ internal channels directly into bridges The patch to catch
+ channels being shoehorned into Stasis() via external mechanisms
+ also happens to catch Announcer and Recorder channels because
+ they aren't known to be stasis-controlled channels in the usual
+ sense. This marks those channels as Stasis()-internal channels
+ and allows them directly into bridges. Review:
+ https://reviewboard.asterisk.org/r/3903/
+
+2014-08-11 10:37 +0000 [r420656-420716] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/utils.c, /: general: Fix memory Corruption in
+ __ast_string_field_ptr_build_va. If the space left in a
+ stringfield is between 0 and
+ (alignof(ast_string_field_allocation)-1) adding new data would
+ cause memory corruption, because we would assume enough space
+ (unsigned underrun). Thanks Arnd Schmitter for reporting and
+ finding out the cause! ASTERISK-23508 #close Reported by: Arnd
+ Schmitter Tested by: Arnd Schmitter, JoshE Review:
+ https://reviewboard.asterisk.org/r/3898/ ........ Merged
+ revisions 420680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 420715 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode.
+ ........ Merged revisions 420654 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 420655 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-08 12:31 +0000 [r420533] Matthew Jordan <mjordan at digium.com>
+
+ * main/message.c: main/message: remove debug message
+
+2014-08-08 02:51 +0000 [r420513] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_cel.c: CEL: Update unit tests for additional
+ information This updates the CEL unit tests for the new
+ information contained in the attended transfer CEL extra field.
+
+2014-08-07 21:48 +0000 [r420436] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
+ resolve the large SDP poll issue. Replace sip_tls_read() and
+ sip_tcp_read() with a single function and resolve the poll/wait
+ issue with large SDP payloads. ASTERISK-18345 #close Reported by:
+ Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
+ patch uploaded by Elazar Broad Review:
+ https://reviewboard.asterisk.org/r/3882/ ........ Merged
+ revisions 420434 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 420435 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-07 21:16 +0000 [r420408-420414] Kinsey Moore <kmoore at digium.com>
+
+ * main/stasis_bridges.c: Stasis: Correct blind transfer message
+ generation This fixes the json object creation format string and
+ key name for the BridgeBlindTransfer Stasis event allowing it to
+ be published properly.
+
+ * main/stasis_bridges.c: Stasis: Ensure transfer messages follow
+ validation rules This makes Stasis() event generation for
+ transfer messages follow validation rules. Currently,
+ ast_json_null() is being used in place of omitting a key entirely
+ which falls afoul of these validation rules.
+ https://reviewboard.asterisk.org/r/3892/
+
+2014-08-07 19:43 +0000 [r420385-420387] Mark Michelson <mmichelson at digium.com>
+
+ * main/bridge.c: Ensure bridges exist when trying to determine
+ bridged parties when publishing transfer information.
+
+ * res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.c, main/bridge.c,
+ res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c,
+ res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c:
+ Revert previous patch since it had some unreviewed content in it.
+
+ * res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c,
+ res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.c, main/bridge.c,
+ res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c: Ensure bridges actually
+ exist when trying to determine the bridged peer.
+
+2014-08-07 15:19 +0000 [r420325] Kinsey Moore <kmoore at digium.com>
+
+ * res/ari/ari_model_validators.c, main/cel.c, apps/app_queue.c,
+ main/stasis_bridges.c, main/channel.c,
+ res/ari/ari_model_validators.h, include/asterisk/datastore.h,
+ tests/test_cel.c, include/asterisk/bridge_features.h,
+ res/res_stasis.c, res/stasis/command.c,
+ rest-api/api-docs/events.json, res/stasis/app.c,
+ res/stasis/control.c, main/bridge.c, res/stasis/stasis_bridge.c,
+ main/bridge_basic.c, res/stasis/command.h,
+ include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h,
+ res/stasis/app.h, res/stasis/control.h: Stasis: Convey transfer
+ information to applications This fixes a class of issues where
+ Stasis applications were not made aware that their channels were
+ being manipulated or replaced by external entitiessuch as
+ transfers, AMI commands, or dialplan applications such as
+ Bridge(). Inconsistent information such as StasisEnd events with
+ unknown channels as a result of masquerades has also been
+ corrected. To accomplish these fixes, several new fields were
+ added to blind and attended transfer messages as well as
+ StasisStart and BridgeAttendedTransfer Stasis events.
+ ASTERISK-23941 #close Review:
+ https://reviewboard.asterisk.org/r/3865/ Review:
+ https://reviewboard.asterisk.org/r/3857/ Review:
+ https://reviewboard.asterisk.org/r/3852/ Review:
+ https://reviewboard.asterisk.org/r/3816/ Review:
+ https://reviewboard.asterisk.org/r/3731/ Review:
+ https://reviewboard.asterisk.org/r/3729/ Review:
+ https://reviewboard.asterisk.org/r/3728/
+
+2014-08-06 21:47 +0000 [r420211-420262] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format.c: Change comment.
+
+ * contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
+ contrib/ast-db-manage/config.ini.sample,
+ contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py
+ (added),
+ contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py
+ (added), contrib/ast-db-manage/cdr.ini.sample,
+ contrib/ast-db-manage/voicemail.ini.sample,
+ contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py
+ (added): alembic: Adjust sippeers, queue_members, and
+ voicemail_messages tables. * Increased the sippeers useragent max
+ string size to 255. * Changed the queue_members uniqueid to an
+ auto incremented integer instead of a string. * Increased the
+ voicemail_messages BLOB size to LONGBLOB on mysql. * Fixed the
+ add_tables_for_pjsip config change version downgrade actions to
+ drop a table it created. * Adjusted the sample alembic.ini files
+ cdr.ini.sample, config.ini.sample, and voicemail.ini.sample to
+ give a mysql and postgres sqlalchemy.url lines. ASTERISK-23847
+ #close Reported by: Stephen More ASTERISK-23825 #close Reported
+ by: Stephen More ASTERISK-23909 #close Reported by: Stephen More
+ Review: https://reviewboard.asterisk.org/r/3870/
+
+2014-08-06 16:10 +0000 [r420148] George Joseph <george.joseph at fairview5.com>
+
+ * main/pbx.c, /, pbx/pbx_lua.c: pbx_lua: fix regression with global
+ sym export and context clash by pbx_config. ASTERISK-23818 (lua
+ contexts being overwritten by contexts of the same name in
+ pbx_config) surfaced because pbx_lua, having the
+ AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before
+ pbx_config. Since I couldn't find any reason for pbx_lua to
+ export it's symbols to the rest of Asterisk, I simply changed the
+ flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
+ realize was that the symbols need to be exported not because
+ Asterisk needs them but because any external Lua modules like
+ luasql.mysql need the base Lua language APIs exported
+ (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's
+ an issue in pbx.c where context_merge was only merging includes,
+ switches and ignore patterns if the context was already existing
+ AND has extensions, or if the context was brand new. If pbx_lua
+ is loaded before pbx_config, the context will exist BUT pbx_lua,
+ being implemented as a switch, will never place extensions in it,
+ just the switch statement. The result is that when pbx_config
+ loads, it never merges the switch statement created by pbx_lua
+ into the final context. This patch sets pbx_lua's modflag back to
+ AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge
+ that catches the case where an existing context has includes,
+ switchs or ingore patterns but no actual extensions.
+ ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo
+ Teräs Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3891/ ........ Merged
+ revisions 420146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 420147 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-05 21:47 +0000 [r420089-420099] Matthew Jordan <mjordan at digium.com>
+
+ * res/stasis/messaging.c: stasis: Fix compilation issue with ao2
+ tagged objects
+
+ * tests/test_message.c: test_message: Fix strict-aliasing
+ compilation issue
+
+ * /: Remove automerge properties :-(
+
+ * res/ari/resource_channels.c, res/res_stasis.c, main/message.c,
+ res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json,
+ res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /,
+ include/asterisk/vector.h, channels/chan_sip.c, res/stasis/app.c,
+ res/stasis/messaging.h (added), res/ari/resource_endpoints.h,
+ res/res_pjsip_messaging.c, tests/test_message.c (added),
+ res/res_xmpp.c, include/asterisk/json.h,
+ res/ari/ari_model_validators.c, include/asterisk/manager.h,
+ CHANGES, res/ari/ari_model_validators.h, main/json.c,
+ res/res_ari_endpoints.c, include/asterisk/message.h: ARI: Add
+ channel technology agnostic out of call text messaging This patch
+ adds the ability to send and receive text messages from various
+ technology stacks in Asterisk through ARI. This includes chan_sip
+ (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages
+ are sent using the endpoints resource, and can be sent directly
+ through that resource, or to a particular endpoint. For example,
+ the following would send the message "Hello there" to PJSIP
+ endpoint alice with a display URI of
+ sip:asterisk at mycooldomain.org:
+ ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk at mycooldomain.org&body=Hello+There
+ This is equivalent to the following as well:
+ ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk at mycooldomain.org&body=Hello+There
+ Both forms are available for message technologies that allow for
+ arbitrary destinations, such as chan_sip. Inbound messages can
+ now be received over ARI as well. An ARI application that
+ subscribes to endpoints will receive messages from those
+ endpoints: { "type": "TextMessageReceived", "timestamp":
+ "2014-07-12T22:53:13.494-0500", "endpoint": { "technology":
+ "PJSIP", "resource": "alice", "state": "online", "channel_ids":
+ [] }, "message": { "from": "\"alice\" <sip:alice at 127.0.0.1>",
+ "to": "pjsip:asterisk at 127.0.0.1", "body": "Watson, come here.",
+ "variables": [] }, "application": "testsuite" } The above was
+ made possible due to some rather major changes in the message
+ core. This includes (but is not limited to): - Users of the
+ message API can now register message handlers. A handler has two
+ callbacks: one to determine if the handler has a destination for
+ the message, and another to handle it. - All dialplan
+ functionality of handling a message was moved into a message
+ handler provided by the message API. - Messages can now have the
+ technology/endpoint associated with them. Various other
+ properties are also now more easily accessible. - A number of ao2
+ containers that weren't really needed were replaced with vectors.
+ Iteration over ao2_containers is expensive and pointless when the
+ lifetime of things is well defined and the number of things is
+ very small. res_stasis now has a new file that makes up its
+ structure, messaging. The messaging functionality implements a
+ message handler, and passes received messages that match an
+ interested endpoint over to the app for processing. Note that
+ inadvertently while testing this, I reproduced ASTERISK-23969.
+ res_pjsip_messaging was incorrectly parsing out the 'to' field,
+ such that arbitrary SIP URIs mangled the endpoint lookup. This
+ patch includes the fix for that as well. Review:
+ https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close
+ Reported by: Matt Jordan ASTERISK-23969 #close Reported by:
+ Andrew Nagy
+
+2014-08-05 19:12 +0000 [r420060] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format.c, /: format.c: Add reason comments for the
+ format_list ordering. ........ Merged revisions 420054 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-08-04 19:45 +0000 [r419944] Rusty Newton <rnewton at digium.com>
+
+ * main/manager.c, /: Manager - Improve documentation for manager
+ commands Getvar and Setvar. The documentation for these commands
+ did not make it clear that they could accept expressions and
+ functions. Modified to make this clear, but tried not to be
+ overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton
+ Tested by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/3854 ........ Merged revisions
+ 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 419943 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-31 11:57 +0000 [r419823-419824] Matthew Jordan <mjordan at digium.com>
+
+ * /: Get rid of automerge properties
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /, res/res_hep_rtcp.c
+ (added), CHANGES, channels/chan_pjsip.c: res_hep_rtcp: Add module
+ that sends RTCP information to a Homer Server This patch adds a
+ new module to Asterisk, res_hep_rtcp. The module subscribes to
+ the RTCP topics in Stasis and receives RTCP information back from
+ the message bus. It encodes into HEPv3 packets and sends the
+ information to the res_hep module for transmission. Using this,
+ someone with a Homer server can get live call quality monitoring
+ for all RTP-based channels in their Asterisk 12+ systems. In
+ addition, there were a few bugs in the RTP engine,
+ res_rtp_asterisk, and chan_pjsip that were uncovered by the tests
+ written for the Asterisk Test Suite. This patch fixes the
+ following: 1) chan_pjsip failed to set its channel unique ids on
+ its RTP instance on outbound calls. It now does this in the
+ appropriate location, in the serialized call callback. 2) The
+ rtp_engine was overflowing some values when packed into JSON.
+ Specifically, some longs and unsigned ints can't be be packed
+ into integer values, for obvious reasons. Since libjansson only
+ supports integers, floats, strings, booleans, and objects, we
+ print these values into strings. 3) res_rtp_asterisk had a few
+ problems: (a) it would emit a source IP address of 0.0.0.0 if
+ bound to that IP address. We now use ast_find_ourip to get a
+ better IP address, and properly marshal the result into an
+ ast_strdupa'd string. (b) Reports can be generated with no report
+ bodies. In particular, this occurs when a sender is transmitting
+ information to a receiver (who will send no RTP back to the
+ sender). As such, the sender has no report body for what it
+ received. We now properly handle this case, and the sender will
+ emit SR reports with no body. Likewise, if we receive an RTCP
+ packet with no report body, we will still generate the
+ appropriate events. ASTERISK-24119 #close
+
+2014-07-29 10:52 +0000 [r419750-419764] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_session.c: res_pjsip_session: Fix race condition
+ where redirecting information may not be set. Since the PJSIP
+ INVITE session module is invoked before any session supplements
+ it was possible for it to handle a redirect before the
+ res_pjsip_diversion module interpreted and set redirecting
+ information on the channel. This would cause the redirecting
+ information to get lost. This patch ensures that session
+ supplements are *always* invoked before a redirect occurs by
+ explicitly calling them in the redirect handler. Review:
+ https://reviewboard.asterisk.org/r/3850/
+
+ * res/res_pjsip_pidf_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c:
+ res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator:
+ Ensure local entity is unquoted. The local entity as provided by
+ PJSIP is quoted within '<' and '>'. As a result placing this
+ value into XML will result in malformed XML being produced. This
+ patch now unquotes the local entity so it can go safely into the
+ XML. Review: https://reviewboard.asterisk.org/r/3851/
+
+2014-07-28 18:50 +0000 [r419686] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /, funcs/func_frame_trace.c, main/abstract_jb.c,
+ apps/app_speech_utils.c: datastores: Audit
+ ast_channel_datastore_remove usage. Audit of v1.8 usage of
+ ast_channel_datastore_remove() for datastore memory leaks. *
+ Fixed leaks in app_speech_utils and func_frame_trace. * Fixed
+ app_speech_utils not locking the channel when accessing the
+ channel datastore list. Review:
+ https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of
+ ast_channel_datastore_remove() for datastore memory leaks. *
+ Fixed leak in func_jitterbuffer. (Was not in v12) Review:
+ https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of
+ ast_channel_datastore_remove() for datastore memory leaks. *
+ Fixed leaks in abstract_jb. * Fixed leak in
+ ast_channel_unsuppress(). Used by ARI mute control and
+ res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used
+ by ARI mute control and res_mutestream. Review:
+ https://reviewboard.asterisk.org/r/3861/ ........ Merged
+ revisions 419684 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 419685 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-25 14:46 +0000 [r419565-419566] Matthew Jordan <mjordan at digium.com>
+
+ * CHANGES: Update CHANGES for r419565
+
+ * res/res_stasis_recording.c, res/ari/ari_model_validators.c,
+ rest-api/api-docs/recordings.json,
+ res/ari/ari_model_validators.h: ARI: report duration values in
+ LiveRecording objects This patch adds three new fields to the
+ LiveRecording model: - total_duration: the total length of the
+ live recording - talking_duration: optional. The duration of
+ talking energy that was detected while the recording was made. -
+ silence_duration: optional. The duration of silence that was
+ detected while the recording was made. These values are reported
+ in the RecordingFinished ARI event. When a DSP is enabled on the
+ channel during the recording - which occurs when the recording is
+ created with max_silence_seconds (indicating that the user
+ actually cares about how much silence is in the file), we will
+ report the talking_duration and silence_duration in addition to
+ the total_duration. Review:
+ https://reviewboard.asterisk.org/r/3770/ ASTERISK-24037 #close
+ Reported by: Samuel Galarneau
+
+2014-07-25 10:53 +0000 [r419536-419538] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_bridgewait.c: app_bridgewait: Remove possibility of race
+ condition between channels leaving/joining. Bridges created by
+ app_bridgewait previously had the "dissolve when empty" flag set.
+ This caused the bridge core to destroy them when the last channel
+ had left. This introduced a race condition where we may have a
+ reference to the bridge but it is not actually joinable when we
+ try to join it. This flag has now been removed and the bridge is
+ guaranteed to be joinable at all times. ASTERISK-23987 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3836/
+
+ * main/bridge.c: bridge: Make "bridge destroy" only available in
+ developer mode and add "all" to "bridge kick". The "bridge
+ destroy" CLI command is invasive to bridges and can leave them in
+ an unexpected state for the users of them. Since this command may
+ be useful for developers it is now only available when developer
+ mode is available. To take its place "all" has been added as a
+ valid option to the "bridge kick" CLI command. It will kick all
+ of the channels in the bridge out. ASTERISK-23987 Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/
+
+2014-07-24 17:57 +0000 [r419442] Corey Farrell <git at cfware.com>
+
+ * /, channels/chan_sip.c: chan_sip: sip_subscribe_mwi_destroy
+ should not call sip_destroy sip_subscribe_mwi_destroy calls
+ sip_destroy on the reference counted mwi->call. This results in
+ the fields of mwi->call being freed, but mwi->call itself it
+ leaked. If other code is still using mwi->call it can cause
+ problems. This change uses dialog_unref instead, to balance the
+ ref provided by sip_alloc(). ASTERISK-24087 #close Reported by:
+ Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/
+ ........ Merged revisions 419440 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 419441 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-24 16:50 +0000 [r419376] Jason Parker <jparker at digium.com>
+
+ * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if
+ ooh323.conf not found. (closes issue ASTERISK-23814) ........
+ Merged revisions 419374 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 419375 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-23 16:45 +0000 [r419316-419318] Matthew Jordan <mjordan at digium.com>
+
+ * main/endpoints.c, tests/test_stasis_endpoints.c: endpoints: Fix
+ failing unit tests from r419196 This patch does two things: (1)
+ It updates the unit tests to expect additional stasis messages.
+ More messages are now sent to the endpoint topic, due to
+ forwarding all channel messages and the forwarding relationship
+ set up between endpoints themselves. (2) Remove the technology
+ forwarding subscription during ast_endpoint_shutdown. This
+ prevents an improper double shutdown of an endpoint from
+ occurring.
+
+ * res/res_pjsip_refer.c: res_pjsip_refer: remove stray debugging
+ line How'd those @ symbols get in there...
+
+2014-07-23 13:58 +0000 [r419285] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * apps/app_voicemail.c, /: app_voicemail: use a consistent
+ generator string When updating voicemail.conf when a user changes
+ their pin, change the generator string to be the same as the
+ module name when reading so that the same config_hook will be
+ called. Review: https://reviewboard.asterisk.org/r/3837/ ........
+ Merged revisions 419284 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-22 16:12 +0000 [r419196] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/channel.h, res/ari/resource_applications.h,
+ res/res_xmpp.c, channels/chan_iax2.c, main/endpoints.c,
+ channels/chan_pjsip.c, main/channel.c,
+ res/ari/resource_endpoints.c, channels/chan_sip.c,
+ include/asterisk/endpoints.h,
+ rest-api/api-docs/applications.json, include/asterisk/xmpp.h,
+ main/channel_internal_api.c, channels/chan_motif.c: ARI: Fix
+ endpoint/channel subscription issues; allow for subscriptions to
+ tech This patch serves two purposes: (1) It fixes some bugs with
+ endpoint subscriptions not reporting all of the channel events
+ (2) It serves as the preliminary work needed for ASTERISK-23692,
+ which allows for sending/receiving arbitrary out of call text
+ messages through ARI in a technology agnostic fashion. The
+ messaging functionality described on ASTERISK-23692 requires two
+ things: (1) The ability to send/receive messages associated with
+ an endpoint. This is relatively straight forwards with the
+ endpoint core in Asterisk now. (2) The ability to send/receive
+ messages associated with a technology and an arbitrary technology
+ defined URI. This is less straight forward, as endpoints are
+ formed from a tech + resource pair. We don't have a mechanism to
+ note that a technology that *may* have endpoints exists. This
+ patch provides such a mechanism, and fixes a few bugs along the
+ way. The first major bug this patch fixes is the forwarding of
+ channel messages to their respective endpoints. Prior to this
+ patch, there were two problems: (1) Channel caching messages
+ weren't forwarded. Thus, the endpoints missed most of the
+ interesting bits (such as channel creation, destruction, state
+ changes, etc.) (2) Channels weren't associated with their
+ endpoint until after creation. This resulted in endpoints missing
+ the channel creation message, which limited the usefulness of the
+ subscription in the first place (a major use case being 'tell me
+ when this endpoint has a channel'). Unfortunately, this meant
+ another parameter to ast_channel_alloc. Since not all channel
+ technologies support an ast_endpoint, this patch makes such a
+ call optional and opts for a new function,
+ ast_channel_alloc_with_endpoint. When endpoints are created, they
+ will implicitly create a technology endpoint for their technology
+ (if one does not already exist). A technology endpoint is special
+ in that it has no state, cannot have channels created for it,
+ cannot be created explicitly, and cannot be destroyed except on
+ shutdown. It does, however, have all messages from other
+ endpoints in its technology forwarded to it. Combined with the
+ bug fixes, we now have Stasis messages being properly forwarded.
+ Consider the following scenario: two PJSIP endpoints (foo and
+ bar), where bar has a single channel associated with it and foo
+ has two channels associated with it. The messages would be
+ forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint
+ PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP /
+ channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the
+ applications resource, can: - subscribe to endpoint:PJSIP/foo and
+ get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and
+ endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get
+ notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar -
+ subscribe to endpoint:PJSIP and get notifications for channels
+ PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints
+ PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes,
+ it never has events itself. It merely provides an aggregation
+ point for all other endpoints in its technology (which in turn
+ aggregate all channel messages associated with that endpoint).
+ This patch also adds endpoints to res_xmpp and chan_motif,
+ because the actual messaging work will need it (messaging without
+ XMPP is just sad). Review:
+ https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692
+
+2014-07-22 14:13 +0000 [r419163] Kinsey Moore <kmoore at digium.com>
+
+ * addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c,
+ addons/ooh323c/src/ooq931.c, tests/test_json.c,
+ tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /,
+ tests/test_abstract_jb.c, apps/app_meetme.c,
+ tests/test_optional_api.c, tests/test_logger.c,
+ tests/test_event.c, tests/test_format_api.c,
+ tests/test_hashtab_thrash.c, channels/chan_gtalk.c,
+ res/res_mwi_external_ami.c, res/res_jabber.c,
+ tests/test_sorcery.c, channels/chan_jingle.c, res/res_corosync.c,
+ tests/test_voicemail_api.c, tests/test_aoc.c,
+ tests/test_astobj2.c, tests/test_config.c: Fix more dev-mode
+ build issues ........ Merged revisions 419129 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 419162 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-18 21:25 +0000 [r419021] Matthew Jordan <mjordan at digium.com>
+
+ * CHANGES, rest-api/api-docs/recordings.json,
+ res/ari/resource_recordings.c, res/stasis_recording/stored.c,
+ res/res_ari_recordings.c,
+ include/asterisk/stasis_app_recording.h,
+ res/ari/resource_recordings.h: ari: Add a copy operation for
+ stored recordings This patch adds a new operation for stored
+ recordings, copy. It takes an existing stored recording and makes
+ a copy of it in the same directory or a relative directory under
+ the stored recording directory.
+ /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
+ This is particularly useful for voicemail-esque applications,
+ which may need to copy or move recordings around a directory
+ structure. Review: https://reviewboard.asterisk.org/r/3768/
+ ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam
+ Galarneau
+
+2014-07-18 21:24 +0000 [r418996-419019] Corey Farrell <git at cfware.com>
+
+ * main/stasis_message_router.c: stasis: fix call to ao2_t_alloc for
+ stasis_message_router_create This fixes a build failure
+ introduced by r3821. struct stasis_topic is opaque, so
+ topic->name is unavailable. Switch to using stasis_topic_name().
+
+ * main/stasis.c, main/stasis_cache_pattern.c,
+ main/stasis_message.c, main/stasis_message_router.c: stasis: use
+ ao2_t_alloc for certain object allocators Add tags to stasis
+ objects using the name. This makes it easier to track the source
+ of certain stasis ref leaks. Review:
+ https://reviewboard.asterisk.org/r/3821/
+
+2014-07-18 16:46 +0000 [r418937] Richard Mudgett <rmudgett at digium.com>
+
+ * funcs/func_audiohookinherit.c: func_audiohookinherit.c: Fixup
+ some XML documentation wording.
+
+2014-07-18 16:01 +0000 [r418914] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/audiohook.h, main/framehook.c, res/res_fax.c,
+ main/bridge_basic.c, include/asterisk/res_fax.h,
+ bridges/bridge_native_rtp.c, main/audiohook.c, CHANGES,
+ include/asterisk/framehook.h, res/res_pjsip_refer.c,
+ main/channel.c, funcs/func_audiohookinherit.c: Channels:
+ Masquerades to automatically move frame/audio hooks Whenever
+ possible, audiohooks and framehooks will now be copied over to
+ the channel that the masquerading channel gets cloned into. This
+ should occur for all audiohooks and most framehooks. As a result,
+ in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
+ deprecated and its behavior is essentially the new default for
+ all audiohooks, plus some additional audiohooks/framehooks.
+ Review: https://reviewboard.asterisk.org/r/3721/
+
+2014-07-17 22:17 +0000 [r418886] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/features_config.c: feature_config: insure featuregroups and
+ applicationmaps are initialized If the features.conf is missing,
+ the cfg->featurgroups and cfg->applicationmaps is not
+ initialized, resulting in assert on ao2_find of a null container.
+ This patch changes the initialization call and adds asserts for a
+ safeguard. Review: https://reviewboard.asterisk.org/r/3809/
+
+2014-07-17 14:27 +0000 [r418810] Kinsey Moore <kmoore at digium.com>
+
+ * main/bridge_channel.c: TEST_FRAMEWORK: Fix threewaytransfer
+ reporting Ensure that three-way transfers can be reported even if
+ featuremap is non-NULL.
+
+2014-07-16 23:06 +0000 [r418787] Corey Farrell <git at cfware.com>
+
+ * channels/dahdi/bridge_native_dahdi.c: Remove include of astobj.h
+ from channels/dahdi/bridge_native_dahdi.c. The include was
+ unneeded, this is split off from r3758 as it applies to 12.
+
+2014-07-16 13:58 +0000 [r418756] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_pjsip.c, include/asterisk/res_pjsip.h,
+ contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py
+ (added), configs/pjsip.conf.sample,
+ res/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c:
+ res_pjsip: Support setting a default accountcode on endpoints
+ Most channel drivers let you specify a default accountcode to be
+ set on channels associated with a particular
+ peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip
+ did not support such a setting. This patch adds a new setting to
+ the res_pjsip endpoint object, 'accountcode'. When a channel is
+ created that is associated with an endpoint with this value set,
+ the channel will automatically have its accountcode property set
+ to the value configured for the endpoint. Review:
+ https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close
+ Reported by: Matt Jordan
+
+2014-07-15 23:03 +0000 [r418715] Kinsey Moore <kmoore at digium.com>
+
+ * main/bridge_channel.c: TEST_FRAMEWORK: Fix ref leak in feature
+ activation This fixes two reference leaks that would occur when
+ TEST_FRAMEWORK was enabled and features were successfully
+ executed.
+
+2014-07-15 22:20 +0000 [r418714] Matthew Jordan <mjordan at digium.com>
+
+ * main/manager.c, /: manager: Return ActionID on nominal responses
+ to PresenceState action When the PresenceState action is
+ executed, the nominal path fails to include the ActionID in the
+ successful response. This patch adds a call to astman_start_ack,
+ which guarantees that an ActionID (if provided) will be sent back
+ to the AMI client. Review:
+ https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close
+ ........ Merged revisions 418713 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-15 17:45 +0000 [r418650] Jonathan Rose <jrose at digium.com>
+
+ * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty
+ strings as argument Previously these two dialplan functions would
+ issue warnings and return failure when an empty string is used as
+ the argument. Now they will not issue a warning and will
+ successfully return an empty string. ASTERISK-23911 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3745/ ........ Merged
+ revisions 418641 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 418649 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-15 17:14 +0000 [r418636] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * channels/chan_sip.c: media formats: fix ref leak of peer for mwi
+ subscription Holding a reference to the peer during mwi
+ subscriptions resulted in a circular reference because the final
+ event message would not be sent until destruction of the peer.
+ Instead, pass the name of the peer to the event callback so that
+ it can fail gracefully after the peer has gone. ASTERISK-23959
+ Review: https://reviewboard.asterisk.org/r/3754/
+
+2014-07-14 14:46 +0000 [r418586] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/logger.h: logger.h: Extract DEBUG_ATLEAST() to
+ complement VERBOSITY_ATLEAST().
+
+2014-07-13 21:55 +0000 [r418466-418506] Corey Farrell <git at cfware.com>
+
+ * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work
+ around REF_DEBUG race which causes out of order log entries *
+ Update refcounter.py to use delta's to track the current
[... 32748 lines stripped ...]
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