[asterisk-commits] bebuild: tag 13.0.0-beta3 r426071 - in /tags/13.0.0-beta3: ./ contrib/realtim...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Oct 20 11:08:18 CDT 2014


Author: bebuild
Date: Mon Oct 20 11:08:14 2014
New Revision: 426071

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=426071
Log:
Importing files for 13.0.0-beta3 release.

Added:
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    tags/13.0.0-beta3/.version   (with props)
    tags/13.0.0-beta3/ChangeLog   (with props)
    tags/13.0.0-beta3/contrib/realtime/mysql/mysql_cdr.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/oracle/oracle_cdr.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/postgresql/postgresql_cdr.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/sqlserver/mssql_cdr.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/13.0.0-beta3/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

Added: tags/13.0.0-beta3/.lastclean
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Added: tags/13.0.0-beta3/ChangeLog
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--- tags/13.0.0-beta3/ChangeLog (added)
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+2014-10-20  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 13.0.0-beta3 Released.
+
+2014-10-20 14:15 +0000 [r425991]  Matthew Jordan <mjordan at digium.com>
+
+	* main/tcptls.c, /, res/res_xmpp.c: AST-2014-011: Fix POODLE
+	  security issues There are two aspects to the vulnerability: (1)
+	  res_jabber/res_xmpp use SSLv3 only. This patch updates the module
+	  to use TLSv1+. At this time, it does not refactor
+	  res_jabber/res_xmpp to use the TCP/TLS core, which should be done
+	  as an improvement at a latter date. (2) The TCP/TLS core, when
+	  tlsclientmethod/sslclientmethod is left unspecified, will default
+	  to the OpenSSL SSLv23_method. This method allows for all
+	  encryption methods, including SSLv2/SSLv3. A MITM can exploit
+	  this by forcing a fallback to SSLv3, which leaves the server
+	  vulnerable to POODLE. This patch adds WARNINGS if a user uses
+	  SSLv2/SSLv3 in their configuration, and explicitly disables
+	  SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk
+	  will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly
+	  chosen. For TLS servers, Asterisk will no longer support SSLv2 or
+	  SSLv3. Much thanks to abelbeck for reporting the vulnerability
+	  and providing a patch for the res_jabber/res_xmpp modules.
+	  Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425
+	  #close Reported by: abelbeck Tested by: abelbeck, opsmonitor,
+	  gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by
+	  abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch
+	  uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff
+	  uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded
+	  by mjordan (License 6283) ........ Merged revisions 425987 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-19 17:07 +0000 [r425965]  George Joseph <george.joseph at fairview5.com>
+
+	* configure.ac, makeopts.in, Makefile, /, configure,
+	  include/asterisk/autoconfig.h.in: build: Force -fsigned-char on
+	  platforms where the default for char is unsigned gcc on the ARM
+	  platform defaults 'char' to 'unsigned char' whereas Intel and
+	  SPARC default to 'signed char'. This is only an issue in the rare
+	  cases where negative values are assigned to a 'char' but this
+	  this patch insures compatibility by detecting platforms that
+	  default to 'unsigned' and adding an '-fsigned-char' flag to
+	  _ASTCFLAGS. If compiling for ARM (native or cross-compile) be
+	  sure to run ./bootstrap.sh and ./configure to regenerate the
+	  build files. You shouldn't have to do this for Intel or SPARC.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4091/ ........ Merged
+	  revisions 425964 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-19 04:01 +0000 [r425923-425944]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922
+	  This patch for r425922 introduced a bug, wherein sending an
+	  INVITE request with no SDP would cause Asterisk to not send an
+	  SDP Offer in the 200 OK. The current structure of
+	  res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as
+	  create_outgoing_sdp has no knowledge of whether or not it is
+	  creating an SDP as a new Offer or an Answer. This is something of
+	  an oversight in the callback definition, as the caller of it does
+	  have this information.
+
+	* res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over
+	  reference to override_prefs The usage of the local override_prefs
+	  variable in create_outgoing_sdp_stream was previously to track an
+	  override format preference set by PJSIP_MEDIA_OFFER. Now,
+	  however, that function simply sets the joint capabilities
+	  structure, session->req_caps. During the media format rework, the
+	  override_prefs was instead used to check if there were any
+	  formats in session->req_caps. However, this usage isn't useful in
+	  create_outgoing_sdp_stream. session->req_caps contains the
+	  negotiated formats for *all* streams, not just the current one
+	  being created. Thus, so long as any stream of any type has
+	  provided a format, override_prefs will be non-zero. Hence, its
+	  usage in checking whether or not we should look at the formats on
+	  the endpoint or the joint capabilities is generally useless.
+	  There's only two things useful to check: (1) Does the endpoint
+	  have a format for the media type? (2) Did we negotiate a format
+	  for the media type? If either of those is a 'no', then we must
+	  kill the media stream.
+
+2014-10-17 22:43 +0000 [r425905]  Jonathan Rose <jrose at digium.com>
+
+	* configs/samples/cli_aliases.conf.sample: Sample Configurations:
+	  make 'pjsip reload' reload all reloadable pjsip modules AST-1432
+	  #close Reported by: John Bigelow
+
+2014-10-17 13:35 +0000 [r425821-425879]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_pjsip.c, res/res_pjsip_session.c, /,
+	  res/res_pjsip_sdp_rtp.c: res_pjsip_session/res_pjsip_sdp_rtp: Be
+	  more tolerant of offers When an inbound SDP offer is received,
+	  Asterisk currently makes a few incorrection assumptions: (1) If
+	  the offer contains more than a single audio/video stream,
+	  Asterisk will reject the entire stream with a 488. This is an
+	  overly strict response; generally, Asterisk should accept the
+	  media streams that it can accept and decline the others. (2) If
+	  the offer contains a declined media stream, Asterisk will attempt
+	  to process it anyway. This can result in attempting to match
+	  format capabilities on a declined media stream, leading to a 488.
+	  Asterisk should simply ignore declined media streams. (3)
+	  Asterisk will currently attempt to handle offers with AVPF with
+	  use_avpf=No/AVP with use_avpf=Yes. This mismatch results in
+	  invalid SDP answers being sent in response. If there is a
+	  mismatch between the media type being offered and the
+	  configuration, Asterisk must reject the offer with a 488. This
+	  patch does the following: * Asterisk will accept SDP offers with
+	  at least one media stream that it can use. Some WARNING messages
+	  have been dropped to NOTICEs as a result. * Asterisk will not
+	  accept an offer with a media type that doesn't match its
+	  configuration. * Asterisk will ignore declined media streams
+	  properly. #SIPit31 Review:
+	  https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close
+	  Reported by: James Van Vleet ASTERISK-24381 #close Reported by:
+	  Matt Jordan ........ Merged revisions 425868 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
+	  setting when sending qualify requests The outboundproxy setting
+	  is currently ignored when sending OPTIONS requests as a result of
+	  the qualify setting. This means that if an Asterisk server is
+	  unable to send the packet directly to a peer, it is unable to
+	  qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
+	  This patch grabs the outboundproxy information for a peer when a
+	  qualify attempt is being constructed and, if it finds the
+	  information, uses it when sending the OPTIONS request. Review:
+	  https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
+	  Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
+	  uploaded by Damian Ivereigh (License 6632) ........ Merged
+	  revisions 425818 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425819 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425820 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-17 02:41 +0000 [r425783]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/core_unreal.c, main/channel.c: AMI: Add missing VarSet
+	  events when a channel inherits variables. There should be AMI
+	  VarSet events when channel variables are inherited by an outgoing
+	  channel. Also local;2 should generate VarSet events when it gets
+	  all of its channel variables from channel local;1. ASTERISK-24415
+	  #close Reported by: Richard Mudgett Patches:
+	  jira_asterisk_24415_v12.patch (license #5621) patch uploaded by
+	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/
+	  ........ Merged revisions 425782 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-17 01:57 +0000 [r425736-425761]  Matthew Jordan <mjordan at digium.com>
+
+	* /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio
+	  issues when moving from remote bridge to softmix When a native
+	  RTP bridge that is remotely bridging its participants switches to
+	  a softmix bridge, it may not properly re-INVITE the media for one
+	  or both participants back to Asterisk. This is due to the current
+	  bridge_native_rtp code only re-INVITEs if it believes the channel
+	  will survive the bridge operation. Currently, that code is
+	  failing, as it expects the channels to have a soft hangup flag
+	  set on it indicating that a redirect has occurred or that the
+	  channel is going to leave the bridge. (The code did not take into
+	  account a smart bridge operation). This patch also renames a few
+	  things to be more reflective of the underlying types. Review:
+	  https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close
+	  ........ Merged revisions 425760 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* /, tests/test_cel.c: test_cel: Update pickup test to expect
+	  CANCEL instead of ANSWSER The CEL pickup test previously looked
+	  for a disposition of ANSWER between the original caller/peer when
+	  the call is picked up. This is actually incorrect: the
+	  disposition should, at the very least, not be ANSWER as the call
+	  was never ANSWERed. The disposition is now CANCEL; this patch
+	  updates the test accordingly. ........ Merged revisions 425757
+	  from http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched
+	  CDRs as opposed to 'size' When refactoring CDRs to use the
+	  configuration framework, a 'whoops' was introduced where the CDR
+	  batch size was used when rescheduling a batch, as opposed to the
+	  time duration. This patch corrects that obvious mistake.
+	  ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged
+	  revisions 425735 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-16 17:30 +0000 [r425714]  George Joseph <george.joseph at fairview5.com>
+
+	* /, include/asterisk/config.h, tests/test_config.c, main/config.c:
+	  config: Fix inf loop using ast_category_browse and
+	  ast_variable_retrieve Fix infinite loop when calling
+	  ast_variable_retrieve inside an ast_category_browse loop when
+	  there is more than 1 category with the same name. Tested-by:
+	  George Joseph Review: https://reviewboard.asterisk.org/r/4089/
+	  ........ Merged revisions 425713 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-16 14:35 +0000 [r425691]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_pjsip_send_to_voicemail.c,
+	  include/asterisk/res_pjsip_pubsub.h,
+	  res/res_pjsip_header_funcs.c,
+	  res/res_pjsip_outbound_authenticator_digest.c,
+	  res/res_pjsip_outbound_registration.c,
+	  res/res_pjsip_endpoint_identifier_anonymous.c,
+	  res/res_pjsip_phoneprov_provider.c,
+	  res/res_pjsip_one_touch_record_info.c,
+	  res/res_pjsip_dialog_info_body_generator.c,
+	  res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_acl.c,
+	  res/res_pjsip_pidf_eyebeam_body_supplement.c,
+	  res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
+	  res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
+	  include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c,
+	  res/res_pjsip_endpoint_identifier_ip.c,
+	  res/res_pjsip_publish_asterisk.c, res/res_pjsip_sdp_rtp.c,
+	  res/res_hep_pjsip.c, res/res_pjsip_messaging.c,
+	  res/res_pjsip_registrar_expire.c, res/res_pjsip_caller_id.c,
+	  res/res_pjsip_mwi_body_generator.c,
+	  res/res_pjsip_endpoint_identifier_user.c, res/res_pjsip_nat.c,
+	  res/res_pjsip_session.c, res/res_pjsip_exten_state.c,
+	  res/res_pjsip_rfc3326.c, res/res_pjsip_mwi.c,
+	  res/res_pjsip_path.c, res/res_pjsip_pubsub.c,
+	  res/res_pjsip_registrar.c, channels/chan_pjsip.c,
+	  res/res_pjsip_transport_websocket.c,
+	  include/asterisk/res_pjsip.h, res/res_pjsip_multihomed.c, /,
+	  res/res_pjsip_authenticator_digest.c,
+	  res/res_pjsip_pidf_digium_body_supplement.c, res/res_pjsip_t38.c,
+	  res/res_pjsip_logger.c: PJSIP: Enforce module load dependencies
+	  This enforces that res_pjsip, res_pjsip_session, and
+	  res_pjsip_pubsub have loaded properly before attempting to load
+	  any modules that depend on them since the module loader system is
+	  not currently capable of resolving module dependencies on its
+	  own. ASTERISK-24312 #close Reported by: Dafi Ni Review:
+	  https://reviewboard.asterisk.org/r/4062/ ........ Merged
+	  revisions 425690 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-16 06:11 +0000 [r425669]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* channels/chan_unistim.c, /: Fix loss of voice after second call
+	  drops (on a second line) in case using multiple lines on unistim
+	  phones. There is regression was introduced in r391379. Reported
+	  by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........
+	  Merged revisions 425667 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425668 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-16 01:25 +0000 [r425646]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE
+	  state would get reset when it shouldn't. In the case where the
+	  ICE negotiation had not yet started current state would get wiped
+	  when it shouldn't. This also removes channel binding as in
+	  practice this does not work well with other implementations.
+	  ........ Merged revisions 425644 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425645 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-15 19:31 +0000 [r425627]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_motif.c: chan_motif: Cleanup
+	  jingle_tech.capabilities only once.
+
+2014-10-15 19:05 +0000 [r425611]  Jonathan Rose <jrose at digium.com>
+
+	* res/parking/parking_tests.c: parking_tests: Fix assertions and
+	  possibly crashes in res_parking unit tests Assertions were caused
+	  by attempting to play music on hold to a channel with no formats.
+	  Parking unit test channels were given formats and a technology so
+	  that they would be able to pretend to read/write frames.
+	  ASTERISK-24413 #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/4075/
+
+2014-10-15 09:59 +0000 [r425590]  Alexandr Anikin <may at telecom-service.ru>
+
+	* /, addons/chan_ooh323.c: chan_ooh323: fix rtptimeout general
+	  value checking correct condition to check rtptimeout in [general]
+	  config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
+	  Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........
+	  Merged revisions 425547 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425548 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425589 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 20:46 +0000 [r425526]  George Joseph <george.joseph at fairview5.com>
+
+	* tests/test_config.c, main/config.c, /, include/asterisk/config.h:
+	  config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG
+	  the /main/config config_basic_ops test was causing a SEGV while
+	  doing an ast_category_delete in an ast_category_browse loop.
+	  Apparently this never worked but was also never tested. I removed
+	  the test, added 2 notes to config.h indicating that it's not
+	  supported and added a few lines of code to ast_category_delete to
+	  prevent the SEGV should someone attempt it in the future.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4078/ ........ Merged
+	  revisions 425525 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 19:00 +0000 [r425504]  Jonathan Rose <jrose at digium.com>
+
+	* main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug
+	  which makes new tasks not execute Tasks that were marked for
+	  pending deletion in the scheduler would be moved to the cache for
+	  later reuse, but after being recycled the deleted mark wouldn't
+	  be removed resulting in fresh tasks being deleted without
+	  reason... and immediately moved back into the cache where they
+	  could be reused again. This could cause horrendous things to
+	  happen in just about anything that used a scheduler.
+	  ASTERISK-24321 #close Reported by: Steve Pitts Review:
+	  https://reviewboard.asterisk.org/r/4071/ ........ Merged
+	  revisions 425503 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 18:12 +0000 [r425481]  George Joseph <george.joseph at fairview5.com>
+
+	* include/asterisk/phoneprov.h, /,
+	  res/res_pjsip_phoneprov_provider.c, res/res_phoneprov.c:
+	  res_phoneprov: Create accessor for
+	  ast_phoneprov_std_variable_lookup Based on feedback from Richard,
+	  I created an accessor for
+	  res_phoneprov/ast_phoneprov_std_variable_lookup and added load
+	  priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by:
+	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/
+	  ........ Merged revisions 425480 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 16:46 +0000 [r425459]  Corey Farrell <git at cfware.com>
+
+	* /, res/res_fax.c: res_fax: Fix reference leak caused by gateway
+	  sessions Fax gateway session objects can be re-used, causing the
+	  same gateway session to be added to faxregistry.container more
+	  than once. This change causes fax_session_new to remove the
+	  reserved session from the container before it's id is changed,
+	  ensuring it's possible for the session to be freed.
+	  ASTERISK-24392 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/4049/ ........ Merged
+	  revisions 425457 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425458 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 16:35 +0000 [r425455]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished
+	  Dials when doing masquerades (Part 2) Masquerades into and out of
+	  channels that are involved in a dial operation don't create the
+	  expected dial end event. The missing dial end event goes against
+	  the model for things like CDRs and generating Dial end manager
+	  actions and such. There are four cases: 1) A channel masquerades
+	  into the caller channel. The case happens when performing a
+	  blonde transfer using the channel driver's protocol. 2) A channel
+	  masquerades into a callee channel. The case happens when
+	  performing a directed call pickup. 3) The caller channel
+	  masquerades out of dial. The case happens when using the Bridge
+	  application on the caller channel. 4) A callee channel
+	  masquerades out of dial. The case happens when using the Bridge
+	  application on a peer channel. As it turned out, all four cases
+	  need to be handled instead of just the first one. ASTERISK-24237
+	  Reported by: Richard Mudgett ASTERISK-24394 #close Reported by:
+	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/
+	  ........ Merged revisions 425430 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-14 16:19 +0000 [r425415]  Corey Farrell <git at cfware.com>
+
+	* /, res/res_fax.c: res_fax: Resolve module reference leak caused
+	  by reserved sessions Remove reference to module providing
+	  reserved session after adding a reference to the final module.
+	  This re-reference is done to ensure that module references are
+	  correct even if the final session selects a different module than
+	  the reserved session. ASTERISK-18923 #close Reported by: Grigoriy
+	  Puzankin Review: https://reviewboard.asterisk.org/r/4048/
+	  ........ Merged revisions 425405 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425407 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425411 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-13 16:10 +0000 [r425384]  George Joseph <george.joseph at fairview5.com>
+
+	* res/res_sorcery_config.c, main/manager.c, /,
+	  include/asterisk/config.h, pbx/pbx_realtime.c,
+	  tests/test_config.c, apps/app_directory.c, tests/test_sorcery.c,
+	  main/config.c, tests/test_sorcery_realtime.c,
+	  res/res_sorcery_realtime.c, apps/app_voicemail.c: manager/config:
+	  Support templates and non-unique category names via AMI This
+	  patch provides the capability to manipulate templates and
+	  categories with non-unique names via AMI. Summary of changes:
+	  GetConfig and GetConfigJSON: Added "Filter" parameter: A comma
+	  separated list of name_regex=value_regex expressions which will
+	  cause only categories whose variables match all expressions to be
+	  considered. The special variable name TEMPLATES can be used to
+	  control whether templates are included. Passing 'include' as the
+	  value will include templates along with normal categories.
+	  Passing 'restrict' as the value will restrict the operation to
+	  ONLY templates. Not specifying a TEMPLATES expression results in
+	  the current default behavior which is to not include templates.
+	  UpdateConfig: NewCat now includes options for allowing duplicate
+	  category names, indicating if the category should be created as a
+	  template, and specifying templates the category should inherit
+	  from. The rest of the actions now accept a filter string as
+	  defined above. If there are non-unique category names, you can
+	  now update specific ones based on variable values. To facilitate
+	  the new capabilities in manager, corresponding changes had to be
+	  made to config, most notably the addition of filter criteria to
+	  many of the APIs. In some cases it was easy to change the
+	  references to use the new prototype but others would have
+	  required touching too many files for this patch so a wrapper with
+	  the original prototype was created. Macros couldn't be used in
+	  this case because it would break binary compatibility with
+	  modules such as res_digium_phone that are linked to real symbols.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4033/ ........ Merged
+	  revisions 425383 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-12 21:09 +0000 [r425362]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Make the ICE
+	  transport check case insensitive as some implementations use
+	  'udp'. ........ Merged revisions 425360 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425361 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-12 08:15 +0000 [r425289-425299]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send
+	  reINVITE after a BYE. After a reINVITE glare situation, Asterisk
+	  would re-send the reINVITE even though the call had been hung up
+	  in the mean time. This patch unschedules the reinvite when
+	  handling the BYE. ASTERISK-22791 #close Reported by: Paolo
+	  Compagnini Tested by: Paolo Compagnini Review:
+	  https://reviewboard.asterisk.org/r/4056/ (testcase is in review
+	  r4055) ........ Merged revisions 425296 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425297 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425298 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* /, Makefile: build: Relax badshell tilde test to allow for ~ in
+	  middle of DESTDIR. The main Makefile has a target test called
+	  'badshell' that tests if DESTDIR does not happen to have an
+	  an-expanded tilde (~). This might be the case if you run: make
+	  install DESTDIR=~/somewhere/ That test also disallowed valid
+	  tildes in directory names. The test is now changed to only
+	  trigger on a tilde at the start of the path. ASTERISK-13797
+	  #close Reported by: Tzafrir Cohen Review:
+	  https://reviewboard.asterisk.org/r/4064/ ........ Merged
+	  revisions 425291 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425292 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425293 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* res/res_calendar_ews.c, /: res_calendar_ews: Relax neon version
+	  check to work with 0.30 too. Allow res_calendar_ews to work not
+	  only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close
+	  Reported by: Tzafrir Cohen Review:
+	  https://reviewboard.asterisk.org/r/4068/ ........ Merged
+	  revisions 425286 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425287 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425288 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-11 21:08 +0000 [r425265]  George Joseph <george.joseph at fairview5.com>
+
+	* /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error
+	  handling Tested module load/reload interaction between
+	  res_phoneprov and res_pjsip_phoneprov_provider in cases where
+	  res_phoneprov didn't load correctly (usually misconfiguration or
+	  missing phoneprov.conf) Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4069/ ........ Merged
+	  revisions 425264 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 20:48 +0000 [r425243]  Joshua Colp <jcolp at digium.com>
+
+	* main/bridge.c, bridges/bridge_native_rtp.c, /: bridge: During a
+	  smart bridge operation provide a more complete bridge to the old
+	  technology. When a smart bridge operation occurs and a bridge
+	  transitions from one technology to another the old technology is
+	  provided the channels formerly in it and told that they are
+	  leaving. Unfortunately the bridge provided along with them is
+	  incomplete. The bridge, despite there being channels in it,
+	  contains none. This forces technology implementations to have
+	  additional logic when channels are leaving or to store their own
+	  duplicated state. This change makes the bridge more complete so
+	  it contains the expected channels. Now that the bridge is
+	  complete special logic within bridge_native_rtp is no longer
+	  needed and has been removed. Review:
+	  https://reviewboard.asterisk.org/r/4057/ ........ Merged
+	  revisions 425242 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 14:31 +0000 [r425221]  Matthew Jordan <mjordan at digium.com>
+
+	* /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration
+	  if res_phoneprov didn't load If res_phoneprov failed to fully
+	  load (due to not being configured), the providers container will
+	  be NULL. If a module attempts to register a phone provisioning
+	  provider, it should check for the presence of the container. If
+	  there is no providers container, it should return an error. This
+	  patch makes the ast_phoneprov_provider_register function do
+	  that... otherwise this would be a silly commit message. ........
+	  Merged revisions 425220 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 14:23 +0000 [r425217]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_pjsip_phoneprov_provider.c:
+	  res_pjsip_phoneprov_provider: Add missing dependency on
+	  pjproject. ........ Merged revisions 425216 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 13:01 +0000 [r425155]  Kinsey Moore <kmoore at digium.com>
+
+	* /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing
+	  regression This fixes a regression in callerid parsing introduced
+	  when another bug was fixed. This bug occurred when the name was
+	  composed entirely of DTMF keys and quoted without a number
+	  section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard
+	  Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by
+	  Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/
+	  ........ Merged revisions 425152 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425153 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425154 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 12:10 +0000 [r425132]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into
+	  rport of responses if 'force_rport' is on. When the 'force_rport'
+	  option is enabled the behavior should be the same as if the
+	  remote side placed rport into the message themselves. Therefore
+	  any responses we send should include the source port of the
+	  request in the rport of the Via header. #SIPit31 ASTERISK-24387
+	  #close Reported by: Matt Jordan ........ Merged revisions 425131
+	  from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-10 07:32 +0000 [r425071]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from
+	  missing ACK to re-INVITE. If a device re-INVITEs at the same time
+	  as the dialog is hung up, and if then the ACK to the re-INVITE
+	  never reaches Asterisk, chan_sip would fail to destroy the dialog
+	  after a while. This resulted in (most prominently) file handle
+	  leaks. (Patch reindented by me.) ASTERISK-20784 #close
+	  ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal
+	  Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle
+	  (License #5334) patch_asterisk_20784.txt uploaded by Nitesh
+	  Bansal (License #6418) Reviewboard:
+	  https://reviewboard.asterisk.org/r/4052/ (testcase can be found
+	  at r4051) ........ Merged revisions 425068 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 425069 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 425070 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 23:35 +0000 [r425052]  George Joseph <george.joseph at fairview5.com>
+
+	* res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider:
+	  fix compile breakage on AST_VECTOR endpoint->inbound_auths was
+	  changed to a vector in 13 and I committed the 12 patch instead of
+	  the 13 patch. Tested-by: George Joseph
+
+2014-10-09 21:38 +0000 [r425031]  Kevin Harwell <kharwell at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no
+	  candidates received for component When starting ice if there is
+	  not at least one remote ice candidate with an RTP component
+	  asterisk will crash. This is due to an assertion in pjnath as it
+	  expects at least one candidate with an RTP component. Added a
+	  check to make sure at least one candidate contains an RTP
+	  component and at least one candidate has an RTCP component.
+	  ASTERISK-24383 #close Review:
+	  https://reviewboard.asterisk.org/r/4039/ ........ Merged
+	  revisions 425030 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 20:54 +0000 [r425008]  George Joseph <george.joseph at fairview5.com>
+
+	* /, res/res_pjsip_phoneprov_provider.c (added),
+	  configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider:
+	  Provides pjsip integration with res_phoneprov This module allows
+	  res_pjsip to integrate with res_phoneprov. It handles the pjsip
+	  'phoneprov' object type. Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3976/ ........ Merged
+	  revisions 425007 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 18:37 +0000 [r424986]  Matthew Jordan <mjordan at digium.com>
+
+	* /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk
+	  load on module load failure ........ Merged revisions 424985 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 17:45 +0000 [r424964]  George Joseph <george.joseph at fairview5.com>
+
+	* include/asterisk/chanvars.h, res/res_phoneprov.c,
+	  res/res_phoneprov.exports.in (added), main/chanvars.c,
+	  include/asterisk/phoneprov.h (added), /,
+	  configs/samples/phoneprov.conf.sample: res_phoneprov: Refactor
+	  phoneprov to allow pluggable config providers This patch makes
+	  res_phoneprov more modular so other modules (like pjsip) can
+	  provide configuration information instead of res_phoneprov
+	  relying solely on users.conf and sip.conf. To accomplish this a
+	  new ast_phoneprov public API is now exposed which allows config
+	  providers to register themselves, set defaults (server profile,
+	  etc) and add user extensions. * ast_phoneprov_provider_register
+	  registers the provider and provides callbacks for loading default
+	  settings and loading users. * ast_phoneprov_provider_unregister
+	  clears the defaults and users. * ast_phoneprov_add_extension
+	  should be called once for each user/extension by the provider's
+	  load_users callback to add them. * ast_phoneprov_delete_extension
+	  deletes one extension. * ast_phoneprov_delete_extensions deletes
+	  all extensions for the provider. Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3970/ ........ Merged
+	  revisions 424963 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 16:36 +0000 [r424942]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/cdr.c, /: cdr.c: Make turning on CDR debug a one step
+	  process instead of two. Now "cdr set debug on" doesn't also
+	  require "core set verbose 1" to see CDR debug output. ........
+	  Merged revisions 424941 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-09 08:08 +0000 [r424880]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* contrib/scripts/safe_asterisk, /: safe_asterisk: Don't
+	  automatically exceed MAXFILES value of 2^20. On systems with lots
+	  of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can
+	  exceed the per-process file limit of 2^20. This patch ensures the
+	  value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close
+	  Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff
+	  uploaded by Michael Myles (License #6626) ........ Merged
+	  revisions 424875 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 424878 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 424879 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-08 18:46 +0000 [r424854]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Allow only UDP ICE
+	  candidates. The underlying library, pjnath, that res_rtp_asterisk
+	  uses for ICE support does not have support for ICE-TCP. As
+	  candidates are passed through directly to it this can cause error
+	  messages to occur when it receives something unexpected (such as
+	  a TCP candidate). This change merely ignores all non-UDP
+	  candidates so they never reach pjnath. ASTERISK-24326 #close
+	  Reported by: Joshua Colp ........ Merged revisions 424852 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 424853 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-10-08 18:24 +0000 [r424769-424850]  Kinsey Moore <kmoore at digium.com>
+
+	* main/stasis.c: Stasis: Relegate log message to dev-mode This
+	  error message primarily applies to development tasks and will now
+	  only show up when dev-mode is enabled via configure.
+
+	* main/sounds_index.c: Indexer: Format message types may not exist
+	  In Asterisk 13+, any given message type is not guaranteed to
+	  exist even if Asterisk comes up correctly since creation of the
+	  message type could be declined. The indexer should not prevent
+	  Asterisk from starting under these conditions.
+
+	* main/stasis.c: Stasis: Only log errors for non-declined types
+	  When message type creation is declined via stasis.conf, certain
+	  operations log errors assuming that the declined type is being
+	  used before initialization or after destruction. These error
+	  messages get quite spammy for oft used message types and should
+	  not be logged in the first place since the message type is
+	  validly NULL. Reported by: Matt DiMeo
+

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