[svn-commits] bebuild: tag 13.0.0-beta1 r420834 - in /tags/13.0.0-beta1: ./ contrib/realtim...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 11 14:48:40 CDT 2014
Author: bebuild
Date: Mon Aug 11 14:48:28 2014
New Revision: 420834
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=420834
Log:
Importing files for 13.0.0-beta1 release.
Added:
tags/13.0.0-beta1/.lastclean (with props)
tags/13.0.0-beta1/.version (with props)
tags/13.0.0-beta1/ChangeLog (with props)
tags/13.0.0-beta1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/13.0.0-beta1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/13.0.0-beta1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/13.0.0-beta1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/13.0.0-beta1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/13.0.0-beta1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/13.0.0-beta1/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/13.0.0-beta1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/13.0.0-beta1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/13.0.0-beta1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/13.0.0-beta1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/13.0.0-beta1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
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--- tags/13.0.0-beta1/ChangeLog (added)
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+2014-08-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 13.0.0-beta1 Released.
+
+2014-08-11 18:50 +0000 [r420808] Matthew Jordan <mjordan at digium.com>
+
+ * 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,
+ rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+ rest-api/resources.json, include/asterisk/manager.h: 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. ........ Merged revisions 420805 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-11 18:46 +0000 [r420796-420803] 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
+ ........ Merged revisions 420802 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, 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/ ........ Merged
+ revisions 420795 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-11 18:32 +0000 [r420758-420794] Mark Michelson <mmichelson at digium.com>
+
+ * main/stasis_channels.c, res/ari/resource_channels.c, CHANGES,
+ res/res_pjsip_pubsub.c, main/manager_channels.c, apps/app_dial.c,
+ res/stasis/app.c, res/stasis/control.c,
+ include/asterisk/stasis_app.h: Improve call forwarding reporting,
+ especially with regards to ARI. This patch addresses a few
+ issues: 1) The order of Dial events have been changed when
+ performing a call forward. The order has now been altered to 1)
+ Dial begins dialing channel A. 2) When A forwards the call to B,
+ we issue the dial end event to channel A, indicating the dial is
+ being canceled due to a forward to B. 3) When the call to channel
+ B occurs, we then issue a new dial begin to channel B. 2) Call
+ forwards are now reported on the calling channel, not the peer
+ channel. 3) AMI DialEnd events have been altered to display the
+ extension the call is being forwarded to when relevant. 4) You
+ can now get the values of channel variables for channels that are
+ not currently in the Stasis application. This brings the
+ retrieval of channel variables more in line with the rest of
+ channel read operations since they may be performed on channels
+ not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan
+ ASTERISK-24138 #close Reported by Matt Jordan Patches:
+ forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
+ Review: https://reviewboard.asterisk.org/r/3899
+
+ * res/res_pjsip_pubsub.c: Fix crashing unit tests with regards to
+ RLS. The unit tests require a sorcery.conf file that has been set
+ up to store resource lists in memory rather than retrieving from
+ configuration. With a setup that is not conducive to running the
+ tests, a fault in sorcery currently causes Asterisk to crash when
+ attempting to run any of the tests. To get around the crash, this
+ adds a function that verifies the current environment and marks
+ the tests as "not run" if the setup is not correct.
+
+ * res/res_pjsip_pubsub.c: Fix crash encountered by the testsuite.
+ Running testsuite tests locally produced no errors, but when run
+ using the continuous integration framework, crashes occurred. The
+ crashes occurred due to a refcounting error that had been fixed
+ for a similar situation.
+
+2014-08-11 13:57 +0000 [r420742] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_hep_pjsip.c, res/res_hep_rtcp.c, res/res_hep.c: res_hep:
+ Remove disabling of modules These modules were originally
+ specified as being disabled, as they were introduced midstream in
+ Asterisk 12. That makes it nicer for folks who are upgrading to a
+ new release in the middle of Asterisk 12. That's not the case for
+ Asterisk 13: it's a brand new release. There's no reason to have
+ the modules disabled by default in that case.
+
+2014-08-11 10:40 +0000 [r420657-420717] 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 ........ Merged
+ revisions 420716 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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 ........ Merged
+ revisions 420656 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-11 01:31 +0000 [r420607-420639] Matthew Jordan <mjordan at digium.com>
+
+ * funcs/func_jitterbuffer.c: funcs/func_jitterbuffer: Tweak
+ documentation This patch merely reformats and cleans up a bit of
+ the jitterbuffer documentation for the wiki.
+
+ * contrib/ast-db-manage/config/versions/d39508cb8d8_create_queue_rules.py
+ (added), configs/samples/queuerules.conf.sample, UPGRADE.txt,
+ configs/samples/extconfig.conf.sample, CHANGES, apps/app_queue.c:
+ app_queue: Add RealTime support for queue rules This patch gives
+ the optional ability to keep queue rules in RealTime. It is
+ important to note that with this patch: (a) Queue rules in
+ RealTime are only examined on module load/reload (b) Queue rules
+ are loaded both from the queuerules.conf file as well as the
+ RealTime backend To inform app_queue to examine RealTime for
+ queue rules, a new setting has been added to queuerules.conf's
+ general section "realtime_rules". RealTime queue rules will only
+ be used when this setting is set to "yes". The schema for the
+ database table supports a rule_name, time, min_penalty, and
+ max_penalty columns. min_penalty and max_penalty can be relative,
+ if a '-' or '+' literal is provided. Otherwise, the penalties are
+ treated as constants. For example: rule_name, time, min_penalty,
+ max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55'
+ 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333'
+ 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which
+ would result in : Rule: default - After 10 seconds, adjust
+ QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule:
+ test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and
+ adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust
+ QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 -
+ After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
+ QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust
+ QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564
+ Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to
+ 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the
+ queue rules will be always reloaded on a module reload, even if
+ the underlying file did not change. With the option disabled, the
+ rules will only be reloaded if the file was modified. Review:
+ https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close
+ Reported by: Michael K patches: app_queue.c_realtime_trunk.patch
+ uploaded by Michael K (License 6621)
+
+ * CHANGES: Update CHANGES file
+
+ * UPGRADE.txt: Update UPGRADE.txt file
+
+2014-08-08 20:08 +0000 [r420577-420592] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: Fix build in devmode.
+
+ * CHANGES, configs/samples/voicemail.conf.sample,
+ apps/app_voicemail.c: app_voicemail: Add the ability to specify
+ multiple email addresses. ASTERISK-24045 Reported by: Jacob
+ Barber Review: https://reviewboard.asterisk.org/r/3833/
+
+2014-08-08 17:53 +0000 [r420534-420562] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c, channels/sip/security_events.c,
+ channels/sip/dialplan_functions.c, channels/sip/reqresp_parser.c,
+ channels/sip/route.c, channels/sip/utils.c,
+ channels/sip/config_parser.c: chan_sip: Mark chan_sip and its
+ files as extended support
+
+ * rest-api-templates/make_ari_stubs.py: make_ari_stubs: Update wiki
+ prefix to '13'
+
+ * rest-api-templates/res_ari_resource.c.mustache:
+ res_ari_resource.c.mustache: Update template to emit module
+ support level
+
+ * /, main/message.c: main/message: remove debug message ........
+ Merged revisions 420533 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-08 03:03 +0000 [r420514] 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.
+ ........ Merged revisions 420513 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-08 01:31 +0000 [r420494-420496] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt: Update UPGRADE file for 13 branch
+
+ * /: Remove old properties
+
+ * / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\
+ \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| /
+ __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_|
+ |_|___/\__\___|_| |_|___|_|\_\ \___\____/
+
+2014-08-07 21:58 +0000 [r420437] 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 ........ Merged
+ revisions 420436 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-07 21:17 +0000 [r420389-420415] 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. ........ Merged revisions 420414 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, 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/ ........ Merged
+ revisions 420408 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * res/res_pjsip_pubsub.c: Fix build in dev mode
+
+2014-08-07 19:44 +0000 [r420384-420388] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/bridge.c: Ensure bridges exist when trying to determine
+ bridged parties when publishing transfer information. ........
+ Merged revisions 420387 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/strings.c, include/asterisk/res_pjsip_presence_xml.h,
+ res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c, include/asterisk/strings.h,
+ res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c,
+ include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pidf_body_generator.c: Add support for RFC 4662
+ resource list subscriptions. This commit adds the ability for a
+ user to configure a resource list in pjsip.conf. Subscribing to
+ this list simultaneously subscribes the subscriber to all
+ resources listed. This has the potential to reduce the amount of
+ SIP traffic when loads of subscribers on a system attempt to
+ subscribe to each others' states.
+
+2014-08-07 18:51 +0000 [r420364] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/format_compatibility.h,
+ channels/iax2/format_compatibility.c,
+ channels/iax2/include/codec_pref.h, main/format_compatibility.c,
+ channels/chan_iax2.c, channels/iax2/codec_pref.c,
+ channels/iax2/include/format_compatibility.h: chan_iax2: Several
+ media format fixes. * Fixed the iax.conf bandwidth option. This
+ is the root cause of ASTERISK-24150. * Added checks in
+ iax2_request() to ensure that there are actual formats requested
+ for the new channel to prevent any more fracks from issues like
+ ASTERISK-24150. This is a consequence of the iax.conf bandwidth
+ option not working. * Fixed struct iax2_codec_pref.order member
+ size mismatch issue when converting to and from the codec
+ preference order list passed over the wire. In addition the
+ values sent over the wire are now compatible with previous
+ Asterisk versions. * Fixed several issues dealing with the struct
+ iax2_codec_pref members. Off-by-one, array limit errors, and the
+ order/framing members always need to be updated together. * Made
+ iax2_request() setup the channel's native format preference order
+ according to the user's wishes. The new media format strategy
+ needs the order specified earler. * Fixed usage of
+ ast_format_compatibility_bitfield2format(). The function can
+ return NULL if the bitfield was not associated with a function. *
+ Deleted dead code iax2_codec_pref_getsize() and
+ iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and
+ iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of
+ inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH,
+ IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants
+ again as they were in Asterisk v1.8. * Renamed prefs to
+ prefs_global so it won't get confused with the local pref
+ versions. * Fixed too small buffer in
+ handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in
+ handle_cli_iax2_show_peer() to output complete lines. * Changed
+ struct create_addr_info.prefs to be struct iax2_codec_pref as an
+ optimization so iax2_request() and iax2_call() do less work. *
+ Fixed a potential deadlock in ast_iax2_new() on an off-nominal
+ path when the pbx could not get started. * Made set_config()
+ setup a local prefs list along side the local capability format
+ bitfield. Once the config is loaded, then the local copies are
+ put into the global versions. * Fix unininialized codec_buf in
+ function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott
+ Griepentrog Review: https://reviewboard.asterisk.org/r/3890/
+
+2014-08-07 15:30 +0000 [r420338] Kinsey Moore <kmoore at digium.com>
+
+ * main/stasis_bridges.c, res/ari/ari_model_validators.h,
+ main/channel.c, 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,
+ main/bridge_basic.c, res/stasis/stasis_bridge.c,
+ include/asterisk/stasis_bridges.h, res/stasis/command.h,
+ include/asterisk/stasis_app.h, res/stasis/app.h,
+ res/stasis/control.h, apps/app_queue.c,
+ res/ari/ari_model_validators.c, main/cel.c: 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/ ........ Merged
+ revisions 420325 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-07 14:37 +0000 [r420314-420315] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_publish_asterisk.c (added), res/res_pjsip_pubsub.c,
+ include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_pubsub.exports.in: res_pjsip_publish_asterisk: Add
+ support for exchanging device and mailbox state using SIP. This
+ module uses the inbound and outbound PUBLISH support to exchange
+ device and mailbox state between Asterisk instances. Each
+ instance is configured to publish to the other and requires no
+ intermediary server. The functionality provided is similar to the
+ XMPP and Corosync support. Review:
+ https://reviewboard.asterisk.org/r/3780/
+
+ * res/res_pjsip_outbound_publish.exports.in (added),
+ res/res_pjsip_outbound_publish.c (added),
+ include/asterisk/res_pjsip_outbound_publish.h (added):
+ res_pjsip_outbound_publish: Add module which provides outbound
+ PUBLISH support. This module implements the core parts required
+ for doing outbound PUBLISH. It takes care of configuration,
+ lifetime management, and authentication. Additional modules
+ implement the specific events that are published. Review:
+ https://reviewboard.asterisk.org/r/3780/
+
+2014-08-07 14:17 +0000 [r420289-420309] Matthew Jordan <mjordan at digium.com>
+
+ * main/pbx.c: pbx: Filter out pattern matching hints in responses
+ sent to ExtensionStateList Hints that are a pattern match are
+ technically stored in the hint container in the same fashion as
+ concrete implementations of hints. The pattern matching hints,
+ however, are not "real" in the sense that things can subscribe to
+ them: rather, they are stored in the hints container so that when
+ a subscription is made a "real" hint can be generated for the
+ subscription if one does not yet exist. The extension state core
+ takes care of this correctly by matching against non-pattern
+ matching extensions prior to pattern matching extensions. Because
+ of this, however, the ExtensionStateList AMI action was returning
+ pattern matching hints when executed. These hints are meaningless
+ from the perspective of AMI clients: their state will never
+ change, they cannot be subscribed to, and events would never
+ normally be generated from them. As such, we now filter these out
+ of the response.
+
+ * build_tools/post_process_documentation.py: build_tools: Skip
+ managerEvent combining for AMI action responses AMI action
+ responses can (and will) reference AMI events that they return.
+ These event references and definitions should not be combined
+ with AMI events raised elsewhere in the code, as they are
+ specifically tied to the AMI action that raised them.
+ ASTERISK-24156 #close Reported by: Rusty Newton
+
+2014-08-06 18:12 +0000 [r420212-420237] Richard Mudgett <rmudgett 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. ........ Merged revisions 407567 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /,
+ contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
+ (added): Add alembic script that adds contact user_agent and
+ endpoint message_context. ........ Merged revisions 411514 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * 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/ ........ Merged
+ revisions 420211 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-06 16:12 +0000 [r420149] 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 ........ Merged
+ revisions 420148 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-06 15:32 +0000 [r420144] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * funcs/func_channel.c: Add documentation to the ability to
+ retrieve the source port of a SIP call. (belongs with r419970)
+ ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by
+ dtryba Review: https://reviewboard.asterisk.org/r/3781/
+
+2014-08-06 12:55 +0000 [r420124] Kinsey Moore <kmoore at digium.com>
+
+ * main/stasis_endpoints.c, main/rtp_engine.c,
+ main/security_events.c, main/ccss.c, main/bridge.c,
+ main/devicestate.c, res/res_stasis_snoop.c, main/endpoints.c,
+ main/stasis_bridges.c, main/presencestate.c, main/loader.c,
+ main/stasis.c, main/cdr.c, main/channel.c, main/stasis_message.c,
+ main/stasis_system.c, main/manager.c, main/app.c,
+ pbx/pbx_realtime.c, main/stasis_channels.c,
+ res/res_stasis_test.c, main/stasis_cache.c, main/pickup.c,
+ tests/test_stasis_channels.c, include/asterisk/stasis.h,
+ configs/samples/stasis.conf.sample (added), main/core_local.c,
+ main/named_acl.c, apps/app_queue.c, apps/app_forkcdr.c,
+ funcs/func_cdr.c, res/res_corosync.c, res/res_stun_monitor.c,
+ main/test.c, main/file.c, tests/test_stasis.c, res/res_stasis.c,
+ apps/app_chanspy.c, res/parking/parking_manager.c: Stasis: Allow
+ message types to be blocked This introduces stasis.conf and a
+ mechanism to prevent certain message types from being published.
+ Internally, this works by preventing the chosen message types
+ from being created which ensures that those message types can
+ never be published. This patch also adjusts message publishers
+ such that message payloads are not created if the related message
+ type is not available. ASTERISK-23943 #close Review:
+ https://reviewboard.asterisk.org/r/3823/
+
+2014-08-05 21:48 +0000 [r420098-420100] Matthew Jordan <mjordan at digium.com>
+
+ * res/stasis/messaging.c, /: stasis: Fix compilation issue with ao2
+ tagged objects ........ Merged revisions 420099 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * tests/test_message.c (added), res/res_xmpp.c,
+ include/asterisk/json.h, CHANGES, include/asterisk/manager.h,
+ res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
+ main/json.c, res/res_ari_endpoints.c, include/asterisk/message.h,
+ res/ari/resource_channels.c, main/message.c, res/res_stasis.c,
+ res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json,
+ res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /,
+ channels/chan_sip.c, res/stasis/app.c, res/stasis/messaging.h
+ (added), res/ari/resource_endpoints.h, res/res_pjsip_messaging.c:
+ Multiple revisions 420089-420090,420097 ........ r420089 |
+ mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
+ 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 ........ r420090 | mjordan | 2014-08-05 15:16:37
+ -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties
+ :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue,
+ 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing
+ compilation issue ........ Merged revisions 420089-420090,420097
+ from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-05 13:59 +0000 [r420028] Jonathan Rose <jrose at digium.com>
+
+ * main/format.c: chan_iax2: Fix a crash that occurs when using
+ allow=all for an IAX2 peer Or any combination of codecs that
+ includes Opus. ASTERISK-24107 #close Review:
+ https://reviewboard.asterisk.org/r/3885/
+
+2014-08-04 21:00 +0000 [r420007] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format_cache.c, include/asterisk/format_cache.h: Remove
+ duplicate definitions of ast_format_vp8.
+
+2014-08-04 20:25 +0000 [r419970] Mark Michelson <mmichelson at digium.com>
+
+ * channels/sip/dialplan_functions.c: Add the ability to retrieve
+ the source port of a SIP call. This adds the ability to call
+ CHANNEL(recvport) on chan_sip channels to see the port on which
+ an INVITE was received. ASTERISK-24040 #close Reported by dtryba
+ Patches: dialplan_functions.patch uploaded by dtryba (License
+ #6628) Review: https://reviewboard.asterisk.org/r/3781
+
+2014-08-04 19:47 +0000 [r419945] 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 ........ Merged
+ revisions 419944 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-02 03:37 +0000 [r419914] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_pjsip.c: Manager: Add PJSIPShowEndpoint[s] documentation
+ This adds a large swath of response documentation for
+ PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies
+ heavily on the existing text in the configInfo documentation via
+ xi:include tags to avoid documentation duplication. Review:
+ https://reviewboard.asterisk.org/r/3888/
+
+2014-08-01 14:48 +0000 [r419888] Mark Michelson <mmichelson at digium.com>
+
+ * CHANGES, res/res_pjsip/pjsip_options.c: Add ContactStatusDetail
+ to PJSIPShowEndpoint AMI output. Now when running
+ PJSIPShowEndpoint, you will receive a ContactStatusDetail for
+ each bound contact that Asterisk is qualifying. This information
+ includes the URI of the contact, current reachability, and
+ roundtrip time. AFS-91 #close Reported by Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/3797
+
+2014-07-31 16:19 +0000 [r419851] Jonathan Rose <jrose at digium.com>
+
+ * res/res_pjsip_notify.c, CHANGES: PJSIP: Send Notify AMI and CLI
+ commands can now send to URI instead of endpoint Review:
+ https://reviewboard.asterisk.org/r/3817/
+
+2014-07-31 11:57 +0000 [r419822-419825] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_hep_rtcp.c (added), CHANGES, channels/chan_pjsip.c,
+ res/res_rtp_asterisk.c, main/rtp_engine.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 ........
+ Merged revisions 419823 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * funcs/func_jitterbuffer.c, doc/appdocsxml.dtd, main/xmldoc.c:
+ xmldocs: Add support for an <example> tag in the Asterisk XML
+ Documentation This patch adds support for an <example /> tag in
+ the XML documentation schema. For CLI help, this doesn't change
+ the formatting too much: - Preceeding white space is removed -
+ Unlike with para elements, new lines are preserved However,
+ having an <example /> tag in the XML schema allows for the wiki
+ documentation generation script to surround the documentation
+ with {code} or {noformat} tags, generating much better content
+ for the wiki - and allowing us to put dialplan examples (and
+ other code snippets, if desired) into the documentation for an
+ application/function/AMI command/etc. Review:
[... 19717 lines stripped ...]
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