[svn-commits] bebuild: tag 12.5.0-rc1 r420813 - in /tags/12.5.0-rc1: ./ contrib/realtime/my...

SVN commits to the Digium repositories svn-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
==============================================================================
--- tags/12.5.0-rc1/.lastclean (added)
+++ tags/12.5.0-rc1/.lastclean Mon Aug 11 13:54:48 2014
@@ -1,0 +1,1 @@
+40

Propchange: tags/12.5.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/12.5.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/12.5.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/12.5.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/12.5.0-rc1/.version?view=auto&rev=420813
==============================================================================
--- tags/12.5.0-rc1/.version (added)
+++ tags/12.5.0-rc1/.version Mon Aug 11 13:54:48 2014
@@ -1,0 +1,1 @@
+12.5.0-rc1

Propchange: tags/12.5.0-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/12.5.0-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/12.5.0-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/12.5.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.5.0-rc1/ChangeLog?view=auto&rev=420813
==============================================================================
--- tags/12.5.0-rc1/ChangeLog (added)
+++ tags/12.5.0-rc1/ChangeLog Mon Aug 11 13:54:48 2014
@@ -1,0 +1,29599 @@
+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 ...]



More information about the svn-commits mailing list