[asterisk-commits] bebuild: tag 13.2.0-rc1 r431516 - in /tags/13.2.0-rc1: ./ contrib/realtime/my...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jan 30 15:48:18 CST 2015


Author: bebuild
Date: Fri Jan 30 15:48:16 2015
New Revision: 431516

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=431516
Log:
Importing files for 13.2.0-rc1 release.

Added:
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    tags/13.2.0-rc1/.version   (with props)
    tags/13.2.0-rc1/ChangeLog   (with props)
    tags/13.2.0-rc1/contrib/realtime/mysql/mysql_cdr.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/oracle/oracle_cdr.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

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Added: tags/13.2.0-rc1/ChangeLog
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--- tags/13.2.0-rc1/ChangeLog (added)
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+2015-01-30  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 13.2.0-rc1 Released.
+
+2015-01-30 17:44 +0000 [r431492]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_agent_pool.c: app_agent_pool: Fix initial module load
+	  agent device state reporting. When the app_agent_pool module
+	  initially loads there is a race condition between the thread
+	  loading agents.conf and the device state internal processing
+	  thread. If the device state internal processing thread handles
+	  the agent creation state updates before the thread that loaded
+	  agents.conf registers the device state provider callback then the
+	  cached agent state is "Invalid". When a consumer module like
+	  app_queue asks for the agent state it gets the cached "Invalid"
+	  state instead of the real state from the provider. * Moved
+	  loading the agents.conf configuration to the last thing setup by
+	  app_agent_pool in load_module(). Now the device state provider
+	  callback is registered before the config is loaded so the agent
+	  creation state updates are guaranteed to get the initial device
+	  state. * Removed some now redundant config cleanup on error in
+	  load_config(). * Added lock protection when accessing the device
+	  state in agent_pvt_devstate_get() and eliminated the RAII_VAR()
+	  usage. ASTERISK-24737 #close Reported by: Steve Pitts Review:
+	  https://reviewboard.asterisk.org/r/4390/
+
+2015-01-30 17:38 +0000 [r431490]  Kevin Harwell <kharwell at digium.com>
+
+	* res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish:
+	  eventually crashes when no response is ever received When
+	  Asterisk attempts to send SIP outbound publish information and no
+	  response is ever received (no 200 okay, 412, 423) the system
+	  eventually crashes. A response is never received because the
+	  system Asterisk is attempting to send publish information to is
+	  not available. The underlying pjsip framework attempts to send
+	  publish information. After several attempts it calls back into
+	  the Asterisk outbound publish code. At this point if the
+	  "client->queue" is empty Asterisk attempts to schedule a refresh
+	  which utilizes "rdata" and since no response was received the
+	  given "rdata" struture is NULL. Attempting to dereference a NULL
+	  object of course results in a crash. The fix here removes the
+	  dependency on rdata for schedule_publish_refresh. Instead
+	  param->expiration is now passed to it as this is set to -1 if no
+	  response is received. Also added a notification when no response
+	  is received. ASTERISK-24635 #close Reported by: Marco Paland
+	  Review: https://reviewboard.asterisk.org/r/4384/
+
+2015-01-30 16:52 +0000 [r431471]  asanders <asanders at localhost>:
+
+	* include/asterisk/http.h, configs/samples/http.conf.sample,
+	  main/http.c: HTTP: For httpd server, need option to define server
+	  name for security purposes Added a new config property
+	  [servername] to the http.conf file; updated the http server to
+	  use the new property when sending responses, for showing http
+	  status through the CLI and when reporting status through the
+	  'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew
+	  Nagy Review: https://reviewboard.asterisk.org/r/4374/
+
+2015-01-30 16:47 +0000 [r431468]  Mark Michelson <mmichelson at digium.com>
+
+	* main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c,
+	  res/res_pjsip_refer.c, main/pbx.c, main/manager.c,
+	  pbx/pbx_spool.c, main/bridge_after.c: Fix some memory leaks.
+	  These memory leaks were found and fixed by John Hardin. I'm just
+	  committing them for him. ASTERISK-24736 #close Reported by Mark
+	  Michelson Review: https://reviewboard.asterisk.org/r/4389
+
+2015-01-29 23:02 +0000 [r431450]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* include/asterisk/bridge.h, main/bridge.c,
+	  res/stasis/stasis_bridge.c: stasis transfer: fix stasis bridge
+	  push race part two When swapping a Local channel in place of one
+	  already in a bridge (to complete a bridge attended transfer), the
+	  channel that was swapped out can actually be hung up before the
+	  stasis bridge push callback executes on the independant transfer
+	  thread. This results in the stasis app loop dropping out and
+	  removing the control that has the the app name which the local
+	  replacement channel needs so it can re-enter stasis. To avoid
+	  this race condition a new push_peek callback has been added, and
+	  called from the ast_bridge_impart thread before it launches the
+	  independant thread that will complete the transfer. Now the
+	  stasis push_peek callback can copy the stasis app name before the
+	  swap channel can hang up. ASTERISK-24649 Review:
+	  https://reviewboard.asterisk.org/r/4382/
+
+2015-01-29 20:58 +0000 [r431420-431426]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_pjsip.c, res/res_pjsip_sips_contact.c (added): Use SIPS
+	  URIs in Contact headers when appropriate. RFC 3261 sections
+	  8.1.1.8 and 12.1.1 dictate specific scenarios when we are
+	  required to use SIPS URIs in Contact headers. Asterisk's
+	  non-compliance with this could actually cause calls to get
+	  dropped when communicating with clients that are strict about
+	  checking the Contact header. Both of the SIP stacks in Asterisk
+	  suffered from this issue. This changeset corrects the behavior in
+	  res_pjsip/chan_pjsip.c Review:
+	  https://reviewboard.asterisk.org/r/4345
+
+	* /, channels/chan_sip.c: Use SIPS URIs in Contact headers when
+	  appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate
+	  specific scenarios when we are required to use SIPS URIs in
+	  Contact headers. Asterisk's non-compliance with this could
+	  actually cause calls to get dropped when communicating with
+	  clients that are strict about checking the Contact header. Both
+	  of the SIP stacks in Asterisk suffered from this issue. This
+	  changeset corrects the behavior in chan_sip. ASTERISK-24646
+	  #close Reported by Stephan Eisvogel Review:
+	  https://reviewboard.asterisk.org/r/4346 ........ Merged revisions
+	  431423 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* res/res_pjsip/pjsip_configuration.c: Allow disabling of 100rel
+	  support on PJSIP endpoints. Due to an inversion error, setting
+	  100rel=no would not actually change the current value of the
+	  setting (which defaulted to "yes"). With this fix, the inversion
+	  is corrected.
+
+2015-01-29 16:46 +0000 [r431403]  George Joseph <george.joseph at fairview5.com>
+
+	* res/res_pjsip_exten_state.c: res_pjsip_exten_state: Reduce log
+	  clutter... change a WARNING to a VERBOSE/2 Reduce log clutter by
+	  changing the "Watcher for hint %s (removed|deactivated)" message
+	  from WARNING to VERBOSE/2. Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4387/
+
+2015-01-29 12:09 +0000 [r431385]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix DTLS when used
+	  with OpenSSL 1.0.1k A recent security fix for OpenSSL broke DTLS
+	  negotiation for many applications. This was caused by read ahead
+	  not being enabled when it should be. While a commit has gone into
+	  OpenSSL to force read ahead on for DTLS it may take some time for
+	  a release to be made and the change to be present in
+	  distributions (if at all). As enabling read ahead is a simple one
+	  line change this commit does that and fixes the issue.
+	  ASTERISK-24711 #close Reported by: Jared Biel ........ Merged
+	  revisions 431384 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-28 17:37 +0000 [r431301-431303]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c,
+	  res/res_pjsip_session.c: Fix file descriptor leak in RTP code.
+	  SIP requests that offered codecs incompatible with configured
+	  values could result in the allocation of RTP and RTCP ports that
+	  would not get reclaimed later. ASTERISK-24666 #close Reported by
+	  Y Ateya Review: https://reviewboard.asterisk.org/r/4323
+	  AST-2015-001 ........ Merged revisions 431300 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* funcs/func_curl.c, /: Multiple revisions 431297-431298 ........
+	  r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan
+	  2015) | 17 lines Mitigate possible HTTP injection attacks using
+	  CURL() function in Asterisk. CVE-2014-8150 disclosed a
+	  vulnerability in libcURL where HTTP request injection can be
+	  performed given properly-crafted URLs. Since Asterisk makes use
+	  of libcURL, and it is possible that users of Asterisk may get
+	  cURL URLs from user input or remote sources, we have made a patch
+	  to Asterisk to prevent such HTTP injection attacks from
+	  originating from Asterisk. ASTERISK-24676 #close Reported by Matt
+	  Jordan Review: https://reviewboard.asterisk.org/r/4364
+	  AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49
+	  -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from
+	  previous patch. ........ Merged revisions 431297-431298 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 431299 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2015-01-28 12:18 +0000 [r431267]  Sean Bright <sean at malleable.com>
+
+	* res/res_format_attr_silk.c, res/res_format_attr_opus.c: media
+	  formats: update res_format_attr_opus & silk In r419044, we
+	  changed how formats were handled, but the return value of the
+	  format_parse_sdp_fmtp functions in res_format_attr_opus and
+	  res_format_attr_silk were not updated, causing calls to fail. Ran
+	  into this when getting codec_opus working with Asterisk 13. Once
+	  the return value was corrected, we were crashing in opus_getjoint
+	  because of NULL format attributes. I've fixed this as well in
+	  this patch. Review: https://reviewboard.asterisk.org/r/4371/
+
+2015-01-28 04:09 +0000 [r431243]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/sorcery.c, res/res_pjsip_outbound_registration.c,
+	  res/res_pjsip.c: res_pjsip_outbound_registration: Fix reload race
+	  condition. Performing a CLI "module reload" command when there
+	  are new pjsip.conf registration objects defined frequently failed
+	  to load them correctly. What happens is a race condition between
+	  res_pjsip pushing its reload into an asynchronous task processor
+	  task and the thread that does the rest of the reloads when it
+	  gets to reloading the res_pjsip_outbound_registration module. A
+	  similar race condition happens between a reload and the CLI/AMI
+	  show registrations commands. The reload updates the
+	  current_states container and the CLI/AMI commands call
+	  get_registrations() which builds a new current_states container.
+	  * Made res_pjsip.c reload_module() use
+	  ast_sip_push_task_synchronous() instead of ast_sip_push_task() to
+	  eliminate two threads processing config reloads at the same time.
+	  * Made get_registrations() not replace the global current_states
+	  container so the CLI/AMI show registrations command cannot
+	  interfere with reloading. You could never add/remove objects in
+	  the container without the possibility of the container being
+	  replaced out from under you by get_registrations(). * Added a
+	  registration loaded sorcery instance observer to purge any dead
+	  registration objects since get_registrations() cannot do this job
+	  anymore. The struct ast_sorcery_instance_observer callbacks must
+	  be used because the callback happens inline with the load
+	  process. The struct ast_sorcery_observer callbacks are pushed to
+	  a different thread. * Added some global current_states NULL
+	  pointer checks in case the container disappears because of
+	  unload_module(). * Made sorcery's struct
+	  ast_sorcery_instance_observer.object_type_loaded callbacks
+	  guaranteed to be called before any struct
+	  ast_sorcery_observer.loaded callbacks will be called. * Moved the
+	  check for non-reloadable objects to before the sorcery instance
+	  loading callbacks happen to short circuit unnecessary work.
+	  Previously with non-reloadable objects, the sorcery instance
+	  loading/loaded callbacks would always happen, the individual
+	  wizard loading/loaded would be prevented, and the non-reloadable
+	  type logging message would be logged for each associated wizard.
+	  ASTERISK-24729 #close Review:
+	  https://reviewboard.asterisk.org/r/4381/
+
+2015-01-27 22:56 +0000 [r431179-431219]  Kevin Harwell <kharwell at digium.com>
+
+	* /, main/tcptls.c: tcptls: Bad file descriptor error when
+	  reloading chan_sip While running through some scenarios using
+	  chan_sip and tcp a problem would occur that resulted in a flood
+	  of bad file descriptor messages on the cli: tcptls.c:712
+	  ast_tcptls_server_root: Accept failed: Bad file descriptor The
+	  message is received because the underlying socket has been
+	  closed, so is valid. This is probably happening because unloading
+	  of chan_sip is not atomic. That however is outside the scope of
+	  this patch. This patch simply stops the logging of multiple
+	  occurrences of that message. ASTERISK-24728 #close Reported by:
+	  Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/
+	  ........ Merged revisions 431218 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, channels/chan_sip.c: chan_sip: stale nonce causes failure When
+	  refreshing (with a small expiration) a registration that was sent
+	  to chan_sip the nonce would be considered stale and reject the
+	  registration. What was happening was that the initial
+	  registration's "dialog" still existed in the dialogs container
+	  and upon refresh the dialog match algorithm would choose that as
+	  the "dialog" instead of the newly created one. This occurred
+	  because the algorithm did not check to see if the from tag
+	  matched if authentication info was available after the 401. So,
+	  it ended up assuming the original "dialog" was a match and
+	  stopped the search. The old "dialog" of course had an old nonce,
+	  thus the stale nonce message. This fix attempts to leave the
+	  original functionality alone except in the case of a REGISTER. If
+	  a REGISTER is received if searches for an existing "dialog"
+	  matching only on the callid. If the expires value is low enough
+	  it will reuse dialog that is there, otherwise it will create a
+	  new one. ASTERISK-24715 #close Reported by: John Bigelow Review:
+	  https://reviewboard.asterisk.org/r/4367/ ........ Merged
+	  revisions 431187 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c,
+	  main/stasis_message_router.c, res/res_pjsip/location.c,
+	  res/res_pjsip/pjsip_configuration.c,
+	  res/res_pjsip/pjsip_distributor.c,
+	  res/res_pjsip/include/res_pjsip_private.h,
+	  res/res_pjsip/pjsip_global_headers.c,
+	  res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
+	  res/res_pjsip/config_transport.c: res_pjsip: make it unloadable
+	  (take 2) Due to the original patch causing memory corruptions it
+	  was removed until the problem could be resolved. This patch is
+	  the original patch plus some added locking around stasis router
+	  subcription that was needed to avoid the memory corruption.
+	  Description of the original problem and patch (still applicable):
+	  The res_pjsip module was previously unloadable. With this patch
+	  it can now be unloaded. This patch is based off the original
+	  patch on the issue (listed below) by Corey Farrell with a few
+	  modifications. Namely, removed a few changes not required to make
+	  the module unloadable and also fixed a bug that would cause
+	  asterisk to crash on unloading. This patch is the first step
+	  (should hopefully be followed by another/others at some point) in
+	  allowing res_pjsip and the modules that depend on it to be
+	  unloadable. At this time, res_pjsip and some of the modules that
+	  depend on res_pjsip cannot be unloaded without causing problems
+	  of some sort. The goal of this patch is to get res_pjsip and only
+	  res_pjsip to be able to unload successfully and/or shutdown
+	  without incident (crashes, leaks, etc...). Other dependent
+	  modules may still cause problems on unload. Basically made sure,
+	  with the patch applied, that res_pjsip (with no other dependent
+	  modules loaded) could be succesfully unloaded and Asterisk could
+	  shutdown without any leaks or crashes that pertained directly to
+	  res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell
+	  Review: https://reviewboard.asterisk.org/r/4363/ patches:
+	  pjsip_unload-broken-r1.patch submitted by Corey Farrell (license
+	  5909)
+
+2015-01-27 17:36 +0000 [r431160]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
+	  app_confbridge: Repeatedly starting and stopping recording ref
+	  leaks the recording channel. Starting and stopping conference
+	  recording more than once causes the recording channels to be
+	  leaked. For v13 the channels also show up in the CLI "core show
+	  channels" output. * Reworked and simplified the recording channel
+	  code to use ast_bridge_impart() instead of managing the recording
+	  thread in the ConfBridge code. The recording channel's ref
+	  handling easily falls into place and other off nominal code paths
+	  get handled better as a result. ASTERISK-24719 #close Reported
+	  by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/
+	  Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged
+	  revisions 431135 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-27 17:32 +0000 [r431157]  Joshua Colp <jcolp at digium.com>
+
+	* main/bridge_channel.c, res/res_pjsip_sdp_rtp.c: bridge /
+	  res_pjsip_sdp_rtp: Fix issues with media not being reinvited
+	  during direct media. This change fixes two issues: 1. During a
+	  swap operation bridging added the new channel before having the
+	  swap channel leave. This was not handled in bridge_native_rtp and
+	  could result in a channel not getting reinvited back to Asterisk.
+	  After this change the swap channel will leave first and the new
+	  channel will then join. 2. If a re-invite was received after a
+	  session had been established any upstream elements (such as
+	  bridge_native_rtp) were not notified that they may want to
+	  re-evaluate things. After this change an UPDATE_RTP_PEER control
+	  frame is queued when this situation occurs and upstream can
+	  react. AST-1524 #close Review:
+	  https://reviewboard.asterisk.org/r/4378/
+
+2015-01-27 17:22 +0000 [r431153]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c: Manager: Fix Manager Action ModuleLoad to give
+	  correct response when reloading Prior to this patch, ModuleLoad
+	  would respond with an error indicating that the requested module
+	  wasn't found in spite of finding and reloading the module.
+	  Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721
+	  #close
+
+2015-01-27 17:20 +0000 [r431134-431145]  Matthew Jordan <mjordan at digium.com>
+
+	* res/ari/resource_bridges.c,
+	  rest-api-templates/asterisk_processor.py,
+	  res/ari/resource_channels.h, res/res_ari_bridges.c,
+	  res/ari/resource_bridges.h, rest-api-templates/api.wiki.mustache,
+	  rest-api/api-docs/channels.json,
+	  rest-api-templates/swagger_model.py,
+	  rest-api/api-docs/bridges.json: ARI: Improve wiki documentation
+	  This patch improves the documentation of ARI on the wiki.
+	  Specifically, it addresses the following: * Allowed values and
+	  allowed ranges weren't documented. This was particularly
+	  frustrating, as Asterisk would reject query parameters with
+	  disallowed values - but we didn't tell anyone what the allowed
+	  values were. * The /play/id operation on /channels and /bridges
+	  failed to document all of the added media resource types. *
+	  Documentation for creating a channel into a Stasis application
+	  failed to note when it occurred, and that creating a channel into
+	  Stasis conflicts with creating a channel into the dialplan. *
+	  Some other minor tweaks in the mustache templates, including
+	  italicizing the parameter type, putting the default value on its
+	  own sub-bullet, and some other nicities. Review:
+	  https://reviewboard.asterisk.org/r/4351
+
+	* apps/confbridge/conf_config_parser.c,
+	  apps/confbridge/include/confbridge.h: app_confbridge: Restore
+	  user's menu name to CLI output of 'confbridge list' When issuing
+	  a 'confbridge list XXXX' CLI command, the resulting output no
+	  longer displays the menu associated with a ConfBridge
+	  participant. The issue was caused by ASTERISK-22760. When that
+	  patch was done, it removed the copying of the menu name
+	  associated with the user from the actual user profile. This patch
+	  fixes the issue by copying the menu name over to the user profile
+	  when the menu hooks are applied to the user. Since that function
+	  now does a little bit more than just apply the hooks, the name of
+	  the function has been changed to cover the copying of the menu
+	  name over as well. In addition, there is a disparity between the
+	  menu name length as it is stored on the conf_menu structure and
+	  the confbridge_user structure; this patch makes the lengths match
+	  so that a strcpy can be used. Review:
+	  https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close
+	  Reported by: Steve Pitts
+
+2015-01-27 11:47 +0000 [r431114]  Joshua Colp <jcolp at digium.com>
+
+	* res/parking/parking_manager.c: res_parking: Fix crash due to race
+	  condition when unloading. There is currently a race condition
+	  when unloading the res_parking module. Depending on the will of
+	  the universe the subscription invocation may occur AFTER the
+	  module is unloaded. This is because the module does NOT use
+	  stasis_unsubscribe_and_join when terminating the subscription. It
+	  merely uses stasis_unsubscribe. This change makes it use
+	  stasis_unsubscribe_and_join which is documented for usage in this
+	  exact scenario. AST-1520 #close Review:
+	  https://reviewboard.asterisk.org/r/4375/
+
+2015-01-26 14:49 +0000 [r431092]  David M. Lee <dlee at digium.com>
+
+	* channels/sip/include/route.h, funcs/func_presencestate.c,
+	  main/rtp_engine.c, configure, include/asterisk/autoconfig.h.in,
+	  include/asterisk/sem.h, configure.ac, main/app.c,
+	  main/bridge_channel.c, main/sem.c, res/res_timing_kqueue.c,
+	  main/asterisk.c: Various fixes for OS X This patch addresses
+	  compilation errors on OS X. It's been a while, so there's quite a
+	  few things. * Fixed __attribute__ decls in route.h to be
+	  portable. * Fixed htonll and ntohll to work when they are defined
+	  as macros. * Replaced sem_t usage with our ast_sem wrapper. *
+	  Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC
+	  4.9 warnings using sig*set() functions. * Fixed some format
+	  strings for portability. * Fixed compilation issues with
+	  res_timing_kqueue (although tests still fail on OS X). * Fixed
+	  menuconfig /sbin/launchd detection, which disables
+	  res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by:
+	  George Joseph ASTERISK-24544 #close Reported by: George Joseph
+	  Review: https://reviewboard.asterisk.org/r/4327/
+
+2015-01-25 13:42 +0000 [r431072]  Matthew Jordan <mjordan at digium.com>
+
+	* main/config.c: dynamic realtime: Updates fail to work due to
+	  update fields being passed over When a crash was fixed due to
+	  usage of the REALTIME function in r423003, a regression was
+	  introduced into ast_update2_realtime where the update fields
+	  passed to the function would be skipped and the lookup field
+	  processed twice. The use of this function is a bit interesting: A
+	  variable argument list is used with two sentinel values - the
+	  first marks the end of the lookup fields/values; the second marks
+	  the end of the update fields/values. Unfortunately,
+	  ast_update2_realtime parses over the lookup fields twice, as
+	  opposed to parsing over the update fields. This causes the
+	  lookups to succeed, but the updates itself to have no effect.
+	  Note that the most common instance of this problem occurred in
+	  app_voicemail during the updating of a mailbox password. Thanks
+	  to the issue reporter, Paddy Grice, for pointing out the problem.
+	  Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231
+	  ASTERISK-24626 #close Reported by: Paddy Grice
+
+2015-01-23 20:13 +0000 [r431050-431052]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_chan_record.c: app_confbridge: Make CBRec
+	  channel names more unique. Channel names should be different from
+	  other channels in the system while the channel exists. * Use a
+	  sequence number for CBRec channels instead of a random number
+	  because the same random number could be picked again for the next
+	  CBRec channel.
+
+	* /, apps/app_confbridge.c: app_confbridge: Whitespace Because
+	  there is sometimes no sence to any whitespace. ........ Merged
+	  revisions 431049 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-23 17:08 +0000 [r431030]  David M. Lee <dlee at digium.com>
+
+	* res/res_pjsip_config_wizard.c: Add depend on pjproject to
+	  res_pjsip_config_wizard.c
+
+2015-01-23 15:12 +0000 [r430999]  Kevin Harwell <kharwell at digium.com>
+
+	* res/parking/parking_applications.c, channels/chan_iax2.c,
+	  res/res_pjsip/pjsip_global_headers.c, res/res_pjsip_pubsub.c,
+	  res/res_ari_channels.c, res/res_stasis.c,
+	  rest-api-templates/param_parsing.mustache,
+	  res/res_ari_endpoints.c, res/res_ari_events.c,
+	  include/asterisk/stasis_app.h, res/res_pjsip_mwi.c: Investigate
+	  and fix memory leaks in Asterisk Fixed memory leaks that were
+	  found in Asterisk. ASTERISK-24693 #close Reported by: Kevin
+	  Harwell Review: https://reviewboard.asterisk.org/r/4347/
+
+2015-01-23 15:03 +0000 [r430994-430998]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* apps/app_voicemail.c, channels/chan_unistim.c,
+	  funcs/func_hangupcause.c, main/manager_bridges.c,
+	  channels/chan_misdn.c, funcs/func_groupcount.c, /,
+	  addons/ooh323c/src/ooh245.c, channels/chan_sip.c, res/res_fax.c,
+	  res/res_pjsip_outbound_registration.c, apps/app_minivm.c,
+	  apps/app_alarmreceiver.c, include/asterisk/channel.h,
+	  contrib/utils/eagi_proxy.c: Fix typo's (retrieve, specified,
+	  address). ........ Merged revisions 430996 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, channels/chan_sip.c: chan_sip: Case insensitive comparison of
+	  "defaultuser" parameter. All the other configuration options are
+	  case insensitive, so this one should be too. ASTERISK-24355
+	  #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded
+	  by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions
+	  430993 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-22 19:24 +0000 [r430957-430975]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/bridge.h,
+	  include/asterisk/bridge_channel_internal.h, main/bridge.c,
+	  include/asterisk/bridge_internal.h, main/bridge_channel.c: Bridge
+	  core: Pass a ref with the swap channel when joining a bridge.
+	  When code imparts a channel into a bridge to swap with another
+	  channel, a ref needs to be held on the swap channel to ensure
+	  that it cannot dissapear before finding it in the bridge. * The
+	  ast_bridge_join() swap channel parameter now always steals a ref
+	  for the swap channel. This is the only change to the bridge
+	  framework's public API semantics. *
+	  bridge_channel_internal_join() now requires the
+	  bridge_channel->swap channel to pass in a ref. ASTERISK-24649
+	  Reported by: John Bigelow Review:
+	  https://reviewboard.asterisk.org/r/4354/
+
+	* res/res_pjsip_outbound_registration.c:
+	  res_pjsip_outbound_registration.c: Minor code cleanup. * Add an
+	  allocation failure check and assert in
+	  sip_outbound_registration_response_cb(). * Made
+	  sip_outbound_registration_state_destroy() handle partially
+	  created state objects from
+	  sip_outbound_registration_state_alloc(). Review:
+	  https://reviewboard.asterisk.org/r/4366/
+
+2015-01-22 18:09 +0000 [r430939]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* res/stasis/app.c, res/stasis/stasis_bridge.c: stasis transfer:
+	  fix a race condition on stasis bridge push After a bridge
+	  transfer completes where a local replacement channel is used, a
+	  stasis transfer message with the details of the transfer is sent.
+	  This is processed by stasis which then sets the stasis app name
+	  and replaced channel snapshot on the replacement channel.
+	  However, since a separate thread was already started to run
+	  stasis on the new replacement channel, a race was on to see if
+	  the message processing would be completed before the app name was
+	  needed, otherwise the channel would be hung up. This change moves
+	  the calls used to set the stasis app name and the replace
+	  snapshot to the bridge_stasis_push function callback from the
+	  bridge transfer logic, allowing the steps to be completed earlier
+	  and more deterministically, and the race elimianted. NOTE: the
+	  swap channel parameter to bridge_stasis_push (and thus all bridge
+	  push callbacks) must always be present when performing a swap
+	  with another channel. ASTERISK-24649 #close Reported by: John
+	  Bigelow Review: https://reviewboard.asterisk.org/r/4341/
+
+2015-01-22 14:23 +0000 [r430921]  Matthew Jordan <mjordan at digium.com>
+
+	* /, apps/app_voicemail.c: apps/app_voicemail: Trigger MWI
+	  notification with MixMonitor m() option The MixMonitor m() option
+	  allows a recording to be pushed to a specific voicemail mailbox.
+	  If the message is delivered to the mailbox's INBOX, however, no
+	  MWI notification is currently raised. This patch corrects the
+	  issue by properly calling notify_new_state from the
+	  msg_create_from_file function. This will cause MWI to be
+	  triggered if the message was placed in the mailbox's INBOX.
+	  ASTERISK-24709 #close Reported by: Gareth Palmer patches:
+	  app_voicemail-430919.patch uploaded by Gareth Palmer (License
+	  5169) ........ Merged revisions 430920 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-21 21:53 +0000 [r430902]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_pjsip_outbound_registration.c:
+	  res_pjsip_outbound_registration.c: Move unref to a better place.
+	  Move an unconditional unref of client_state so it doesn't look
+	  like it could be used after the last ref has destroyed it.
+
+2015-01-21 13:33 +0000 [r430840-430864]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: channels/chan_sip: Fix registration leak
+	  during reload When the SIP registrations were migrated to using
+	  ao2 in what was then trunk, the explicit destruction of the
+	  registrations on module reload was removed and not replaced with
+	  an ao2 equivalent. Debugging done by Stefan Engström, the issue
+	  reporter, on ASTERISK-24673 confirmed that the reference in the
+	  registry_list container was being leaked. Since the purpose of
+	  cleanup_all_regs is to prep a registration for destruction, this
+	  function now calls an ao2_callback function callback with the
+	  OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the
+	  registrations. This cleans up each registration, and also removes
+	  it from the registration container registry_list. Review:
+	  https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close
+	  Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan
+	  Engström Tested by: Stefan Engström
+
+	* cdr/cdr_manager.c, cel/cel_manager.c: AMI: Add documentation for
+	  the missing Cdr/CEL events. This patch adds AMI event
+	  documentation for the Cdr and CEL AMI events. Note that while
+	  these events do share fields with each other and with other
+	  channel related events, they do not contain all of the fields in
+	  a standard channel snapshot, nor is the description of the fields
+	  identical. As such, the patch opts for documentation for each
+	  field, for each event. Review:
+	  https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close
+	  Reported by: Dan Jenkins
+
+	* apps/app_dial.c: apps/app_dial: Don't publish DialEnd twice on
+	  unexpected GoSub/Macro values The Dial application has some
+	  interesting options with the mid-call Macro (M) and GoSub (U)
+	  options. If the MACRO_RESULT/GOSUB_RESULT returns specific
+	  values, the Dial application will take some action upon the
+	  channels involved in the dial operation (such as hanging up a
+	  particular party, etc.) The Dial application ensures that a
+	  Stasis message is published in the event that
+	  MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial
+	  operation, so that there is a corresponding DialEnd event
+	  published in AMI/ARI for the DialBegin event that preceeded it. A
+	  bug exists where that same DialEnd event will be published on
+	  Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is
+	  not one that the Dial application cares about. This causes two
+	  DialEnd events to be published - one with the
+	  MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is
+	  all sorts of wrong. This patch fixes the bug by ensuring that we
+	  only publish a DialEnd message to Stasis if the Dial
+	  application's mid-call Macro/GoSub returns something that Dial
+	  cares about. Review: https://reviewboard.asterisk.org/r/4336
+	  ASTERISK-24682 #close Reported by: Matt Jordan
+
+	* main/rtp_engine.c: main/rtp_engine: Format NTP timestamps as
+	  unsigned longs When the RTCP reports are created, the NTP
+	  timestamps are stored as strings, as JSON does not have an
+	  integer type long enough to store the value. However, on 32-bit
+	  systems, a signed long may overflow for some portion of the
+	  timestamp. This patch corrects the overflow by formatting the
+	  timestamps as unsigned longs.
+
+2015-01-20 16:51 +0000 [r430818]  asanders <asanders at localhost>:
+
+	* res/ari/resource_bridges.c: ARI: Fixed crash that occurred when
+	  updating a bridge when the optional query parameter 'name' was
+	  not supplied. Prior to this changeset, posting to the:
+	  /ari/bridges/{bridgeId} endpoint without specifying a value for
+	  the [name] query parameter, would crash Asterisk if the bridge
+	  you are attempting to create (or update) had the same ID as an
+	  existing bridge. The internal mechanism of the POST operation
+	  interpreted a null value for name, thus resulting in an error
+	  condition that crashed Asterisk. ASTERISK-24560 #close Reported
+	  By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/
+
+2015-01-20 16:46 +0000 [r430817]  Richard Mudgett <rmudgett at digium.com>
+
+	* configs/samples/iax.conf.sample, res/res_fax.c,
+	  funcs/func_channel.c, UPGRADE.txt, res/snmp/agent.c,
+	  channels/chan_iax2.c: CHANNEL(peer), chan_iax2, res_fax, SNMP
+	  agent: Fix deadlock from reaching across a bridge. Calling
+	  ast_channel_bridge_peer() cannot be done while holding any
+	  channel locks. The reported issue hit the deadlock in chan_iax2,
+	  but an audit of the ast_channel_bridge_peer() calls found three
+	  more locations where the same deadlock can occur. * Made
+	  CHANNEL(peer), res_fax, and the SNMP agent not call
+	  ast_channel_bridge_peer() with any channel locked. For
+	  CHANNEL(peer) I had to rework the logic to not hold the channel
+	  lock. * Made chan_iax2 no longer call ast_channel_bridge_peer().
+	  It was done for legacy reasons that no longer apply. * Removed
+	  the iax.conf forcejitterbuffer option. It is now always enabled
+	  when the jitterbuffer option is enabled. If you put a jitter
+	  buffer on a channel it will be on the channel. ASTERISK-24600
+	  #close Reported by: Jeff Collell Review:
+	  https://reviewboard.asterisk.org/r/4342/
+
+2015-01-20 02:39 +0000 [r430796-430799]  Matthew Jordan <mjordan at digium.com>
+
+	* contrib/scripts/install_prereq, /:
+	  contrib/scripts/install_prereq: Don't install 32-bit packages on
+	  64-bit hosts On Debian based systems, the install_prereq tool
+	  uses a search command on Debian that results in selecting both
+	  64-bit and 32-bit packages. Besides the waste of disk space, this
+	  can actually cause aptitude use 100% of memory on a VM with 1GB
+	  of RAM as it tried to work out all of the 32-bit package
+	  dependencies. This patch filters out the 32-bit packages on a
+	  64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048
+	  #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan
+	  patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang
+	  (License 5876) ........ Merged revisions 430798 from

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