[asterisk-commits] bebuild: tag 12.2.0-rc1 r411558 - in /tags/12.2.0-rc1: ./ contrib/realtime/my...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 28 13:36:57 CDT 2014


Author: bebuild
Date: Fri Mar 28 13:36:53 2014
New Revision: 411558

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411558
Log:
Importing files for 12.2.0-rc1 release.

Added:
    tags/12.2.0-rc1/.lastclean   (with props)
    tags/12.2.0-rc1/.version   (with props)
    tags/12.2.0-rc1/ChangeLog   (with props)
    tags/12.2.0-rc1/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/12.2.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

Added: tags/12.2.0-rc1/.lastclean
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Added: tags/12.2.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.2.0-rc1/ChangeLog?view=auto&rev=411558
==============================================================================
--- tags/12.2.0-rc1/ChangeLog (added)
+++ tags/12.2.0-rc1/ChangeLog Fri Mar 28 13:36:53 2014
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+2014-03-28  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 12.2.0-rc1 Released.
+
+2014-03-28 18:09 +0000 [r411534]  Matthew Jordan <mjordan at digium.com>
+
+	* include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added),
+	  res/res_hep.exports.in (added), CHANGES, configs/hep.conf.sample
+	  (added), res/res_hep.c (added): res_hep/res_hep_pjsip: Add a
+	  HEPv3 capture agent module and a logger for PJSIP This patch adds
+	  the following: (1) A new module, res_hep, which implements a
+	  generic packet capture agent for the Homer Encapsulation Protocol
+	  (HEP) version 3. Note that this code is based on a patch provided
+	  by Alexandr Dubovikov; I basically just wrapped it up, added
+	  configuration via the configuration framework, and threw in a
+	  taskprocessor. (2) A new module, res_hep_pjsip, which forwards
+	  all SIP message traffic that passes through the res_pjsip stack
+	  over to res_hep for encapsulation and transmission to a HEPv3
+	  capture server. Much thanks to Alexandr for his Asterisk patch
+	  for this code and for a *lot* of patience waiting for me to port
+	  it to 12/trunk. Due to some dithering on my part, this has taken
+	  the better part of a year to port forward (I still blame CDRs for
+	  the delay). ASTERISK-23557 #close Review:
+	  https://reviewboard.asterisk.org/r/3207/
+
+2014-03-28 17:52 +0000 [r411532]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/oochannels.c,
+	  addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c,
+	  addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+	  addons/chan_ooh323.c, /: process stack command even if gatekeeper
+	  client isn't register don't destroy gatekeeper client if it is
+	  not started don't destroy gatekeeper client in some sort of
+	  gatekeeper errors signal rtp create condition when call cleared
+	  before rtp structure created (closes issue ASTERISK-23460)
+	  Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch
+	  Tested by: Dmitry Melekhov ........ Merged revisions 411531 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 17:35 +0000 [r411529]  Matthew Jordan <mjordan at digium.com>
+
+	* rest-api/api-docs/applications.json,
+	  rest-api/api-docs/playbacks.json, UPGRADE.txt,
+	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+	  rest-api/resources.json, CHANGES, include/asterisk/manager.h,
+	  rest-api/api-docs/bridges.json,
+	  rest-api/api-docs/recordings.json,
+	  rest-api/api-docs/deviceStates.json,
+	  rest-api/api-docs/endpoints.json,
+	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
+	  rest-api/api-docs/asterisk.json: Update API versions and
+	  UPGRADE/CHANGES for 12.2.0 This patch does the following: * It
+	  updates the AMI version to 2.2.0 to indicate backwards compatible
+	  changes have been made since the last release * It updates the
+	  ARI version to 1.2.0 to indicate backwards compatible changes
+	  have been made since the last release * It updates the
+	  UPGRADE/CHANGES files with changes that were not mentioned
+
+2014-03-28 17:08 +0000 [r411514]  Mark Michelson <mmichelson at digium.com>
+
+	* 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.
+
+2014-03-28 16:48 +0000 [r411512]  Matthew Jordan <mjordan at digium.com>
+
+	* /, res/res_odbc.exports.in, UPGRADE.txt, res/res_odbc.c,
+	  configs/res_odbc.conf.sample, include/asterisk/res_odbc.h,
+	  res/res_config_odbc.c: res_config_odbc/res_odbc: Fix handling of
+	  non-text columns updates with empty values. This patch fixes
+	  setting nullable integer columns to NULL instead of an empty
+	  string, which fails for PostgreSQL, for example. The current code
+	  is supposed to do so, but the check is broken. The patch also
+	  allows the first column in the list to be a nullable integer.
+	  This patch also adds a compatibility setting in res_odbc.conf,
+	  allow_empty_string_in_nontext. It is enabled by default. It
+	  should be disabled for database backends (such as PostgreSQL)
+	  that require NULL instead of an empty string for Integer columns.
+	  Review: https://reviewboard.asterisk.org/r/3375 (issue
+	  ASTERISK-23459) Reported by: zvision patches:
+	  res_config_odbc.diff uploaded by zvision (License 5755) ........
+	  Merged revisions 411399 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411408 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 16:17 +0000 [r411465]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/tcptls.c, main/manager.c, /, main/http.c: http: response
+	  body often missing after specific request This patch works around
+	  a problem with the HTTP body being dropped from the response to a
+	  specific client and under specific circumstances: a) Client
+	  request comes from node.js user agent "Shred" via use of
+	  swagger-client library. b) Asterisk and Client are *not* on the
+	  same host or TCP/IP stack In testing this problem, it has been
+	  determined that the write of the HTTP body is lost, even if the
+	  data is written using low level write function. The only solution
+	  found is to instruct the TCP stack with the shutdown function to
+	  flush the last write and finish the transmission. See review for
+	  more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
+	  Reported by: Sam Galarneau Review:
+	  https://reviewboard.asterisk.org/r/3402/ ........ Merged
+	  revisions 411462 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411463 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-28 16:00 +0000 [r411374-411461]  Matthew Jordan <mjordan at digium.com>
+
+	* /: Remove block on 411408
+
+	* /, UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between
+	  1.4 and 1.8+ systems. ........ Merged revisions 411457 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411458 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* contrib/realtime/mysql/voicemail_messages.sql (removed),
+	  contrib/realtime/postgresql/realtime.sql (removed),
+	  contrib/realtime/mysql/voicemail_data.sql (removed),
+	  contrib/realtime/mysql/musiconhold.sql (removed),
+	  contrib/realtime/mysql/queue_log.sql (removed),
+	  contrib/realtime/mysql/voicemail.sql (removed),
+	  contrib/realtime/mysql/sippeers.sql (removed),
+	  contrib/realtime/mysql/iaxfriends.sql (removed),
+	  contrib/realtime/mysql/meetme.sql (removed): contrib/realtime:
+	  Remove empty SQL script files Since the relatime scripts are now
+	  managed by Alembic, the previous realtime scripts were previously
+	  removed. However, the removal process messed up, as the files
+	  were still in the repository. The contents were just empty. This
+	  removes the files from the tree.
+
+	* channels/sip/include/sip.h, /: chan_sip: Add MESSAGE request to
+	  allowed methods The allowed methods advertised by chan_sip did
+	  not previously note the MESSAGE request. Even in Asterisk 1.8, we
+	  do accept in-dialog MESSAGE requests; we should advertise that we
+	  support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
+	  #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
+	  Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
+	  Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged
+	  revisions 411372 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411373 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-27 19:15 +0000 [r411311-411315]  Corey Farrell <git at cfware.com>
+
+	* main/message.c, apps/app_jack.c, funcs/func_dialplan.c,
+	  channels/chan_sip.c, funcs/func_math.c,
+	  funcs/func_jitterbuffer.c, res/res_mutestream.c,
+	  funcs/func_global.c, apps/app_speech_utils.c,
+	  res/res_pjsip_header_funcs.c, funcs/func_callcompletion.c,
+	  funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c,
+	  apps/app_stack.c, funcs/func_callerid.c, res/res_calendar.c,
+	  apps/app_voicemail.c, funcs/func_speex.c, /,
+	  funcs/func_strings.c, res/res_xmpp.c, res/res_jabber.c,
+	  main/features_config.c, channels/chan_iax2.c,
+	  apps/confbridge/conf_config_parser.c,
+	  channels/pjsip/dialplan_functions.c, funcs/func_groupcount.c,
+	  funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
+	  funcs/func_frame_trace.c: Fix dialplan function NULL channel
+	  safety issues (closes issue ASTERISK-23391) Reported by: Corey
+	  Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........
+	  Merged revisions 411313 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411314 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* include/asterisk.h, /, main/format.c: main/formats: Fix crash in
+	  ast_format_cmp during non-clean shutdown. * Update asterisk.h to
+	  reflect availability of ast_register_cleanup in 11.9. * Use
+	  ast_register_cleanup for format_attr_shutdown. (closes issue
+	  ASTERISK-23103) Reported by: JoshE ........ Merged revisions
+	  411310 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-27 14:20 +0000 [r411295]  Mark Michelson <mmichelson at digium.com>
+
+	* main/sorcery.c: Give sorcery instances a reference to their
+	  wizards. On graceful shutdown, sorcery wizards are all killed
+	  off, but it is possible for sorcery instances to still have
+	  dangling pointers after this, possibly causing a crash. Giving
+	  the sorcery instances a reference to their wizards ensures that
+	  the wizard reference will remain valid for the lifetime of the
+	  sorcery instance. Review: https://reviewboard.asterisk.org/r/3401
+
+2014-03-26 22:44 +0000 [r411245]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/say.c: say: Fix a bug where SayNumber in Polish tries to
+	  play incorrect sound. This change fixes a bug where calling
+	  SayNumber with a number divisible by 100 using the Polish
+	  language would cause the code to attempt to play a sound file
+	  with an empty name. (closes issue ASTERISK-23509) Reported by:
+	  zvision Review: https://reviewboard.asterisk.org/r/3378/ ........
+	  Merged revisions 411243 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411244 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-26 16:07 +0000 [r411193]  Jonathan Rose <jrose at digium.com>
+
+	* configs/sip.conf.sample, /, channels/chan_sip.c: chan_sip: Send
+	  real CallerID information with P-Assserted-Identity (RFC-3325)
+	  Prior too this patch, the P-Asserted-Identity header would
+	  include anonymous caller id information which seems to go against
+	  the point of the P-Asserted-Identity header. Now the real caller
+	  ID information will be included in this header. Also, no privacy
+	  header would be included. This patch adds 'Privacy: id' to
+	  outgoing SIP messages that include the P-Asserted-Identity
+	  header. (closes issue AST-1301) ........ Merged revisions 411189
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 411190 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-26 16:03 +0000 [r411191]  Richard Mudgett <rmudgett at digium.com>
+
+	* contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py:
+	  Fix 'alembic branches' merge conflict as described by the web
+	  page.
+
+2014-03-25 18:43 +0000 [r411173]  Sean Bright <sean at malleable.com>
+
+	* res/ari/config.c: ARI: Don't complain about missing ARI users
+	  when we aren't enabled Currently, if ARI is not enabled it will
+	  still complain that there are no configured users. This patch
+	  checks to see if ARI is enabled before logging and error or
+	  iterating the container to validate the users. Review:
+	  https://reviewboard.asterisk.org/r/3391/
+
+2014-03-25 17:52 +0000 [r411157-411159]  Mark Michelson <mmichelson at digium.com>
+
+	* tests/test_sorcery.c, tests/test_sorcery_realtime.c,
+	  main/sorcery.c, res/res_mwi_external.c,
+	  res/res_pjsip/config_system.c, configs/sorcery.conf.sample,
+	  main/bucket.c, include/asterisk/sorcery.h,
+	  res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c:
+	  Prevent duplicate sorcery wizards from being applied to sorcery
+	  object types. This commit contains several changes to sorcery: 1)
+	  Application of sorcery configuration based on module name is
+	  automatically performed when sorcery is opened for a module. 2)
+	  Sorcery will not attempt to apply the same wizard to an object
+	  type more than once. 3) Sorcery gives more exact results when
+	  attempting to apply a wizard, whether as the default or based on
+	  configuration. Sorcery unit tests still pass for me after making
+	  these changes. Review: https://reviewboard.asterisk.org/r/3326
+
+	* res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
+	  res/res_pjsip_messaging.c, res/res_pjsip.c,
+	  include/asterisk/res_pjsip.h: Add a "message_context" option for
+	  PJSIP endpoints.
+
+2014-03-25 16:55 +0000 [r411141]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/res_pjsip.h, res/res_pjsip/pjsip_options.c,
+	  res/res_pjsip.c: res_pjsip: Fix contact authenticate_qualify
+	  endpoint lookup when qualifing a contact. * Fixed bad use of
+	  ao2_find() in on_endpoint(). * Replaced use of find_endpoints()
+	  with find_an_endpoint() since only the first found endpoint is
+	  ever needed. * Fixed qualify_contact_cb() to update the contact
+	  with the aor authenticate_qualify setting. Otherwise, permanent
+	  contacts in the aor type sections would have a config line order
+	  dependancy. * Fixed off nominal path contact ref leak in
+	  qualify_contact(). The comment saying the unref is not needed was
+	  wrong. * Fixed off nominal path use of the endpoint parameter if
+	  it is NULL in send_out_of_dialog_request(). * Added missing off
+	  nominal path unref of pjsip tdata in
+	  send_out_of_dialog_request(). * Fixed off nominal path failing to
+	  call the callback in send_request_cb() when the request is
+	  challenged for authentication. * Eliminated silly RAII_VAR() use
+	  in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen
+	  to better reflect reality. (closes issue ASTERISK-23254) Reported
+	  by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/
+
+2014-03-25 16:04 +0000 [r411091]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
+	  update_provisional_keepalive() is called while
+	  send_provisional_keepalive_full() is waiting on the PVT lock,
+	  then pvt->provisional_keepalive_sched_id will be changed to a new
+	  sched_id value by update_provisional_keepalive(), but that new
+	  sched_id then may be overwritten with -1 by
+	  send_provisional_keepalive_full(), killing the pvt's reference to
+	  a schedule and "leaking" the reference. (closes issue
+	  ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
+	  Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
+	  Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
+	  (license 5012) ........ Merged revisions 411088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411089 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-25 15:44 +0000 [r411086]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_stasis.c: ARI: Resolve a subscription leak against
+	  implicit bridge subscriptions When a channel in a stasis
+	  application is joined to a bridge, a subscription for that bridge
+	  is created implicitly for the stasis application serving the
+	  channel. Prior to this patch, subsequent removals of the channel
+	  from the bridge would leave the subscription open. Review:
+	  https://reviewboard.asterisk.org/r/3380/
+
+2014-03-24 21:38 +0000 [r411023]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Always use fromdomain if set
+	  for domain, even if callerid is set to restricted. (closes issue
+	  ASTERISK-20841) Reported by: Kelly Goedert ........ Merged
+	  revisions 411021 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 411022 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-21 16:01 +0000 [r410995]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_pjsip_registrar.c: res_pjsip_registrar.c: Miscellaneous
+	  cleanup in rx_task(). * Fix variable shadowing of 'updated' by
+	  renaming it to 'contact_update'. * Checked 'contact_update' for
+	  ast_sorcery_copy() failure. * Removed silly use of RAII_VAR() for
+	  'contact_update'.
+
+2014-03-20 22:54 +0000 [r410966]  Jonathan Rose <jrose at digium.com>
+
+	* /, apps/app_confbridge.c: app_confbridge: Fix bug - users with
+	  startmuted set don't start muted (closes issue ASTERISK-23461)
+	  Reported by: Chico Manobela Review:
+	  https://reviewboard.asterisk.org/r/3373/ ........ Merged
+	  revisions 410965 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-20 16:27 +0000 [r410949]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/channel.h, res/ari/resource_channels.c,
+	  res/res_stasis_snoop.c, include/asterisk/rtp_engine.h,
+	  main/dial.c, main/manager.c, main/channel_internal_api.c,
+	  main/core_unreal.c: assigned-uniqueids: Miscellaneous cleanup and
+	  fixes. * Fix memory leak in ast_unreal_new_channels(). Made it
+	  generate the ;2 uniqueid on a stack variable instead of mallocing
+	  it. * Made send error response to ARI and AMI requests instead of
+	  just logging excessive uniqueid length and allowing truncation.
+	  action_originate() and ari_channels_handle_originate_with_id(). *
+	  Fixed minor truncating uniqueid hole when generating the ;2
+	  uniqueid string length. Created public and internal lengths of
+	  uniqueid. The internal length can handle a max public uniqueid
+	  plus an appended ;2. * free() and ast_free() are NULL tolerant so
+	  they don't need a NULL test before calling. * Made use better
+	  struct initialization format instead of the position dependent
+	  initialization format. Also anything not explicitly initialized
+	  in the struct is initialized to zero by the compiler. * Made
+	  ast_channel_internal_set_fake_ids() use the safer
+	  ast_copy_string() instead of strncpy(). Review:
+	  https://reviewboard.asterisk.org/r/3371/
+
+2014-03-19 17:26 +0000 [r410933]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for identify
+	  sections to be specified in sorcery.conf. "identify" is a special
+	  type of configuration object in PJSIP because unlike the other
+	  objects, it is not provided by the base res_pjsip module.
+	  Instead, it is provided by the res_pjsip_endpoint_identifier_ip
+	  module. If using the default sorcery wizard
+	  (config,criteria=type=identify) then things work because the
+	  module that applies the default wizard is the correct module.
+	  However, if attempting to use sorcery.conf to apply an alternate
+	  wizard, it was not possible. If you attempted to specify the
+	  identify object type in the res_pjsip section, then the object
+	  could not be registered since the object was undocumented for the
+	  res_pjsip module. There was no alternate configuration section
+	  defined for it, so you were out of luck if you wanted to override
+	  the default wizard. With this change, the identify section will
+	  properly have a sorcery.conf-based wizard applied when the
+	  identify definition is within the
+	  res_pjsip_endpoint_identifier_ip section.
+
+2014-03-19 14:24 +0000 [r410904-410918]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_stasis.c: res_stasis: Fix a bug where the default bridge
+	  type was not set.
+
+	* CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json,
+	  res/ari/resource_bridges.h: res_stasis: Extend bridge type to be
+	  a comma separated list of bridge attributes. This change turns
+	  the bridge type field into a comma separated list of attributes.
+	  These attributes include: mixing, holding, dtmf_events, and
+	  proxy_media. By setting the various attributes a user can control
+	  the type of bridge created with the behavior they need for their
+	  application. (closes issue ASTERISK-23437) Reported by: Matt
+	  Jordan Review: https://reviewboard.asterisk.org/r/3359/
+
+2014-03-19 02:29 +0000 [r410890]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_ari.c: res_ari: Fix documentation schema error
+
+2014-03-18 23:31 +0000 [r410876]  Rusty Newton <rnewton at digium.com>
+
+	* res/res_ari.c: res_ari: Add notes about Asterisk HTTP server to
+	  the "enabled" config option for the res_ari general section Added
+	  note and see-also reminding user to enable the HTTP server.
+	  (closes issue ASTERISK-22499) Reported by: Rusty Newton
+
+2014-03-18 15:28 +0000 [r410861]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cdr.c: cdr: Add asserts for when we don't know about a CDR
+	  for a channel In the CDR core, every channel should either be
+	  filtered out (due to being an 'internal' channel used as an
+	  implementation detail, such as playing media back into a bridge)
+	  or it should get a CDR. Even if that CDR ends up being discarded,
+	  we still give the channel a CDR in case we end up needing it. If
+	  we hit a situation where a channel does not have a CDR, we should
+	  blow up in -dev-mode. Asserts are appropriate for that. This
+	  patch adds those asserts, as they would have quickly caught the
+	  error fixed by r410814.
+
+2014-03-18 14:51 +0000 [r410858]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/http.c: ARI: allow json content type with zero length body
+	  When a request was received with a Content-type of json, the body
+	  was sent for json parsing - even if it was zero length. This
+	  resulted in ARI requests failing that were valid, such as a
+	  channel DELETE with no parameters. The code has now been changed
+	  to skip json parsing with zero content length. (closes issue
+	  SWP-6748) Reported by: Samuel Galarneau Review:
+	  https://reviewboard.asterisk.org/r/3360/
+
+2014-03-18 12:45 +0000 [r410844]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of
+	  nameservers in off-nominal resolver creation failure. Thanks
+	  Walter Doekes!
+
+2014-03-18 11:51 +0000 [r410830]  Sean Bright <sean at malleable.com>
+
+	* res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when
+	  available. Per Johann Steinwendtner on the asterisk-dev mailing
+	  list:
+	  http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
+	  g711_free() was introduced in spandsp 0.0.6pre4 and
+	  g711_release() became a noop. I opted not to remove the call to
+	  g711_release() since it is harmless and to call g711_free() if we
+	  have a sufficiently recent version of spandsp. (issue
+	  ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged
+	  revisions 410829 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-18 02:02 +0000 [r410813]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/stasis_cache.c: stasis_cache: Use the right variable in the
+	  cache entry ao2 cmp function.
+
+2014-03-17 22:53 +0000 [r410793-410795]  Joshua Colp <jcolp at digium.com>
+
+	* CHANGES, res/res_pjsip/include/res_pjsip_private.h,
+	  res/res_pjsip.c, main/dns.c, res/res_pjsip/config_system.c,
+	  include/asterisk/dns.h: res_pjsip: Enable PJSIP DNS client
+	  support. This change enables DNS client support within PJSIP.
+	  System nameservers are automatically discovered using res_init or
+	  res_ninit. If this fails then PJSIP will resort to using
+	  gethostbyname for resolution. By enabling this support we gain
+	  SRV support, failover, and weight support. (closes issue
+	  ASTERISK-23435) Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3343/
+
+	* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Make address
+	  replacement less aggressive. This change makes the
+	  res_pjsip_multihomed module less aggressive when changing the
+	  address in messages. It will now only occur if the transport in
+	  use is bound to the any address OR if the system determined
+	  source address matches the bound address of the transport in use.
+	  Review: https://reviewboard.asterisk.org/r/3369/
+
+2014-03-17 21:56 +0000 [r410747-410750]  Russ Meyerriecks <rmeyerreicks at digium.com>
+
+	* /, main/callerid.c: !fixup: callerid: Logic error in checksum
+	  processing Fixes syntax error in previous commit :-( ........
+	  Merged revisions 410748 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 410749 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/callerid.c, /: callerid: Logic error in checksum processing
+	  Callerid checksum-ing was being handled incorrectly here. When
+	  the checksum is calculated to be 0x00, it will perform 0x100-0x00
+	  which results in 0x100. This value will then fail the otherwise
+	  correct callerid message. This patch changes the logic to simply
+	  add the calculated checksum to the transmitted 2's compliment
+	  checksum. Review: https://reviewboard.asterisk.org/r/3356/
+	  (closes issue ASTERISK-23488) ........ Merged revisions 410710
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 410717 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-17 18:36 +0000 [r410673-410696]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_mwi_external.c, res/res_pjsip/config_system.c,
+	  configs/sorcery.conf.sample, include/asterisk/sorcery.h,
+	  res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
+	  tests/test_sorcery.c, tests/test_sorcery_realtime.c,
+	  main/sorcery.c: Revert changes to sorcery that accidentally got
+	  committed. These changes were still up for review and have not
+	  been approved yet. I must have had the changes in my working copy
+	  when making a different change.
+
+	* tests/test_sorcery.c, main/channel.c,
+	  res/res_pjsip/config_system.c, res/res_mwi_external.c,
+	  include/asterisk/bridge_channel.h, funcs/func_frame_trace.c,
+	  configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c,
+	  include/asterisk/sorcery.h, tests/test_sorcery_astdb.c,
+	  include/asterisk/frame.h, main/bridge_channel.c,
+	  tests/test_sorcery_realtime.c, main/sorcery.c,
+	  res/res_stasis_playback.c, main/frame.c,
+	  bridges/bridge_softmix.c: Fix stuck channel in ARI through the
+	  introduction of synchronous bridge actions. Playing back a file
+	  to a channel in an ARI bridge would attempt to wait until the
+	  playback concluded before returning. The method used involved
+	  signaling the waiting thread in the ARI custom playback function.
+	  The problem with this is that there were some corner cases that
+	  were not accounted for: * If a bridge channel could not be found,
+	  then we never would attempt the playback but would still attempt
+	  to wait for the playback to complete. * If the bridge playfile
+	  action failed to queue, we would still attempt to wait for the
+	  playback to complete. * If the bridge playfile action were queued
+	  but some circumstance caused the playback not to occur (the
+	  bridge dies, the channel is removed from the bridge), then we
+	  would never be notified. The solution to this is to move the
+	  waiting logic into the bridge code. A new bridge API function is
+	  added to queue a synchronous action on a bridge. The waiting
+	  thread is notified when the queued frame has been freed, either
+	  due to an error occurring or due to successful playback. As a
+	  failsafe, the waiting thread has a 10 minute timeout just in case
+	  there is a frame leak somewhere. Review:
+	  https://reviewboard.asterisk.org/r/3338
+
+2014-03-17 16:42 +0000 [r410671]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_chan_announce.c: app_confbridge: Add missing
+	  destructor call to announcer channel destructor.
+
+2014-03-16 20:20 +0000 [r410650]  Matthew Jordan <mjordan at digium.com>
+
+	* res/stasis/app.c: stasis/app.c: Add some extra debugging for
+	  subscription counts Events are sent to a connected ARI
+	  application based on the things that ARI application cares about.
+	  These subscriptions can be set up implicitly - such as when that
+	  ARI application creates a new object - or explicitly, via the
+	  application resource's subscription operations. Debugging *why*
+	  something was being sent to an application - or why something was
+	  not being sent to an application - was a bit tricky, as there was
+	  no debug information for the subscriptions. This patch adds some
+	  debug level 3 statements that show the subscription counts for
+	  applications. (Level 3 was chosen as it matches the verbose level
+	  3 statements elsewhere)
+
+2014-03-14 21:55 +0000 [r410625]  Mark Michelson <mmichelson at digium.com>
+
+	* tests/test_sorcery_realtime.c: Fix failing realtime sorcery
+	  tests. The store realtime callback needs to return a positive
+	  value for sorcery to treat the store as a success.
+
+2014-03-14 21:28 +0000 [r410623]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c, /: manager: fix memory leak in manager_add_filter
+	  function (closes issue ASTERISK-23420) Reported by: Etienne
+	  Lessard Patches: manager_eventfilter_leak uploaded by Etienne
+	  Lessard (license 6394) ........ Merged revisions 410609 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-14 20:53 +0000 [r410590-410607]  Mark Michelson <mmichelson at digium.com>
+
+	* main/db.c, /: Remove an extra ast_cond_wait() that slipped
+	  through the patch. ........ Merged revisions 410606 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/config.c, res/res_sorcery_realtime.c: Handle the return
+	  values of realtime updates and stores more accurately. Realtime
+	  backends' update and store callbacks return the number of rows
+	  affected, or -1 if there was a failure. There were a couple of
+	  issues: * The config API was treating 0 as a successful return,
+	  and positive values as a failure. Now the config API treats
+	  anything >= 0 as a success. * res_sorcery_realtime was treating 0
+	  as a successful return from the store procedure, and any positive
+	  values as a failure. Now sorcery treats anything > 0 as a
+	  success. It still considers 0 a "failure" since there is no
+	  change to report to observers. Review:
+	  https://reviewboard.asterisk.org/r/3341
+
+	* res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited and
+	  solicited MWI to an endpoint. If an endpoint is receiving
+	  unsolicited MWI for a mailbox and then attempts to subscribe to
+	  an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
+	  is rejected with a 500 response. Review:
+	  https://reviewboard.asterisk.org/r/3345
+
+2014-03-14 17:56 +0000 [r410588]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* CHANGES: uniqueid: Update CHANGES to reflect new features Note
+	  the new features provided by uniqueid in the CHANGES file. (issue
+	  ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/
+
+2014-03-14 16:26 +0000 [r410574]  Jonathan Rose <jrose at digium.com>
+
+	* CHANGES, res/res_pjsip/config_transport.c,
+	  include/asterisk/acl.h, main/acl.c,
+	  res/res_pjsip/pjsip_configuration.c: PJSIP: TOS values should be
+	  represented as decimals in sorcery objects (closes issue
+	  ASTERISK-23235) Reported by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3324/
+
+2014-03-14 16:11 +0000 [r410559]  Mark Michelson <mmichelson at digium.com>
+
+	* main/db.c, /: Prevent delayed astdb syncs. The syncing thread
+	  sleeps for a second before waiting to be told to attempt to sync
+	  again. If a signal were sent during this sleeping period, we
+	  would end up having to wait until the next sync signal occurred
+	  in order to sync up the astdb. This code rearrangement also
+	  ensures that any pending transactions will be synced prior to
+	  Asterisk shutting down. Patches: db_sync.patch by John Hardin
+	  (License #6512) ........ Merged revisions 410556 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-14 16:05 +0000 [r410558]  Jonathan Rose <jrose at digium.com>
+
+	* res/ari/resource_bridges.c: ARI/bridges: Forward
+	  Playback/Recording Started/Finished to bridge topic (closes issue
+	  ASTERISK-23444) Reported by: Ben Merrills Review:
+	  https://reviewboard.asterisk.org/r/3340/
+
+2014-03-14 15:55 +0000 [r410541-410555]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/app.h, res/res_mwi_external.c, main/app.c:
+	  res_mwi_external: Clear the stasis cache entry when the external
+	  MWI is deleted. One of the things missing when external MWI
+	  support was added was the ability to clear the stasis cache entry
+	  of deleted external MWI mailboxes. Review:
+	  https://reviewboard.asterisk.org/r/3325/
+
+	* main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal
+	  path of handle_dial_message(). * Trivial common code hoisting in
+	  handle_bridge_leave_message(). * Some whitespace fixing.
+
+2014-03-13 19:30 +0000 [r410527]  Kinsey Moore <kmoore at digium.com>
+
+	* res/stasis/control.c, res/stasis/control.h, res/res_stasis.c:
+	  ARI: Ensure managing application receives ChannelEnteredBridge
+	  messages This fixes an issue where a Stasis application running
+	  over ARI and subscribed to ari/events could miss the
+	  ChannelEnteredBridge event because it did not subscribe to the
+	  new bridge fast enough. To accomplish this, it subscribes the
+	  application controlling the channel to the new bridge before
+	  adding it to that bridge which required the stasis_app_control
+	  structure to maintain a reference to the stasis_app. (closes
+	  issue ASTERISK-23295) Review:
+	  https://reviewboard.asterisk.org/r/3336/
+
+2014-03-13 13:24 +0000 [r410509-410510]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Remove change
+	  for testing fix.
+
+	* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Fix a bug where
+	  the 200 OK for a REGISTER would contain the wrong contact.
+
+2014-03-12 19:05 +0000 [r410491-410493]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_musiconhold.c, main/channel.c: res_musiconhold.c:
+	  Generate MOH start/stop events whenever the MOH stream is
+	  started/stopped. * Made res_musiconhold.c always post the
+	  MusicOnHoldStart/MusicOnHoldStop events when it actually
+	  starts/stops the music streams. This allows the events to always
+	  happen when MOH starts/stops. The event posting code was moved to
+	  the MOH alloc/release routines. * Made channel_do_masquerade()
+	  stop any MOH on the original channel before masquerading so the
+	  original channel will get a stop event with correct information.
+	  * Cleaned up a couple odd codings in moh_files_alloc() and
+	  moh_alloc() dealing with the music state variable. (issue
+	  ASTERISK-23311) Reported by: Benjamin Keith Ford Review:
+	  https://reviewboard.asterisk.org/r/3306/
+
+	* apps/confbridge/conf_state.c,
+	  apps/confbridge/conf_state_single.c,
+	  apps/confbridge/conf_state_inactive.c,
+	  apps/confbridge/conf_state_single_marked.c, /: app_confbridge:
+	  Make explicitly stop MOH if a user is kicked or hangs up while
+	  MOH is playing. When MOH is playing to a user in a conference and
+	  the user is kicked or hangs up from the conference then the AMI
+	  MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
+	  MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
+	  by: Benjamin Keith Ford Review:
+	  https://reviewboard.asterisk.org/r/3306/ ........ Merged
+	  revisions 410490 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-12 12:50 +0000 [r410451-410471]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_multihomed.c: res_pjsip_multihomed: Fix a bug where
+	  outgoing messages for TCP would go out using UDP. This change
+	  fixes a bug where the code which changes the transport did not
+	  check whether the message is going out over UDP or not before
+	  changing it. For TCP and TLS transports we don't need to change
+	  the transport as the correct one is already chosen.
+
+	* res/res_pjsip_multihomed.c (added): res_pjsip_multihomed: Add
+	  module which places the correct address within messages. Due to
+	  how messages are handled within PJSIP it is not until a message
+	  is actually sent that the destination is reliably known. This
+	  means that the addresses placed within the message may not be of
+	  the interface the message is being sent out on. This module
+	  determines what interface a message is being sent on and updates
+	  the message to contain the correct address if applicable. This
+	  module was tested by myself in a virtualized environment with
+	  multiple interfaces and also by Kinsey Moore in the following
+	  configuration: Networks: * 10.24.16.0/21 ** hard phone ** default

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