[svn-commits] bebuild: tag 13.0.0 r426118 - in /tags/13.0.0: ./ contrib/realtime/mysql/ con...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Oct 24 09:42:54 CDT 2014
Author: bebuild
Date: Fri Oct 24 09:42:47 2014
New Revision: 426118
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=426118
Log:
Importing files for 13.0.0 release.
Added:
tags/13.0.0/.lastclean (with props)
tags/13.0.0/.version (with props)
tags/13.0.0/ChangeLog (with props)
tags/13.0.0/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/13.0.0/contrib/realtime/mysql/mysql_config.sql (with props)
tags/13.0.0/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/13.0.0/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/13.0.0/contrib/realtime/oracle/oracle_config.sql (with props)
tags/13.0.0/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/13.0.0/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/13.0.0/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/13.0.0/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/13.0.0/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/13.0.0/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/13.0.0/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/13.0.0/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0/.lastclean?view=auto&rev=426118
==============================================================================
--- tags/13.0.0/.lastclean (added)
+++ tags/13.0.0/.lastclean Fri Oct 24 09:42:47 2014
@@ -1,0 +1,1 @@
+40
Propchange: tags/13.0.0/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/13.0.0/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/13.0.0/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/13.0.0/.version
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0/.version?view=auto&rev=426118
==============================================================================
--- tags/13.0.0/.version (added)
+++ tags/13.0.0/.version Fri Oct 24 09:42:47 2014
@@ -1,0 +1,1 @@
+13.0.0
Propchange: tags/13.0.0/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/13.0.0/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/13.0.0/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/13.0.0/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0/ChangeLog?view=auto&rev=426118
==============================================================================
--- tags/13.0.0/ChangeLog (added)
+++ tags/13.0.0/ChangeLog Fri Oct 24 09:42:47 2014
@@ -1,0 +1,19275 @@
+2014-10-24 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 13.0.0 Released.
+
+2014-10-22 21:27 +0000 [r426097] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: Cannot use struct
+ ast_translator.core_{src,src}_codec. This fixes a Segmentation
+ fault introduced in r419044 "media formats: re-architect handling
+ of media for performance improvements". The problem is that
+ codec_dahdi was using core_src_codec and core_dst_codec in the
+ ast_translator structure when these fields were never set. Now
+ instead of trying to map the new core codec descriptions to the
+ way DAHDI defines different codecs, we will store the DAHDI
+ specific formats in 'struct translator' directly so we can refer
+ to them without mapping. This also allows us to remove the
+ "global_format_map" structure, since we can now query the list of
+ translators directly to make sure we do not ever register a DAHDI
+ based translator for a specific path more than once and eliminate
+ the need to keep the list and the map in sync. ASTERISK-24435
+ #close Reported by: Marian Koniuszko Review:
+ https://reviewboard.asterisk.org/r/4105/
+
+2014-10-21 17:47 +0000 [r426079] Richard Mudgett <rmudgett at digium.com>
+
+ * main/translate.c: translage.c: Fix regression when generating
+ translation path strings. Fix the AMI Status action read and
+ write translation path strings from growing for each channel in
+ the status event list by reseting the ast string given to
+ ast_translate_path_to_str() to fill in the given translation
+ path.
+
+2014-10-20 14:15 +0000 [r425991] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_xmpp.c, main/tcptls.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>
+
+ * Makefile, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, makeopts.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_sdp_rtp.c, res/res_pjsip.c,
+ res/res_pjsip_session.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_t38.c, res/res_pjsip_registrar_expire.c,
+ res/res_pjsip_mwi_body_generator.c,
+ res/res_pjsip_endpoint_identifier_user.c,
+ res/res_pjsip_send_to_voicemail.c,
+ include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_outbound_authenticator_digest.c,
+ res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c,
+ res/res_pjsip_acl.c, res/res_pjsip_pubsub.c,
+ res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
+ include/asterisk/res_pjsip.h,
+ res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
+ res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c,
+ res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c,
+ res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
+ res/res_pjsip_logger.c, res/res_pjsip_nat.c,
+ res/res_pjsip_session.c, res/res_pjsip_exten_state.c,
+ res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c,
+ res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c,
+ res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c,
+ channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c,
+ res/res_pjsip_pidf_eyebeam_body_supplement.c,
+ include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c,
+ res/res_pjsip_pidf_digium_body_supplement.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ res/res_pjsip_publish_asterisk.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>
+
+ * /, include/asterisk/config.h, tests/test_config.c, main/config.c:
+ 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>
+
+ * res/res_phoneprov.c, include/asterisk/phoneprov.h, /,
+ res/res_pjsip_phoneprov_provider.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>
+
+ * 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, res/res_sorcery_config.c, main/manager.c,
+ /, include/asterisk/config.h, pbx/pbx_realtime.c,
+ tests/test_config.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/phoneprov.h (added), /,
+ configs/samples/phoneprov.conf.sample,
+ include/asterisk/chanvars.h, res/res_phoneprov.c,
+ res/res_phoneprov.exports.in (added), main/chanvars.c:
+ 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
[... 22548 lines stripped ...]
More information about the svn-commits
mailing list