--- /dev/null
+2013-10-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 12.0.0-beta1 Released.
+
+2013-10-08 22:58 +0000 [r400771] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_pjsip_header_funcs.c (added): Add PJSIP_HEADER function
+ for manipulation of SIP headers in the PJSIP stack This patch
+ adds support to the PJSIP stack in Asterisk for SIP header
+ manipulation. Note that this is analagous to
+ SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming
+ supplemental session callback is registered that takes the
+ pjsip_hdrs from the incoming session and stores them in a linked
+ list in the session datastore. Calls to PJSIP_HEADER traverse
+ over the list and return the nth matching header where 'n' is the
+ 'number' argument to the function. When adding a header, the
+ first call creates a datastore and linked list and adds the
+ datastore to the session. The header is then created as a
+ pjsip_hdr and added to the list. An outgoing supplemental session
+ callback then traverses the list and adds the headers to the
+ outgoing pjsip_msg. When removing a header, the list created with
+ PJSIP_HEADER(add,...) is traversed and all matching entries are
+ removed. (closes issue ASTERISK-22498) Reported by: George Joseph
+ patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph
+ (License 6322)
+
+2013-10-08 22:30 +0000 [r400769] Kinsey Moore <kmoore@digium.com>
+
+ * /, configure, configure.ac: Add warning when compiling with iODBC
+ support When running configure, libiodbc2 development headers
+ will fulfill the requirement for ODBC development headers, but
+ will not function properly. This adds a warning when libiodbc2
+ development headers are detected instead of unixodbc development
+ headers. (closes issue ASTERISK-22459) Reported by: Patrick
+ Maille Tested by: Walter Doekes Patches:
+ issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
+ (License 5674) ........ Merged revisions 400767 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400768 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-08 21:19 +0000 [r400754] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_agent_pool.c: app_agent_pool: Fix AMI/CLI AgentLogoff
+ soft preventing agents from logging back in. * Clear the
+ deferred_logoff flag when an agent logs in. (closes issue
+ ASTERISK-22669) Reported by: John Bigelow
+
+2013-10-08 20:51 +0000 [r400749] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip/config_transport.c: Switch from
+ using pjsip_strerror to pj_strerror. pjsip_strerror is only aware
+ of PJSIP-specific error codes. pj_strerror() is aware of all
+ PJProject error codes and OS-specific error codes. This
+ specifically fixes an oft-seen error in transport configuration
+ code where EADDRINUSE would result in "Unknown PJSIP error
+ 120098" instead of a useful message.
+
+2013-10-08 20:16 +0000 [r400724-400742] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/confbridge.conf.sample, /,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
+ CHANGES, apps/confbridge/conf_config_parser.c: app_confbridge:
+ Can now set the language used for announcements to the
+ conference. ConfBridge now has the ability to set the language of
+ announcements to the conference. The language can be set on a
+ bridge profile in confbridge.conf or by the dialplan function
+ CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983)
+ Reported by: Jonathan White Patches: M19983_rev2.diff (license
+ #5138) patch uploaded by junky (modified) Tested by: rmudgett
+ ........ Merged revisions 400741 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix
+ duplicate default_user profile. * Fixed looking in the wrong
+ profiles container to see if the default_user profile is already
+ created in verify_default_profiles(). The bridge profile
+ container is never going to hold user profiles. :) ........
+ Merged revisions 400723 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-08 18:19 +0000 [r400682-400701] Kinsey Moore <kmoore@digium.com>
+
+ * /, funcs/func_config.c: Fix func_config list entry allocation The
+ AST_CONFIG dialplan function defined in func_config.c allocates
+ its config file list entries using ast_malloc. List entry
+ allocations destined for use with Asterisk's linked list API must
+ be ast_calloc()d or otherwise initialized so that list pointers
+ are set to NULL. These uses of ast_malloc have been replaced by
+ ast_calloc to prevent dereferencing of uninitialized pointer
+ values when traversing the list. (closes issue ASTERISK-22483)
+ Reported by: Brian Scott ........ Merged revisions 400694 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400697 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_rtp_asterisk.c, /: Fix STUN crash when using IPv6 any
+ address Ensure that when chan_sip binds to the IPv6 any address
+ ([::]), IPv4 candidates are also added. (closes issue
+ ASTERISK-21917) Reported by: Torrey Searle Patches:
+ 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License
+ 5334) ........ Merged revisions 400681 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-08 15:36 +0000 [r400680] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/pjsip_options.c: Push CLI qualify into the
+ threadpool. If you run Asterisk in the background and then
+ connect to it through a separate console, the thread that runs
+ CLI commands is not registered with PJLIB. Thus PJLIB does not
+ like it when you attempt to send OPTIONS requests from that
+ thread. So now we push the task into the threadpool, which we
+ know to be registered with PJLIB. Thanks to Antti Yrjola for
+ reporting this.
+
+2013-10-08 15:11 +0000 [r400661-400671] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_agi.c, apps/app_queue.c: Make app_queue and res_agi
+ independent of AMI being enabled. The
+ https://reviewboard.asterisk.org/r/2888/ review changes manager
+ to not subscribe to stasis when it is disabled for performance
+ reasons. When manager is disabled app_queue and res_agi decline
+ to load and fail to clean up what they have already allocated. *
+ Made app_queue and res_agi clean up allocated resources when they
+ decline to load. * Made app_queue and res_agi use their own
+ subscriptions to the stasis topics instead of borrowing manager's
+ message router structure inappropriately. (closes issue
+ ASTERISK-22604) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2902/
+
+ * include/asterisk/stasis.h, apps/app_queue.c,
+ include/asterisk/manager.h: Miscellaneous stand alone comment
+ cleanups.
+
+2013-10-06 17:11 +0000 [r400624] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_queue.c, /: Fix Regression With Queuelog EXITWITHKEY
+ Only Logging Two Out Of Four Fields Commit r62462 added two extra
+ fields for logging "the original position the caller entered the
+ queue at, and the amount of time the caller was waiting in the
+ queue." But when r75969 was merged from 1.4 into trunk (r75977),
+ these two fields disappeared. Those two extra fields were not
+ logged in 1.4 and when the patch was merged, those fields went
+ away. Therefore, this is a regression and was caught by the
+ reporter because he was reading the awesome "Asterisk: The
+ Definitive Guide" book. (closes issue ASTERISK-22197) Reported
+ by: Dalius M. Tested by: Dalius M. Patches:
+ asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2901/ ........ Merged
+ revisions 400622 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400623 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-05 00:41 +0000 [r400588] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/iax2/include/parser.h: chan_iax2: Fix compile error.
+
+2013-10-04 21:40 +0000 [r400567] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/netsock2.c, channels/iax2/include/parser.h, main/acl.c,
+ include/asterisk/netsock2.h, CHANGES, channels/chan_iax2.c,
+ channels/iax2/parser.c, main/netsock.c: Add IPv6 Support To
+ chan_iax2 This patch adds IPv6 support to chan_iax2. Yay! (closes
+ issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2660/
+
+2013-10-04 19:31 +0000 [r400552] David M. Lee <dlee@digium.com>
+
+ * rest-api/api-docs/applications.json (added): Added missing file
+ from r400522
+
+2013-10-04 18:42 +0000 [r400532-400542] Jonathan Rose <jrose@digium.com>
+
+ * res/res_pjsip_logger.c: chan_pjsip: Make logger togglable without
+ loading/unloading This patch makes the res_pjsip_logger do a few
+ things... First, it will be built and installed by default now,
+ so end users won't need to enable it in menuselect. Second, while
+ it is loaded, it no longer will immediately issue log messages.
+ Upon loading, it is in the disabled state and must be turned on
+ with the new CLI command. The CLI command 'pjsip set logger
+ <on/off/host> has been added and can be used to do the following:
+ pjsip set logger on: Enables logger for all PJSIP traffic pjsip
+ set logger off: Disables logger for all PJSIP traffic pjsip set
+ logger host <host>: Enables logger for the specific host Review:
+ https://reviewboard.asterisk.org/r/2900/
+
+ * configs/extconfig.conf.sample, configs/sorcery.conf.sample,
+ contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py
+ (added),
+ contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py:
+ chan_pjsip: Add alembic scripts for generating db tables for
+ PJSIP Also updates sample configurations for sorcery and
+ extconfig to demonstrate how to use databases created by that
+ alembic script. (closes issue ASTERISK-22133) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2892/
+
+2013-10-04 15:54 +0000 [r400522] Matthew Jordan <mjordan@digium.com>
+
+ * rest-api/resources.json, include/asterisk/_private.h,
+ main/endpoints.c, res/ari/ari_model_validators.c,
+ res/ari/ari_model_validators.h, res/res_ari_model.c, main/json.c,
+ res/ari.make, res/ari/resource_applications.c (added),
+ res/ari/resource_applications.h (added), res/res_stasis.c,
+ main/asterisk.c, rest-api/api-docs/endpoints.json,
+ rest-api/api-docs/events.json, res/stasis/app.c,
+ include/asterisk/endpoints.h,
+ rest-api-templates/ari_model_validators.h.mustache,
+ res/res_ari_applications.c (added), res/ari/resource_endpoints.h,
+ include/asterisk/stasis_app.h, res/stasis/app.h: ARI: Add
+ subscription support This patch adds an /applications API to ARI,
+ allowing explicit management of Stasis applications. * GET
+ /applications - list current applications * GET
+ /applications/{applicationName} - get details of a specific
+ application * POST /applications/{applicationName}/subscription -
+ explicitly subscribe to a channel, bridge or endpoint * DELETE
+ /applications/{applicationName}/subscription - explicitly
+ unsubscribe from a channel, bridge or endpoint Subscriptions work
+ by a reference counting mechanism: if you subscript to an event
+ source X number of times, you must unsubscribe X number of times
+ to stop receiveing events for that event source. Review:
+ https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451)
+ Reported by: Matt Jordan
+
+2013-10-04 15:48 +0000 [r400510-400520] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip.c: Enclose the To URI and update its user portion
+ if a request user has been specified.
+
+ * res/res_pjsip_session.c: Replace the connection address at the
+ SDP level if altering the SDP with the external media address.
+
+2013-10-04 04:54 +0000 [r400508] David M. Lee <dlee@digium.com>
+
+ * rest-api/api-docs/playback.json, res/res_ari_playback.c:
+ Corrected response class for stopPlayback
+
+2013-10-03 23:11 +0000 [r400471] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Don't ignore expires value in
+ contact header if it lacks semicolon (closes issue
+ ASTERISK-22574) Reported by: Filip Jenicek Patches:
+ chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
+ ........ Merged revisions 400469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400470 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-03 21:40 +0000 [r400460] Matthew Jordan <mjordan@digium.com>
+
+ * main/channel_internal_api.c: Remove publication of a channel
+ snapshot when the technology is set This patch removes said
+ publication for a few reasons: (1) It is unnecessary. Association
+ of the channel technology with a specific channel is an
+ implementation detail that should be assumed to "just happen",
+ and consumers of Stasis don't need to be informed about it. (2)
+ Publication of said message can now cause crashes, as the actual
+ creation of a channel in normal locations now stages its
+ messages. As a result, things that create dummy channels (such as
+ the SIP RTP QOS unit test) and associate them with a channel
+ technology were now crashing, as the channel itself was not known
+ by Stasis.
+
+2013-10-03 19:31 +0000 [r400442] Joshua Colp <jcolp@digium.com>
+
+ * main/cdr.c: When serializing CDR variables (like for "core show
+ channels") don't output an error if CDRs aren't enabled.
+
+2013-10-03 19:29 +0000 [r400440] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/security_events.c: Fix security events for AMI invalid
+ password In r337595, additional security events were added for
+ chan_sip authentication failures. The new IEs added to the
+ existing invalid password event were defined as required IEs, but
+ existing users of the event did not set the new IEs and could not
+ since they didn't apply to existing uses. They are now marked as
+ optional IEs. (closes issue ASTERISK-22578) Reported by: Matt
+ Jordan ........ Merged revisions 400421 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-03 19:11 +0000 [r400403] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/bridge_technology.h,
+ bridges/bridge_native_rtp.c: Fix assumption in
+ bridge_native_rtp.c regarding number of participants in a bridge.
+ When a party leaves a bridge, there may be more participants in
+ the bridge than expected. As such, it is important not to make
+ assumptions regarding the list of channels in a bridge. This
+ change makes it so that when a party leaves a native RTP bridge,
+ we unbridge it and the party it was bridged with. Previously, the
+ first and last channels in the list were unbridged since it was
+ assumed that these were the two channels that had been bridged.
+ As previously stated, a new party had been inserted into the
+ bridge, so this logic did not work properly. (closes issue
+ ASTERISK-22615) reported by Matt Jordan (closes issue
+ ASTERISK-22532) reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2899
+
+2013-10-03 19:05 +0000 [r400401] Joshua Colp <jcolp@digium.com>
+
+ * res/ari/resource_channels.c: Fix a crash caused by muting and
+ unmuting a channel in ARI without specifying a direction. (closes
+ issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by
+ Matt Jordan, whose office I have taken over in the name of
+ Canada.
+
+2013-10-03 18:44 +0000 [r400398] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cel.c: cel: Some whitespace cleanups
+
+2013-10-03 18:28 +0000 [r400384-400395] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_multicast.c, /: Ensure res_rtp_mutlicast sets SSRC
+ properly This fixes a bug where the SSRC field on multicast RTP
+ can be stuck at 0 which can cause problems for endpoints trying
+ to make sense of incoming streams. (closes issue ASTERISK-22567)
+ Reported by: Simone Camporeale Patches:
+ 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
+ (License 6536) ........ Merged revisions 400393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400394 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/xml.c: Detect and use xsltCleanupGlobals when available This
+ introduces usage of an additional libxslt cleanup function,
+ xsltCleanupGlobals, when the configure script detects that it is
+ available. Early versions of the library did not include this
+ function. (closes issue ASTERISK-22570) Reported by: Corey
+ Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey
+ Farrell (License 5909)
+
+2013-10-03 17:55 +0000 [r400383] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/ast-db-manage/config/env.py,
+ contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
+ contrib/ast-db-manage/voicemail/env.py: Update Alembic database
+ scripts for external scripting and PostgreSQL, Oracle This patch
+ does the following: 1) The env scripts have been updated to be
+ tolerant of a NULL configuration file. This occurs when
+ configuration is provided by an external script, such that the
+ actual config.ini file is not used. 2) Enum types have all been
+ given names. This is needed for PostgreSQL script generation. 3)
+ The identifier meetme_confno_starttime_endtime is greater than 30
+ characters, and hence invalid for Oracle databases. This has been
+ truncated down to meetme_confno_start_end.
+
+2013-10-03 16:22 +0000 [r400373] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_vpb.cc: chan_vpb: Make compile again.
+
+2013-10-03 14:56 +0000 [r400362] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_cel.c: Get rid of uses of stasis_topic_wait()
+
+2013-10-03 14:51 +0000 [r400360] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_t38.c, res/res_pjsip_sdp_rtp.c: Fix crashes in
+ res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and
+ external_media_address is set. The callback function for changing
+ the media address in streams wrongly assumes that a connection
+ line will always be present. This is false as no line is present
+ if a stream has been rejected. (closes issue ASTERISK-22645)
+ Reported by: Rusty Newton
+
+2013-10-02 22:34 +0000 [r400318-400356] Mark Michelson <mmichelson@digium.com>
+
+ * res/ari/resource_bridges.c, channels/chan_jingle.c,
+ channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c,
+ pbx/pbx_spool.c, channels/dahdi/bridge_native_dahdi.c,
+ main/format_cap.c, channels/chan_motif.c, res/res_agi.c,
+ channels/chan_h323.c, apps/app_confbridge.c, res/res_stasis.c,
+ addons/chan_ooh323.c, channels/chan_sip.c,
+ bridges/bridge_holding.c, res/res_pjsip_sdp_rtp.c,
+ tests/test_format_api.c, bridges/bridge_simple.c,
+ bridges/bridge_softmix.c, main/core_local.c,
+ channels/chan_console.c, channels/chan_iax2.c,
+ channels/chan_oss.c, include/asterisk/format_cap.h,
+ res/res_pjsip_session.c, main/media_index.c, main/channel.c,
+ channels/chan_misdn.c, main/manager.c, channels/chan_skinny.c,
+ main/file.c, res/res_pjsip/pjsip_configuration.c,
+ channels/chan_alsa.c, tests/test_config.c, channels/chan_nbs.c,
+ bridges/bridge_native_rtp.c, addons/chan_mobile.c,
+ channels/chan_pjsip.c, channels/chan_mgcp.c,
+ res/res_clioriginate.c, channels/chan_unistim.c,
+ main/rtp_engine.c, channels/chan_multicast_rtp.c, main/ccss.c,
+ channels/chan_bridge_media.c, apps/app_meetme.c,
+ main/bridge_basic.c, apps/app_originate.c,
+ res/parking/parking_applications.c, channels/chan_gtalk.c: Cache
+ string values of formats on ast_format_cap() to save processing.
+ Channel snapshots have string representations of the channel's
+ native formats. Prior to this change, the format strings were
+ re-created on ever channel snapshot creation. Since channel
+ native formats rarely change, this was very wasteful. Now, string
+ representations of formats may optionally be stored on the
+ ast_format_cap for cases where string representations may be
+ requested frequently. When formats are altered, the string cache
+ is marked as invalid. When strings are requested, the cache
+ validity is checked. If the cache is valid, then the cached
+ strings are copied. If the cache is invalid, then the string
+ cache is rebuilt and copied, and the cache is marked as being
+ valid again. Review: https://reviewboard.asterisk.org/r/2879
+
+ * /: Remove svn:mergeinfo property.
+
+ * main/stasis_endpoints.c, main/stasis_wait.c (removed),
+ res/ari/resource_endpoints.c, /, include/asterisk/stasis.h,
+ tests/test_cel.c, include/asterisk/stasis_endpoints.h,
+ channels/chan_pjsip.c, main/stasis.c: Remove unnecessary waits
+ from stasis. Since caches are updated on publisher threads, there
+ is no need to wait for the cache updates to occur after a stasis
+ message is published. In the case of chan_pjsip device state
+ changes, this set of changes caused an improvement to
+ performance. Review: https://reviewboard.asterisk.org/r/2890
+
+2013-10-02 21:32 +0000 [r400316] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_iax2.c, /: Cast Integer Argument To Unsigned Char
+ The member reg in the peercnt structure is an unsigned char and
+ peercnt_modify() is expecting an unsigned char argument which
+ gets assigned to peercnt->reg. This patch fixes that by casting
+ the integer argument being passed to peercnt_modify to unsigned
+ char. ........ Merged revisions 400314 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400315 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-02 21:25 +0000 [r400312] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c, main/manager.c, main/cel.c: Only create Stasis
+ subscriptions when enabled Subscribing to Stasis isn't free. As
+ such, this patch makes AMI, CDR, and CEL - the "big 3" - only
+ subscribe when enabled. Toggling their availability via a .conf
+ file will unsubscribe/subscribe as appropriate. Review:
+ https://reviewboard.asterisk.org/r/2888/
+
+2013-10-02 20:30 +0000 [r400303] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: Originate: Make setting caller id on outgoing call
+ use either name or number. Previous code was requiring both name
+ and number to be available. Also restored a comment block on why
+ caller id is also set on an outgoing call leg in addition to
+ connected line from earlier versions of Asterisk.
+
+2013-10-02 19:19 +0000 [r400291] Kinsey Moore <kmoore@digium.com>
+
+ * rest-api/api-docs/asterisk.json: Correct allowable values for ARI
+ general information filter
+
+2013-10-02 18:57 +0000 [r400286] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Fix the CDR CLI command 'cdr show active {channel}'
+ When the switch from channel names to channel unique IDs
+ happened, the poor CLI command got left in the dust. This fixes
+ the command so that users can once again see how Asterisk is
+ messing up your billing information.
+
+2013-10-02 18:42 +0000 [r400284] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_t38.c: Fix a crash in res_pjsip_t38 caused by the
+ wrong assumption that a session will always have a channel. When
+ starting up or shutting down this assumption is false.
+
+2013-10-02 18:25 +0000 [r400281] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * Makefile, doc/astdb2sqlite3.8 (added), /, doc/astdb2bdb.8
+ (added): man pages for astdb2bdb and astdb2sqlite3 Review:
+ https://reviewboard.asterisk.org/r/2898/ ........ Merged
+ revisions 400279 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-10-02 17:11 +0000 [r400268-400270] Richard Mudgett <rmudgett@digium.com>
+
+ * main/json.c, main/stasis_cache.c, res/res_ari.c, main/utils.c,
+ apps/app_stack.c, res/stasis_recording/stored.c: MALLOC_DEBUG:
+ Fix some misuses of free() when MALLOC_DEBUG is enabled. * There
+ were several places in ARI where an external library was
+ mallocing memory that must always be released with free(). When
+ MALLOC_DEBUG is enabled, free() is redirected to the MALLOC_DEBUG
+ version. Since the external library call still uses the normal
+ malloc(), MALLOC_DEBUG complains that the freed memory block is
+ not registered and will not free it. These cases must use
+ ast_std_free(). * Changed calls to asprintf() and vasprintf() to
+ the equivalent ast_asprintf() and ast_vasprintf() versions
+ respectively.
+
+ * channels/sig_ss7.c: sig_ss7: Fix compiler warnings.
+
+2013-10-02 16:20 +0000 [r400245-400265] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/channel.h, channels/chan_gtalk.c,
+ channels/chan_console.c, channels/sig_pri.c,
+ channels/chan_iax2.c, channels/chan_jingle.c, main/channel.c,
+ main/dial.c, channels/chan_dahdi.c,
+ include/asterisk/stasis_channels.h, channels/chan_skinny.c,
+ channels/chan_motif.c, channels/chan_alsa.c,
+ main/stasis_channels.c, channels/chan_pjsip.c,
+ channels/sig_ss7.c, channels/chan_mgcp.c,
+ channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
+ channels/chan_sip.c, main/bridge.c: Reduce channel snapshot
+ creation and publishing by up to 50%. This change introduces the
+ ability to stage channel snapshot creation and publishing by
+ suppressing the implicit creation and publishing that some
+ functions have. Once all operations are executed the staging is
+ marked as done and a single snapshot is created and published.
+ Review: https://reviewboard.asterisk.org/r/2889/
+
+ * res/res_pjsip_session.c: Fix a random one way audio issue in
+ PJSIP. Due to the asynchronous design of the PJMEDIA SDP
+ negotiator it was possible for the SDP to be negotiated *after* a
+ channel was created and after it was being wait on by an
+ application. It is only after negotiation occurs that the file
+ descriptors for RTP are placed on the channel. Since the channel
+ was already being waited on these file descriptors were not
+ monitored, causing incoming media to never be read. This change
+ wakes up any application waiting on the channel so that added
+ file descriptors end up being monitored. (closes issue AST-1227)
+ Reported by: John Bigelow
+
+ * include/asterisk/stasis_app.h, res/ari/resource_channels.c,
+ res/stasis/control.c: Allow specifying a channel to dial an
+ extension and context in an ARI dial operation. (issue
+ ASTERISK-22625) Reported by: Scott Griepentrog
+
+ * res/res_pjsip_session.c: Retrieve and store the hostname only
+ once so multiple threads do not potentially initialize it at the
+ same time.
+
+2013-10-01 21:17 +0000 [r400227-400236] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: chan_dahdi: Fix
+ analog parking using flash-hook. Transferring an analog call
+ using a flash-hook to parking would fail to park the call and
+ result in an invalid ao2 object unref. * Park the correct bridged
+ channel.
+
+ * main/features_config.c: Features: Rearm the parking config
+ options have moved warning for each reload.
+
+2013-10-01 15:48 +0000 [r400217] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Filter out internal channels for bridge leave
+ messages and parked call messages Granted, if you manage to park
+ a Conference announcer channel, something has gone horrifically
+ wrong.
+
+2013-09-30 21:31 +0000 [r400205] Jonathan Rose <jrose@digium.com>
+
+ * configs/res_parking.conf.sample, configs/features.conf.sample:
+ configuration samples: Pull all parking related stuff out of
+ features.conf This patch also adds documentation for parking from
+ features.conf to res_parking.conf
+
+2013-09-30 19:57 +0000 [r400194-400196] Matthew Jordan <mjordan@digium.com>
+
+ * funcs/func_cdr.c: Parse arguments passed to the CDR_PROP function
+ correctly I can only blame this on a bad merge, because this in
+ no way worked properly the way it was written. Mea culpa. The
+ function should now parse its arguments correctly and function
+ properly. (Note that the API used by the CDR_PROP function has
+ working unit tests... this was merely bad coding of the actual
+ registered function) (closes issue ASTERISK-22613) Reported by:
+ Private Name
+
+ * main/cdr.c: Remove spurious event raised when CDRs are reloaded
+ The Reload event is now raised by the module loading core. As
+ such, the Reload event in the CDR engine was a duplicate and not
+ needed.
+
+2013-09-30 18:48 +0000 [r400178-400181] David M. Lee <dlee@digium.com>
+
+ * res/res_chan_stats.c, main/stasis.c, main/cdr.c,
+ main/manager_bridges.c, channels/chan_dahdi.c, main/manager.c,
+ channels/chan_skinny.c, tests/test_devicestate.c, res/res_agi.c,
+ tests/test_taskprocessor.c, res/res_stasis_test.c,
+ main/manager_channels.c, channels/chan_mgcp.c,
+ res/res_pjsip_refer.c, res/res_security_log.c,
+ main/stasis_cache.c, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/taskprocessor.h, include/asterisk/stasis.h,
+ res/parking/parking_applications.c, main/sounds_index.c,
+ channels/sig_pri.c, apps/app_queue.c, main/cel.c,
+ res/parking/parking_bridge_features.c,
+ main/stasis_message_router.c, funcs/func_presencestate.c,
+ apps/confbridge/confbridge_manager.c, res/res_pjsip_mwi.c,
+ tests/test_stasis.c, res/parking/parking_manager.c,
+ main/manager_mwi.c, apps/app_voicemail.c, main/stasis_wait.c,
+ res/stasis/app.c, main/ccss.c, apps/app_meetme.c,
+ include/asterisk/stasis_internal.h, main/manager_endpoints.c,
+ main/devicestate.c, res/res_xmpp.c, main/taskprocessor.c,
+ main/endpoints.c, channels/chan_iax2.c, res/res_jabber.c: Remove
+ dispatch object allocation from Stasis publishing While looking
+ for areas for performance improvement, I realized that an unused
+ feature in Stasis was negatively impacting performance. When a
+ message is sent to a subscriber, a dispatch object is allocated
+ for the dispatch, containing the topic the message was published
+ to, the subscriber the message is being sent to, and the message
+ itself. The topic is actually unused by any subscriber in
+ Asterisk today. And the subscriber is associated with the
+ taskprocessor the message is being dispatched to. First, this
+ patch removes the unused topic parameter from Stasis subscription
+ callbacks. Second, this patch introduces the concept of
+ taskprocessor local data, data that may be set on a taskprocessor
+ and provided along with the data pointer when a task is pushed
+ using the ast_taskprocessor_push_local() call. This allows the
+ task to have both data specific to that taskprocessor, in
+ addition to data specific to that invocation. With those two
+ changes, the dispatch object can be removed completely, and the
+ message is simply refcounted and sent directly to the
+ taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/
+
+ * include/asterisk/vector.h (added), res/stasis/app.c,
+ main/channel_internal_api.c, include/asterisk/stasis.h,
+ apps/app_queue.c, main/cel.c, main/stasis.c,
+ tests/test_stasis_endpoints.c, main/cdr.c,
+ main/manager_bridges.c, main/manager.c, main/manager_system.c,
+ tests/test_stasis.c, main/manager_channels.c, main/manager_mwi.c,
+ main/stasis_cache_pattern.c: Optimize how Stasis forwards are
+ dispatched This patch optimizes how forwards are dispatched in
+ Stasis. Originally, forwards were dispatched as subscriptions
+ that are invoked on the publishing thread. This did not account
+ for the vast number of forwards we would end up having in the
+ system, and the amount of work it would take to walk though the
+ forward subscriptions. This patch modifies Stasis so that rather
+ than walking the tree of forwards on every dispatch, when
+ forwards and subscriptions are changed, the subscriber list for
+ every topic in the tree is changed. This has a couple of
+ benefits. First, this reduces the workload of dispatching
+ messages. It also reduces contention when dispatching to
+ different topics that happen to forward to the same aggregation
+ topic (as happens with all of the channel, bridge and endpoint
+ topics). Since forwards are no longer subscriptions, the bulk of
+ this patch is simply changing stasis_subscription objects to
+ stasis_forward objects (which, admittedly, I should have done in
+ the first place.) Since this required me to yet again put in a
+ growing array, I finally abstracted that out into a set of
+ ast_vector macros in asterisk/vector.h. Review:
+ https://reviewboard.asterisk.org/r/2883/
+
+ * configs/stasis.conf.sample (removed), include/asterisk/sem.h
+ (added), configure.ac, include/asterisk/stasis.h,
+ main/taskprocessor.c, main/sem.c (added), main/stasis.c,
+ main/stasis_config.c (removed), include/asterisk/taskprocessor.h,
+ configure, include/asterisk/autoconfig.h.in: Taskprocessor
+ optimization; switch Stasis to use taskprocessors This patch
+ optimizes taskprocessor to use a semaphore for signaling, which
+ the OS can do a better job at managing contention and waiting
+ that we can with a mutex and condition. The taskprocessor
+ execution was also slightly optimized to reduce the number of
+ locks taken. The only observable difference in the taskprocessor
+ implementation is that when the final reference to the
+ taskprocessor goes away, it will execute all tasks to completion
+ instead of discarding the unexecuted tasks. For systems where
+ unnamed semaphores are not supported, a really simple semaphore
+ implementation is provided. (Which gives identical performance as
+ the original taskprocessor implementation). The way we ended up
+ implementing Stasis caused the threadpool to be a burden instead
+ of a boost to performance. This was switched to just use
+ taskprocessors directly for subscriptions. Review:
+ https://reviewboard.asterisk.org/r/2881/
+
+2013-09-30 15:55 +0000 [r400141] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c, configs/pjsip.conf.sample,
+ res/res_pjsip_outbound_registration.c, configs/sip.conf.sample,
+ CHANGES: Allow Asterisk to retry after 403 on register This adds
+ a global option in chan_sip to allow it to continue attempting
+ registration if a 403 is received, clearing the cached nonce and
+ treating it as a non-fatal response. Normally, this would cause
+ registration attempts to that endpoint to stop. This also adds a
+ similar per-outbound-registration option to chan_pjsip which
+ allows the retry interval to be altered for 403 responses to
+ REGISTER requests. (closes issue ASTERISK-17138) Review:
+ https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi
+ ........ Merged revisions 400137 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400140 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-30 15:24 +0000 [r400138] David M. Lee <dlee@digium.com>
+
+ * main/astobj2.c, main/stasis.c, main/stasis_message_router.c,
+ main/taskprocessor.c, include/asterisk/stasis_message_router.h,
+ res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c:
+ Stasis performance improvements This patch addresses several
+ performance problems that were found in the initial performance
+ testing of Asterisk 12. The Stasis dispatch object was allocated
+ as an AO2 object, even though it has a very confined lifecycle.
+ This was replaced with a straight ast_malloc(). The Stasis
+ message router was spending an inordinate amount of time
+ searching hash tables. In this case, most of our routers had 6 or
+ fewer routes in them to begin with. This was replaced with an
+ array that's searched linearly for the route. We more heavily
+ rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref()
+ actually became noticeable on the profile. This was #ifdef'ed to
+ only run when AO2_DEBUG was enabled. After being misled by an
+ erroneous comment in taskprocessor.c during profiling, the wrong
+ comment was removed. Review:
+ https://reviewboard.asterisk.org/r/2873/
+
+2013-09-28 22:56 +0000 [r400058-400121] Matthew Jordan <mjordan@digium.com>
+
+ * configs/pjsip_notify.conf.sample (added), res/res_pjsip_notify.c:
+ res_pjsip_notify: Add documentation We forgot to add
+ documentation for res_pjsip_notify, which would prevent it from
+ being loaded. Whoops. This patch also updates res_pjsip_notify to
+ use pjsip_notify.conf, which now has its own sample file in the
+ configs directory as well. Review:
+ https://reviewboard.asterisk.org/r/2835/
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Correct erroneous
+ lost packet information in RTCP reports RTCP's calculation of the
+ number of lost packets in an RTP stream is based on that stream's
+ sequence number count, the number of received packets, and how
+ many packets we expect to receive. When the SSRC for an RTP
+ stream changes, there can - and almost always will be - a large
+ jump in the next packet's timestamp and sequence number. If we
+ don't reset the number of received packets, sequence number
+ count, and other metrics used by RTCP, the next RR/SR report will
+ use the previous SSRC's values to calculate the lost packet count
+ for the new SSRC - resulting in a very large number of lost
+ packets. This patch modifies res_rtp_asterisk such that, if it
+ detects a SSRC change, it will reset the various values used by
+ the RTCP calculations. From the perspective of RTCP, this appears
+ as a new media stream - which is what it is. Review:
+ https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
+ Reported by: Thomas Arimont ........ Merged revisions 400089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400093 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, configure, configure.ac: Add check for openSUSE when detecting
+ bfd library In ASTERISK-17842, some additional library checks
+ were added to the configure script so that the bfd library could
+ be found on CentOS and Fedora systems. As it turns out, openSUSE
+ requires an additional library. This patch adds another check to
+ the configure script for openSUSE that will add that library.
+ Review: https://reviewboard.asterisk.org/r/2885/ (closes issue
+ AST-1169) Reported by: Guenther Kelleter ........ Merged
+ revisions 400073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400075 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cdr.c: CDR: Improve handling of parking; resolve assertion
+ when originating into park This patch covers two problems: 1)
+ Currently, when a call is transferred into a parking lot from a
+ bridge (using either the blind transfer or one touch parking
+ mechanisms), the application fails to be set to "Park" in the
+ resulting CDR record for the parked channel. This is due to the
+ ParkedCall message arriving before the BridgeEnter for the
+ channel entering the parking bridge. The ParkedCall message isn't
+ handled as the CDR for the channel has already been finalized
+ (due to the channel having left its two party bridge), and the
+ BridgeEnter - which creates the new CDR - doesn't have the
+ parking information. This patch modifies the behavior so that
+ reception of a ParkedCall message will - if not handled by a CDR
+ chain - cause a new CDR to be created and put into the Parking
+ state. 2) It fixes a FRACK that occurred when a channel is
+ originated into a parking space. The DialedPending state - which
+ occurs for both Dialed and Originated channels - assumed that it
+ couldn't handle the parking transitions due to it having a Party
+ B; however, Originated channels don't have a Party B. As such,
+ the existing CDR needs to transition into the parking state -
+ this patch does that. Review:
+ https://reviewboard.asterisk.org/r/2877/ (closes issue
+ ASTERISK-22482) Reported by: Richard Mudgett
+
+ * apps/app_queue.c: app_queue: Make manager events tolerant of
+ Local channel shenanigans app_queue currently attempts to handle
+ Local channel optimizations in an effort to provide accurate
+ information in Stasis messages (and their corresponding AMI
+ events) as well as the Queue log. Sometimes, however, things
+ don't go as planned. Consider the following scenario: SIP/foo <->
+ L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local
+ channel optimization. app_queue will normally do the following: *
+ Listen for the Local optimization events and update our agent
+ accordingly to SIP/agent in the queue log and messages * When we
+ get a hangup, publish the AgentComplete event based on our
+ information (SIP/foo and SIP/agent) However, as with all things
+ that depend on sanity from something as capricious as Local
+ channels, things can go wrong: (1) SIP/agent immediately hangs up
+ upon answering. This triggers a race condition between
+ termination messages coming from SIP/agent and the ongoing Local
+ channel optimization messages. (Note that this can also occur
+ with SIP/foo) (2) In a race condition, Asterisk can (rarely)
+ deliver the hangup messages prior to the Local channel
+ optimization. In that case, the messages *may* arrive to
+ app_queue in the following order: * Hangup SIP/Agent * Hangup
+ SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When
+ app_queue receives the hangup of the agent or the caller, it will
+ attempt to publish the AgentComplete event. However, it now has a
+ problem - it thinks its agent is the ;1 side of the Local
+ channel, as it never received the optimization event. At the same
+ time, that channel is already gone. This results in getting NULL
+ from the Stasis cache. What's more, we can't really wait for the
+ optimization message, as we are currently handling the hangup of
+ the channel that the optimization event would tell us to use.
+ This patch modifies the behavior in app_queue such that, since we
+ still have a lot of pertinent queue information (interface, queue
+ name, etc.), we now raise the event with what information we
+ know. The channels involved now may or may not be present. Users
+ will still at least get the "AgentComplete" event, which
+ "completes" the known Agent information. Review:
+ https://reviewboard.asterisk.org/r/2878/ (closes issue
+ ASTERISK-22507) Reported by: Richard Mudgett
+
+ * main/manager.c: manager: Fix crash when appending a manager
+ channel variable In r399887, a minor performance improvement was
+ introduced by not allocating the manager variable struct if it
+ wasn't used. Unfortunately, when directly accessing an
+ ast_channel struct, manager assumed that the struct was always
+ allocated. Since this was no longer the case, things got a bit
+ crashy. This fixes that problem by simply bypassing appending
+ variables if the manager channel variable struct isn't there.
+
+2013-09-27 21:56 +0000 [r400015-400020] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_cdr.c, res/res_parking.c: app_cdr and res_parking: Fix
+ some resource leaks. * app_cdr left the ResetCDR application
+ registered. * res_parking leaked a ref to config global. (closes
+ issue ASTERISK-22566) Reported by: Corey Farrell Patches:
+ ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey
+ Farrell
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: chan_sip:
+ Increase some scratch buffer sizes dealing with caller id. *
+ Eliminated an unnecessary initialization in check_user_full().
+ (closes issue ASTERISK-22477) Reported by: Michael Shepelev
+ ........ Merged revisions 400013 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 400014 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-27 18:26 +0000 [r399990] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_session.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip.exports.in:
+ res_pjsip: crash when using localnet and
+ external_signaling_address options There was a collision of
+ mod_data use on the transaction between using a nat hook and an
+ session response callback. During state change it was assumed
+ what was in the mod_data was nothing or the response callback.
+ However, it was possible for it to also contain a nat hook thus
+ resulting in a bad cast and a crash. Added the ability to store
+ multiple data elements in mod_data via a hash table. In this
+ instance, mod_data now stores a hash table of the two values that
+ can be retrieved using an associated string key. (closes issue
+ ASTERISK-22394) Reported by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/2843/
+
+2013-09-27 17:34 +0000 [r399976] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Reject calls on 200 OKs if no SDP has been received When Asterisk
+ receives a 200 OK in response to an invite, that peer should have
+ sent an SDP at some point by then. If the channel has never
+ received an SDP, media won't have been set and the remote address
+ won't be known. Endpoints in general should not be doing this.
+ This patch makes it so that Asterisk will simply hang up a call
+ if it sends a 200 OK at this point. So far this odd behavior for
+ endpoints has only been observed in tests which involved manually
+ created SIP transactions in SIPp. (closes issue ASTERISK-22424)
+ Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/ ........ Merged
+ revisions 399939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399962 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-27 17:03 +0000 [r399937] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_astobj2.c, main/astobj2.c, include/asterisk/astobj2.h:
+ astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a strange
+ feature that came into the world under suspicious circumstances
+ to support an abuse of the ao2_container by chan_iax2. Since
+ chan_iax2 no longer uses OBJ_CONTINUE, it is safe to remove it.
+ The simplified code should help performance slightly and make
+ understanding the code easier. Review:
+ https://reviewboard.asterisk.org/r/2887/
+
+2013-09-27 14:29 +0000 [r399924] Mark Michelson <mmichelson@digium.com>
+
+ * bridges/bridge_native_rtp.c: Fix refleaks of ast_rtp_instance
+ structures. These refleaks were causing bridged calls not to
+ close their RTP ports. Thus a call would leave open 4 ports (RTP
+ for party A, RTCP for party A, RTP for party B, and RTCP for
+ party B). This led to an eventual depletion of available RTP
+ ports.
+
+2013-09-27 14:01 +0000 [r399912] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/cel.h, tests/test_cel.c, main/cel.c: Restore
+ usefulness of the CEL Peer field This change makes the CEL peer
+ field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and
+ fills the field with a comma-separated list of all channels in
+ the bridge other than the channel that is entering or exiting the
+ bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes
+ issue ASTERISK-22393)
+
+2013-09-26 18:48 +0000 [r399897] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip/security_events.c, res/res_pjsip_registrar.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip.exports.in: pjsip:
+ race condition in registrar While handling a registration request
+ a race condition could occur if/when two+ clients registered at
+ the same time. This happened when one request obtained a copy of
+ the current contacts for an AOR and another request did the same
+ before the first request updated. Thus the second would update
+ and overwrite the first (or vice-versa depending on which
+ actually updated first). In the case of it being the same contact
+ two "add" events would be raised. pjsip registration handling is
+ now serialized to alleviate this issue. (closes issue AST-1213)
+ Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2860/
+
+2013-09-26 15:41 +0000 [r399887] David M. Lee <dlee@digium.com>
+
+ * main/channel.c: Minor performance bump by not allocate manager
+ variable struct if we don't need it
+
+2013-09-26 14:12 +0000 [r399874] Rusty Newton <rnewton@digium.com>
+
+ * apps/app_dial.c: Adding a few words to the Dial option 'r' help
+ text to clarify its tone argument description
+
+2013-09-25 20:36 +0000 [r399842] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, /: chan_dahdi: CLI
+ "core stop gracefully" has needless delay for PRI and SS7. The
+ PRI and SS7 link control threads are not stopped correctly when
+ the chan_dahdi.so module is unloaded. The link control threads
+ pri_dchannel() and ss7_linkset() are not awakened from a poll()
+ to cancel the thread. * Added a SIGURG signal after requesting
+ the thread cancel to break the link control thread poll()
+ immediately. For SS7 it was slightly worse, the link poll()
+ timeout would always be whatever was the last libss7 scheduled
+ event time used. If no libss7 scheduled event was pending, the
+ thread could run more often than necessary. * Set nextms to 60
+ seconds for the ss7_linkset() poll() if there is no other libss7
+ scheduled event. ........ Merged revisions 399818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399834 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-25 19:40 +0000 [r399798] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip.c: Broke the build - Fixing XML DTD violation added
+ in r399782, missing <para> tags inside a <note>
+
+2013-09-25 19:28 +0000 [r399796] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Realtime Peer Update Problem When
+ Un-registering And Expires Header In 200ok 1st Issue When a
+ realtime peer sends an un-REGISTER request, Asterisk un-registers
+ the peer but the database table record still has regseconds and
+ fullcontact for the peer. This results in calls attempting to be
+ routed to the peer which is no longer registered. The expected
+ behavior is to get busy/congested when attempting to call an
+ un-registered peer through the dialplan. What was discovered is
+ that we are clearing out the peer's registration in the database
+ in parse_register_contact() when calling expire_register() but
+ then upon returning from parse_register_contact(), update_peer()
+ is run which stores back in the database table regseconds and
+ fullcontact. 2nd Issue The reporter pointed out that the 200 ok
+ being returned by Asterisk after un-registering a peer contains a
+ Contact header with ;expires= and the Expires header is not set
+ to 0. This is actually a regression. Tests were created for this
+ second issue (ASTERISK-22548). The tests have been reviewed and a
+ Ship It! was received on those tests. This patch does the
+ following: * Do not ignore the Expires header value even when it
+ is set to 0. The patch sets the pvt->expiry earlier on in the
+ function so that it is set properly and used. * If pvt->expiry is
+ 0, do not call update_peer since that means the peer has already
+ been un-registered and there is no need to update the database
+ record again since nothing has changed. (closes issue
+ ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben
+ Smithurst, Michael L. Young Patches:
+ asterisk-22428-rt-peer-update-and-expires-header.diff by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2869/ ........ Merged
+ revisions 399794 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399795 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-25 18:36 +0000 [r399781] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip.c: Fixing documentation for the configOption
+ "external_media_address" of both Endpoints and Transports
+ Re-using some of Mark Michelson's text from an E-mail discussion
+ for: * Modifying synopsis for both options * Adding description
+ to both options * Changing name of "external_media_address" for
+ Endpoint configuration to "media_address" in anticipation of the
+ option name being changed. (As it is not really specific to
+ external destinations) (issue ASTERISK-22405) (closes issue
+ ASTERISK-22405) Reported by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/2850/
+
+2013-09-24 22:50 +0000 [r399736-399749] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: astobj2: Made use OBJ_SEARCH_xxx identifiers as
+ field enum values internally. * Made ao2_unlink to protect itself
+ from stray OBJ_SEARCH_xxx values passed in.
+
+ * /, channels/chan_iax2.c: chan_iax2: Prevent some needless
+ breaking of the native IAX2 bridge. * Clean up some twisted code
+ in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and
+ AST_CONTROL_SRCCHANGE to a list of frames to prevent the native
+ bridge loop from breaking. * Passing the
+ AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a
+ native IAX2 bridge. (issue ABE-2912) Review:
+ https://reviewboard.asterisk.org/r/2870/ ........ Merged
+ revisions 399697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399708 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and
+ above this is really just documentation until IAX2 native
+ bridging is restored.
+
+2013-09-24 19:22 +0000 [r399666-399695] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c: app_queue: Don't be quite so aggressive in
+ initializing the array We only need the first character.
+
+ * apps/app_queue.c: app_queue: Initialize array holding MixMonitor
+ exec options If the channel variable MONITOR_EXEC is set,
+ app_queue will pass the specified execution parameters to the
+ MixMonitor application when a queue is recorded. If that channel
+ variable is not set, the buffer that holds the escaped value was
+ not being initialized to NULL, and so would be passed to the
+ MixMonitor application with garbage. Hilarity ensued as
+ app_mixmonitor attempted to execute gobeldy-gook.
+
+ * main/stasis_bridges.c, tests/test_cdr.c, main/cdr.c: Fix a
+ performance problem CDRs There is a large performance price
+ currently in the CDR engine. We currently perform two
+ ao2_callback calls on a container that has an entry for every
+ channel in the system. This is done to create matching pairs
+ between channels in a bridge. As such, the portion of the CDR
+ logic that this patch deals with is how we make pairings when a
+ channel enters a mixing bridge. In general, when a channel enters
+ such a bridge, we need to do two things: (1) Figure out if anyone
+ in the bridge can be this channel's Party B. (2) Make pairings
+ with every other channel in the bridge that is not already our
+ Party B. This is a two step process. In the first step, we look
+ through everyone in the bridge and see if they can be our Party B
+ (single_state_process_bridge_enter). If they can - yay! We mark
+ our CDR as having gotten a Party B. If not, we keep searching. If
+ we don't find one, we wait until someone joins who can be our
+ Party B. Step 2 is where we changed the logic
+ (handle_bridge_pairings and bridge_candidate_process).
+ Previously, we would first find candidates - those channels in
+ the bridge with us - from the active_cdrs_by_channel container.
+ Because a channel could be a candidate if it was Party B to an
+ item in the container, the code implemented multiple
+ ao2_container callbacks to get all the candidates. We also had to
+ store them in another container with some other meta information.
+ This was rather complex and costly, particularly if you have 300
+ Local channels (600 channels!) going at once. Luckily, none of it
+ is needed: when a channel enters a bridge (which is when we're
+ figuring all this stuff out), the bridge snapshot tells us the
+ unique IDs of everyone already in the bridge. All we need to do
+ is: For all channels in the bridge: If the channel is us or our
+ Party B that we got in step 1, skip it Compare us and the
+ candidate to figure out who is Party A (based on some specific
+ rules) If we are Party A: Make a new CDR for us, append it to our
+ chain, and set the candidate as Party B If they are Party A: If
+ they don't have a Party B: Make a new CDR for them, append us to
+ their chain, and us as Party B Otherwise: Copy us over as Party B
+ on their existing CDR. This patch does that. Because we now use
+ channel unique IDs to find the candidates during bridging,
+ active_cdrs_by_channel now looks up things using uniqueid instead
+ of channel name. This makes the more complex code simpler; it
+ does, however, have the drawback that dialplan applications and
+ functions will be slightly slower as they have to iterate through
+ the container looking for the CDR by name. That's a small price
+ to pay however as the bridging code will be called a lot more
+ often. This patch also does two other minor changes: (1) It
+ reduces the container size of the channels in a bridge snapshot
+ to 1. In order to be predictable for multi-party bridges, the
+ order of the channels in the container must be stable; that is,
+ it must always devolve to a linked list. (2) CDRs and the
+ multi-party test was updated to show the relationship between two
+ dialed channels. You still want to know if they talked -
+ previously, dialed channels were always ignored, which is wrong
+ when they have managed to get a Party B. (closes issue
+ ASTERISK-22488) Reported by: Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/2861/
+
+2013-09-23 12:02 +0000 [r399624] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_session.c: Fix crash in res_pjsip
+ on load if error occurs, and prevent unloading of res_pjsip and
+ res_pjsip_session. During load time in res_pjsip if an error
+ occurred the operation would attempt to rollback all operations
+ done during load. This is not permitted by PJSIP as it will
+ assert if the operation has not been done. This fix changes the
+ code so it will only rollback what has been initialized already.
+ Further changes also prevent res_pjsip and res_pjsip_session from
+ being unloaded. This is due to limitations within PJSIP itself.
+ The library environment can only be changed to a certain extent
+ and does not provide the ability, currently, to deinitialize
+ certain required functionality. (closes issue ASTERISK-22474)
+ Reported by: Corey Farrell
+
+2013-09-21 04:48 +0000 [r399576-399607] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix ref leaks in
+ ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the
+ loop so it is unref'ed after every loop. Moved message_blob to
+ loop and switched it to a regular variable. The regular variable
+ was used since message_blob is used in a very contained way.
+ (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches:
+ rtcp_report-leak.patch (license #5909) patch uploaded by Corey
+ Farrell Tested by: Corey Farrell
+
+ * main/media_index.c: media_index: Fix process_description_file()
+ memory leak of file_id_persist.
+
+ * main/features_config.c: features_config: Fix config ref leak of
+ parkinglots. This leak happend for just about every channel
+ created.
+
+ * apps/app_queue.c: app_queue: Fix json blob ref leak. The json ref
+ from queue_member_blob_create() was never released.
+
+ * main/json.c: json: Make it obvious that ast_json_unref() is NULL
+ safe. It looked like the safety check was done after the NULL
+ pointer was used.
+
+2013-09-20 22:41 +0000 [r399565] Kinsey Moore <kmoore@digium.com>
+
+ * main/config_options.c, /: Ensure global types in the config
+ framework are initialized If a config object was allocated but
+ one of its global objects was never encountered, then the global
+ object's defaults were never applied. Ensure that global objects
+ are initialized properly upon allocation instead of on
+ configuration. Review: https://reviewboard.asterisk.org/r/2866/
+ ........ Merged revisions 399564 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-20 22:04 +0000 [r399553] Jonathan Rose <jrose@digium.com>
+
+ * main/dial.c: originate/call forwarding: Fix a crash when
+ forwarding a call from originate (closes issue ASTERISK-22487)
+ Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2868/
+
+2013-09-20 16:17 +0000 [r399531] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_pjsip.c: Add a missing session supplement
+ unregistration in chan_pjsip for ACKs. (closes issue
+ ASTERISK-22453) Reported by: Corey Farrell Patches:
+ chan_pjsip_session_unregister_supplement.patch uploaded by Corey
+ Farrell (license 5909)
+
+2013-09-20 14:25 +0000 [r399514] Kevin Harwell <kharwell@digium.com>
+
+ * /, main/logger.c: Memory leak in logger. Fixed a memory leak
+ discovered in the logger where a temporary string buffer was not
+ being freed. (closes issue ASTERISK-22540) Reported by: John
+ Hardin ........ Merged revisions 399513 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-19 23:16 +0000 [r399501] Richard Mudgett <rmudgett@digium.com>
+
+ * main/optional_api.c: optional_api: Make always use the standard
+ malloc functions even with MALLOC_DEBUG.
+
+2013-09-19 16:53 +0000 [r399458] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Make direct media reinvites for
+ T38 put Asterisk in the media path Prior to this patch, Asterisk
+ would incorrectly use the previous endpoint addresses in SDP in
+ spite of providing its own port. T38 is never meant to be done
+ through directmedia and Asterisk should always be in the media
+ path for these streams. (closes issue ASTERISK-17273) Reported
+ by: Kevin Stewart (closes issue ASTERISK-18706) Reported by:
+ Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/
+ ........ Merged revisions 399456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-18 19:59 +0000 [r399404] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/abstract_jb.c: Fix jitter buffer log file creation This
+ adjusts '/'-to-'#' replacement to replace all instances of '/'
+ instead of just the first to ensure that the jitter buffer log
+ file gets the correct name as per Richard Kenner's suggestion.
+ (closes issue ASTERISK-21036) Reported by: Richard Kenner
+ ........ Merged revisions 399402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399403 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-18 17:23 +0000 [r399365-399376] Matthew Jordan <mjordan@digium.com>
+
+ * /, build_tools/prep_tarball: Update prep_tarball with new
+ documentation files on the Asterisk wiki This will now pull both
+ a command reference for the version being prepared, as well as an
+ Admin Guide that applies to all versions of Asterisk. (issue
+ ASTERISK-22439) Reported by: Olle Johansson ........ Merged
+ revisions 399351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399373 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when
+ a timing module isn't loaded If bridge_softmix fails to be
+ created because no timing source is present in Asterisk, this
+ will currently fail gracefully but with (most likely) a generic
+ error message by whatever module tried to create the softmix
+ bridge. This patch adds a more explicit warning so you can
+ actually diagnose and fix the problem. Review:
+ https://reviewboard.asterisk.org/r/2857/ ........ Merged
+ revisions 399353 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-18 14:34 +0000 [r399339] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip_messaging.c: res_pjsip_messaging: Register message
+ technology as pjsip pjsip's message technology was being
+ registered as 'sip', which was causing it to not load due it
+ conflicting with chan_sip's registered 'sip' technology for
+ messaging. It now registers as 'pjsip'. However, due to this
+ change the "to" field for outgoing pjsip messages need to be
+ prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to
+ res_pjsip_messaging will automatically have their "to" fields
+ altered in order to accommodate the change. Outgoing messages
+ also handle changing it back to 'sip' before being sent so the
+ pjsip library will properly handle it. (closes issue
+ ASTERISK-22445) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2833/
+
+2013-09-18 00:12 +0000 [r399294] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/features_config.c: Fix Segfault In features-config.c When
+ Application Has No Arguments Some applications do not require
+ arguments. Therefore, when parsing application maps in
+ features.conf, it is possible that app_data will be set to NULL.
+ * This patch sets app_data to "" if it is NULL. Review:
+ https://reviewboard.asterisk.org/r/2804
+
+2013-09-17 23:08 +0000 [r399283] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_t38.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip_sdp_rtp.c: Change the
+ "external_media_address" PJSIP endpoint option to
+ "media_address". The endpoint option does not apply to
+ communication with external entities. Rather, the option is
+ applied to all communications with the endpoint. The
+ external_media_address transport configuration option may
+ override the endpoint option if it turns out that we are going to
+ be communicating with an external entity. Two things of note: 1)
+ I have not updated the XML documentation. This is being taken
+ care of by Rusty as part of his work on issue ASTERISK-22405 2)
+ This commit is likely to cause testsuite failures since there are
+ tests that use the external_media_address endpoint option, and
+ they will need to be changed over. Well, I'm planning to get that
+ updated ASAP after this commit. (closes issue ASTERISK-22528)
+ reported by Rusty Newton
+
+2013-09-17 18:37 +0000 [r399268] Kevin Harwell <kharwell@digium.com>
+
+ * main/asterisk.c, /, main/logger.c: Remote console: more output
+ discrepancies The remote console continued to have issues with
+ its output. In this case CLI command output would either not show
+ up (if verbose level = 0) or would contain verbose prefixes (if
+ verbose level > 0) once log messages were sent to the remote
+ console. The fix now now adds verbose prefix data to all new
+ lines contained in a verbose log string. (closes issue
+ ASTERISK-22450) Reported by: David Brillert (closes issue
+ AST-1193) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2825/ ........ Merged
+ revisions 399267 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-17 17:54 +0000 [r399257] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features_config.h: Fix doxygen to use correct
+ units of features.conf options.
+
+2013-09-17 17:09 +0000 [r399237-399247] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridge_basic.c, main/features_config.c: Fix other timeouts
+ (atxferloopdelay and atxfernoanswertimeout) to use seconds
+ instead of milliseconds. Thanks to Richard Mudgett for pointing
+ this out.
+
+ * include/asterisk/features_config.h, main/bridge_basic.c,
+ main/features_config.c: Switch transferdigittimeout to be
+ configured as seconds instead of milliseconds. This was an
+ unintentional consequence of the update of features.conf to use
+ the config framework in Asterisk 12. Thanks to Marco Signorini on
+ the Asterisk developers list for pointing out the problem.
+
+2013-09-17 14:48 +0000 [r399225] Kevin Harwell <kharwell@digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c, /: Confbridge: empty
+ conference not being torn down Confbridge would not properly tear
+ down an empty conference bridge when all users were kicked via
+ end_marked=yes and at least one user was also set to wait_marked.
+ This occurred because while end_marked users were being kicked
+ and at least one was also set to wait_marked then the leave
+ wait_marked handler would be called on that user, but there would
+ be no waiting user (still considered active). The waiting users
+ would decrement and now be negative. The conference would remain,
+ but be put into an inactive state. The solution was to move from
+ the active list to the wait list, those users with wait_marked
+ set right before kicking. This allows both the active and wait
+ users to decrement correctly and the confbridge to tear down
+ properly. A crashed also occurred when trying to list the
+ specific conference from the CLI. This happened because the
+ conference specified was invalid. Since the conference properly
+ tears down now there is no way to reference it thus alleviating
+ the crash as well. (closes issue ASTERISK-21859) Reported by:
+ Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/
+ ........ Merged revisions 399222 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-16 18:34 +0000 [r399160-399207] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_ari_model.c: Fix module load errors for
+ test_ari_model.so. You cannot use a function pointer variable
+ with an external function from another dynamically loaded module
+ because data variables are always resolved even with RTLD_LAZY. *
+ Added wrapper functions for ast_ari_validate_int() and
+ ast_ari_validate_string() to use instead for the function pointer
+ variable. (closes issue ASTERISK-22457) Reported by: David M. Lee
+
+ * apps/app_speech_utils.c, res/res_speech.exports.in:
+ app_speech_utils: Fix unresolved symbol ast_speech_get_setting().
+ Fixes regression introduced by -r374096. * Made
+ res_speech.export.in export ast_* symbols instead of specific
+ functions. * Made app_speech_utils.c declare that it is dependent
+ upon res_speech. (issue ASTERISK-17136) Reported by: Richard
+ Kenner
+
+ * /, channels/chan_iax2.c: chan_iax2: Fix saving the wrong expiry
+ time in astdb. When a new IAX2 client registers, the astdb
+ database is updated with the value of minregexpire defined in
+ iax.conf instead of using the expiry time that is provided by the
+ client. The provided expiry time of the client is updated after
+ inserting the astdb entry. As a consequence, restarting or
+ reloading asterisk creates clients whose registration may expire
+ before they reregister. The clients are therefore unavailable
+ after minregexpire seconds until they reregister. * Move updating
+ of the expiry time to before inserting into the astdb. (closes
+ issue ASTERISK-22504) Reported by: Stefan Wachtler Patches:
+ chan_iax2.c.patch (license #6533) patch uploaded by Stefan
+ Wachtler ........ Merged revisions 399158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399159 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-16 02:33 +0000 [r399146] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Filter internal channels out of bridge enter/leave
+ message handling Some channels exist merely as an implementation
+ detail in Asterisk, such as ConfBridge's announcer/recorder
+ channels. These channels should never be exposed to the outside
+ world, or to interfaces that report on Asterisk. We already
+ filter out such channels in snapshot processing; however, we
+ failed to filter out bridge related messages that involved these
+ channels. This patch filters out bridge related messages that are
+ for such channels. This prevents a spurious WARNING message from
+ being displayed when those channels move in and out of bridges.
+
+2013-09-13 22:05 +0000 [r399136] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge_channel.c, include/asterisk/bridge.h,
+ apps/confbridge/conf_chan_announce.c, tests/test_cdr.c,
+ res/res_pjsip_refer.c, channels/chan_sip.c, res/stasis/control.c,
+ main/bridge.c, main/bridge_basic.c, main/core_unreal.c,
+ res/parking/parking_applications.c, main/core_local.c,
+ res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
+ include/asterisk/features.h, main/channel.c,
+ include/asterisk/bridge_channel.h, res/parking/parking_tests.c,
+ main/features.c, tests/test_cel.c: Restore Dial, Queue, and
+ FollowMe 'I' option support. The Dial, Queue, and FollowMe
+ applications need to inhibit the bridging initial connected line
+ exchange in order to support the 'I' option. * Replaced the
+ pass_reference flag on ast_bridge_join() with a flags parameter
+ to pass other flags defined by enum ast_bridge_join_flags. *
+ Replaced the independent flag on ast_bridge_impart() with a flags
+ parameter to pass other flags defined by enum
+ ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe
+ applications are now the only callers of ast_bridge_call() and
+ ast_bridge_call_with_flags(), changed the calling contract to
+ require the initial COLP exchange to already have been done by
+ the caller. * Made all callers of ast_bridge_impart() check the
+ return value. It is important. As a precaution, I also made the
+ compiler complain now if it is not checked. * Did some cleanup in
+ parking_tests.c as a result of checking the ast_bridge_impart()
+ return value. An independent, but associated change is: * Reduce
+ stack usage in ast_indicate_data() and add a dropping redundant
+ connected line verbose message. (closes issue ASTERISK-22072)
+ Reported by: Joshua Colp Review:
+ https://reviewboard.asterisk.org/r/2845/
+
+2013-09-13 20:54 +0000 [r399100] David M. Lee <dlee@digium.com>
+
+ * /, main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
+ defined. If MALLOC_DEBUG is enabled, then the debug destructor
+ for the container is used, which would erroneously write to
+ /tmp/refs. This patch only uses the debug destructor if ref_debug
+ is used. (closes issue ASTERISK-22536) ........ Merged revisions
+ 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 399099 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-13 14:49 +0000 [r399083] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
+ include/asterisk/res_pjsip.h, res/res_pjsip.exports.in: Create
+ more accurate Contact headers for dialogs when we are the UAS.
+ (closes issue AST-1207) reported by John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2842
+
+2013-09-13 14:25 +0000 [r399064] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: Broke the build! Forgot
+ para tags within my description.
+ https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304
+
+2013-09-13 14:24 +0000 [r399059] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/config_auth.c,
+ res/res_pjsip_outbound_authenticator_digest.c,
+ res/res_pjsip_authenticator_digest.c: Change how realms are
+ handled for outbound authentication. With this change, if no
+ realm is specified in an outbound auth section, then we will
+ simply match the realm that was present in the 401/407 challenge.
+ (closes issue ASTERISK-22471) Reported by George Joseph (closes
+ issue ASTERISK-22386) Reported by Rusty Newton Patches:
+ outbound_auth_realm_v4.patch uploaded by George Joseph (License
+ #6322)
+
+2013-09-13 14:21 +0000 [r399039-399049] David M. Lee <dlee@digium.com>
+
+ * res/res_pjsip_log_forwarder.c (added), res/res_pjsip_logger.c,
+ res/res_rtp_asterisk.c: res_pjsip: Forward PJSIP logging to
+ Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to
+ forward PJSIP's log messages to Asterisk's logger. This is done
+ in a new module: res_pjsip_log_forwarder.so. This patch sets
+ defaultenabled on the existing res_pjsip_logger.so to no, since
+ logging every SIP packet seems a bit odd to do by default, and is
+ (hopefully) less necessary with regular PJSIP logging. It also
+ removes res_rtp_asterisk's disabling of PJSIP logging. (closes
+ issue ASTERISK-22360) Reported by: Joshua Colp Review:
+ https://reviewboard.asterisk.org/r/2830/
+
+ * res/res_http_websocket.c: ARI: Fix WebSocket response when
+ subprotocol isn't specified When I moved the ARI WebSocket from
+ /ws to /ari/events, I added code to allow a WebSocket to connect
+ without specifying the subprotocol if there's only one
+ subprotocol handler registered for the WebSocket. Naively, I
+ coded it to always respond with the subprotocol in use.
+ Unfortunately, according to RFC 6455, if the server's response
+ includes a subprotocol header field that "indicates the use of a
+ subprotocol that was not present in the client's handshake [...],
+ the client MUST _Fail the WebSocket Connection_.", emphasis
+ theirs. This patch correctly omits the Sec-WebSocket-Protocol if
+ one is not specified by the client. (closes issue ASTERISK-22441)
+ Review: https://reviewboard.asterisk.org/r/2828/
+
+2013-09-13 13:54 +0000 [r399035] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This
+ change ensures that MeetMeAdmin commands requiring a user
+ actually get a user and fixes another issue where an extra
+ dereference could occur for a last-entered user being ejected if
+ a user identifier was also provided. (closes issue
+ ASTERISK-21907) Reported by: Alex Epshteyn Review:
+ https://reviewboard.asterisk.org/r/2844/ ........ Merged
+ revisions 399033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 399034 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-13 13:27 +0000 [r399031] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: 'identify' configObject
+ doesn't have a synopsis Add a straightforward synopsis and
+ description to the identify config object in XML documentation.
+ (issue ASTERISK-22311) (closes issue ASTERISK-22311) Reported By:
+ Rusty Newton
+
+2013-09-12 23:41 +0000 [r399019-399021] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge.c: CLI bridge: Fix "bridge destroy <id>" and "bridge
+ kick <id> <chan>" tab completion. These two commands must deal
+ with the live bridges container for tab completion and not the
+ stasis cache.
+
+ * main/bridge.c: astobj2: Register the bridges container for debug
+ inspection.
+
+2013-09-12 23:21 +0000 [r399017] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip_acl.c: Documentation fix and improvements to XML
+ configuration help res_pjsip_acl * One bug fix. Made the synopsis
+ for "type" to accurate. * changing the usage of "IP-domains" to
+ "IP addresses" * clarifying the usage for the options, by adding
+ a relevant description for each * modified other areas of the XML
+ help for clarity, such as the module description and a few
+ synopsis changes here and there. See the patch. (issue
+ ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty
+ Newton Review: https://reviewboard.asterisk.org/r/2823/
+
+2013-09-12 20:20 +0000 [r398991] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
+ Revert r398835 due to failing tests involving originate (issue
+ ASTERISK-22424) Reported by: Jonathan Rose ........ Merged
+ revisions 398977 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398986 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-12 16:38 +0000 [r398938] Richard Mudgett <rmudgett@digium.com>
+
+ * main/core_unreal.c: core_local: Fix memory corruption race
+ condition. The masquerade super test is failing on v12 with high
+ fence violations and crashing. The fence violations are showing
+ that party id allocated memory strings are somehow getting
+ corrupted in the bridge_reconfigured_connected_line_update()
+ function. The invalid string values happen to be the freed memory
+ fill pattern. After much puzzling, I deduced that the
+ bridge_reconfigured_connected_line_update() is copying a string
+ out of the source channel's caller party id struct just as
+ another thread is updating it with a new value. The copying
+ thread is using the old string pointer being freed by the
+ updating thread. A search of the code found the
+ unreal_colp_redirect_indicate() routine updating the caller party
+ id's without holding the channel lock. A latent bug in v1.8 and
+ v11 hatched in v12 because of the bridging and connected line
+ changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2839/
+
+2013-09-12 15:23 +0000 [r398927] David M. Lee <dlee@digium.com>
+
+ * res/res_pjsip.c: Fix symbol collision with pjsua. We shouldn't be
+ exporting any symbols that start with pjsip_.
+
+2013-09-12 00:04 +0000 [r398882-398886] Rusty Newton <rnewton@digium.com>
+
+ * apps/app_queue.c, /: 'queue add member' help text correction You
+ are adding dial strings to the queue, not channels. An aribitrary
+ string could be used, but you are typically referencing a
+ channel. Correcting the command help text. (issue ASTERISK-22263)
+ (closes issue ASTERISK-22263) Reported By: Rusty Newton ........
+ Merged revisions 398884 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398885 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, configs/chan_dahdi.conf.sample: Documentation fix -
+ waitfordialtone is not boolean, it's time in milliseconds
+ Changing text in chan_dahdi.conf sample to be accurate. (issue
+ ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
+ Malcolm Davenport ........ Merged revisions 398880 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398881 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-11 19:56 +0000 [r398837] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
+ Reject calls without prior SDP on 200 OK If we receive a 200 OK
+ without SDP, we will now check to see if the remote address has
+ been established for that channel's RTP session and if the to tag
+ for that channel has changed from the most recent to tag in a
+ response less than 200. If either a change has been made since
+ the last to-tag was received or the remote address is unset, then
+ we will drop the call. (closes issue ASTERISK-22424) Reported by:
+ Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/diff/#index_header
+ ........ Merged revisions 398835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398836 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-11 18:02 +0000 [r398821] Russell Bryant <russell@russellbryant.com>
+
+ * configs/confbridge.conf.sample, /: Fix typo in
+ confbridge.conf.sample The denoise filter requires func_speex,
+ not codec_speex. Fix this in the description of the denoise=yes
+ option in confbridge.conf. ........ Merged revisions 398820 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-11 14:14 +0000 [r398806] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip_caller_id.c, channels/chan_pjsip.c: pjsip: reinvite
+ for connected line updates occurs when it should not Connected
+ line updates are now only sent out if an actual update needs to
+ occur. This happens under the following conditions: 1. The
+ endpoint we are sending to is trusted. 2. Either a
+ P-Asserted-Identity or Remote Party-ID header needs to be
+ added/sent. 3. The connected id's number and name are valid. Also
+ added an SDP when an update is sent out. (closes issue AST-1212)
+ Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2831/
+
+2013-09-10 18:03 +0000 [r398759] Richard Mudgett <rmudgett@digium.com>
+
+ * main/indications.c, main/asterisk.c, main/xmldoc.c, main/cli.c,
+ /, funcs/func_dialgroup.c, main/heap.c,
+ res/res_pjsip/pjsip_configuration.c, main/event.c,
+ res/res_musiconhold.c: Fix incorrect usages of ast_realloc().
+ There are several locations in the code base where this is done:
+ buf = ast_realloc(buf, new_size); This is going to leak the
+ original buf contents if the realloc fails. Review:
+ https://reviewboard.asterisk.org/r/2832/ ........ Merged
+ revisions 398757 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398758 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-10 17:49 +0000 [r398750-398754] David M. Lee <dlee@digium.com>
+
+ * utils/check_expr.c, /: Fixed utils directory breakage from
+ r398748, this time with extra hate. ........ Merged revisions
+ 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 398753 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c, /: Fixed
+ utils directory breakage from r398648 ........ Merged revisions
+ 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 398749 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-09 23:23 +0000 [r398726] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
+ completely different from the freed magic number. Race conditions
+ between freeing a nul terminated string and ast_strdup()'ing it
+ are more likely to be detected if the fence and freed magic
+ numbers are completely different. ........ Merged revisions
+ 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 398721 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-09 21:59 +0000 [r398694] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: Add extra debugging to
+ res_pjsip_endpoint_identifier_ip
+
+2013-09-09 20:12 +0000 [r398638-398651] David M. Lee <dlee@digium.com>
+
+ * /, main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
+ DEBUG_THREADS when lock is acquired in __constructor__ This patch
+ fixes some long-standing bugs in debug threads that were
+ exacerbated with recent Optional API work in Asterisk 12. With
+ debug threads enabled, on some systems, there's a lock ordering
+ problem between our mutex and glibc's mutex protecting its module
+ list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
+ thread, the module list will be locked before acquiring our
+ mutex. In another thread, our mutex will be locked before locking
+ the module list (which happens in the depths of calling
+ backtrace()). This patch fixes this issue by moving backtrace()
+ calls outside of critical sections that have the mutex acquired.
+ The bigger change was to reentrancy tracking for
+ ast_cond_{timed,}wait, which wrongly assumed that waiting on the
+ mutex was equivalent to a single unlock (it actually suspends all
+ recursive locks on the mutex). (closes issue ASTERISK-22455)
+ Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged
+ revisions 398648 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398649 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/ari/resource_channels.h: Added note about expected behavior
+ of originate (the rest of the commit)
+
+ * rest-api/api-docs/channels.json: Added note about expected
+ behavior of originate
+
+2013-09-08 23:25 +0000 [r398628] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_cdr.c: Update CDR Unit tests to reflect container
+ changes in r398579 When a channel joins a multi-party bridge, the
+ ordering of the CDRs that is created is determined by the
+ ordering of the channels who happen to be in that bridge. When
+ r398579 changed the number of buckets in the container to
+ something sensible, it changed the ordering that the CDRs was
+ created in, causing one of the multiparty tests to fail. This
+ fixes the test with the now expected ordering.
+
+2013-09-07 01:02 +0000 [r398580-398619] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_xmpp.c: Prevent XMPP timeout on blank responses
+ Sometimes the Google Voice servers have a bad habit of sending
+ out 1 byte replies to the xmpp resource. When a blank 1 byte
+ reply is received from the socket the buffer attempts to wait
+ (endlessly) for the rest of the reply from google which
+ effectively blocks the socket and google voice calls will no
+ longer come into the server. This patch allows the xmpp module to
+ correctly detect empty packets and send out ping replies to
+ google. It also sets a socket timeout on the default socket which
+ prevents the xmpp socket from closing and preventing future
+ google voice calls from coming into the server. Furthermore
+ instead of sending an empty reply back to google we send a proper
+ xmpp ping reply back. This also adds several more socket
+ messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy
+ Review: https://reviewboard.asterisk.org/r/2771 Patches:
+ xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........
+ Merged revisions 398618 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_xmpp.c, res/res_jabber.c: Multiple revisions
+ 398558,398577 ........ r398558 | kmoore | 2013-09-06 14:28:16
+ -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed
+ MWI The mailbox and context are swapped on the receiving end for
+ all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and
+ all more recent versions. This swaps those values to be correct
+ when publishing to the internal event system from Jabber/XMPP
+ distributed MWI state. (closes issue ASTERISK-22435) Reported by:
+ abelbeck Tested by: Michael Keuter Patches:
+ asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
+ abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
+ uploaded by abelbeck ........ Merged revisions 398523 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) |
+ 10 lines Commit the remainder of r398523 This is a missing part
+ of the commit in revision 398523 that corrects the name of a
+ variable. (issue ASTERISK-22435) ........ Merged revisions 398576
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 398558,398577 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-06 21:16 +0000 [r398579] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cdr.c: cdr: Change the number of container buckets to be
+ similar to the channels container. * Fix the temporary cdr
+ candidate containers to use a prime number of buckets.
+
+2013-09-06 21:03 +0000 [r398578] Kinsey Moore <kmoore@digium.com>
+
+ * /: Unblock r398558
+
+2013-09-06 20:20 +0000 [r398533-398572] Richard Mudgett <rmudgett@digium.com>
+
+ * main/core_local.c: core_local: Fix LocalOptimizationBegin AMI
+ event missing Source channel snapshot. * Fix the
+ LocalOptimizationBegin AMI event by eliminating an artificial
+ buffer size limitation that is too small anyway.
+
+ * main/cdr.c: cdr: Fix some ref leaks. * Added missing unregister
+ of the cdr container in cdr_engine_shutdown(). * Fixed ref leak
+ in off nominal path of cdr_object_alloc(). * Removed some
+ unnecessary NULL checks in cdr_object_dtor().
+
+ * main/cdr.c, main/udptl.c, main/parking.c, main/stasis_config.c,
+ include/asterisk/astobj2.h, main/cel.c, main/features_config.c,
+ apps/app_agent_pool.c: astobj2: Add warn unused attribute to some
+ functions. * Fixed resulting warnings with improper use of
+ ao2_global_obj_replace(). * Made a couple uses of
+ ao2_global_obj_replace_unref(x, NULL) into the equivalent and
+ more appropriate ao2_global_obj_release() call.
+
+2013-09-06 18:49 +0000 [r398511-398521] Kinsey Moore <kmoore@digium.com>
+
+ * main/http.c, res/stasis/app.c: Fix build warnings When
+ AST_DEVMODE is not defined, ast_asserts are not compiled into the
+ binary. In some cases, this means variables are not referenced or
+ are set but unused which causes warnings to show up. (closes
+ issue ASTERISK-22446) Reported by: Jason Parker (qwell)
+
+ * /, channels/chan_h323.c: Fix chan_h323 compilation This fixes the
+ things in chan_h323 that were missed or ignored in the great
+ channel opaquification and gets chan_h323 back into a compiling
+ state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov
+ Patches: chan_h323.patch uploaded by Dmitry Melekhov ........
+ Merged revisions 398510 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-05 21:46 +0000 [r398381-398498] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: astobj2: Only define ao2_bt() once. * Make
+ ao2_bt() not use single char variable names. * Fix ao2_bt()
+ formatting.
+
+ * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in
+ __attempt_transmit(). * Reduce indentation in
+ __attempt_transmit(). * Don't update the static last error time
+ variable every time in __schedule_action() and socket_read().
+ ........ Merged revisions 398456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker
+ thread idle_list. * Fix stray reference to idle_list in
+ cleanup_thread_list(). This may be the reason for the note in
+ iax2_process_thread() about threads not being removed from the
+ task lists. * Move cleanup_thread_list(&idle_list) to after the
+ other lists are cleaned up. ........ Merged revisions 398416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398417 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock
+ avoidance. * Fix bridgecallno deadlock avoidance. When doing
+ deadlock avoidance, you need to retest the status of values for
+ each loop to see if you still need the lock for bridgecallno. *
+ As a safety check, after acquiring the bridgecallno lock you
+ should check if iaxs[bridgecallno] is NULL just like the current
+ callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
+ to after processing any deferred frames to ensure that the
+ iostate is IDLE when it is placed back into the idle list.
+ defer_full_frame() tries to ensure iax2_process_thread() wakes up
+ to process the frame. ........ Merged revisions 398379 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398380 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-05 14:09 +0000 [r398368] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_outbound_registration.c: Clarify server_uri and
+ client_uri registration settings. Used some of Rusty's suggested
+ language plus also included more SIPesque descriptions of where
+ the URIs are actually used in an outgoing REGISTER. (closes issue
+ ASTERISK-22390) reported by Rusty Newton
+
+2013-09-04 23:06 +0000 [r398303] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/iax2/parser.c: chan_iax2: Add missing control frame
+ names to debug frame decode output. ........ Merged revisions
+ 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 398302 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-04 22:28 +0000 [r398299] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_outbound_authenticator_digest.c: Give more detail
+ regarding failures to create request with auth credentials.
+ (issue ASTERISK-22386)
+
+2013-09-04 21:36 +0000 [r398283-398286] Jonathan Rose <jrose@digium.com>
+
+ * /, tests/test_voicemail_api.c: unit tests: test_voicemail_api
+ leaks stringfields from snapshots (closes issue ASTERISK-22414)
+ Reported by: Corey Farrell Patches:
+ test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
+ (license 5909) ........ Merged revisions 398285 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_voicemail.c, /: app_voicemail: Fix leaking config
+ objects when msg_id doesn't match (issues ASTERISK-22414)
+ Reported by: Corey Farrell Patch:
+ test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
+ (license 5909) ........ Merged revisions 398281 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-04 16:00 +0000 [r398237] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output
+ printed with arbitrary verbose levels. Fix the misdn debug output
+ to remote consoles. chan_misdn uses ast_console_puts() which
+ doesn't know about verbose levels. Better to use ast_verbose()
+ instead. Without this patch the misdn debug messages are appended
+ to the verbose level which ever was set by the message sent to
+ the console before, i.e. any undefined level. (closes issue
+ AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch
+ (license #6372) patch uploaded by Guenther Kelleter ........
+ Merged revisions 398235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398236 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-04 14:29 +0000 [r398226] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip_outbound_registration.c: Debug messages for pjsip
+ outbound registration Added debug messages indicating that an
+ outbound registration attempt was made and it was successful in
+ pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton
+
+2013-09-03 19:49 +0000 [r398215] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling
+ on empty tcs received ........ Merged revisions 398214 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-09-03 18:08 +0000 [r398206] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_pjsip_dtmf_info.c: Prevent a crash in
+ res_pjsip_dtmf_info.c This change makes sure that a content type
+ header exists before checking the contents of the header against
+ known SIP INFO DTMF content types.
+
+2013-09-03 14:36 +0000 [r398198] David M. Lee <dlee@digium.com>
+
+ * Makefile: Fixed 'make clean' for wiki docs
+
+2013-09-03 14:27 +0000 [r398196] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, cel/cel_custom.c: Be a little more verbose when loading
+ cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
+ ........ Merged revisions 398167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398168 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-30 20:58 +0000 [r398149] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/optional_api.h, main/optional_api.c,
+ main/asterisk.c: Fix graceful shutdown crash. The cleanup code
+ for optional_api needs to happen after all of the optional API
+ users and providers have unused/unprovided. Unfortunately,
+ regsitering the atexit() handler at the beginning of main() isn't
+ soon enough, since module destructors run after that.
+
+2013-08-30 20:34 +0000 [r398147] Rusty Newton <rnewton@digium.com>
+
+ * configs/pjsip.conf.sample: New pjsip.conf.sample (issue
+ ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2811/
+
+2013-08-30 19:51 +0000 [r398116-398139] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip_outbound_registration.c,
+ include/asterisk/sorcery.h, res/res_pjsip.c,
+ res/res_pjsip/config_transport.c, main/sorcery.c: Add a
+ reloadable option for sorcery type objects Some configuration
+ objects currently won't place nice if reloaded. Specifically, in
+ this case the pjsip transport objects. Now when registering an
+ object in sorcery one may specify that the object is allowed to
+ be reloaded or not. If the object is set to not reload then upon
+ reloading of the configuration the objects of that type will not
+ be reloaded. The initially loaded objects of that type however
+ will remain. While the transport objects will not longer be
+ reloaded it is still possible for a user to configure an endpoint
+ to an invalid transport. A couple of log messages were added to
+ help diagnose this problem if it occurs. (closes issue
+ ASTERISK-22382) Reported by: Rusty Newton (closes issue
+ ASTERISK-22384) Reported by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/2807/
+
+ * res/res_security_log.c, /, channels/chan_sip.c, main/translate.c,
+ main/named_acl.c, main/indications.c, main/config.c: Fix various
+ memory leaks main/config.c - cleanup cache fie includes
+ res/res_security_log.c - unregister logger level
+ channesl/chan_sip.c - cleanup io context and notify_types
+ main/translator.c - cleanup at shutdown main/named_acl.c -
+ cleanup cli commands main/indications.c -
+ ast_get_indication_tone() unref default_tone_zone if used (closes
+ issues ASTERISK-22378) Reported by: Corey Farrell Patches:
+ config_shutdown.patch uploaded by coreyfarrell (license 5909)
+ res_security_log.patch uploaded by coreyfarrell (license 5909)
+ chan_sip-11.patch uploaded by coreyfarrell (license 5909)
+ indications_refleak.patch uploaded by coreyfarrell (license 5909)
+ named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license
+ 5909) translate_shutdown.patch uploaded by coreyfarrell (license
+ 5909) ........ Merged revisions 398102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398103 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-30 18:35 +0000 [r398100] Matthew Jordan <mjordan@digium.com>
+
+ * UPGRADE.txt: Update UPGRADE.txt file for Asterisk 12 This simply
+ pulls in the changes that were breaking from the CHANGES file and
+ updates a few other areas accordingly. It also removes the 10 =>
+ 11 notes, which are traditionally removed from each major version
+ and stored in the appropriate UPGRADE-X.txt file.
+
+2013-08-30 18:18 +0000 [r398068] Jonathan Rose <jrose@digium.com>
+
+ * main/config_options.c, main/features_config.c: features_config:
+ Ignore parkinglots in features.conf instead of failing to load
+ Parkinglots are defined in res_features.conf now, but this patch
+ fixes features_config so that features don't fail to load when
+ parkinglots are present in features.conf Review:
+ https://reviewboard.asterisk.org/r/2801/
+
+2013-08-30 17:57 +0000 [r398062] Kevin Harwell <kharwell@digium.com>
+
+ * main/manager.c, /, res/res_agi.c: Memory leak fix
+ ast_xmldoc_printable returns an allocated block that must be
+ freed by the caller. Fixed manager.c and res_agi.c to stop
+ leaking these results. (closes issue ASTERISK-22395) Reported by:
+ Corey Farrell Patches: manager-leaks-12.patch uploaded by
+ coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+ by coreyfarrell (license 5909) ........ Merged revisions 398060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 398061 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-30 17:10 +0000 [r398023-398025] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_substitution.c: test_substitution: Fix failing test.
+ Revert the -r392190 change. The original test was correct. The
+ CDR code was actually returning an unititialized buffer.
+
+ * /, tests/test_substitution.c: test_substituition: Fix failed test
+ reporting to actually report failure. You cannot put the "Testing
+ <blah> pass/fail" on a single line before actually performing the
+ test. Now any additional failure information is logged before the
+ test pass/fail announcement. * Added an additional CDR(answer,u)
+ test. ........ Merged revisions 398018 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 398019 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-30 16:57 +0000 [r398020] Jonathan Rose <jrose@digium.com>
+
+ * main/features_config.c, main/udptl.c: features_config: Don't
+ require features.conf to be present for Asterisk to load (closes
+ issue ASTERISK-22426) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2806/
+
+2013-08-30 16:26 +0000 [r398002-398016] Kevin Harwell <kharwell@digium.com>
+
+ * apps/app_mixmonitor.c, /: Fix memory leaks (closes issue
+ ASTERISK-22368) Reported by: Corey Farrell Patches:
+ issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes
+ (license 5674) ........ Merged revisions 398004 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 398011 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/asterisk.c: Check return value on fwrite ........ Merged
+ revisions 398000 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-30 13:39 +0000 [r397985-397989] David M. Lee <dlee@digium.com>
+
+ * res/res_ari.c, tests/test_optional_api.c (added),
+ channels/chan_sip.c, include/asterisk/autoconfig.h.in,
+ configure.ac, rest-api-templates/res_ari_resource.c.mustache,
+ res/ari/internal.h, res/res_http_websocket.c, CHANGES,
+ include/asterisk/compiler.h, include/asterisk/ari.h,
+ main/loader.c, include/asterisk/optional_api.h,
+ build_tools/cflags.xml, configure, res/res_ari_events.c,
+ include/asterisk/http_websocket.h, main/optional_api.c (added),
+ rest-api-templates/swagger_model.py, res/ari/ari_websockets.c,
+ main/asterisk.c, channels/sip/include/sip.h: optional_api: Fix
+ linking problems between modules that export global symbols With
+ the new work in Asterisk 12, there are some uses of the
+ optional_api that are prone to failure. The details are rather
+ involved, and captured on [the wiki][1]. This patch addresses the
+ issue by removing almost all of the magic from the optional API
+ implementation. Instead of relying on weak symbol resolution, a
+ new optional_api.c module was added to Asterisk core. For modules
+ providing an optional API, the pointer to the implementation
+ function is registered with the core. For modules that use an
+ optional API, a pointer to a stub function, along with a
+ optional_ref function pointer are registered with the core. The
+ optional_ref function pointers is set to the implementation
+ function when it's provided, or the stub function when it's now.
+ Since the implementation no longer relies on magic, it is now
+ supported on all platforms. In the spirit of choice, an
+ OPTIONAL_API flag was added, so we can disable the optional_api
+ if needed (maybe it's buggy on some bizarre platform I haven't
+ tested on) The AST_OPTIONAL_API*() macros themselves remained
+ unchanged, so existing code could remain unchanged. But to help
+ with debugging the optional_api, the patch limits the #include of
+ optional API's to just the modules using the API. This also
+ reduces resource waste maintaining optional_ref pointers that
+ aren't used. Other changes made as a part of this patch: * The
+ stubs for http_websocket that wrap system calls set errno to
+ ENOSYS. * res_http_websocket now properly increments module use
+ count. * In loader.c, the while() wrappers around dlclose() were
+ removed. The while(!dlclose()) is actually an anti-pattern, which
+ can lead to infinite loops if the module you're attempting to
+ unload exports a symbol that was directly linked to. * The
+ special handling of nonoptreq on systems without weak symbol
+ support was removed, since we no longer rely on weak symbols for
+ optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ
+ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2797/
+
+ * res/res_stasis_playback.c,
+ include/asterisk/stasis_app_recording.h,
+ res/ari/resource_recordings.h, res/res_stasis_recording.c,
+ res/Makefile, res/ari/ari_model_validators.c,
+ rest-api/api-docs/recordings.json, res/stasis_recording (added),
+ res/ari/resource_recordings.c, res/ari/ari_model_validators.h,
+ res/res_ari_recordings.c: ARI: Implement /recordings/stored API's
+ his patch implements the ARI API's for stored recordings. While
+ the original task only specified deleting a recording, it was
+ simple enough to implement the GET for all recordings, and for an
+ individual recording. The recording playback operation was
+ modified to use the same code for accessing the recording as the
+ REST API, so that they will behave consistently. There were
+ several problems with the api-docs that were also fixed, bringing
+ the ARI spec in line with the implementation. There were some
+ 'wishful thinking' fields on the stored recording model (duration
+ and timestamp) that were removed, because I ended up not
+ implementing a metadata file to go along with the recording to
+ store such information. The GET /recordings/live operation was
+ removed, since it's not really that useful to get a list of all
+ recordings that are currently going on in the system. (At least,
+ if we did that, we'd probably want to also list all of the
+ current playbacks. Which seems weird.) (closes issue
+ ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/
+
+2013-08-30 01:19 +0000 [r397975-397977] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: pbx.c: Make pbx_substitute_variables_helper_full()
+ not mask variables.
+
+ * main/pbx.c, tests/test_substitution.c, funcs/func_cdr.c: Revert
+ last commit.
+
+ * main/pbx.c, tests/test_substitution.c, funcs/func_cdr.c: pbx.c:
+ Make ast_str_substitute_variables_full() not mask variables.
+
+2013-08-30 00:10 +0000 [r397960-397968] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_pidf.c: Sanitize XML output for PIDF bodies.
+ PJSIP's PIDF API does not replace angle brackets with their
+ appropriate counterparts for XML. So we have to do it ourself. In
+ this particular case, the problem had to do with attempting to
+ place an unsanitized SIP URI into an XML node. Now we don't get a
+ 488 from recipients of our PIDF NOTIFYs.
+
+ * res/res_pjsip_pidf.c: Fix method for creating activities string
+ in PIDF bodies. The previous method did not allocate enough space
+ to create the entire string, but adjusted the string's slen value
+ to be larger than the actual allocation. This resulted in garbled
+ text in NOTIFY requests from Asterisk. This method allocates the
+ proper amount of space first and then writes the content into the
+ buffer.
+
+2013-08-29 22:45 +0000 [r397958] Kevin Harwell <kharwell@digium.com>
+
+ * channels/chan_misdn.c, /, apps/app_dumpchan.c, main/logger.c,
+ apps/app_verbose.c, main/asterisk.c: Verbose logging
+ discrepancies Refactored cases where a combination of
+ ast_verbose/options_verbose were present. Also in general tried
+ to eliminate, in as many places as possible, where the
+ options_verbose global variable was being used. Refactored the
+ way local and remote consoles handle verbose message logging in
+ an attempt to solve the various discrepancies that sometimes
+ would show between the two. (closes issue AST-1193) Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2798/ ........ Merged
+ revisions 397948 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-29 22:24 +0000 [r397955] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_pubsub.c: Fix when the subscription_terminated
+ callback is called for subscription handlers. The previous
+ placement would result in the resubscribe() callback called
+ instead of the subscription_terminated() callback being called
+ when a subscription was ended via a SUBSCRIBE request. This would
+ result in confusing PJSIP and having it throw an assertion.
+
+2013-08-29 21:34 +0000 [r397946] Kevin Harwell <kharwell@digium.com>
+
+ * main/manager.c, main/stasis_config.c, main/file.c, main/app.c,
+ main/config_options.c, main/cel.c, main/asterisk.c, main/cdr.c:
+ Memory leaks fix (closes ASTERISK-22376) Reported by: John Hardin
+ Patches: memleak.patch uploaded by jhardin (license 6512)
+ memleak2.patch uploaded by jhardin (license 6512)
+
+2013-08-29 21:33 +0000 [r397945] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_session.c: Fix a race condition where a canceled
+ call was answered. RFC 5407 section 3.1.2 details a scenario
+ where a UAC sends a CANCEL at the same time that a UAS sends a
+ 200 OK for the INVITE that the UAC is canceling. When this
+ occurs, it is the role of the UAC to immediately send a BYE to
+ terminate the call. This scenario was reproducible by have a
+ Digium phone with two lines place a call to a second phone that
+ forwarded the call to the second line on the original phone. The
+ Digium phone, upon realizing that it was connecting to itself,
+ would attempt to cancel the call. The timing of this happened to
+ trigger the aforementioned race condition about 80% of the time.
+ Asterisk was not doing its job of sending a BYE when receiving a
+ 200 OK on a cancelled INVITE. The result was that the ast_channel
+ structure was destroyed but the underlying SIP session, as well
+ as the PJSIP inv_session and dialog, were still alive. Attempting
+ to perform an action such as a transfer, once in this state,
+ would result in Asterisk crashing. The circumstances are now
+ detected properly and the session is ended as recommended in RFC
+ 5407. (closes issue AST-1209) reported by John Bigelow
+
+2013-08-29 20:21 +0000 [r397938] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/scripts/safe_asterisk, Makefile,
+ configs/safe_asterisk.conf.sample (removed), CHANGES: Revert
+ r394939 due to (numerous) objections The patch from
+ ASTERISK-21965 was committed perhaps a bit too hastily. Walter
+ and Tzafrir have pointed out numerous issues with the approach
+ and have propsed an alternative in r/2757. Since it's not a time
+ critical issue and is not worth holding up the release of 12 for
+ it, I've gone ahead and reverted r394939 from 12/trunk and
+ re-opened ASTERISK-21965.
+
+2013-08-29 16:18 +0000 [r397927] David M. Lee <dlee@digium.com>
+
+ * rest-api-templates/asterisk_processor.py,
+ rest-api-templates/make_ari_stubs.py,
+ rest-api-templates/api.wiki.mustache: Account for {} in Swagger
+ notes
+
+2013-08-29 16:04 +0000 [r397924] Matthew Jordan <mjordan@digium.com>
+
+ * Makefile: Recursively search for '.c' files when making
+ documentation with 'make full' Without this, documentation
+ defined in sub-folders is ignored. Since having properly
+ generated documentation is especially important in Asterisk 12 -
+ not having it can cause a module to not load - 'make full' needs
+ to look in all .c files.
+
+2013-08-29 15:42 +0000 [r397921-397922] Mark Michelson <mmichelson@digium.com>
+
+ * main/cel.c: Remove extra debug message.
+
+ * main/stasis_bridges.c, apps/app_queue.c, main/cel.c: Resolve
+ assumptions that bridge snapshots would be non-NULL for transfer
+ stasis events. Attempting to transfer an unbridged call would
+ result in crashes in either CEL code or in the conversion to AMI
+ messages.
+
+2013-08-29 12:27 +0000 [r397911] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/ast-db-manage/voicemail (added),
+ contrib/ast-db-manage/voicemail/script.py.mako (added),
+ contrib/ast-db-manage/README.md (added),
+ contrib/ast-db-manage/config/versions (added),
+ contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py
+ (added), contrib/ast-db-manage (added),
+ contrib/ast-db-manage/voicemail/versions (added),
+ contrib/ast-db-manage/config.ini.sample (added),
+ contrib/ast-db-manage/config/env.py (added),
+ contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py
+ (added), contrib/ast-db-manage/config (added),
+ contrib/ast-db-manage/config/script.py.mako (added),
+ contrib/ast-db-manage/voicemail.ini.sample (added),
+ contrib/ast-db-manage/voicemail/env.py (added): Actually *add*
+ the database schema management utilities In r397874, the scripts
+ were removed... but not replaced. Thanks to Michael Young for
+ noticing this!
+
+2013-08-28 23:14 +0000 [r397885-397902] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cdr.c, funcs/func_cdr.c, main/stdtime/localtime.c: Fix some
+ uninitialized buffers for CDR handling valgrind found. * Made
+ ast_strftime_locale() ensure that the output buffer is
+ initialized. The std library strftime() returns 0 and does not
+ touch the buffer if it has an error. However, the function can
+ also return 0 without an error. (closes issue ASTERISK-22412)
+ Reported by: rmudgett
+
+ * main/cdr.c: Fixed problems with ast_cdr_serialize_variables(). *
+ Fixed return value of ast_cdr_serialize_variables() on error. It
+ needs to return 0 indicating no CDR variables found. * Made
+ ast_cdr_serialize_variables() check the return value of
+ cdr_object_format_property() and assert if nonzero. A member of
+ the cdr_readonly_vars[] was not handled. * Removed unused
+ elements from cdr_readonly_vars[]: total_duration, total_billsec,
+ first_start, and first_answer.
+
+ * main/cdr.c: Made the on/off in CLI "cdr set debug [on|off]" case
+ insensitive.
+
+ * main/cdr.c: Make CDR variable name chandling consistently case
+ insensitive.
+
+ * main/cdr.c: Make CDR code deal with channel names case
+ insensitively.
+
+ * funcs/func_cdr.c, main/cdr.c: Some CDR code optimization.
+
+ * funcs/func_cdr.c: Whitespace and curly braces.
+
+2013-08-28 21:05 +0000 [r397876] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_refer.c: Improve detection of answer on SIP blind
+ transfer. A problem encountered during testing was that
+ res_pjsip_refer would not ever send a NOTIFY with a 200 OK
+ sipfrag. This is because the framehook that was supposed to send
+ the NOTIFY would never be told that an answer had occurred. This
+ happened for two reasons: 1) The transferee channel on which the
+ framehook was on was already up. 2) Answers are rarely if ever
+ written to channels. Rather, the ast_answer() or ast_raw_answer()
+ function is used to answer channels. Thanks to a suggestion by
+ Matt Jordan, the best way to detect that the call had been
+ answered was to find out when the transferee channel joined a
+ bridge. With stasis this is an easy task. So now, in addition to
+ the framehook logic, there is a stasis subscription used to
+ determine when the transferee has entered a bridge. Once it has
+ entered, an appropriate NOTIFY is sent.
+
+2013-08-28 20:55 +0000 [r397870-397874] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/realtime/mysql/voicemail_messages.sql,
+ contrib/realtime/postgresql/realtime.sql,
+ contrib/realtime/mysql/voicemail_data.sql, CHANGES,
+ contrib/realtime/mysql/musiconhold.sql,
+ contrib/realtime/mysql/queue_log.sql,
+ contrib/realtime/mysql/voicemail.sql,
+ contrib/realtime/mysql/sippeers.sql,
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql: Add database schema management
+ using Alembic This patch replaces contrib/realtime/ with a new
+ setup for managing the database schema required for database
+ integration with Asterisk. In addition to initializing a database
+ with the proper schema, alembic can do a database migration to
+ assist with upgrading Asterisk in the future. Hopefully this
+ helps make setting up and operating Asterisk with a database
+ easier. With this the schema only needs to be maintained in one
+ place instead of once per database. The schemas I have added here
+ have a bit of improvement over the examples that were there
+ before (some added consistency and added some missing indexes).
+ Managing the schema in one place here also applies to all
+ databases supported by SQLAlchemy. See
+ contrib/ast-db-manage/README.md for more details. Review:
+ https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant
+ (license 6300)
+
+ * CHANGES: Update CHANGES file for Asterisk 12 This updates the
+ Asterisk 12 CHANGES file with the things that were missed during
+ the development cycle. Review:
+ https://reviewboard.asterisk.org/r/2795/
+
+2013-08-28 16:12 +0000 [r397856-397859] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: pbx.c: Make ast_str_substitute_variables_full() not
+ mask variables.
+
+ * include/asterisk/threadstorage.h: Match use of ast_free() with
+ ast_calloc() and add some curly braces.
+
+2013-08-28 15:40 +0000 [r397854] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/pjsip_distributor.c: Fix dialog matching in the SIP
+ distributor. Dialog matching is performed in the distributor for
+ the sole purpose of retrieving an associated serializer so the
+ request may be serialized. This patch fixes two problems. First,
+ incoming CANCEL requests that had no to-tag (which really should
+ be *all* CANCEL requests) would not match with a dialog. An
+ earlier bug fix to deal with early CANCEL requests would result
+ in the CANCEL being replied to with a 481. The fix for this is to
+ find the matching INVITE transaction and get the dialog from that
+ transaction. Second, no SIP responses were matching dialogs. This
+ is because we were inverting the tags that we were passing into
+ PJSIP's dialog finding function. This logic has been corrected by
+ setting local and remote tag variables based on whether the
+ incoming message is a request or response.
+
+2013-08-27 19:15 +0000 [r397816] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis.c, main/stasis_bridges.c,
+ rest-api-templates/param_parsing.mustache, res/res_ari_bridges.c,
+ res/stasis/app.c, res/res_ari_events.c, res/res_ari_asterisk.c,
+ rest-api-templates/res_ari_resource.c.mustache, res/stasis/app.h:
+ ARI: WebSocket event cleanup Stasis events (which get distributed
+ over the ARI WebSocket) are created by subscribing to the
+ channel_all_cached and bridge_all_cached topics, filtering out
+ events for channels/bridges currently subscribed to. There are
+ two issues with that. First was a race condition, where messages
+ in-flight to the master subscribe-to-all-things topic would get
+ sent out, even though the events happened before the channel was
+ put into Stasis. Secondly, as the number of channels and bridges
+ grow in the system, the work spent filtering messages becomes
+ excessive. Since r395954, individual channels and bridges have
+ caching topics, and can be subscribed to individually. This patch
+ takes advantage, so that channels and bridges are subscribed to
+ on demand, instead of filtering the global topics. The one case
+ where filtering is still required is handling BridgeMerge
+ messages, which are published directly to the bridge_all topic.
+ Other than the change to how subscriptions work, this patch
+ mostly just moves code around. Most of the work generating JSON
+ objects from messages was moved to .to_json handlers on the
+ message types. The callback functions handling app subscriptions
+ were moved from res_stasis (b/c they were global to the model) to
+ stasis/app.c (b/c they are local to the app now). (closes issue
+ ASTERISK-21969) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2754/
+
+2013-08-27 18:49 +0000 [r397809] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astmm.c: Made MALLOC_DEBUG less CPU intensive by default.
+ Storing a backtrace for each allocation in anticipation of a
+ memory management problem is very CPU intensive. * Added the CLI
+ "memory backtrace {on|off}" command to request that the backtrace
+ be gathered only on request. The backtrace is off by default.
+ (issue ASTERISK-22221) Reported by: Matt Jordan
+
+2013-08-27 18:05 +0000 [r397759] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
+ SDP If the SIP channel driver processes an invalid SDP that
+ defines media descriptions before connection information, it may
+ attempt to reference the socket address information even though
+ that information has not yet been set. This will cause a crash.
+ This patch adds checks when handling the various media
+ descriptions that ensures the media descriptions are handled only
+ if we have connection information suitable for that media. Thanks
+ to Walter Doekes, OSSO B.V., for reporting, testing, and
+ providing the solution to this problem. (closes issue
+ ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
+ issueA22007_sdp_without_c_death.patch uploaded by wdoekes
+ (License 5674) ........ Merged revisions 397756 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397757 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 397758 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-27 16:47 +0000 [r397745] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c, channels/chan_motif.c, channels/chan_iax2.c,
+ channels/sig_pri.c, channels/sig_ss7.c: Fix uninitialized value
+ in struct ast_control_pvt_cause_code usage. ........ Merged
+ revisions 397744 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-27 16:03 +0000 [r397690-397713] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK
+ on dialog that has no channel A remote exploitable crash
+ vulnerability exists in the SIP channel driver if an ACK with SDP
+ is received after the channel has been terminated. The handling
+ code incorrectly assumed that the channel would always be
+ present. This patch adds a check such that the SDP will only be
+ parsed and applied if Asterisk has a channel present that is
+ associated with the dialog. Note that the patch being applied was
+ modified only slightly from the patch provided by Walter Doekes
+ of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
+ Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
+ issueA21064_fix.patch uploaded by wdoekes (License 5674) ........
+ Merged revisions 397710 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397711 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 397712 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/bridge_channel.c: Better handle clearing the OUTGOING flag
+ when a channel leaves a bridge When a channel with the OUTGOING
+ flag leaves a bridge, and it will survive being pulled from the
+ bridge (either because it will execute dialplan, go into another
+ bridge, or live in a friendly autoloop), we have to clear the
+ OUTGOING flag. This is the signal to the CDR engine that this
+ channel is no longer a second class citizen, i.e., it is not
+ "dialed". The soft hangup flags are only half the picture. If a
+ channel is being moved from one bridge to another, the soft
+ hangup flags aren't set; however, the state of the bridge_channel
+ will not be hung up. Since the channel does not have one of the
+ two hang up states, that implies that the channel is still
+ technically alive. This patch modifies the check so that it
+ checks both the soft hangup flags as well as the bridge_channel
+ state. If either suggests that the channel is going to persist,
+ we clear the OUTGOING flag.
+
+2013-08-26 21:30 +0000 [r397673] David M. Lee <dlee@digium.com>
+
+ * main/bucket.c: Fixed bucket.c for systems where tv_usec is not an
+ unsigned long.
+
+2013-08-26 16:24 +0000 [r397643-397650] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridge_channel.h, main/bridge_channel.c:
+ bridging: Fix a livelock with local channel optimization. Use a
+ better means of waking up the bridge channel thread.
+
+ * channels/Makefile: chan_dahdi: Add some missing build cleanup.
+
+2013-08-25 18:12 +0000 [r397621-397630] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_bucket.c: Fix bucket unit tests After the review for
+ buckets was completed (r2715), the handling of names in the
+ bucket core was deferred to the wizards. As such, the bucket unit
+ tests cannot expect that passing a URI with a scheme specified
+ but no actual resource name will automatically fail. The tests
+ have been updated to not make this check.
+
+ * include/asterisk/config_options.h, main/config_options.c,
+ tests/test_config.c: Fix the config_options_test The config
+ options test requires the entire configuration item to be
+ transparent from the documentation system. So we let it do that
+ too. As an aside, please do not use this power for evil.
+ Documentation is your friend, and you really should document your
+ configurations. Hiding your module's configuration information
+ from the system attempting to enforce some sanity in the universe
+ is something only a Bond villain would contemplate.
+
+ * res/res_pjsip/pjsip_configuration.c: Add rtpengine configuration
+ parameter The rtpengine configuration parameter was documented in
+ the XML documentation, but it was not actually registered with
+ the sorcery object. This adds the parameter with a default of
+ "asterisk", such that res_rtp_asterisk is chosen as the default
+ RTP implementation. (closes issue ASTERISK-22380) Reported by:
+ Rusty Newton Tested by: Rusty Newton
+
+2013-08-23 22:36 +0000 [r397614] Matthew Jordan <mjordan@digium.com>
+
+ * / (added): __________ | \ |_______ | | | ______| | / | _ _ _ _ _
+ | _______| / \ ___| |_ ___ _ __(_)___| | __ / || | / _ \ / __|
+ __/ _ \ '__| / __| |/ / | || |_______ / ___ \__ \| | __/ | | \__
+ \ < | || | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| \__________|
+
+2013-08-23 22:20 +0000 [r397613] Joshua Colp <jcolp@digium.com>
+
+ * main/bucket.c: Fix building of trunk. Note: This is why I commit
+ on the weekend.
+
+2013-08-23 22:12 +0000 [r397606] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c: Fix channel reference leak in Originated channels
+ When originating channels, ast_pbx_outgoing_* caused the dialed
+ channel reference to be bumped twice. Ostensibly, this routine is
+ bumping the channel lifetime such that the channel doesn't get
+ nuked in between locks/unlocks; however, since the routine should
+ return the dialed channel with its reference bumped, it only
+ needs to do this one time.
+
+2013-08-23 21:53 +0000 [r397603] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip.c: Add some clarifying documentation to the
+ rewrite_contact endpoint option.
+
+2013-08-23 21:51 +0000 [r397602] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge_channel.c: Blank line tweaks.
+
+2013-08-23 21:49 +0000 [r397599-397600] Joshua Colp <jcolp@digium.com>
+
+ * main/config_options.c, makeopts.in, main/asterisk.c,
+ include/asterisk/bucket.h (added), main/sorcery.c,
+ include/asterisk/config_options.h, tests/test_bucket.c (added),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, main/bucket.c
+ (added), configure.ac: Add the bucket API. Bucket is a URI based
+ API for the creation, retrieval, updating, and deletion of
+ "buckets" and files contained within them. Review:
+ https://reviewboard.asterisk.org/r/2715/
+
+ * include/asterisk/sorcery.h: Fix a bug where the argc value was
+ passed as no_doc when registering custom sorcery types. This also
+ adds a _nodoc equivalent.
+
+2013-08-23 21:02 +0000 [r397593] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridge_channel.c: Add test events necessary for bridge tests
+ to pass in the test suite. (closes issue AST-1200) reported by
+ John Bigelow Review: https://reviewboard.asterisk.org/r/2790/
+
+2013-08-23 20:14 +0000 [r397585] Matthew Jordan <mjordan@digium.com>
+
+ * main/stasis_channels.c: Fix error in using
+ ast_channel_snapshot_type before initialization Starting Asterisk
+ would kick back an ERROR message stating that the Stasis message
+ type ast_channel_snapshot_type was used prior to initialization.
+ This occurred due to the caching topic being created prior to the
+ message type that it depended on. This patch re-orders the start
+ up such that the message type is initialized prior to the caching
+ topic. It also checks the return value of the initialization of
+ the agent login/logoff types.
+
+2013-08-23 19:05 +0000 [r397578] Jonathan Rose <jrose@digium.com>
+
+ * bridges/bridge_native_rtp.c: bridge_native_rtp: Fix hold chain
+ bugs caused by native RTP bridge framehook Issuing hold/unhold
+ would lead to odd behavior. Between two chan_sip devices, a hold
+ could cause an endless chain of updates while with pjsip a
+ similar chain would begin but then end somewhat randomly. This
+ patch fixes that by no longer tweaking the RTP glue on both sides
+ of the call for every HOLD/UNHOLD/UPDATE_RTP_PEER frame. (issue
+ ASTERISK-22217) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2794/
+
+2013-08-23 18:33 +0000 [r397577] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_musiconhold.c, main/bridge_channel.c, main/channel.c,
+ include/asterisk/bridge_channel_internal.h, main/bridge.c,
+ include/asterisk/bridge_channel.h, main/channel_internal_api.c,
+ bridges/bridge_builtin_interval_features.c,
+ include/asterisk/channel.h: Handle DTMF and hold wrapup when a
+ channel leaves the bridging system. DTMF start/end and
+ hold/unhold events have state because a DTMF begin event and hold
+ event must be ended by something. The following cases need to be
+ handled when a channel is moved around in the system. * When a
+ channel leaves a bridge it may owe a DTMF end event to the
+ bridge. * When a channel leaves a bridge it may owe an UNHOLD
+ event to the bridge. (This case is explicitly ignored because
+ things like transfers need explicit control over this.) * When a
+ channel leaves the bridging system it may need to simulate a DTMF
+ end event to the channel. * When a channel leaves the bridging
+ system it may need to simulate an UNHOLD event to the channel.
+ The patch also fixes the following: * Fixes playing a file and
+ restarting MOH using the latest MOH class used. (closes issue
+ ASTERISK-22043) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2791/
+
+2013-08-23 18:10 +0000 [r397571] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_sorcery_astdb.c, tests/test_sorcery.c,
+ tests/test_sorcery_realtime.c: Fix sorcery unit tests When strict
+ XML documentation checking was re-enabled, the test objects used
+ in sorcery would fail to register as the types were not marked
+ internal and the nodoc option wasn't used for the options. This
+ fixes that problem, such that, as one would hope, they once again
+ pass.
+
+2013-08-23 18:07 +0000 [r397570] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, main/astobj2.c, include/asterisk/backtrace.h,
+ main/lock.c, include/asterisk/utils.h, include/asterisk/astmm.h,
+ /, main/backtrace.c, main/logger.c, main/utils.c,
+ include/asterisk/lock.h, main/astmm.c: Fix memory corruption when
+ trying to get "core show locks". Review
+ https://reviewboard.asterisk.org/r/2580/ tried to fix the
+ mismatch in memory pools but had a math error determining the
+ buffer size and didn't address other similar memory pool
+ mismatches. * Effectively reverted the previous patch to go in
+ the same direction as trunk for the returned memory pool of
+ ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols()
+ when BETTER_BACKTRACES is defined. * Fixed some formatting in
+ ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated
+ by libpri when MALLOC_DEBUG is enabled. * Fixed
+ __dump_backtrace() freeing memory from ast_bt_get_symbols() when
+ MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of
+ compile issues with the utils directory. (closes issue
+ ASTERISK-22221) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2778/ ........ Merged
+ revisions 397525 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397528 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-23 18:02 +0000 [r397568] Matthew Jordan <mjordan@digium.com>
+
+ * main/config_options.c: Prevent seg fault in off nominal path when
+ registered option fails to validate If an option is registered to
+ a type and it is the last known type in the list of registered
+ types, and the option fails to register, an overrun of the types
+ array can occur due to the index variable having been already
+ incremented.
+
+2013-08-23 17:45 +0000 [r397567] Kevin Harwell <kharwell@digium.com>
+
+ * contrib/scripts/sip_to_res_sip/astconfigparser.py,
+ contrib/scripts/sip_to_res_sip/astdicts.py,
+ contrib/scripts/sip_to_res_sip/sip_to_res_sip.py: PSJIP -
+ sip.conf to res_sip.conf script Most, if not all, of the backing
+ features of a conf file should now be implemented (e.g.
+ multi-line comments, includes, templates, etc...). A few of the
+ options still need to be mapped. Those are currently listed in
+ the 'sip_to_res_sip.py' file. Things to do: (1) There is more
+ work to do here, at least for the sip.conf items that aren't
+ currently parsed. An issue will be created for that. (2) All of
+ the scripts should probably be passed through pylint and have as
+ many PEP8 issues fixed as possible. (3) A public review is
+ probably warranted at that point of the entire script. Reported
+ by: Matt Jordan
+
+2013-08-23 17:19 +0000 [r397565] David M. Lee <dlee@digium.com>
+
+ * rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
+ res/res_ari_bridges.c, res/stasis/control.c,
+ include/asterisk/stasis_app.h,
+ include/asterisk/stasis_app_impl.h: ARI: Correct error codes for
+ bridge operations This patch adds error checking to ARI bridge
+ operations, when adding/removing channels to/from bridges. In
+ general, the error codes fall out as follows: * Bridge not found
+ - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel
+ not found - 400 Bad Request * Channel not in Stasis - 422
+ Unprocessable Entity * Channel not in this bridge (on remove) -
+ 422 Unprocessable Entity (closes issue ASTERISK-22036) Review:
+ https://reviewboard.asterisk.org/r/2769/
+
+2013-08-23 15:49 +0000 [r397524-397527] Matthew Jordan <mjordan@digium.com>
+
+ * CHANGES: Update CHANGES file to reflect pass through support for
+ Opus/VP8
+
+ * include/asterisk/format.h, channels/chan_pjsip.c,
+ res/res_format_attr_opus.c (added), main/channel.c,
+ main/format.c, res/res_rtp_asterisk.c, main/frame.c,
+ main/rtp_engine.c, channels/chan_sip.c, res/res_pjsip_sdp_rtp.c,
+ include/asterisk/opus.h (added): Add pass through support for
+ Opus and VP8; Opus format attribute negotiation This patch adds
+ pass through support for Opus and VP8. That includes: * Format
+ attribute negotiation for Opus. Note that unlike some other
+ codecs, the draft RFC specifies having spaces delimiting the
+ attributes in addition to ';', so you have "attra=X; attrb=Y".
+ This broke the attribute parsing in chan_sip, so a small tweak
+ was also included in this patch for that. * A format attribute
+ negotiation module for Opus, res_format_attr_opus * Fast picture
+ update for VP8. Since VP8 uses a different RTCP packet number
+ than FIR, this really is specific to VP8 at this time. Note that
+ the format attribute negotiation in res_pjsip_sdp_rtp was written
+ by mjordan. The rest of this patch was written completely by
+ Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/
+ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches:
+ asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero
+ (License 6518)
+
+ * main/sorcery.c, include/asterisk/config_options.h,
+ include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
+ main/config_options.c, main/features_config.c,
+ res/res_pjsip/pjsip_options.c, res/res_pjsip.c: Update config
+ framework/sorcery with types/options without documentation There
+ are times when a configuration option should not have
+ documentation. 1. Some options are registered with a particular
+ object merely as a warning to users. These options aren't even
+ really 'deprecated' - which has its own separate API call - they
+ are actually provided by a different configuration file. The
+ options are merely registered so that the user gets a warning
+ that a different configuration file provides the item. 2. Some
+ object types - most notably some used by modules that use sorcery
+ - are completely internal and should never be shown to the user.
+ 3. Sorcery itself has several 'hidden' fields that should never
+ be shown to a user. This patch updates the configuration
+ framework and sorcery with additional API calls that allow a
+ module to register types as internal and options as not requiring
+ documentation. This bypasses the XML documentation checking. This
+ patch also re-enables the strict XML documentation checking in
+ trunk, as well as updates some documentation that was missing.
+ Review: https://reviewboard.asterisk.org/r/2785/ (closes issue
+ ASTERISK-22359) Reported by: Matt Jordan (closes issue
+ ASTERISK-22112) Reported by: Rusty Newton
+
+2013-08-23 13:58 +0000 [r397515] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_pjsip.c: Fix crash when answering after a transport
+ error occurs. If a response to an initial incoming INVITE results
+ in a transport error the INVITE transaction is removed from the
+ INVITE session. Any attempts to answer the INVITE session after
+ this results in a crash as it requires the INVITE transaction to
+ exist. This change explicitly locks the dialog and checks to
+ ensure that the INVITE transaction exists before answering.
+ (closes issue AST-1203) Reported by: John Bigelow
+
+2013-08-23 13:18 +0000 [r397514] Kinsey Moore <kmoore@digium.com>
+
+ * configs/cel.conf.sample: Update CEL sample config
+
+2013-08-23 00:26 +0000 [r397505] Jonathan Rose <jrose@digium.com>
+
+ * res/res_stasis.c, rest-api/api-docs/bridges.json,
+ res/ari/resource_bridges.c, res/res_ari_bridges.c,
+ res/ari/resource_bridges.h, include/asterisk/stasis_app.h: ARI:
+ Music on Hold/Background Music for bridges Adds ARI functions to
+ be able to turn on/off music on hold in a bridge. It actually
+ functions more as a background music without further actions on
+ the bridge since if the rest of the channels in the bridge aren't
+ explicitly muted, they will still be able to communicate. (closes
+ issue ASTERISK-21974) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2688/
+
+2013-08-22 23:15 +0000 [r397494] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, bridges/bridge_holding.c, apps/app_followme.c:
+ Minor tweaks with ast_moh_start() callers.
+
+2013-08-22 22:33 +0000 [r397493] Kinsey Moore <kmoore@digium.com>
+
+ * CHANGES, apps/app_directory.c, apps/app_chanspy.c,
+ include/asterisk/say.h, apps/app_voicemail.c, main/channel.c,
+ main/pbx.c, main/say.c, res/res_agi.c: Add SayAlphaCase and
+ similar functionality for AGI This adds a new dialplan
+ application, SayAlphaCase, that performs much the same function
+ as SayAlpha except that it takes additional options which allow
+ the user to specify whether the case of each letter should be
+ announced for uppercase, lowercase, or all letters. Similar
+ functionality has been added to the SAY ALPHA AGI command via an
+ optional parameter. Original Patch by: Kevin Scott Adams Reported
+ by: Kevin Scott Adams Review:
+ https://reviewboard.asterisk.org/r/2725/ (closes issue
+ ASTERISK-20782)
+
+2013-08-22 22:09 +0000 [r397484] Kevin Harwell <kharwell@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_dtmf_info.c: res_sip_dtmf_info:
+ Support sending of 'raw' DTMF Added the ability to handle 'raw'
+ DTMF within the body of an INFO message. Also made it so values
+ 10-16 are mapped to valid DTMF values. (closes issue
+ ASTERISK-22144) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2776/
+
+2013-08-22 21:39 +0000 [r397483] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_pjsip.c: Add missing configOption close tags
+
+2013-08-22 21:29 +0000 [r397482] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/musiconhold.h: Update MOH start/stop routine
+ doxygen.
+
+2013-08-22 21:21 +0000 [r397481] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip.c: Fix missing xml doc configOption 'type' for for
+ both 'system' and 'global' configObjects (issue ASTERISK-22344)
+ (closes issue ASTERISK-22344)
+
+2013-08-22 21:09 +0000 [r397472] Richard Mudgett <rmudgett@digium.com>
+
+ * res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
+ res/res_parking.c, bridges/bridge_builtin_features.c,
+ include/asterisk/bridge_channel.h, main/features.c,
+ bridges/bridge_builtin_interval_features.c,
+ include/asterisk/bridge_internal.h, apps/app_confbridge.c,
+ main/bridge_channel.c, res/res_stasis.c,
+ include/asterisk/bridge.h, apps/app_dial.c, main/bridge.c,
+ main/bridge_basic.c, apps/app_bridgewait.c,
+ res/parking/parking_applications.c: Bridge API: Set a cause code
+ on a channel when it is ejected from a bridge. The cause code
+ needs to be passed from the disconnecting channel to the bridge
+ peers if the disconnecting channel dissolves the bridge. * Made
+ the call to an app_agent_pool agent disconnect with the busy
+ cause code if the agent does not ack the call in time or hangs up
+ before acking the call. (closes issue ASTERISK-22042) Reported
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/
+
+2013-08-22 20:29 +0000 [r397471] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c: Ensure CEL creates a default config if it isn't
+ provided with one
+
+2013-08-22 20:18 +0000 [r397466] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Remove set but unused variable 'meid'.
+
+2013-08-22 19:52 +0000 [r397461] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c: Fix crash when getting CEL config
+
+2013-08-22 18:52 +0000 [r397441-397451] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridge.c, main/bridge_basic.c, main/features.c, main/app.c,
+ main/core_local.c, CHANGES, apps/app_queue.c,
+ include/asterisk/bridge_basic.h, include/asterisk/core_unreal.h,
+ include/asterisk/features.h, include/asterisk/app.h: Massively
+ clean up app_queue. This essentially makes app_queue usable
+ again. From reviewboard: * Reporting of transfers and call
+ completion is done by creating stasis subscriptions and listening
+ for specific events in order to determine when the call is
+ finished (either via a transfer or hangup). * Dial end messages
+ have been added where they were previously missing. * Queue stats
+ are properly being updated again once calls have finished. *
+ AgentComplete stasis messages and AMI events are now occurring
+ again. * Mixmonitor starting has been factored into its own
+ function and uses the Mixmonitor API now instead of using
+ ast_pbx_run() In addition to the changes in app_queue, there are
+ several supplementary changes as well: * Queue logging now
+ differentiates between attended and blind transfers. A note about
+ this is in the CHANGES file. * Local channel optimization events
+ now report more information. This includes which of the two local
+ channels involved is the destination of the optimization, the
+ channel that is replacing the destination local channel, and an
+ identifier so that begin and end events can be matched to each
+ other. The end events are now sent whether the optimization was
+ successful or not and includes an indicator of whether the
+ optimization was successful. * Changes were made to features and
+ bridging_basic so that additional flags may be set on a bridge.
+ This is necessary because the queue requires that its bridge only
+ allows move-swap local channel optimizations into the bridge.
+ (closes issue ASTERISK-21517) Reported by Matt Jordan (closes
+ issue ASTERISK-21943) Reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2694
+
+ * res/res_pjsip_mwi.c, res/res_pjsip_pubsub.c,
+ res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h:
+ Handle default body types for SIP event packages in
+ res_pjsip_pubsub Prior to this change, we would reject SUBSCRIBE
+ requests that had no Accept headers. Now event package handlers
+ that handle the default type for the event package indicate that
+ they do so. Therefore, if we have a handler that can handle the
+ default type, we can allow SUBSCRIBEs for the handler's event
+ package that have no Accept headers. (closes issue
+ ASTERISK-22067) reported by Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2774
+
+2013-08-22 17:34 +0000 [r397440] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge_channel.c, main/abstract_jb.c: Made the abstract
+ jitter buffer resync on some more control frames. Resync the
+ abstract jitter buffer on the following additional control
+ frames: AST_CONTROL_HOLD AST_CONTROL_UNHOLD
+ AST_CONTROL_T38_PARAMETERS
+
+2013-08-22 17:13 +0000 [r397431] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c, include/asterisk/cel.h, tests/test_cel.c: Make CEL
+ behavior conform to the documentation This modifies the behavior
+ of the CEL engine to conform to documented behavior for Asterisk
+ 12 as defined on the wiki
+ https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification
+ The primary changes deal with removal of the peer field from
+ function calls since it is no longer directly relevant to the
+ bridging system and removal of the layer of CDR-like business
+ logic that was providing a partial emulation of Asterisk 11 CEL
+ functionality. With this change, there is no longer a distinction
+ between "bridges" and "conferences" and all participation changes
+ are denoted with bridge enter and bridge exit messages. This
+ updates the CEL unit tests to handle these changes and simplifies
+ some of the macros used in the process. This also fixes a
+ segfault when attempting to ref a configuration that failed to
+ load. Review: https://reviewboard.asterisk.org/r/2788/ (issue
+ ASTERISK-21567)
+
+2013-08-22 16:46 +0000 [r397426] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge.c: Update BUGBUG comment.
+
+2013-08-22 12:28 +0000 [r397379-397415] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/asterisk.c: Don't store repeated commands in the editline
+ history buffer. The equivalent of bash HISTCONTROL=ignoredups.
+ Review: https://reviewboard.asterisk.org/r/2775/
+
+ * /, main/asterisk.exports.in, default.exports: Add _IO_stdin_used
+ in version-script to fix SIGBUSes on Sparc. The
+ --version-script,asterisk.exports linker flag (and the module
+ exports) didn't provide _IO_stdin_used in the list of exported
+ symbols. That causes some kind of libc compatibility mode to kick
+ in, where stdio file structures (stdout/stderr) land somewhere
+ else. In the case of the Sparc, they landed on misaligned memory.
+ This became apparent first after r376428 (Reorder startup
+ sequence) when a lot of ast_log's were replaced with fprintf's.
+ Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
+ architectures, the Sparc is very picky about memory alignment.)
+ (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
+ Kister Review: https://reviewboard.asterisk.org/r/2760/ ........
+ Merged revisions 397377 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397378 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-21 23:09 +0000 [r397366] Jonathan Rose <jrose@digium.com>
+
+ * main/udptl.c, /: UDPTL: Fix a regression where UDPTL won't load
+ default settings If the file udptl.conf is unavailable at
+ startup, UDPTL will fail to initialize and while it makes some
+ noise, it isn't immediately obvious why consumers start to fail
+ when using it. This patch makes UDPTL load as though an empty
+ config was provided when udptl is unavailable at startup. (closes
+ issue ASTERISK-22349) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2773/ ........ Merged
+ revisions 397365 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-21 20:02 +0000 [r397346-397355] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridge_basic.h, main/bridge_basic.c,
+ main/features.c: * Move ast_bridge_channel_setup_features() into
+ bridge_basic.c. * Made application map hooks be removed on a
+ basic bridge personality change.
+
+ * main/bridge.c, main/bridge_channel.c: Deferred some more BUGBUG
+ comments to a JIRA issue or XXX comment.
+
+2013-08-21 17:12 +0000 [r397310] David M. Lee <dlee@digium.com>
+
+ * main/http.c, /: Complete http_shutdown. This patch frees up some
+ resources allocated in http.c. * tcp listeners stopped * tls
+ settings freed * uri redirects freed * unregister internal http.c
+ uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
+ Patches: http.patch uploaded by Corey Farrell (license 5909)
+ ........ Merged revisions 397308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397309 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-21 16:31 +0000 [r397307] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/frame.h, /: Set 14400 as the default max bit
+ rate if T38MaxBitRate is not specified If an endpoint fails to
+ include the T38MaxBitRate attribute during negotiation, Asterisk
+ will negotiate a bit rate of 2400 instead of the ITU recommended
+ bit rate of 14400. This patch fixes this by making
+ AST_T38_RATE_14400 the 'default' value of the enum by assigning
+ it a value of 0, such that if an endpoint fails to include the
+ attribute, the default will be 14400. Note that Walter Doekes
+ included the nice comment in frame.h about why we are
+ purposefully assigning AST_T38_RATE_14400 a value of 0. (closes
+ issue ASTERISK-22275) Reported by: Andreas Steinmetz patches:
+ fax-fix.patch uploaded by anstein (License 6523) ........ Merged
+ revisions 397256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397257 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-21 16:23 +0000 [r397295-397306] David M. Lee <dlee@digium.com>
+
+ * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
+ res/res_ari_channels.c, rest-api/api-docs/asterisk.json,
+ res/ari/resource_asterisk.c, res/res_ari_asterisk.c: ARI: Correct
+ segfault with /variable calls are missing ?variable parameter.
+ Both /asterisk/variable and /channel/{channelId}/variable
+ requires a ?variable parameter to be passed into the query. But
+ we weren't checking for the parameter being missing, which caused
+ a segfault. All calls now properly return 400 Bad Request errors
+ when the parameter is missing. The Swagger api-docs were updated
+ accordingly. (closes issue ASTERISK-22273)
+
+ * main/stasis_endpoints.c: ARI: Remove the 'channel:' scheme from
+ endpoint's channel list. For times when a reference in ARI might
+ be ambiguous, the reference is built as an URI (such as
+ channel:1376341790.3). An endpoint's channel list is not
+ ambiguous, and in fact the field is named 'channel_ids', but it
+ had channel URI's instead of channel id's. This patch changes the
+ list to be the raw id instead of the URI. (closes issue
+ ASTERISK-22291)
+
+ * res/stasis/control.h, res/res_stasis.c: res_stasis: remove call
+ to missing function control_continue. In the shuffling around of
+ res_stasis, control_continue was renamed to
+ stasis_app_control_continue, but the call in res_stasis wasn't
+ updated. In looking into it, it turns out it wasn't really the
+ right thing to do in res_stasis anyways. This patch changes the
+ handling of received a AST_CONTROL_HANGUP frame to be the same as
+ receiving a NULL frame, and removed the declaration of
+ control_continue(), since it doesn't exist any more. (closes
+ issue ASTERISK-22292) Reported by: Denis Smirnov
+
+2013-08-21 15:51 +0000 [r397294] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge_channel.c, res/parking/parking_bridge_features.c,
+ apps/app_agent_pool.c, bridges/bridge_holding.c, main/bridge.c,
+ include/asterisk/bridge_channel.h, main/features.c,
+ bridges/bridge_builtin_interval_features.c,
+ apps/app_bridgewait.c, include/asterisk/bridge_features.h: Fix
+ several interrelated issues dealing with the holding bridge
+ technology. * Added an option flags parameter to interval hooks.
+ Interval hooks now can specify if the callback will affect the
+ media path or not. * Added an option flags parameter to the
+ bridge action custom callback. The action callback now can
+ specify if the callback will affect the media path or not. * Made
+ the holding bridge technology reexamine the participant idle mode
+ option whenever the entertainment is restarted. * Fixed
+ app_agent_pool waiting agents needlessly starting and stopping
+ MOH every second by specifying the heartbeat interval hook as not
+ affecting the media path. * Fixed app_agent_pool agent alert from
+ restarting the MOH after the alert beep. The agent entertainment
+ is now changed from MOH to silence after the alert beep. * Fixed
+ holding bridge technology to defer starting the entertainment. It
+ was previously a mixture of immediate and deferred. * Fixed
+ holding bridge technology to immediately stop the entertainment.
+ It was previously a mixture of immediate and deferred. If the
+ channel left the bridging system, any deferred stopping was
+ discarded before taking effect. * Miscellaneous holding bridge
+ technology rework coding improvements. Review:
+ https://reviewboard.asterisk.org/r/2761/
+
+2013-08-21 14:39 +0000 [r397255] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Prevent a crash on outbound SIP MESSAGE
+ requests. If a From header on an outbound out-of-call SIP MESSAGE
+ were malformed, the result could crash Asterisk. In addition, if
+ a From header on an incoming out-of-call SIP MESSAGE request were
+ malformed, the message was happily accepted rather than being
+ rejected up front. The incoming message path would not result in
+ a crash, but the behavior was bad nonetheless. (closes issue
+ ASTERISK-22185) reported by Zhang Lei ........ Merged revisions
+ 397254 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-21 14:08 +0000 [r397244] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_stasis.c: Allow channels in app_stasis to hangup properly
+ This detects hangups that occur while bridged to allow channels
+ to exit app_stasis even if the hangup frame was absorbed by the
+ bridge the channel was in. Reported by: David Lee (closes issue
+ ASTERISK-22297)
+
+2013-08-21 13:41 +0000 [r397243] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Allow the SIP_CODEC family of
+ variables to specify more than one codec The SIP_CODEC family of
+ variables let you set the preferred codec to be offered on an
+ outbound INVITE request. However, for video calls, you need to be
+ able to set both the audio and video codecs to be offered. This
+ patch lets the SIP_CODEC variables accept a comma delineated list
+ of codecs. The first codec in the list is set as the preferred
+ codec; additional codecs are still offered however. This lets a
+ dialplan writer set both audio and video codecs, e.g.,
+ Set(SIP_CODEC=ulaw,h264) Note that this feature was written by
+ both Dennis Guse and Frank Haase Review:
+ https://reviewboard.asterisk.org/r/2728 (closes issue
+ ASTERISK-21976) Reported by: Denis Guse Tested by: mjordan,
+ sysreq patches: patch-channels-chan__sip.c-393919 uploaded by
+ dennis.guse (license 6513)
+
+2013-08-21 02:15 +0000 [r397206] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Not Storing Current Incoming Recv
+ Address In 1.8, r384779 introduced a regression by retrieving an
+ old dialog and keeping the old recv address since recv was
+ already set. This has caused a problem when a proxy is involved
+ since responses to incoming requests from the proxy server, after
+ an outbound call is established, are never sent to the correct
+ recv address. In 11, r382322 introduced this regression. The fix
+ is to revert that change and always store the recv address on
+ incoming requests. Thank you Walter Doekes for helping to point
+ out this error and Mark Michelson for your input/review of the
+ fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
+ Tested by: Alex Zarubin, Karsten Wemheuer Patches:
+ asterisk-22071-store-recvd-address.diff by Michael L. Young
+ (license 5026) ........ Merged revisions 397204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397205 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-20 21:01 +0000 [r397111-397193] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/config_security.c (removed),
+ res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c,
+ include/asterisk/res_pjsip.h: Localize and rename ACL
+ configuration. This is more-or-less a reversion of previous ACL
+ behavior so that it is more self-contained. ACL sections are now
+ only parsed if res_pjsip_acl.so is loaded. Moreover, the
+ configuration section is now "type=acl" instead of
+ "type=security". The original reason for having ACLs configured
+ in a "type=security" section was to lump ACLs and other
+ security-related items into the same section. The problem is that
+ ACLs really should be in their own sections and there are no
+ other security-related options implemented anyways.
+
+ * /, channels/chan_sip.c: Remove REF_DEBUG definition. ........
+ Merged revisions 397156 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397157 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/sip/dialplan_functions.c, /, channels/chan_sip.c: Fix
+ refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
+ the list of pvts. (closes issue ASTERISK-22248) reported by Corey
+ Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
+ (license #5909) ........ Merged revisions 397112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397133 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip.c: Clarify documentation for the "identify_by"
+ option for SIP endpoints. This also removes documentation for the
+ options that no longer exist. (closes issue ASTERISK-22306)
+ reported by Rusty Newton
+
+2013-08-20 15:36 +0000 [r397110] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/threadstorage.c, main/astfd.c: Unregister CLI commands on
+ exit This patch ensures that CLI commands enabled by
+ DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
+ exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
+ Tested by: Corey Farrell Patches: debug_cli_unregister.patch
+ uploaded by Corey Farrell ........ Merged revisions 397106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397107 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-20 15:32 +0000 [r397073-397109] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: Add debug message to
+ res_pjsip_endpoint_identifier_ip to indicate when an endpoint is
+ successfully retrieved. (closes issue ASTERISK-22101) reported by
+ Rusty Newton
+
+ * res/res_pjsip_registrar.c: Add warning messages for registration
+ failure paths. (closes issue ASTERISK-22089) reported by Rusty
+ Newton patches: patch1.txt uploaded by John Bigelow (License
+ #5091)
+
+ * res/res_pjsip.c: Add note to transport configuration that a
+ restart is required to change transports. (closes issue
+ ASTERISK-22094) reported by Rusty Newton
+
+2013-08-20 14:26 +0000 [r397072] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 397067 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix
+ xmldoc memory leak This fixes a single-attribute memory leak that
+ was occurring when the "required" attribute was not true. (closes
+ issue ASTERISK-22249) Reported by: Corey Farrell Tested by: Corey
+ Farrell Patches: xmldoc-free_attr_required.patch uploaded by
+ Corey Farrell ........ Merged revisions 397064 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 11:48 +0000 [r396996] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, configs/sip.conf.sample, configs/h323.conf.sample: Add
+ "autoframing" option to sip.conf.sample and h323.conf.sample. The
+ autoframing option was added to chan_sip.c in r43243 (mogorman,
+ 2006-09-19 01:32:57), but never made its way into the sample
+ configs. Review: https://reviewboard.asterisk.org/r/2768/
+ ........ Merged revisions 396994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396995 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-20 11:33 +0000 [r396993] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_dtmf_info.c: Remove assumption in
+ res_pjsip_dtmf_info that all INFO messages will contain a body.
+ (closes issue ASTERISK-22320) Reported by: Matt Jordan
+
+2013-08-20 00:08 +0000 [r396946-396949] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c, /: Let Queue wrap up time influence member
+ availability Queue members who happen to be in multiple queues at
+ the same time may not have any wrap up time. This problem
+ occurred due to a code change in Asterisk 11.3.0 that unified
+ device state tracking of Queue members in multiple Queues (which
+ fixed some other problems, but unfortunately caused this one).
+ This patch fixes the behavior by having the is_member_available
+ function check the queue's wrap up time and the time of the
+ member's last call, such that for a particular queue, the member
+ won't be considered available if their last call is within the
+ wrap up time. (closes issue ASTERISK-22189) Reported by: Tony
+ Lewis Tested by: Tony Lewis ........ Merged revisions 396948 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_meetme.c: Resolve conflicts between
+ CONFFLAG_DONT_DENOISE and CONFFLAG_INTROUSER_VMREC When r382230
+ added an option to not denoise the MeetMe conference (if a user
+ had a channel whose format's sample rate changed frequently, for
+ example), the value added was the maximum allowed value for the
+ constants that define the options for MeetMe in 1.8. Not so in 11
+ - unfortunately, the option CONFFLAG_DONT_DENOISE conflicts with
+ CONFFLAG_INTROUESR_VMREC. This patch fixes that, and also tweaks
+ one of the way in which the constants was declared for
+ consistency. Thanks to Tony Mountifield for pointing out the
+ problem and solution. (closes issue ASTERISK-22269) Reported by:
+ Tony Mountifield ........ Merged revisions 396944 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-19 16:10 +0000 [r396930] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge.c: Update BUGBUG comment.
+
+2013-08-19 14:54 +0000 [r396923] Jonathan Rose <jrose@digium.com>
+
+ * main/bridge.c: attended transfers: Fix a bug affecting external
+ blond transfers Performing a blond transfer (attended transfer
+ that is completed before the transfer recipient picks up)
+ externally through chan_sip or chan_pjsip would result in lost
+ references to the channels involved with the transfer as well as
+ their bridge. (closes issue ASTERISK-22092) Reported by:
+ mmichelson Review: https://reviewboard.asterisk.org/r/2766/
+
+2013-08-19 14:53 +0000 [r396915-396922] Matthew Jordan <mjordan@digium.com>
+
+ * channels/sip/include/sip.h: Whitespace cleanup Remove some
+ extraneous blobs
+
+ * main/data.c: Fix invalid access to disposed memory in main/data
+ unit test It is not safe to iterate over a macro'd list of ao2
+ objects, deref them such that the item's destructor is called,
+ and leave them in the list. The list macro to iterate over items
+ requires the item to be a valid allocated object in order to
+ proceed to the next item; with MALLOC_DEBUG on the corruption of
+ the linked list is caught in the crash. This patch fixes the
+ invalid access to free'd memory by removing the ao2 item from the
+ list before de-refing it.
+
+2013-08-18 03:05 +0000 [r396908-396909] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_mgcp.c: Update chan_mgcp to the modified parking
+ API
+
+ * res/res_corosync.c: Disable build of res_corosync until it is
+ back in a compiling state
+
+2013-08-17 18:13 +0000 [r396899-396902] Rusty Newton <rnewton@digium.com>
+
+ * res/res_pjsip.c: xml doc changes for 'aor' config object and a
+ few of its options Added or modified text in the xml doc for the
+ 'aor' config object to address a few issues: * help for the
+ 'mailboxes' option didn't make it clear how the "list" should be
+ formatted. * AoR object's involvement in inbound registration
+ wasn't mentioned. * help for the 'contact' option didn't describe
+ how to specify multiple contacts. * help for the 'max_contacts'
+ option didn't tell whether it limited the amount of contacts
+ defined through static configuration. (issue ASTERISK-22118)
+ (closes issue ASTERISK-22118)
+
+ * res/res_pjsip.c: 'domain_alias' config object XML help doesn't
+ make it clear that the name used for the object is the domain
+ alias (issue ASTERISK-22114) (closes issue ASTERISK-22114)
+
+ * res/res_pjsip.c: xml doc changes for clarity - 'auth' config
+ object and auth's 'auth_type' config option (issue
+ ASTERISK-22108) (closes issue ASTERISK-22108)
+
+ * res/res_pjsip.c: xml doc change for transport config object -
+ remove non-applicable warning and add text regarding Asterisk
+ restart (closes issue ASTERISK-22105)
+
+2013-08-17 15:01 +0000 [r396887-396890] Kinsey Moore <kmoore@digium.com>
+
+ * res/parking/parking_applications.c, include/asterisk/parking.h,
+ main/bridge_channel.c, res/parking/parking_bridge_features.c,
+ channels/chan_dahdi.c, res/parking/res_parking.h,
+ res/res_parking.c, channels/sig_analog.c, channels/chan_skinny.c,
+ main/parking.c, main/bridge.c: Allow res_parking to be unloadable
+ This change protects accesses of res_parking such that it can
+ unload safely once transient uses of its registered functions are
+ complete. The parking API has been restructured such that its
+ consumers do not have access to the vtable exposed by the parking
+ provider, but instead route through stubs to prevent consumers
+ from holding on to function pointers. This adds calls to all the
+ parking unload functions and moves application loading and
+ unloading into functions in parking_applications.c similar to the
+ rest of the parts of res_parking. Review:
+ https://reviewboard.asterisk.org/r/2763/ (closes issue
+ ASTERISK-22142)
+
+ * tests/test_cel.c, cel/cel_sqlite3_custom.c, main/event.c,
+ main/asterisk.c, cel/cel_pgsql.c, cel/cel_radius.c,
+ include/asterisk/cel.h, cel/cel_tds.c, tests/test_event.c,
+ include/asterisk/_private.h, main/cel.c, cel/cel_odbc.c,
+ include/asterisk/event.h, include/asterisk/event_defs.h,
+ cel/cel_manager.c, cel/cel_custom.c: Refactor CEL to avoid using
+ the event system core This removes usage of the event system for
+ CEL backend data distribution and strips unused pieces out of the
+ event system. Review: https://reviewboard.asterisk.org/r/2732/
+
+ * channels/sig_pri.c, channels/chan_iax2.c, apps/app_queue.c,
+ res/res_jabber.c, main/presencestate.c, channels/sig_pri.h,
+ res/res_parking.c, channels/chan_dahdi.c, main/manager.c,
+ funcs/func_presencestate.c, include/asterisk/event.h,
+ include/asterisk/event_defs.h, channels/chan_skinny.c,
+ tests/test_cel.c, main/event.c,
+ include/asterisk/security_events_defs.h,
+ res/parking/parking_manager.c, channels/chan_mgcp.c,
+ res/res_security_log.c, apps/app_voicemail.c,
+ res/parking/parking_ui.c, channels/chan_unistim.c, main/pbx.c,
+ include/asterisk/devicestate.h, main/security_events.c,
+ channels/chan_sip.c, main/ccss.c, tests/test_event.c,
+ main/devicestate.c, res/parking/parking_applications.c,
+ res/res_xmpp.c: Strip down the old event system This removes
+ unused code, event types, IE pltypes, and event IE types where
+ possible and makes several functions private that were once
+ public. This includes a renumbering of the remaining event and IE
+ types which breaks binary compatibility with previous versions.
+ The last remaining consumers of the old event system (or parts
+ thereof) are main/security_events.c, res/res_security_log.c,
+ tests/test_cel.c, tests/test_event.c, main/cel.c, and the CEL
+ backends. Review: https://reviewboard.asterisk.org/r/2703/
+ (closes issue ASTERISK-22139)
+
+2013-08-16 20:48 +0000 [r396849-396877] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridge.h, main/bridge.c,
+ include/asterisk/bridge_channel.h, main/bridge_channel.c: Fix CLI
+ "bridge kick <bridge> <channel>" to check if the bridge needs
+ dissolving. SIP/foo -- Local;1==Local;2 -- .... --
+ Local;1==Local;2 -- SIP/bar Kick a ;1 channel and the chain
+ toward SIP/foo goes away. Kick a ;2 channel and the chain toward
+ SIP/bar goes away. This can leave a local channel chain between
+ the kicked ;1 and ;2 channels that are orphaned until you
+ manually request one of those channels to hangup or request the
+ bridge to dissolve. * Added ast_bridge_kick() as a companion to
+ ast_bridge_remove(). The functional difference is that
+ ast_bridge_kick() may dissolve the bridge as a result of the
+ channel leaving the bridge. * Made CLI "bridge kick <bridge>
+ <channel>" use ast_bridge_kick() instead of ast_bridge_remove()
+ so the bridge can dissolve if needed. * Renamed
+ bridge_channel_handle_hangup() to ast_bridge_channel_kick() and
+ made it accessible to other files.
+
+ * include/asterisk/doxygen/architecture.h,
+ include/asterisk/bridge_channel_internal.h: Fix some doxygen
+ bridging file references.
+
+ * apps/app_queue.c, main/indications.c,
+ res/parking/parking_bridge_features.c, main/cdr.c, main/data.c,
+ main/manager.c, tests/test_jitterbuf.c, main/features.c,
+ tests/test_voicemail_api.c, main/file.c, tests/test_cel.c,
+ main/stasis_channels.c, main/bridge_channel.c, main/message.c,
+ tests/test_cdr.c, main/db.c, main/xmldoc.c, main/format.c,
+ res/res_rtp_asterisk.c, main/pbx.c, main/rtp_engine.c,
+ tests/test_abstract_jb.c, channels/chan_sip.c, main/pickup.c:
+ Doxygen comment tweaks.
+
+ * main/hashtab.c, main/utils.c: Fix utilities compilation/linking.
+ The horrid structure of the source in the utils directory strikes
+ again. Moved the _ast_mem_backtrace_buffer[] definition from the
+ logical location in utils.c to hashtab.c so the aelparse and
+ conf2ael utilities can link.
+
+ * include/asterisk/utils.h: utils.h: Minor formatting tweaks.
+
+2013-08-16 16:03 +0000 [r396842] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/astobj2.h, main/stasis_channels.c,
+ tests/test_stasis.c, main/stasis.c, main/stasis_cache_pattern.c,
+ main/stasis_cache.c: Stasis: address refcount races;
+ implementation comments Change r395954 reordered some stasis
+ object destruction, which should have been fine. Unfortunately,
+ it caused some hard to reproduce issues related to objects being
+ accessed after they had been destroyed. The patch in r396329
+ fixed the destruction order problem; this patch addresses the
+ underlying issue. A few other stasis-related fixes were also
+ added. * Add ref-bumps around areas where objects may get
+ transitively destroyed. (For example, where we lock a topic,
+ unref a subscription, which unrefs the topic, which explodes the
+ topic when we try to unlock it.) * Wrote an extensive doxygen
+ page about Stasis implementation, relationships between objects,
+ lifecycles of objects, how the refcounting works, etc. Many other
+ comments were added, corrected, or cleaned up. * Added an assert
+ to the topic dtor to catch extra ref decrements. * Fixed type
+ used after destruction errors for graceful shutdown in
+ stasis_channels.c. * I added two unit tests in an attempt to
+ catch destruction order issues. Since the underlying cause is a
+ race condition, though, the tests rarely failed even when the
+ code was wrong. * Fixed a leak in stasis_cache_pattern.c. (closes
+ issue ASTERISK-22243) Review:
+ https://reviewboard.asterisk.org/r/2746/
+
+2013-08-16 12:20 +0000 [r396829] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, main/utils.c, main/sounds_index.c: Improve sounds
+ indexer CLI commands This reworks the CLI commands used to access
+ sounds information from "sounds show[ soundid]" to "core show
+ sounds" and "core show sound <soundid>". This also reworks the
+ "sounds reload" CLI command to fall under normal module reloading
+ ("module reload sounds"). Also, make trunk build when
+ DEBUG_MALLOC is not enabled. Review:
+ https://reviewboard.asterisk.org/r/2745/ (closes issue
+ ASTERISK-22141)
+
+2013-08-16 07:18 +0000 [r396822] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/utils.c, include/asterisk/utils.h, main/pbx.c: Prevent heap
+ alloc functions from running out of stack space. When asterisk
+ has run out of memory (for whatever reason), the alloc function
+ logs a message. Logging requires memory. A recipe for infinite
+ recursion. Stop the recursion by comparing the function call
+ depth for sane values before attempting another OOM log message.
+ Review: https://reviewboard.asterisk.org/r/2743/
+
+2013-08-15 22:10 +0000 [r396783-396814] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridge_channel.c: Bridge: Don't suspend/unspend the channel
+ for interception routines. By their nature, the connected line
+ and redirecting interception routines are not supposed to affect
+ the channel's media. Therefore, they should not suspend and
+ unsuspend the channel while running. The suspend/unsuspend
+ operations could be expensive depending upon the bridge and
+ channel technology involved.
+
+ * res/parking/res_parking.h, res/res_parking.c,
+ res/parking/parking_tests.c, main/features.c: Minor parking
+ cleanup.
+
+ * res/parking/parking_bridge_features.c: Parking: Eliminate local
+ channel name hack to get peer channel. (closes issue
+ ASTERISK-22034) Reported by: Matt Jordan
+
+ * main/features.c, main/bridge_channel.c: Remove early bridge
+ BUGBUG comments. Remove some unneeded features.c comments.
+
+ * configs/features.conf.sample: Update features.conf.sample
+ atxferdropcall option.
+
+ * main/bridge_channel.c, apps/confbridge/conf_config_parser.c,
+ main/bridge.c, include/asterisk/bridge_channel.h,
+ main/config_options.c: Changed some BUGBUG tags to associated
+ JIRA issue tags.
+
+ * bridges/bridge_softmix.c, include/asterisk/bridge.h,
+ main/bridge.c, main/features.c: Resolve some BUGBUG comments.
+
+2013-08-15 16:37 +0000 [r396747] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/asterisk.c, main/cli.c: Remove leading spaces from the
+ CLI command before parsing If you've mistakenly put a space
+ before typing in a command, the leading space will be included as
+ part of the command, and the command parser will not find the
+ corresponding command. This patch rectifies that situation by
+ stripping the leading spaces on commands. Review:
+ https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
+ Lesher ........ Merged revisions 396745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396746 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-15 15:12 +0000 [r396732-396734] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c, include/asterisk/channel.h,
+ channels/chan_iax2.c, channels/chan_vpb.cc: Remove some dead code
+ dealing with: AST_BRIDGE_REC_CHANNEL_0, AST_BRIDGE_REC_CHANNEL_1,
+ and AST_BRIDGE_IGNORE_SIGS.
+
+ * main/bridge_channel.c,
+ include/asterisk/bridge_channel_internal.h, main/manager.c: Fix
+ Bridge API DTMF hook matching for begin and end DTMF events. The
+ Bridge API DTMF hook matching would not deal with DTMF end events
+ only. It required a DTMF begin event to start matching the DTMF
+ hooks. There are many places in Asterisk where code only
+ generates DTMF end events without the corresponding begin event.
+ One such place is the AMI action Atxfer. * Fixed DTMF hook
+ matching if there is a string of DTMF frames in the read queue.
+ We could potentially miss some of them before. * Fixed AMI Atxfer
+ action documentation. (closes issue ASTERISK-22037) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/2752/
+
+2013-08-15 12:17 +0000 [r396722-396724] Kinsey Moore <kmoore@digium.com>
+
+ * main/bridge.c, main/features.c, apps/app_confbridge.c: Fix
+ feature_attended_transfer test The feature_attended_transfer test
+ is failing due to Asterisk not passing DTMF in the bridges
+ created for internal attended transfers. This sets the features
+ initialization routine to set this flag by default and adjusts
+ the basic bridge and confbridge's use of the bridging system
+ accordingly as per Richard's suggestion instead of adjusting this
+ individual case. This change allows the necessary DTMF to pass
+ through the attended transfer bridge and complete the test
+ successfully. Review: https://reviewboard.asterisk.org/r/2759/
+ (closes issue ASTERISK-22222)
+
+ * channels/chan_sip.c, main/utils.c, include/asterisk/lock.h: Fix
+ deadlocks in chan_sip in REFER and BYE handling This resolves
+ several deadlocks in chan_sip relating to usage of
+ ast_channel_bridge_peer and improves accessibility of lock
+ debugging function calls. Review:
+ https://reviewboard.asterisk.org/r/2756/ (closes issue
+ ASTERISK-22215)
+
+ * res/res_stasis.c: Prevent automagic things from happening to
+ Stasis application bridges This prevents swap optimization,
+ merges, and transfers involving Stasis application bridges. It
+ wouldn't be nice if the bridge you thought you owned disappeared
+ from under you. Reported-by: Richard Mudgett
+
+2013-08-15 00:16 +0000 [r396695-396713] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_vpb.cc, include/asterisk/channel.h:
+ Remove unsupported channel technology callbacks.
+
+ * channels/chan_vpb.cc: chan_vpb: Effectively remove native
+ support. Left enough bread crumbs to be able to convert later if
+ needed.
+
+ * channels/chan_iax2.c: chan_iax2: Conditionally remove native
+ support for now. (issue ASTERISK-21944)
+
+ * channels/chan_misdn.c: chan_misdn: Effectively remove native
+ support. Left enough bread crumbs to be able to convert later if
+ needed.
+
+ * apps/app_bridgewait.c: app_bridgewait: Inhibit local channel
+ optimizations to the bridge. Holding bridges can allow local
+ channel move/swap optimization to the bridge. However, we cannot
+ allow it for the BridgeWait holding bridge because the call will
+ lose the channel roles and dialplan location as a result.
+
+2013-08-14 19:06 +0000 [r396621-396658] Joshua Colp <jcolp@digium.com>
+
+ * /, tests/test_hashtab_thrash.c: Tweak comment for why usleep is
+ used. ........ Merged revisions 396656 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396657 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
+ allow the critical threads to execute. Depending on certain
+ conditions it was possible for the hashtab counting thread to
+ starve other threads, preventing them from executing in the
+ expected fashion. This change adds a sleep to allow the others to
+ do what they need to do. While this doesn't thrash the hashtab as
+ much as previously, it at least works. (closes issue
+ ASTERISK-22276) Reported by: Matt Jordan ........ Merged
+ revisions 396619 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396620 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-13 18:47 +0000 [r396581-396584] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Convert 'just did sched_add
+ waitid...' from warning to debug message. Patches:
+ reviewboard-2377.patch uploaded by Paul Belanger Review:
+ https://reviewboard.asterisk.org/r/2377/ ........ Merged
+ revisions 396582 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396583 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
+ rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
+ by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
+ ........ Merged revisions 396579 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396580 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-13 15:27 +0000 [r396559-396568] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis.c, res/ari/resource_bridges.c,
+ res/res_stasis_bridge_add.c (removed), res/res_stasis_playback.c,
+ res/stasis/control.c, res/res_stasis_bridge_add.exports.in
+ (removed), include/asterisk/stasis_app.h,
+ include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c,
+ res/stasis/control.h, include/asterisk/bridge_internal.h,
+ include/asterisk/bridge_features.h: ARI: allow other operations
+ to happen while bridged This patch changes ARI bridging to allow
+ other channel operations to happen while the channel is bridged.
+ ARI channel operations are designed to queue up and execute
+ sequentially. This meant, though, that while a channel was
+ bridged, any other channel operations would queue up and execute
+ only after the channel left the bridge. This patch changes ARI
+ bridging so that channel commands can execute while the channel
+ is bridged. For most operations, things simply work as expected.
+ The one thing that ended up being a bit odd is recording. The
+ current recording implementation will fail when one attempts to
+ record a channel that's in a bridge. Note that the bridge itself
+ may be recording; it's recording a specific channel in the bridge
+ that fails. While this is an annoying limitation, channel
+ recording is still very useful for use cases such as voice mail,
+ and bridge recording makes up much of the difference for other
+ use cases. (closes issue ASTERISK-22084) Review:
+ https://reviewboard.asterisk.org/r/2726/
+
+ * tests/test_hashtab_thrash.c: Missed a spot in r396559
+
+ * tests/test_hashtab_thrash.c: Fix build warnings when printf a
+ tv_usec. The debug logs added in r396528 neglected to account for
+ suseconds_t being an int. See r392076 for more info.
+
+2013-08-12 22:05 +0000 [r396552] John Bigelow <jbigelow@digium.com>
+
+ * res/res_pjsip_registrar.c: Add test suite events for when
+ contacts are added or removed from an AOR These are needed by the
+ pjsip inbound registration test suite tests. (issue
+ ASTERISK-21833) (issue ASTERISK-21834) (issue ASTERISK-21835)
+ (issue ASTERISK-21837) Review:
+ https://reviewboard.asterisk.org/r/2700/ Review:
+ https://reviewboard.asterisk.org/r/2739/
+
+2013-08-12 15:59 +0000 [r396542-396543] Matthew Jordan <mjordan@digium.com>
+
+ * main/bridge_channel.c, main/bridge.c, main/features.c: Fix two
+ race conditions and ref counting issue when joining a bridge
+ These problems were all caught by a test in the Asterisk Test
+ Suite that originated some Local channels and attempted to move
+ the ;2 half of the Local channel into a bridge using the Bridge
+ AMI action. (1) When originating a channel, the Newchannel event
+ is emitted quickly; however, the ;2 channel will not have a pbx
+ thread assigned to it until after the outbound 'dialing' for the
+ ;1 is complete. Thus, there is a period of time where the outside
+ world "knows" of the channel's existence and can influence it but
+ Asterisk has not yet started the dialplan execution thread. If a
+ Bridge AMI action is taken on the channel, the channel appears to
+ be a Dialed channel with no PBX thread; hence, the channel will
+ be imparted into the Bridge by first 'yanking' the channel. At
+ the same time, a race condition can occur after the yank (but
+ before entering the bridge) when ;1 answers and starts a PBX on
+ the ;2. The end result currently is an assertion failure in the
+ Bridging API, as a channel with a PBX is imparted into the
+ Bridge. There's no way to prevent AMI from attempting to Bridge a
+ channel immediately after creation; likewise, holding the channel
+ lock through the entire Dial operation is unwise (and
+ impossible). Instead of treating the presence of a PBX thread as
+ an error, we simply bail out of the adding the channel to the
+ bridge through ast_bridge_impart. The Bridge action will then
+ fail - but we avoid a situation where the channel is both
+ executing a PBX thread and simultaneously being given a separate
+ thread in the bridging system (which would be a "bad thing").
+ Since imparting a channel with a PBX *can* occur and is not a
+ programming error, the asserts have been removed. (2) When the
+ first condition occurs, we have to take one of two actions:
+ either hangup the yanked channel as it did not enter the bridge,
+ or deref it because we don't own it. We can determine if we own
+ it or not by testing for the presence of the PBX thread. If we
+ hung it up directly, we'd crash. (3) bridge_find_channel does not
+ increase the reference count of the ast_bridge_channel object.
+ The RAII_VAR usage in ast_bridge_add_channel thus created a
+ ticking time bomb in whatever bridge the channel moved into, as
+ the destructor for the ast_bridge_channel object would be called.
+ Review: https://reviewboard.asterisk.org/r/2741/
+
+ * main/pbx.c: Unlock outgoing dial lock on off nominal path If the
+ thread servicing the dial request isn't created successfully, the
+ outgoing dial lock will still be held when the function returns.
+ This patch unlocks the lock on this off nominal path.
+
+2013-08-10 20:29 +0000 [r396521-396535] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_hashtab_thrash.c: Pipe test output through test object
+ not stdout Otherwise, it doesn't show up in the automated test
+ failures
+
+ * tests/test_hashtab_thrash.c: Add some debugging when
+ test_hashtab_thrash fails Disabling DEBUG_THREADS caused this
+ test to fail on the 32-bit build agent. Adding some debugging to
+ see why it thinks the test is timing out.
+
+ * main/pbx.c: Unlock the dial operation lock on a failed dial If a
+ dial operation fails, the pbx_outgoing_attempt routine will exit
+ without first having unlocked the outgoing dial lock. This would
+ be a "bad thing".
+
+2013-08-09 21:50 +0000 [r396512] Richard Mudgett <rmudgett@digium.com>
+
+ * bridges/bridge_native_rtp.c: bridge_native_rtp: Remove some
+ unnecessary NULL checks on c1.
+
+2013-08-09 20:29 +0000 [r396505] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/autoservice.c: Don't leak frames when memory is full in
+ autoservice_run. Review: https://reviewboard.asterisk.org/r/2566/
+
+2013-08-09 17:28 +0000 [r396497-396498] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, channels/chan_sip.c: pbx: Make originate threads
+ indicate dial status when synchronous This makes it so that we
+ can detect failures to originate as with earlier versions of
+ Asterisk, which restores the Asterisk 11 behavior for the
+ originate manager action. This was causing the ACL tests for SIP
+ and IAX2 to fail since those tests expected originate failures
+ when ACLs would cause rejections. Also, this patch fixes crashes
+ in chan_sip when ACLs rejected peers during registration
+ verification. (closes issue ASTERISK-22212) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2753/
+
+ * main/bridge_channel.c, include/asterisk/bridge.h,
+ res/ari/resource_bridges.c, include/asterisk/core_unreal.h,
+ main/core_unreal.c: bridge_channel: Support the lonely flag and
+ make ARI use it. The lonely flag is an optional flag for bridge
+ channels that will make them leave a bridge when a channel leaves
+ if only lonely channels are in the bridge at that point. This is
+ useful for things like ending recording and playback channels
+ when they cease to be interacting with other channels in the
+ bridge. (closes issue ASTERISK-22117) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2721/
+
+2013-08-09 13:58 +0000 [r396490] Matthew Jordan <mjordan@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Update documentation for
+ ConfBridge with some additional markup Add some additional markup
+ for items that needed it, e.g., replaceable tags, literal tags,
+ etc.
+
+2013-08-08 22:57 +0000 [r396480] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_stasis.c: Fix stasis/core unit test. Should have had
+ the CR/LF.
+
+2013-08-08 22:09 +0000 [r396474] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: create channels at run-time
+ This code adds chan_dahdi the command 'dahdi create channels
+ <range>' (where <range> is a single <n>-<m> or 'new') and updates
+ 'dahdi destroy channel' with a similar 'dahdi destroy channels'.
+ It allows DAHDI channels and spans to be added after the initial
+ channel load (without destroying all other channels as in 'dahdi
+ restart'). It also includes some fixes to the D-Channel / span
+ destruction code (r394552). This change is intended to provide a
+ hook for a script running from udev once a span has been assigned
+ ("registered") / unassigned ("unregistered") for its channels.
+ The udev hook configures the span's channels with dahdi_cfg -S,
+ and can then ask Asterisk to create ethe channels. See the
+ scripts added to DAHDI-tools in 2.7.0. Review:
+ https://reviewboard.asterisk.org/r/1598/
+
+2013-08-08 20:52 +0000 [r396417-396463] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_stasis.c: Add missing CR/LF to FakeMI stasis test AMI
+ event.
+
+ * main/stasis_bridges.c: Remove extra CR/LF from AMI event.
+
+ * main/manager_bridges.c, apps/confbridge/confbridge_manager.c,
+ include/asterisk/manager.h, main/stasis_bridges.c: Make bridge
+ snapshots use prefixes. * Changed
+ ast_manager_build_bridge_state_string() to assume an empty prefix
+ string just like ast_manager_build_channel_state_string(). *
+ Created ast_manager_build_bridge_state_string_prefix() to work
+ just like ast_manager_build_channel_state_string_prefix(). * Made
+ BridgeMerge AMI event use To/From prefixes.
+
+2013-08-08 18:40 +0000 [r396412] Matthew Jordan <mjordan@digium.com>
+
+ * formats/format_wav_gsm.c: Improve disk writes for wav49 format
+ Writing to a file in the wav49 format performs rather
+ inefficiently. The procedure is approximately: (1) Write GSM
+ frame to the end of the file (2) Seek to the end of the file (3)
+ Seek to the header (4) Update the file size (5) Seek (again) to
+ the end of the file (6) Repeat This pattern negates any attempt
+ to use the stdio buffering setup in ast_writefile. It also
+ results in many small writes that require a seek going to the
+ disk each second which translates to poor disk performance on
+ certain file systems, particularly when there are multiple wav49
+ files being written simultaneously. (closes issue ASTERISK-19595)
+ Reported by: Byron Clark Tested by: Byron Clark patches:
+ gsm_wav_only_update_header_on_close.patch uploaded by byronclark
+ (License 6157)
+
+2013-08-08 17:51 +0000 [r396401] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridge_features.h, main/bridge.c,
+ main/channel_internal_api.c, main/features.c: Remove some
+ resolved or obsolete BUGBUG comments.
+
+2013-08-08 14:13 +0000 [r396391-396392] Matthew Jordan <mjordan@digium.com>
+
+ * apps/confbridge/conf_chan_record.c, main/channel_internal_api.c,
+ include/asterisk/channel.h, main/cel.c,
+ apps/confbridge/conf_chan_announce.c, main/manager_channels.c,
+ main/channel.c, main/manager_bridges.c,
+ channels/chan_bridge_media.c: Hide the Surrogate channels from
+ external consumers; kill Masquerade events This patch does three
+ things: 1. It provides a Surrogate channel technology with a
+ consolidated "implementation detail flag" on the channel
+ technology. This tells consumers of Stasis that the creation of
+ this channel is an implementation detail in Asterisk and can be
+ ignored (if they so choose). This consolidates the conference
+ recorder/announcer flags as well - these flags had no additional
+ meaning beyond "ignore this channel please". 2. It modifies
+ allocation of a channel in two ways: (a) If a channel technology
+ can be determined from the name, we set it directly in the
+ allocation routine. This prevents the initial publication of the
+ message from going out with a NULL channel technology where
+ possible. This lets Stasis consumers get the right channel
+ technology on the first publication. (b) It reorganizes
+ allocation to make use of the 'finalized' property on the
+ channel. This was already used to know that a channel had
+ completely finished its construction in the masquerade routine;
+ now we also use it to know whether or not the setting of certain
+ channel properties is occurring during or post construction. The
+ various set routines were modified accordingly as well. 3. The
+ masquerade event is now dead, Jim. It no longer served any
+ purpose whatsoever - if you perform a call pickup you'll get a
+ Pickup event; if you perform an attended transfer you will still
+ get those events; if you steal a channel to put it elsewhere
+ you'll get the corresponding NewExten or BridgeEnter events.
+ Review: https://reviewboard.asterisk.org/r/2740
+
+ * main/utils.c: Prevent spurious memory error when appending
+ backtrace with MALLOC_DEBUG Backtraces are allocated outside of
+ the usual memory tracking performed by MALLOC_DEBUG. This allows
+ them to be used by the memory tracking enabled by that build
+ option; however, it also means that when backtraces are disposed
+ of they have to be done so outside of the re-defined free. This
+ patch undef's free prior to disposing of the allocated backtrace
+ when a backtrace is appended as a result of 'core show locks'.
+
+2013-08-08 12:38 +0000 [r396385] Kinsey Moore <kmoore@digium.com>
+
+ * main/bridge.c: Prevent unreal channels from optimizing during
+ DTMF emulation This prevents unreal channel optimization during
+ the prequalification phase when either channel is involved in
+ DTMF emulation. This prevents a situation where an emulated digit
+ would be missed because the emulation was never completed.
+ Review: https://reviewboard.asterisk.org/r/2747/ (closes issue
+ ASTERISK-22214)
+
+2013-08-08 07:05 +0000 [r396378] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: - Fix different issues with call
+ transfer cancel. In case 3rd party busy or congestion call was
+ not returned. - Fix displaying soft button 'Redial' in case of no
+ redial number exists ........ Merged revisions 396377 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-08 02:58 +0000 [r396365-396371] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Handle Surrogate channels in Dial message processing
+ Depending on when a Surrogate channel replaces an existing
+ channel, it is possible to get a Dial message for the Surrogate
+ channel. When this occurs, no CDR will exist for the channel as
+ Surrogate channels are ignored. Safely handle the case when a CDR
+ doesn't exist for a Dial message.
+
+ * apps/app_queue.c: Perform Ring-No-Answer checks before processing
+ Hangup logic The rna() routine will raise a Stasis message
+ involving both the caller and the agent. This doesn't work so
+ well if we already hung up the agent channel, as the channel
+ doesn't quite exist. Not surprisingly, this will crash. This
+ patch properly runs the rna subroutine (performing all of the
+ Ring-No-Answer logic) prior to hanging up the agent channel.
+ (closes issue ASTERISK-22258) Reported by: Kiril Valchev Tested
+ by: Kiril Valchev
+
+2013-08-06 21:20 +0000 [r396329-396347] David M. Lee <dlee@digium.com>
+
+ * apps/app_meetme.c: Fixed app_meetme for cache split changes
+
+ * main/app.c, res/res_stasis_recording.c, include/asterisk/frame.h,
+ rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
+ apps/app_voicemail.c, main/channel.c, res/res_ari_recordings.c,
+ include/asterisk/app.h, include/asterisk/stasis_app_recording.h,
+ res/ari/resource_recordings.h, funcs/func_frame_trace.c,
+ apps/app_minivm.c: ARI: Add recording controls This patch
+ implements the controls from ARI recordings. The controls are: *
+ DELETE /recordings/live/{recordingName} - stop recording and
+ discard it * POST /recordings/live/{recordingName}/stop - stop
+ recording * POST /recordings/live/{recordingName}/pause - pause
+ recording * POST /recordings/live/{recordingName}/unpause -
+ resume recording * POST /recordings/live/{recordingName}/mute -
+ mute recording (record silence to the file) * POST
+ /recordings/live/{recordingName}/unmute - unmute recording. Since
+ this underlying functionality did not already exist, is was added
+ to app.c by a set of control frames, similar to how playback
+ control works. The pause/mute control frames are toggles, even
+ though the ARI controls are idempotent, to be consistent with the
+ playback control frames. (closes issue ASTERISK-22181) Review:
+ https://reviewboard.asterisk.org/r/2697/
+
+ * tests/test_stasis.c, main/stasis_cache_pattern.c,
+ main/stasis_cache.c, include/asterisk/stasis.h: Tweak caching
+ topics to fix CEL tests The Stasis changes in r395954 had an
+ unanticipated side effect: messages published directly to an _all
+ topic does not get forwarded to the corresponding caching topic.
+ This patch fixes that by changing how caching topics forward
+ messages, and how the caching pattern forwards are setup. For the
+ caching pattern, the all_topic is forwarded to the
+ all_topic_cached. This forwards messages published directly to
+ the all_topic to all_topic_cached. In order to avoid duplicate
+ messages on all_topic_cached, caching topics were changed to no
+ longer forward uncached messages. Subscribers to an individual
+ caching topic should only expect to receive cache updates, and
+ subscription change messages. Since individual caching topics are
+ new, this shouldn't be a problem. There are a few minor changes
+ to the pre-cache split behavior. * For topics changed to use the
+ caching pattern, the all_topic_cached will forward snapshots in
+ addition to cache updates. Since subscribers by design ignore
+ unexpected messages, this should be fine. * Caching topics that
+ don't use the caching pattern no longer forward non-cache
+ updates. This makes no difference for the current caching topics.
+ * mwi_topic_cached, channel_by_name_topic and
+ presence_state_topic_cached have no subscribers *
+ device_state_topic_cached's only subscriber only processes cache
+ udpates (issue ASTERISK-22243) Review:
+ https://reviewboard.asterisk.org/r/2738
+
+2013-08-06 13:08 +0000 [r396320-396321] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
+ res/res_pjsip/config_system.c: Expose res_pjsip threadpool
+ options Expose initial size, automatic increment, maximum size,
+ and idle timeout as configurable parameters for the res_pjsip
+ thread pool. Review: https://reviewboard.asterisk.org/r/2704/
+ (closes issue ASTERISK-22143)
+
+ * main/cdr.c: Fix memory leaks in the CDR engine Fix refcount bugs
+ and a possible locking problem in the CDR engine relating to use
+ of ao2_iterators. Review:
+ https://reviewboard.asterisk.org/r/2724/ (closes issue
+ ASTERISK-22126)
+
+2013-08-06 12:39 +0000 [r396319] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_notify.c, res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_messaging.c, res/res_pjsip_exten_state.c: Fix crash
+ in res_pjsip_outbound_registration when the remote server can not
+ be resolved. This crash was caused by decrementing the reference
+ count of a newly created message when it should not be. This
+ change fixes that but also fixes all other cases where this was
+ incorrectly done. (closes issue ASTERISK-22188) Reported by:
+ Kinsey Moore
+
+2013-08-06 08:43 +0000 [r396309-396311] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_strings.c, /: Check result of ast_var_assign() calls
+ for memory allocation failure (2). Missed a spot in the previous
+ commit. ........ Merged revisions 396310 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * utils/extconf.c, apps/app_stack.c, apps/app_playback.c,
+ funcs/func_global.c, main/cdr.c, pbx/pbx_loopback.c, main/pbx.c,
+ /, funcs/func_strings.c, pbx/pbx_dundi.c: Check result of
+ ast_var_assign() calls for memory allocation failure. We try to
+ keep the system running even when all available memory is spent.
+ Review: https://reviewboard.asterisk.org/r/2734/ ........ Merged
+ revisions 396279 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 396287 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-05 20:20 +0000 [r396253] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Registration Failure When A Peer And
+ TLS Are Used If a peer is used in a register line and TLS is
+ defined as the transport, the registration fails since the
+ transport on the dialog is never set properly resulting in UDP
+ being used instead of TLS. This patch sets the dialog's transport
+ based on the transport that was defined in the register line. If
+ the register line does not specify a transport, the parsing
+ function for the register line always defaults back to UDP.
+ (closes issue ASTERISK-21964) Reported by: Doug Bailey Tested by:
+ Doug Bailey Patches: asterisk-21964-set-reg-dialog-transport.diff
+ by Michael L. Young (license 5026) ........ Merged revisions
+ 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 396248 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-05 20:18 +0000 [r396245] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/bridge_basic.h, main/bridge_basic.c,
+ main/features.c: bridge features: Dial and Queue add features
+ instead of replace them. Dial and Queue would previously apply a
+ new set of features whenever bridging. These options would be
+ based purely on the options supplied to the dial/queue
+ applications. This patch changes the function those applications
+ use to bridge calls so that the features will be added to the set
+ of existing features for each channel rather than having them
+ override the existing features. (closes issue ASTERISK-22209)
+ Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2713/
+
+2013-08-05 19:01 +0000 [r396201] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_pjsip_outbound_registration.c: Add AMI registration
+ events for PJSIP outbound registration attempts This patch adds
+ AMI events whenever an outbound registration attempt succeeds or
+ fails from res_pjsip_outbound_registration. This brings it inline
+ with the existing SIP channel driver and IAX channel driver.
+ Review: https://reviewboard.asterisk.org/r/2729/
+
+2013-08-05 18:52 +0000 [r396198-396200] Michael L. Young <elgueromexicano@gmail.com>
+
+ * UPGRADE-11.txt, /: Change "from" to "From". (related to issue
+ ASTERISK-21903) ........ Merged revisions 396199 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, UPGRADE-11.txt: Adding a note to UPGRADE.txt about a change
+ made to res_agi in order to indicate when streaming an audio file
+ fails like it is done in other parts of the code to indicate an
+ error. Note was requested by Paul Belanger:
+ http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
+ (related to issue ASTERISK-21903) ........ Merged revisions
+ 396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 396197 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-08-05 17:48 +0000 [r396175-396189] Jonathan Rose <jrose@digium.com>
+
+ * bridges/bridge_holding.c: bridge_holding: Add suspsend/unsuspend
+ callbacks Suspend and unsuspend callbacks are added to the
+ holding bridge so that entertainment can be disabled and
+ re-enabled when operations would suspend a channel on the bridge
+ (such as playback operations). This fixes entertainment so that
+ when those operations end, the entertainment can pick back up and
+ it also serves as an optimization. Also, this patch fixes a bug
+ caused by triggering ringing frames immediately instead of
+ pushing them to the queue which created a race condition where
+ sometimes parking with ringing during attended transfers would
+ cause the ringing to be interrupted by an unhold frame. (closes
+ issue ASTERISK-22006) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2711/
+
+ * rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
+ res/res_ari_bridges.c, include/asterisk/bridge_roles.h,
+ res/ari/resource_bridges.h, res/stasis/control.c,
+ include/asterisk/stasis_app.h, main/bridge_roles.c: ARI:
+ bridges/{bridgeID}/addChannel: add roles parameter Roles are now
+ cleared with each entry into a bridge with addChannel. If the
+ roles parameter is present, the role specified will be applied to
+ all channels being added with the addChannel command. (closes
+ issue ASTERISK-21973) Reported by: Matt Jordan
+ https://reviewboard.asterisk.org/r/2691/
+
+ * res/parking/parking_bridge.c, res/parking/res_parking.h,
+ res/res_parking.c, res/parking/parking_tests.c (added):
+ res_parking: Unit tests Adds the following unit tests: *
+ create_lot: tests adding and removal of a new parking lot
+ (baseline) * park_extensions: creates a parking lot that
+ registers extensions and then confirms that all of the expected
+ extensions exist * extensions_conflicts: creates numerous parking
+ lots to test that extension conflicts in parking lots result in
+ parking lot creation failing * dynamic_parking_variables: Tests
+ that the creation of dynamic parking lots respects the related
+ channel variables set on the channel that requests them. *
+ park_call: Tests adding a channel to a parking lot's holding
+ bridge by standard parking functions. * retrieve_call: Tests
+ pulling a channel out of a parking lot's holding bridge via
+ parked call retrieval functions. (closes issue ASTERISK-22138)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2714/
+
+2013-08-05 14:35 +0000 [r396166] David M. Lee <dlee@digium.com>
+
+ * main/asterisk.c, main/cli.c, main/channel.c, main/pbx.c,
+ main/manager.c, res/ari/resource_asterisk.c, utils/extconf.c,
+ include/asterisk/options.h: Fix res_ari_asterisk load issue The
+ new res_ari_asterisk.so module presents several config options
+ from asterisk main. Unfortunately, they aren't exported, so the
+ module won't load on Linux. This patch renames the variables,
+ adding the ast_ prefix so they will be exported. Review:
+ https://reviewboard.asterisk.org/r/2737
+
+2013-08-03 03:53 +0000 [r396158] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager_bridges.c: Don't unsubscribe from the AMI message
+ router from manager_bridges The AMI message router is owned
+ wholly by manager.c. Previously, each of the manager_{item}
+ source files had their own message router and they unsubscribed
+ from each; once they moved over to using a single message router
+ only a single unsubscribe became necessary.
+
+2013-08-02 17:50 +0000 [r396145] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sig_pri.c: And get rid of another ast_bridged_channel()
+
+2013-08-02 17:29 +0000 [r396136-396143] David M. Lee <dlee@digium.com>
+
+ * main/stasis_bridges.c: Clean up ast_json with ast_json_unref
+
+ * /: Removed svnmerge-integrated from trunk
+
+2013-08-02 15:01 +0000 [r396126] Mark Michelson <mmichelson@digium.com>
+
+ * res/snmp/agent.c: Get the SNMP code to compile.
+
+2013-08-02 14:46 +0000 [r396119-396125] David M. Lee <dlee@digium.com>
+
+ * res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
+ rest-api/api-docs/asterisk.json, res/ari/resource_asterisk.c: ARI
+ - GET /ari/asterisk/info This patch adds basic system information
+ access to ARI. The results are roughly what you get from 'core
+ show settings', with a few minor differences. * Data is
+ structured, with 'build', 'system', 'config' and 'status'
+ sub-objects. * Each sub-object is selectable, using the ?only=
+ parameter. A comma separated list can be provided to select
+ multiple sections. * A few config options are numeric, for which
+ 0 means 'unlimited'. Instead of having a special interpretation
+ of those fields, they are simply omitted if they're 0. * The
+ information is limited to what might be useful to building
+ external applications. (closes issue ASTERISK-21575) Review:
+ https://reviewboard.asterisk.org/r/2702/
+
+ * res/res_ari_recordings.c, res/ari/resource_bridges.h,
+ res/res_ari_endpoints.c, res/res_ari_events.c,
+ res/ari/resource_asterisk.h, rest-api/api-docs/channels.json,
+ res/res_ari_sounds.c, res/res_ari_bridges.c,
+ rest-api-templates/param_cleanup.mustache (added),
+ rest-api/api-docs/events.json, /, res/ari/resource_events.c,
+ rest-api-templates/ari_resource.h.mustache,
+ res/res_ari_asterisk.c, res/res_ari_playback.c,
+ rest-api-templates/res_ari_resource.c.mustache,
+ res/ari/resource_events.h, rest-api/api-docs/sounds.json,
+ res/res_ari_channels.c, rest-api/api-docs/bridges.json,
+ rest-api-templates/param_parsing.mustache,
+ res/ari/resource_bridges.c, res/ari/resource_sounds.h: ARI -
+ implement allowMultiple for parameters Swagger allows parameters
+ to be specified as 'allowMultiple', meaning that the parameter
+ may be specified as a comma separated list of values. I had
+ written some of the API docs using that, but promptly forgot
+ about implementing it. This patch finally fills in that gap. The
+ codegen template was updated to represent 'allowMultiple' fields
+ as array/size fields in the _args structs. It also parses the
+ comma separated list using ast_app_separate_args(), so quoted
+ strings in the argument will be handled properly. Review:
+ https://reviewboard.asterisk.org/r/2698/
+
+ * res/ari/ari_websockets.c, tests/test_json.c, main/json.c,
+ res/res_sorcery_astdb.c, include/asterisk/json.h, main/cel.c:
+ Address JSON thread safety issues. In tracking down some unit
+ tests failures, I ended up reading the fine print[1] regarding
+ Jansson's thread safety. In short: 1. Ref-counting is non-atomic.
+ 2. json_dumps() and friends are not thread safe. This patch adds
+ locking where necessary to our ast_json_* wrapper API, with
+ documentation in json.h describing the thread safety limitations
+ of the API. [1]:
+ http://www.digip.org/jansson/doc/2.4/portability.html#thread-safety
+ Review: https://reviewboard.asterisk.org/r/2716/
+
+2013-08-02 14:13 +0000 [r396107] Mark Michelson <mmichelson@digium.com>
+
+ * main/cel.c, include/asterisk/parking.h, main/bridge_channel.c,
+ main/stasis_bridges.c, res/parking/parking_manager.c,
+ res/parking/parking_bridge.c, main/manager_bridges.c,
+ include/asterisk/stasis_bridges.h: Make a couple of changes to
+ help AMI events to be more clear in what is occurring. *
+ BridgeEnter now contains the unique ID of the channel that is to
+ be swapped out, if applicable. * There is a ParkedCallSwap event
+ that is sent when a parked channel has a new channel take its
+ place. (closes issue ASTERISK-22193) reported by Mark Michelson
+ Review: https://reviewboard.asterisk.org/r/2712
+
+2013-08-02 14:08 +0000 [r396105] Kinsey Moore <kmoore@digium.com>
+
+ * utils/Makefile, utils/refcounter.c, main/strings.c,
+ include/asterisk/astobj2.h, include/asterisk/strings.h,
+ main/astobj2.c: Move ast_str_container_alloc and friends This
+ moves ast_str_container_alloc, ast_str_container_add,
+ ast_str_container_remove, and related private functions into
+ strings.c/h since they really don't belong in astobj2.c/h. As a
+ result of this move, utils also had to be updated. Review:
+ https://reviewboard.asterisk.org/r/2719/ (closes issue
+ ASTERISK-22041)
+
+2013-08-02 14:05 +0000 [r396102-396103] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_iax2.c, apps/app_chanspy.c, channels/chan_oss.c,
+ channels/chan_mgcp.c, main/channel.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, main/rtp_engine.c, channels/chan_sip.c,
+ channels/chan_skinny.c, funcs/func_channel.c,
+ main/channel_internal_api.c, include/asterisk/channel.h: Get rid
+ of ast_bridged_channel() and the bridged_channel field on
+ ast_channels. This commit is smaller than the initial review
+ placed on review board. This is because a change to allow for
+ channel drivers to access parking functionality externally was
+ committed and invalidated quite a few of the changes initially
+ made. (closes issue ASTERISK-22039) reported by Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2717
+
+ * include/asterisk/pickup.h: Make sure that pickup.h does not use
+ an include guard name used elsewhere.
+
+2013-08-02 13:29 +0000 [r396087-396099] Kinsey Moore <kmoore@digium.com>
+
+ * main/pickup.c: Correct the last of the Newchannel xi:includes
+
+ * res/res_pjsip_notify.c, res/res_pjsip_outbound_registration.c,
+ res/res_pjsip/include/res_pjsip_private.h,
+ res/res_pjsip/pjsip_options.c, res/res_pjsip.c: Add CLI/AMI
+ commands to force chan_pjsip actions For chan_pjsip, this
+ introduces CLI/AMI remote unregistration commands, reworks CLI
+ syntax for sending NOTIFYs, adds AMI qualification support, and
+ adds documentation for PJSIPNotify. This also fixes two
+ refcounting bugs in the outbound registration code. Review:
+ https://reviewboard.asterisk.org/r/2695/ (closes issue
+ ASTERISK-21939)
+
+2013-08-02 04:48 +0000 [r396075] David M. Lee <dlee@digium.com>
+
+ * channels/sig_analog.c: Fixed chan_dahdi compilation failure
+
+2013-08-02 03:12 +0000 [r396060-396062] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_cel.c, tests/test_cdr.c: Fix test modules More missing
+ include files. :-\
+
+ * channels/chan_mgcp.c, channels/chan_dahdi.c: Add pickup.h include
+ lines for chan_dahdi and chan_mgcp
+
+ * apps/app_directed_pickup.c, main/features.c, tests/test_cel.c,
+ include/asterisk/parking.h, include/asterisk/pickup.h (added),
+ main/asterisk.c, res/parking/parking_manager.c, tests/test_cdr.c,
+ channels/chan_unistim.c, main/pbx.c, res/stasis/control.c,
+ main/pickup.c (added), channels/chan_sip.c, main/bridge.c,
+ UPGRADE.txt, res/parking/parking_applications.c,
+ include/asterisk/_private.h, channels/chan_gtalk.c, main/cel.c,
+ CHANGES, include/asterisk/features.h, main/cdr.c,
+ res/res_parking.c, channels/chan_skinny.c: Remove dead code from
+ features.c; refactor pickup code into pickup.c This patch does
+ the following: * It moves the pickup code out of features.c and
+ into pickup.c * It removes the vast majority of dead code out of
+ features.c. In particular, this includes the parking code. (issue
+ ASTERISK-22134)
+
+2013-08-01 23:38 +0000 [r396048] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_registrar.c: Fix a crash due to performing full URI
+ validation on a contact which only contains '*'. (closes issue
+ AST-1198) Reported by: John Bigelow
+
+2013-08-01 21:19 +0000 [r396035] David M. Lee <dlee@digium.com>
+
+ * main/sorcery.c: Fix sorcery for some rather picky regex
+ implementations. Some regex implementations won't compile an
+ empty string. Assuming that it's equivalent of a regex that will
+ match anything, use ".?" instead.
+
+2013-08-01 20:55 +0000 [r396010-396028] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/parking.h,
+ main/bridge_channel.c, res/parking/parking_bridge_features.c,
+ channels/chan_mgcp.c, include/asterisk/features.h,
+ channels/chan_dahdi.c, res/res_parking.c, channels/sig_analog.c,
+ channels/chan_skinny.c, main/parking.c, main/bridge.c,
+ main/features.c: Support externally initiated parking requests;
+ remove some dead code This patch does the following: * It adds
+ support for externally initiated parking requests. In particular,
+ chan_skinny has a protocol level message that initiates a call
+ park. This patch now supports that option, as well as the
+ protocol specific mechanisms in chan_dahdi/sig_analog and
+ chan_mgcp. * A parking bridge features virtual table has been
+ added that provides access to the parking functionality that the
+ Bridging API needs. This includes requests to park an entire
+ 'call' (with little or no additional information, thank you
+ chan_skinny), perform a blind transfer to a parking extension,
+ determine if an extension is a parking extension, as well as the
+ actual "do the parking" request from the Bridging API. *
+ Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use
+ of the new functions * The removal of some - but not all - dead
+ parking code from features.c This also fixed blind transferring a
+ multi-party bridge to a parking lot (which was implemented, but
+ had at least one code path where using the parking features kK
+ might not have worked) Review:
+ https://reviewboard.asterisk.org/r/2710 (closes issue
+ ASTERISK-22134) Reported by: Matt Jordan
+
+ * CHANGES, apps/app_queue.c: Add queue member paused hints This
+ patch adds the ability in Queue to raise a hint when a member's
+ paused state changes. The hint uses the form
+ 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and
+ {member_name} are the name of the queue and the name of the
+ member to subscribe to, respectively. For example: exten =>
+ 8501,hint,Queue:sales_pause_mark. Members will show as In Use
+ when paused. Note that the format of the queue pause hint was
+ changed slightly from what is on the issue to accomodate
+ suggestion on the code review. Review:
+ https://reviewboard.asterisk.org/r/2254 (closes issue
+ ASTERISK-20842) Reported by: Philippe Lindheimer patches:
+ qpause-10-378206.diff uploaded by Philippe Lindheimer (license
+ 5519) qpause-11-378206.diff uploaded by Philippe Lindheimer
+ (license 5519) qpause-trunk-378206.diff uploaded by Philippe
+ Lindheimer (license 5519)
+
+2013-08-01 17:23 +0000 [r395985-395998] Kinsey Moore <kmoore@digium.com>
+
+ * configure: Regenerate configure for configure.ac changes
+
+ * main/xml.c, main/stasis_bridges.c,
+ contrib/scripts/install_prereq, main/manager_bridges.c,
+ channels/chan_dahdi.c, main/manager.c, doc/snapshots.xslt
+ (added), main/features.c, apps/app_minivm.c, res/res_agi.c,
+ main/stasis_channels.c, main/manager_channels.c,
+ channels/chan_sip.c, main/Makefile, configure.ac, UPGRADE.txt,
+ main/aoc.c, main/core_local.c, channels/sig_pri.c,
+ apps/app_queue.c, CHANGES, funcs/func_global.c,
+ apps/app_agent_pool.c, Makefile,
+ apps/confbridge/confbridge_manager.c, makeopts.in,
+ doc/appdocsxml.dtd, apps/app_stack.c,
+ res/parking/parking_manager.c, main/manager_mwi.c,
+ main/rtp_engine.c, apps/app_meetme.c,
+ include/asterisk/autoconfig.h.in: Fix documentation replication
+ issues This prevents XML documentation duplication by expanding
+ channel and bridge snapshot tags into channel and bridge snapshot
+ parameter sets with a given prefix or defaulting to no prefix.
+ This also prevents documentation from becoming fractured and out
+ of date by keeping all variations of the documentation in
+ template form such that it only needs to be updated once and
+ keeps maintenance to a minimum. Review:
+ https://reviewboard.asterisk.org/r/2708/
+
+2013-08-01 16:56 +0000 [r395954-395984] David M. Lee <dlee@digium.com>
+
+ * utils/astman.c: Fixed warning in astman for gcc-4.8.
+
+ * channels/chan_pjsip.c, res/res_pjsip_mwi.c: Fixed compile errors
+ introduced in r395954. Just a merge error due to a file rename.
+ Grrr...
+
+ * channels/chan_unistim.c, main/stasis_endpoints.c, main/pbx.c,
+ include/asterisk/devicestate.h, res/ari/resource_endpoints.c,
+ apps/app_meetme.c, main/bridge.c,
+ include/asterisk/channel_internal.h,
+ include/asterisk/presencestate.h, include/asterisk/channel.h,
+ channels/sig_pri.c, main/cel.c, main/stasis_bridges.c,
+ main/stasis.c, res/ari/resource_bridges.c, channels/chan_dahdi.c,
+ include/asterisk/app.h, include/asterisk/stasis_channels.h,
+ apps/confbridge/confbridge_manager.c, res/res_agi.c,
+ include/asterisk/stasis_cache_pattern.h (added),
+ tests/test_cel.c, res/ari/resource_channels.c,
+ include/asterisk/stasis_endpoints.h, res/res_stasis.c,
+ include/asterisk/bridge.h, main/manager_channels.c,
+ apps/app_voicemail.c, main/stasis_cache.c, main/stasis_wait.c
+ (added), res/stasis/control.c, channels/chan_sip.c,
+ main/manager_endpoints.c, main/channel_internal_api.c,
+ include/asterisk/stasis_bridges.h, include/asterisk/stasis.h,
+ main/devicestate.c, res/res_xmpp.c, main/endpoints.c,
+ channels/chan_iax2.c, res/res_jabber.c, main/presencestate.c,
+ res/res_chan_stats.c, tests/test_stasis_endpoints.c, main/cli.c,
+ main/cdr.c, main/manager_bridges.c, main/manager.c,
+ tests/test_devicestate.c, main/app.c, main/stasis_channels.c,
+ tests/test_stasis.c, channels/chan_mgcp.c,
+ main/stasis_cache_pattern.c (added): Split caching out from the
+ stasis_caching_topic. In working with res_stasis, I discovered a
+ significant limitation to the current structure of
+ stasis_caching_topics: you cannot subscribe to cache updates for
+ a single channel/bridge/endpoint/etc. To address this, this patch
+ splits the cache away from the stasis_caching_topic, making it a
+ first class object. The stasis_cache object is shared amongst
+ individual stasis_caching_topics that are created per
+ channel/endpoint/etc. These are still forwarded to global
+ whatever_all_cached topics, so their use from most of the code
+ does not change. In making these changes, I noticed that we
+ frequently used a similar pattern for bridges, endpoints and
+ channels: single_topic ----------------> all_topic ^ |
+ single_topic_cached ----+----> all_topic_cached | +----> cache
+ This pattern was extracted as the 'Stasis Caching Pattern',
+ defined in stasis_caching_pattern.h. This avoids a lot of
+ duplicate code between the different domain objects. Since the
+ cache is now disassociated from its upstream caching topics, this
+ also necessitated a change to how the 'guaranteed' flag worked
+ for retrieving from a cache. The code for handling the caching
+ guarantee was extracted into a 'stasis_topic_wait' function,
+ which works for any stasis_topic. (closes issue ASTERISK-22002)
+ Review: https://reviewboard.asterisk.org/r/2672/
+
+2013-08-01 11:21 +0000 [r395938] Joshua Colp <jcolp@digium.com>
+
+ * res/res_pjsip_session.c: Answer with multiple codecs if the
+ underlying pjproject supports it.
+
+2013-08-01 00:07 +0000 [r395906-395907] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Raise Registry AMI events on registration
+ failures This patch makes it so that all registration attempts
+ that fail that also permanently modify the registration state
+ will raise an appropriate AMI event. Note that this patch was
+ forward ported to trunk and the Stasis Core message bus by
+ mjordan. (closes issue ASTERISK-21368) Reported by: Dmitriy Serov
+ patches: chan_sip.c.diff uploaded by Demon (license 6479)
+
+ * res/res_agi.c, CHANGES: Update CONTROL STREAM FILE to accept an
+ 'offsetms' parameter This patch allows starting playback of audio
+ through the CONTROL STREAM FILE AGI command to start at a
+ particular offset. It will also return the final position of the
+ file in the 'endpos' attribute. (closes issue ASTERISK-17803)
+ Reported by: Murray Melvin patches: res_agi.c.r316293.diff
+ uploaded by murraytm (license 6221)
+
+2013-07-31 15:43 +0000 [r395884] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip/pjsip_options.c: Found another missed "sip" ->
+ "pjsip" CLI command.
+
+2013-07-31 15:27 +0000 [r395881] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_cel.c: Disable CEL tests that need rearchitecting to
+ operate properly
+
+2013-07-31 14:45 +0000 [r395868] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_constant.c (removed): Remove
+ "constant" endpoint identifier. This was created as a debugging
+ tool before proper endpoint identifiers were created. Using it
+ now can actually lead to harmful results.
+
+2013-07-31 14:29 +0000 [r395866] Joshua Colp <jcolp@digium.com>
+
+ * bridges/bridge_native_rtp.c: Fix hold/unhold in
+ bridge_native_rtp, use tech_pvt instead of bridge_pvt, reduce
+ bridging attempts, and fix breaking native RTP bridges. (closes
+ issue ASTERISK-22128) (closes issue ASTERISK-22104)
+
+2013-07-31 13:31 +0000 [r395837-395851] Kinsey Moore <kmoore@digium.com>
+
+ * configs/pjsip.conf.sample,
+ res/res_pjsip/include/res_pjsip_private.h, channels/chan_pjsip.c,
+ include/asterisk/res_pjsip.h,
+ include/asterisk/res_pjsip_pubsub.h,
+ include/asterisk/res_pjsip_exten_state.h,
+ include/asterisk/res_pjsip_session.h: Fix remnants of the pjsip
+ renaming
+
+ * tests/test_cel.c: Enforce conference exit order for CEL tests
+
+2013-07-30 22:41 +0000 [r395810-395824] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c: Missed a conversion to
+ pjsip.conf in documentation and sorcery.
+
+ * main/abstract_jb.c: Remove ast_bridged_channel call from
+ abstract_jb.c Interestingly, this only happens in dead code.
+
+2013-07-30 20:44 +0000 [r395793] David M. Lee <dlee@digium.com>
+
+ * res/res_pjsip: Setting svn:ignore for res/res_pjsip
+
+2013-07-30 19:10 +0000 [r395748-395779] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_pjsip_endpoint_identifier_constant.c: Update
+ res_pjsip_endpoint_identifier_constant.c to use reorganized
+ endpoint structure.
+
+ * res/res_sip_transport_websocket.c (removed),
+ res/res_pjsip_authenticator_digest.c (added),
+ res/res_pjsip_session.exports.in (added), res/res_pjsip_sdp_rtp.c
+ (added), include/asterisk/res_sip_exten_state.h (removed),
+ res/res_sip_notify.c (removed),
+ res/res_sip_endpoint_identifier_ip.c (removed),
+ res/res_pjsip/include (added), res/res_pjsip_exten_state.c
+ (added), res/res_sip_t38.c (removed),
+ res/res_sip_registrar_expire.c (removed),
+ res/res_pjsip_pubsub.exports.in (added),
+ res/res_pjsip/config_system.c (added), configs/pjsip.conf.sample
+ (added), include/asterisk/res_sip_pubsub.h (removed),
+ res/res_sip_outbound_registration.c (removed), res/res_pjsip
+ (added), res/res_pjsip/include/res_pjsip_private.h (added),
+ res/res_sip_endpoint_identifier_anonymous.c (removed),
+ res/res_sip_endpoint_identifier_constant.c (removed),
+ res/res_pjsip_transport_websocket.c (added),
+ res/res_pjsip/pjsip_outbound_auth.c (added),
+ include/asterisk/res_pjsip_session.h (added),
+ res/res_sip_diversion.c (removed), res/res_sip_dtmf_info.c
+ (removed), res/res_pjsip_t38.c (added), res/Makefile,
+ res/res_sip_authenticator_digest.c (removed),
+ res/res_sip_session.exports.in (removed),
+ res/res_pjsip_endpoint_identifier_user.c (added),
+ res/res_pjsip_pidf.c (added), res/res_sip_messaging.c (removed),
+ include/asterisk/res_pjsip_pubsub.h (added),
+ res/res_sip_caller_id.c (removed), channels/chan_gulp.c
+ (removed), res/res_pjsip/location.c (added), res/res_sip_nat.c
+ (removed), res/res_pjsip_outbound_registration.c (added),
+ res/res_sip_session.c (removed),
+ res/res_pjsip_endpoint_identifier_anonymous.c (added),
+ res/res_sip_rfc3326.c (removed),
+ res/res_pjsip/pjsip_distributor.c (added), res/res_pjsip_pubsub.c
+ (added), res/res_sip_mwi.c (removed), res/res_pjsip_diversion.c
+ (added), include/asterisk/res_pjsip.h (added),
+ res/res_pjsip_dtmf_info.c (added), res/res_pjsip.exports.in
+ (added), res/res_sip_registrar.c (removed),
+ res/res_pjsip_exten_state.exports.in (added),
+ res/res_pjsip/config_security.c (added),
+ include/asterisk/res_sip_session.h (removed),
+ res/res_pjsip_messaging.c (added), res/res_pjsip_caller_id.c
+ (added), res/res_pjsip_logger.c (added),
+ res/res_pjsip/pjsip_global_headers.c (added), res/res_pjsip_nat.c
+ (added), res/res_pjsip_session.c (added),
+ res/res_sip_endpoint_identifier_user.c (removed), res/res_sip.c
+ (removed), res/res_sip_pidf.c (removed), res/res_pjsip_rfc3326.c
+ (added), res/res_pjsip_mwi.c (added),
+ res/res_pjsip/pjsip_configuration.c (added),
+ res/res_sip_outbound_authenticator_digest.c (removed),
+ res/res_pjsip_registrar.c (added), channels/chan_pjsip.c (added),
+ res/res_sip_one_touch_record_info.c (removed),
+ res/res_pjsip/config_global.c (added), res/res_sip_acl.c
+ (removed), res/res_sip_pubsub.c (removed),
+ res/res_pjsip/config_auth.c (added),
+ include/asterisk/res_pjsip_exten_state.h (added),
+ res/res_pjsip_notify.c (added), configs/res_sip.conf.sample
+ (removed), res/res_sip_refer.c (removed),
+ res/res_pjsip_endpoint_identifier_ip.c (added),
+ include/asterisk/res_sip.h (removed),
+ res/res_pjsip/security_events.c (added), res/res_sip.exports.in
+ (removed), res/res_pjsip/config_domain_aliases.c (added),
+ res/res_pjsip_registrar_expire.c (added),
+ res/res_sip_exten_state.exports.in (removed),
+ res/res_pjsip/pjsip_options.c (added), res/res_pjsip.c (added),
+ res/res_sip_sdp_rtp.c (removed), res/res_sip_logger.c (removed),
+ res/res_pjsip_outbound_authenticator_digest.c (added),
+ res/res_sip_exten_state.c (removed),
+ res/res_pjsip_endpoint_identifier_constant.c (added),
+ res/res_pjsip_one_touch_record_info.c (added),
+ res/res_sip_pubsub.exports.in (removed), res/res_pjsip_acl.c
+ (added), res/res_pjsip/config_transport.c (added),
+ res/res_pjsip_refer.c (added), res/res_sip (removed): The large
+ GULP->PJSIP renaming effort. The general gist is to have a clear
+ boundary between old SIP stuff and new SIP stuff by having the
+ word "SIP" for old stuff and "PJSIP" for new stuff. Here's a
+ brief rundown of the changes: * The word "Gulp" in dialstrings,
+ functions, and CLI commands is now "PJSIP" * chan_gulp.c is now
+ chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*"
+ are now "chan_pjsip_*" * All files that were "res_sip*" are now
+ "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files
+ in the "res_pjsip" directory that began with "sip_*" are now
+ "pjsip_*" * The configuration file is now "pjsip.conf" instead of
+ "res_sip.conf" * The module info for all PJSIP-related files now
+ uses "PJSIP" instead of "SIP" * CLI and AMI commands created by
+ Asterisk's PJSIP modules now have "pjsip" as the starting word
+ instead of "sip"
+
+ * res/res_sip_outbound_authenticator_digest.c,
+ res/res_sip_outbound_registration.c, res/res_sip_mwi.c,
+ res/res_sip_one_touch_record_info.c, res/res_sip_pubsub.c,
+ res/res_sip_diversion.c, res/res_sip/sip_configuration.c,
+ include/asterisk/res_sip.h, res/res_sip/sip_distributor.c,
+ res/res_sip.exports.in, res/res_sip_authenticator_digest.c,
+ res/res_sip/sip_outbound_auth.c, res/res_sip_sdp_rtp.c,
+ res/res_sip_messaging.c, res/res_sip_t38.c, channels/chan_gulp.c,
+ res/res_sip_caller_id.c, res/res_sip.c, res/res_sip_nat.c,
+ res/res_sip_session.c, res/res_sip/sip_options.c: Reorganize the
+ ast_sip_endpoint structure into substructures. (closes issue
+ ASTERISK-22135) reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2707
+
+2013-07-30 14:16 +0000 [r395731] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_session.exports.in, channels/chan_gulp.c,
+ res/res_sip_t38.c (added), res/res_sip.c,
+ res/res_sip/sip_configuration.c, res/res_sip_session.c,
+ include/asterisk/res_sip.h, include/asterisk/res_sip_session.h:
+ Add support for T.38 fax to chan_pjsip. Review:
+ https://reviewboard.asterisk.org/r/2692/
+
+2013-07-30 13:46 +0000 [r395728] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_pktccops.c: Fix compilation on gcc 4.8.1
+
+2013-07-29 17:51 +0000 [r395686] David M. Lee <dlee@digium.com>
+
+ * main/mixmonitor.c, res/parking/parking_devicestate.c,
+ include/asterisk/mixmonitor.h: Removed quotes from svn:keywords
+ props on a few files. Subversion doesn't do quote processing, so
+ it actually thinks that the closing quote in 'Revision"' is a
+ part of the keyword.
+
+2013-07-29 16:16 +0000 [r395674] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_sip.c: Clarify documentation for trust of identification.
+ (closes issue ASTERISK-22023) Reported by Rusty Newton
+
+2013-07-29 15:58 +0000 [r395672-395673] Matthew Jordan <mjordan@digium.com>
+
+ * main/loader.c: Put the include in there Mea culpa...
+
+ * main/loader.c: When performing a reload, reload the new
+ features_config and not the old Performing a module reload of
+ core components causes specific functions compiled into the
+ Asterisk binary to be reloaded. The table of said functions was
+ still pointing to the old features reload mechanism, and not the
+ new one.
+
+2013-07-29 14:51 +0000 [r395653] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_cel.c: Clean up and improve test_cel Improve
+ reliability of attended transfer merge and link tests. Stop using
+ ast_log(LOG_ERROR, ...); in favor of ast_test_status_update
+ Remove fred and eve channel helpers since they are not necessary
+
+2013-07-29 14:08 +0000 [r395636] David M. Lee <dlee@digium.com>
+
+ * res/ari: Set svn:ignore in res/ari directory
+
+2013-07-29 12:10 +0000 [r395619] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_sip.c: Remove comment that no longer applies The monitor
+ thread is already properly torn down on unload and load failure.
+
+2013-07-27 23:11 +0000 [r395588-395603] Kinsey Moore <kmoore@digium.com>
+
+ * res/ari (added), res/ari/resource_endpoints.h (added),
+ rest-api-templates/stasis_http.make.mustache (removed),
+ res/stasis_http.make (removed), res/ari/resource_bridges.c
+ (added), tests/test_ari_model.c, res/res_ari_recordings.c
+ (added), rest-api-templates/ari.make.mustache (added),
+ res/ari/resource_bridges.h (added), res/res_ari_events.c (added),
+ res/res_statsd.c, res/res_ari_sounds.c (added),
+ rest-api-templates/stasis_http_resource.h.mustache (removed),
+ res/ari/resource_recordings.c (added), main/stasis_config.c,
+ rest-api-templates/ari_resource.h.mustache (added),
+ res/ari/resource_events.c (added), res/ari/resource_recordings.h
+ (added), rest-api-templates/rest_handler.mustache,
+ res/res_ari_asterisk.c (added), res/ari/resource_events.h
+ (added), res/ari/internal.h (added), res/ari/resource_sounds.c
+ (added), res/res_stasis_http_endpoints.c (removed),
+ res/ari/resource_sounds.h (added), res/res_stasis_http.exports.in
+ (removed), res/ari/resource_asterisk.c (added),
+ res/res_ari_endpoints.c (added), res/ari/resource_asterisk.h
+ (added), res/res_stasis_http_bridges.c (removed),
+ res/ari/config.c (added), res/res_ari_bridges.c (added),
+ include/asterisk/stasis_http.h (removed), res/res_ari_playback.c
+ (added), res/res_ari_model.exports.in,
+ res/res_stasis_http_recordings.c (removed),
+ include/asterisk/ari.h (added), res/res_ari_channels.c (added),
+ res/res_stasis_http_events.c (removed), res/res_ari_model.c,
+ res/ari.make (added), res/ari/resource_playback.c (added),
+ res/res_stasis_http_sounds.c (removed),
+ res/ari/resource_playback.h (added), res/ari/resource_channels.c
+ (added), res/ari/cli.c (added),
+ rest-api-templates/ari_model_validators.c.mustache,
+ res/ari/resource_channels.h (added),
+ res/res_stasis_http_asterisk.c (removed),
+ rest-api-templates/res_stasis_http_resource.c.mustache (removed),
+ rest-api-templates/make_ari_stubs.py,
+ rest-api-templates/res_ari_resource.c.mustache (added),
+ res/Makefile, configs/ari.conf.sample,
+ res/ari/ari_model_validators.c (added),
+ res/ari/ari_model_validators.h (added), res/res_ari.exports.in
+ (added), rest-api-templates/stasis_http_resource.c.mustache
+ (removed), tests/test_stasis_http.c (removed),
+ res/res_stasis_http.c (removed), res/ari/ari_websockets.c
+ (added), rest-api-templates/ari_resource.c.mustache (added),
+ tests/test_ari.c (added), res/res_stasis_http_playback.c
+ (removed), res/stasis_http (removed), res/res_ari.c (added),
+ res/ari/resource_endpoints.c (added),
+ rest-api-templates/ari_model_validators.h.mustache,
+ res/res_stasis_http_channels.c (removed): Rename everything
+ Stasis-HTTP to ARI This renames all files and API calls from
+ several variants of Stasis-HTTP to ARI including: * Stasis-HTTP
+ -> ARI * STASIS_HTTP -> ARI * stasis_http -> ari (ast_ari for
+ global symbols, file names as well) * stasis http -> ARI Review:
+ https://reviewboard.asterisk.org/r/2706/ (closes issue
+ ASTERISK-22136)
+
+ * tests/test_cel.c: Improve reliability of bridge merge CEL test
+
+2013-07-26 21:34 +0000 [r395559-395574] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_bridgewait.c, apps/app_confbridge.c,
+ include/asterisk/bridge_features.h, include/asterisk/parking.h,
+ main/bridge_channel.c, res/parking/parking_bridge_features.c,
+ apps/app_agent_pool.c, apps/confbridge/conf_config_parser.c,
+ bridges/bridge_builtin_features.c, main/parking.c, main/bridge.c,
+ main/bridge_basic.c, main/features.c,
+ bridges/bridge_builtin_interval_features.c: Remove the unsafe
+ bridge parameter from ast_bridge_hook_callback's. Most hook
+ callbacks did not need the bridge parameter. The pointer value
+ could become invalid if the channel is moved to another bridge
+ while it is executing. * Fixed some issues in
+ feature_attended_transfer() as a result. * Reduce the bridge
+ inhibit count in attended_transfer_properties_shutdown() after it
+ has restored the bridge channel hooks. * Removed basic bridge
+ requirement on feature_blind_transfer(). It does not require the
+ basic bridge like feature_attended_transfer().
+
+ * bridges/bridge_builtin_interval_features.c,
+ apps/app_bridgewait.c, include/asterisk/bridge_features.h,
+ res/parking/parking_bridge_features.c, main/bridge.c: Improved
+ feature limits interval hook implementaion. * Fixed feature
+ limits to not use special members of struct ast_bridge_features.
+ * Fixed memory leak in off nominal paths of
+ bridge_builtin_set_limits(). * Fixed off nominal path in
+ ast_bridge_features_limits_construct() freeing unallocated memory
+ if it was not called by bridge_builtin_set_limits(). * Made
+ bridge_builtin_interval_features.so unloadable. * Simplified
+ parking's use of its duration interval hook. * Made BridgeWait S
+ option not depend upon another module being loaded. (closes issue
+ ASTERISK-22107) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2701/
+
+2013-07-26 17:42 +0000 [r395527] David M. Lee <dlee@digium.com>
+
+ * res/stasis_http/resource_events.c, res/stasis/app.c: Fix
+ /stasis/res/app_replaced unit test. A typo in recent changes
+ caused the JSON ApplicationReplaced message to fail to build, so
+ the message wasn't being sent out the WebSocket. Related, the
+ replaced application would also unregister itself when it
+ disconnected, which would actually unregister the new
+ application. This was also fixed.
+
+2013-07-26 16:34 +0000 [r395509] Jonathan Rose <jrose@digium.com>
+
+ * main/bridge_channel.c, include/asterisk/bridge.h,
+ include/asterisk/bridge_channel_internal.h, main/bridge.c,
+ apps/app_bridgewait.c: Add name argument to BridgeWait() so
+ multiple holding bridges may be used Changes arguments for
+ BridgeWait from BridgeWait(role, options) to
+ BridgeWait(bridge_name, role, options). Now multiple holding
+ bridges may be created and referenced by this application.
+ (closes issue ASTERISK-21922) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2642/
+
+2013-07-26 00:03 +0000 [r395466-395477] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_bridgewait.c: Remove some unnecessary parentheses.
+
+ * bridges/bridge_builtin_interval_features.c: Revision
+
+2013-07-25 20:54 +0000 [r395439-395455] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_session.c: Fix crash due to trying to send a
+ re-invite while in the incorrect state. This crash would occur if
+ a re-invite was queued while the initial INVITE transaction was
+ still occurring and the response to the INVITE was not ACKed.
+ This lack of ACK would cause the INVITE session state to never
+ reach confirmed. Once the transaction terminated, however, the
+ queued re-invite would occur and cause a crash due to this lack
+ of state change. This fix checks the INVITE session state before
+ performing the re-invite to ensure it is in the required
+ confirmed state.
+
+ * res/res_sip.c, res/res_sip/sip_configuration.c: Change the
+ default value for "allowsubscribe" to yes to match chan_sip.
+
+2013-07-25 18:27 +0000 [r395430] Richard Mudgett <rmudgett@digium.com>
+
+ * main/stasis_bridges.c, include/asterisk/bridge_after.h,
+ include/asterisk/bridge_channel_internal.h,
+ main/manager_bridges.c, include/asterisk/bridge_channel.h,
+ main/bridge_after.c, include/asterisk/bridge_technology.h,
+ include/asterisk/bridge_internal.h,
+ include/asterisk/bridge_features.h, main/bridge_channel.c,
+ include/asterisk/bridge.h, include/asterisk/bridge_basic.h,
+ include/asterisk/bridge_roles.h, main/bridge.c,
+ main/bridge_basic.c, include/asterisk/stasis_bridges.h,
+ main/bridge_roles.c: Restore bridging files history.
+
+2013-07-25 15:29 +0000 [r395367-395410] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/features.h, main/features.c: Remove some dead
+ parking call Since nothing is using these global parking
+ functions, remove them! The first of many.
+
+ * main/features.c: Remove dead bridging code from features This
+ removes the previously #if 0'd code. The functionality removed
+ has either been subsumed by the Bridging API or is no longer
+ applicable.
+
+ * main/cli.c, main/cdr.c, main/manager_bridges.c, main/manager.c,
+ res/stasis_http/resource_bridges.c, tests/test_cel.c,
+ res/res_stasis.c, main/stasis_bridges.c, tests/test_cdr.c: Fix
+ incorrect reference to stasis/bridging.h
+
+ * res/parking/parking_controller.c, main/core_unreal.c,
+ bridges/bridge_simple.c,
+ include/asterisk/bridging_channel_internal.h (removed),
+ apps/app_queue.c, channels/sig_pri.c, apps/app_agent_pool.c,
+ res/parking/res_parking.h, channels/chan_misdn.c,
+ include/asterisk/bridging_internal.h (removed),
+ main/bridge_after.c (added),
+ apps/confbridge/confbridge_manager.c,
+ include/asterisk/bridging_basic.h (removed), tests/test_cel.c,
+ apps/confbridge/conf_chan_announce.c, channels/chan_unistim.c,
+ main/bridging_roles.c (removed), include/asterisk/bridge_roles.h
+ (added), res/stasis/control.c, bridges/bridge_holding.c,
+ bridges/bridge_softmix.c, channels/chan_iax2.c, main/cli.c,
+ apps/confbridge/conf_config_parser.c,
+ res/res_stasis_bridge_add.c, main/manager_bridges.c (added),
+ include/asterisk/bridging_technology.h (removed),
+ bridges/bridge_builtin_features.c, channels/chan_skinny.c,
+ include/asterisk/bridge_channel.h (added),
+ main/manager_bridging.c (removed),
+ include/asterisk/bridge_technology.h (added),
+ apps/app_confbridge.c, main/stasis_channels.c,
+ include/asterisk/bridge_features.h (added),
+ include/asterisk/bridge.h (added), tests/test_cdr.c,
+ main/bridging_basic.c (removed), include/asterisk/bridge_basic.h
+ (added), channels/chan_sip.c, include/asterisk/stasis_bridges.h
+ (added), include/asterisk/bridging_after.h (removed),
+ res/stasis_http/resource_bridges.c, main/bridge_roles.c (added),
+ res/parking/parking_applications.c, main/core_local.c,
+ main/cel.c, include/asterisk/stasis_bridging.h (removed),
+ res/parking/parking_bridge_features.c,
+ include/asterisk/bridging_channel.h (removed),
+ include/asterisk/features.h,
+ include/asterisk/bridge_channel_internal.h (added),
+ channels/dahdi/bridge_native_dahdi.c, funcs/func_channel.c,
+ include/asterisk/bridging.h (removed),
+ include/asterisk/bridge_internal.h (added),
+ bridges/bridge_native_rtp.c, main/bridge_channel.c (added),
+ res/parking/parking_manager.c, include/asterisk/core_unreal.h,
+ include/asterisk/doxygen/architecture.h, main/parking.c,
+ res/res_sip_refer.c, main/bridge.c (added), main/bridge_basic.c
+ (added), apps/app_bridgewait.c, main/stasis_bridges.c (added),
+ include/asterisk/bridge_after.h (added), main/bridging_after.c
+ (removed), res/parking/parking_bridge.c, main/channel.c,
+ channels/chan_dahdi.c, main/manager.c, channels/sig_analog.c,
+ apps/confbridge/include/confbridge.h, main/stasis_bridging.c
+ (removed), include/asterisk/bridging_features.h (removed),
+ res/stasis_http/resource_channels.c, main/bridging_channel.c
+ (removed), apps/app_dumpchan.c, main/features.c,
+ bridges/bridge_builtin_interval_features.c,
+ include/asterisk/bridging_roles.h (removed), main/bridging.c
+ (removed), channels/chan_mgcp.c, apps/app_dial.c,
+ channels/chan_bridge_media.c, apps/confbridge/conf_chan_record.c:
+ A great big renaming patch This patch renames the bridging* files
+ to bridge*. This may seem pedantic and silly, but it fits better
+ in line with current Asterisk naming conventions: * channel is
+ not "channeling" * monitor is not "monitoring" etc. A bridge is
+ an object. It is a first class citizen in Asterisk. "Bridging" is
+ the act of using a bridge on a set of channels - and the API that
+ fulfills that role is more than just the action. (closes issue
+ ASTERISK-22130)
+
+ * include/asterisk/bridging_technology.h,
+ include/asterisk/bridging_internal.h,
+ bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h, funcs/func_channel.c,
+ main/bridging_channel.c, main/features.c,
+ include/asterisk/bridging.h,
+ bridges/bridge_builtin_interval_features.c, main/bridging.c,
+ main/bridging_basic.c, apps/app_dial.c,
+ include/asterisk/bridging_after.h (added),
+ bridges/bridge_softmix.c,
+ include/asterisk/bridging_channel_internal.h, apps/app_queue.c,
+ res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
+ include/asterisk/bridging_channel.h, main/bridging_after.c
+ (added): Move after bridge callbacks into their own file One more
+ major refactoring to go.
+
+2013-07-25 00:44 +0000 [r395351] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_gulp.c, res/res_sip_session.c,
+ res/res_sip/sip_distributor.c: Improve initial INVITE handling
+ and fix crash due to rapidly arriving CANCEL. (closes issue
+ ASTERISK-22150) Review: https://reviewboard.asterisk.org/r/2696/
+
+2013-07-24 23:40 +0000 [r395316-395340] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging_channel.c,
+ include/asterisk/bridging_channel_internal.h, main/bridging.c,
+ include/asterisk/bridging_features.h: Simplify interval hooks
+ since there is only one bridge threading model now. * Convert
+ interval timers to use the ast_waitfor_nandfds() timeout. *
+ Remove bridge channel action for intervals. Now the main loop
+ handles running interval hooks.
+
+ * include/asterisk/bridging_features.h, main/bridging_channel.c,
+ apps/app_confbridge.c, main/bridging.c: Refactor
+ ast_bridge_features struct. * Reduced the number of hook
+ containers to just dtmf_hooks, interval_hooks, and other_hooks.
+ As a result, several functions dealing with the different hook
+ containers could be combined. * Extended the generic hook struct
+ for DTMF and interval hooks instead of using a variant record. *
+ Merged the special talk detector hook into the other_hooks
+ container. * Replaced ast_bridge_features_set_talk_detector()
+ with ast_bridge_talk_detector_hook(). (issue ASTERISK-22107)
+
+ * main/features.c: * Refactor setup_bridge_features_builtin(). *
+ Add an error message so you know when a feature is not available
+ and you tried to use it. It usually means the module has not been
+ loaded.
+
+2013-07-24 19:32 +0000 [r395295-395298] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.exports.in: Export exports.in as well Because is is
+ rather needed.
+
+ * bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h, main/bridging_channel.c,
+ bridges/bridge_builtin_interval_features.c,
+ include/asterisk/bridging_channel_internal.h, main/bridging.c,
+ res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
+ include/asterisk/bridging_channel.h, main/bridging_basic.c:
+ Update bridge_channel refactorings; export bridge_ symbol
+
+2013-07-24 18:51 +0000 [r395283] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/install_prereq: Add pjproject to install_prereq.
+ Also fixes spacing, in passing. (closes issue ASTERISK-22131)
+
+2013-07-24 18:08 +0000 [r395267-395271] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_sip.c: Tweak another magic number
+
+ * main/manager_bridging.c: Make AMI BridgeInfo action more verbose
+ Ensure that the BridgeInfo command provides adequate state
+ information about channels by publishing the full channel
+ snapshot for BridgeInfoChannel subevents. This prevents a
+ two-stage lookup since most consumers will be keying on channel
+ names instead of uniqueids. (closes issue ASTERISK-22140)
+
+ * res/res_sip/sip_global_headers.c: Tweak a magic number (closes
+ issue ASTERISK-22146)
+
+2013-07-24 16:01 +0000 [r395254-395255] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridging_channel.h, main/channel.c,
+ main/bridging_channel.c,
+ include/asterisk/bridging_channel_internal.h: Add missing
+ end-of-file line terminators.
+
+ * bridges/bridge_native_rtp.c: Add missing line terminator to debug
+ message.
+
+2013-07-24 15:38 +0000 [r395253] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/channel.h,
+ include/asterisk/bridging_channel_internal.h (added),
+ res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
+ include/asterisk/bridging_channel.h (added),
+ res/parking/parking_bridge.c, include/asterisk/features.h,
+ main/channel.c, include/asterisk/bridging_technology.h,
+ include/asterisk/bridging_internal.h,
+ bridges/bridge_builtin_features.c, main/bridging_channel.c
+ (added), main/features.c, include/asterisk/bridging.h,
+ bridges/bridge_builtin_interval_features.c, main/bridging.c,
+ main/bridging_basic.c: Perform the initial renaming of the
+ Bridging API This patch does the following: * It pulls out
+ bridge_channel and puts it into its own translation unit * It
+ adds public and protected headers for bridging_channel. Protected
+ functions are appropriate only for the Bridging API and
+ sub-classes of a bridge. (issue ASTERISK-22130)
+
+2013-07-24 14:35 +0000 [r395243] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Let the compiler do more type checking with
+ bridge hook callbacks.
+
+2013-07-23 22:32 +0000 [r395227] Joshua Colp <jcolp@digium.com>
+
+ * bridges/bridge_native_rtp.c: Fix a check in bridge_native_rtp
+ which determined if attaching the framehook failed or not.
+
+2013-07-23 21:32 +0000 [r395215] Jonathan Rose <jrose@digium.com>
+
+ * main/bridging_basic.c, funcs/func_channel.c,
+ include/asterisk/bridging_basic.h: func_channel: dtmf_features
+ setting Allows reading andsetting dtmf features via a channel
+ function CHANNEL(dtmf_features) (closes issue ASTERISK-21876)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2648/
+
+2013-07-23 21:14 +0000 [r395203-395205] Joshua Colp <jcolp@digium.com>
+
+ * bridges/bridge_native_rtp.c: Add some debug messages to make it
+ clear what RTP bridging functionality is in use.
+
+ * bridges/bridge_native_rtp.c: Fix some logic so native RTP bridge
+ will occur when monitor, audiohooks, or framehooks are not
+ present.
+
+2013-07-23 19:14 +0000 [r395188] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, include/asterisk/bridging.h,
+ bridges/bridge_softmix.c: Pull softmix bridge parameters into a
+ sub structure.
+
+2013-07-23 18:41 +0000 [r395183] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_gulp.c: Drop the reference count on the correct
+ object.
+
+2013-07-23 18:41 +0000 [r395154-395182] Richard Mudgett <rmudgett@digium.com>
+
+ * main/utils.c, channels/chan_dahdi.c: Reinclude sys/stat.h in
+ chan_dahdi.c and remove redundant include in utils.c
+
+ * channels/dahdi/bridge_native_dahdi.c, channels/chan_dahdi.h,
+ channels/chan_mgcp.c, channels/chan_dahdi.c: Some chan_dahdi
+ protected function renaming. analog_lib_handles -->
+ dahdi_analog_lib_handles enable_dtmf_detect -->
+ dahdi_dtmf_detect_enable disable_dtmf_detect -->
+ dahdi_dtmf_detect_disable dahdi_enable_ec --> dahdi_ec_enable
+ dahdi_disable_ec --> dahdi_ec_disable update_conf -->
+ dahdi_conf_update dahdi_link --> dahdi_master_slave_link
+ dahdi_unlink --> dahdi_master_slave_unlink (closes issue
+ ASTERISK-22129) Reported by: rmudgett
+
+ * channels/dahdi (added), channels/dahdi/bridge_native_dahdi.h,
+ bridges/bridge_softmix.c, channels/Makefile, main/bridging.c,
+ channels/chan_dahdi.c, channels/dahdi/bridge_native_dahdi.c,
+ channels/chan_dahdi.h (added): Restore chan_dahdi native bridging
+ and PRI tromboned call elimination. Created a native_dahdi
+ bridging technology for use with the new bridging API. The new
+ bridging technology is part of the chan_dahdi channel driver
+ because it is very specific to that driver. Rather than include
+ the new code directly into chan_dahdi.c the new bridge technology
+ is in its own file and linked into chan_dahdi.so. A large part of
+ this change is the mechanical process of moving declarations
+ around so chan_dahdi.c can be split up into more files later. *
+ Changed the bridging core to pass NULL frames into the channel
+ technologies instead of discarding them. The channel technologies
+ may need the proding to determine if their configuration is still
+ valid. (closes issue ASTERISK-21886) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2681/
+
+2013-07-23 15:28 +0000 [r395151] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stasis_bridging.h, main/bridging.c,
+ main/bridging_basic.c, main/bridging_roles.c,
+ include/asterisk/bridging_internal.h (added),
+ bridges/bridge_builtin_features.c, main/stasis_bridging.c,
+ include/asterisk/bridging_features.h,
+ include/asterisk/features_config.h, include/asterisk/bridging.h,
+ main/features.c, include/asterisk/bridging_roles.h, main/cel.c,
+ main/features_config.c: Make DTMF attended transfer support
+ feature-complete. This greatly modifies the operation of DTMF
+ attended transfers so that the full range of options from
+ features.conf applies. In addition, a new option has been added
+ that allows for a transferer to switch between bridges during a
+ transfer before completing the transfer. (closes issue
+ ASTERISK-21543) reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2654
+
+2013-07-23 14:57 +0000 [r395136] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http_channels.c, res/res_stasis_http_sounds.c,
+ res/res_stasis_http_bridges.c, res/res_stasis_http_recordings.c,
+ res/res_stasis_http.c, res/res_stasis_http_endpoints.c,
+ res/res_stasis_http_asterisk.c, res/res_stasis_http_playback.c,
+ rest-api-templates/res_stasis_http_resource.c.mustache: No more
+ teapots. Now that the ARI implementation is nearing some
+ definition of completeness, we should properly respond with 501's
+ for unimplemented functionality, instead of the almost humorous
+ 418.
+
+2013-07-23 14:49 +0000 [r395135] Matthew Jordan <mjordan@digium.com>
+
+ * main/channel.c: Kill the zombies In previous versions of
+ Asterisk, the zombies roamed freely, unchecked and uncontrolled.
+ They ravaged Asterisk systems with their biting and their nashing
+ and their pointy teeth. Sometimes, you couldn't even hang them
+ up. Now, zombies are rare. They still *technically* exist in
+ certain places, but they are controlled. Kind of like a zombie
+ zoo: you can see them, but you can't touch them, and they can't
+ touch you. Bring your kids! Because zombies are now population
+ controlled with a very short lifespan, there's no reason to
+ rename the channels to '%s<ZOMBIE>'. The channels are guaranteed
+ to die off quickly; the rename really is just confusing at this
+ point. This patch finally removes the renaming. On the plus side:
+ this made my life easier in CDRs during call pickup and attended
+ transfers to an Asterisk application. It will make other folks
+ lives easier as well! Review:
+ https://reviewboard.astierks.org/r/2690/ (closes issue
+ ASTERISK-21699) Reported by: Matt Jordan
+
+2013-07-23 13:52 +0000 [r395121] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_sip_sdp_rtp.c, channels/chan_gulp.c, res/res_sip.c,
+ channels/chan_sip.c, res/res_sip/sip_configuration.c,
+ res/res_sip_session.c, include/asterisk/res_sip.h,
+ include/asterisk/res_sip_session.h: Add DTLS-SRTP support to
+ chan_pjsip This patch introduces DTLS-SRTP support to chan_pjsip
+ and the options necessary to configure it including an option to
+ allow choosing between 32 and 80 byte SRTP tag lengths. During
+ the implementation and testing of this patch, three other bugs
+ were found and their fixes are included with this patch. The two
+ in chan_sip were a segfault relating to DTLS setup and mistaken
+ call rejection. The third bug fix prevents chan_pjsip from
+ attempting to perform bridge optimization between two endpoints
+ if either of them is running any form of SRTP. Review:
+ https://reviewboard.asterisk.org/r/2683/ (closes issue
+ ASTERISK-21419)
+
+2013-07-23 13:42 +0000 [r395118-395120] David M. Lee <dlee@digium.com>
+
+ * res/stasis/app.c, res/stasis/app.h, res/res_stasis.c: Continue
+ events when ARI WebSocket reconnects This patch addresses a bug
+ in the /ari/events WebSocket in handling reconnects. When a
+ Stasis application's associated WebSocket was disconnected and
+ reconnected, it would not receive events for any channels or
+ bridges it was subscribed to. The fix was to lazily clean up
+ Stasis application registrations, instead of removing them as
+ soon as the WebSocket goes away. When an application is
+ unregistered at the WebSocket level, the underlying application
+ is simply deactivated. If the application WebSocket is
+ reconnected, the application is reactivated for the new
+ connection. To avoid memory leaks from lingering, unused
+ application, the application list is cleaned up whenever new
+ applications are registered/unregistered. (closes issue
+ ASTERISK-21970) Review: https://reviewboard.asterisk.org/r/2678/
+
+ * main/cdr.c, main/stasis_message_router.c,
+ main/manager_bridging.c,
+ include/asterisk/stasis_message_router.h, tests/test_stasis.c,
+ main/manager_channels.c: Fix bridge/channel AMI event ordering
+ issues The stasis_cache_update messages are somewhat cumbersome
+ to handle with the stasis_message_router. Since all updates have
+ the same message type, they are normally handled with the same
+ route. Since caching itself is a first class component of
+ stasis-core, it makes sense for the router to handle the cache
+ update messages itself. This patch adds
+ stasis_message_router_add_cache_update() and
+ stasis_message_router_remove_cache_update() to handle the routing
+ of stasis_cache_update messages. This patch also corrects an
+ issue with manager_{bridging,channels}.c, where events might be
+ reordered. The reordering occurs because the components use
+ different message routers, which they needed because they both
+ needed to route cache update messages. They now both use
+ manager's router, and add cache routes for just the cache updates
+ they are interested in. (closes issue ASTERISK-22038) Review:
+ https://reviewboard.asterisk.org/r/2677/
+
+2013-07-23 12:56 +0000 [r395107] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_sip/sip_options.c: Add missing newline
+
+2013-07-23 12:27 +0000 [r395102] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/res_sip_session.h,
+ res/res_sip_session.exports.in, channels/chan_gulp.c,
+ res/res_sip_session.c: Expose the chan_pjsip implementation pvt
+ and session in a defined manner. This allows modules outside of
+ chan_pjsip itself to get the session given only an Asterisk
+ channel. Review: https://reviewboard.asterisk.org/r/2674/
+
+2013-07-23 00:16 +0000 [r395089] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Fix unbalanced lock when serializing CDR variables
+ I'm only surprised that this didn't cause larger problems.
+
+2013-07-23 00:02 +0000 [r395088] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Remove some BUGBUG notes that have been handled.
+
+2013-07-22 20:42 +0000 [r395074] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_cel.c: Make the CEL blind transfer test pass
+ consistently
+
+2013-07-22 13:52 +0000 [r394881-395034] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.c, /: Update copyright year to 2013 in asterisk.c;
+ some whitespace fixes (closes issue ASTERISK-22179) Reported by:
+ Malcolm Davenport ........ Merged revisions 395032 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 395033 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * funcs/func_channel.c, /: Clean up documentation This patch cleans
+ up documentation in func_channel for the following items: *
+ rtpsource * secure_signaling * secure_media * various OOH323
+ parameters (closes issue ASTERISK-20969) Reported by: snuffy
+ patches: func_chan-update.diff uploaded by snuffy (License 5024)
+ ........ Merged revisions 394980 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394981 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, configs/indications.conf.sample: Provide proper ring tone in
+ indications.conf for Malaysia The ring tone provided in the
+ sample indications.conf was incorrect. This patch modifies the
+ sample ring tone to be what it should: ring =
+ 425/400,0/200,425/400,0/2000 This brings it in line with the tone
+ definition in DAHDI 2.7.0. (zonedata.c) (closes issue
+ ASTERISK-21997) Reported by: Filip Jenicek patches:
+ malaysia_ring.patch uploaded by phill (License 6277) ........
+ Merged revisions 394940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394941 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * contrib/scripts/safe_asterisk, Makefile,
+ configs/safe_asterisk.conf.sample (added), CHANGES: Always
+ install safe_asterisk; add configuration file support This patch
+ modifies the behavior of safe_asterisk in two ways: (1) It
+ modifies the Asterisk Makefile such that safe_asterisk is always
+ installed on a 'make install'. This was done as bugfixes in the
+ safe_asterisk script were not applied in previous version of
+ Asterisk without first removing the old version of the script.
+ (2) In order to keep a newly installed version of safe_asterisk
+ from impacting local modifications, a new config file -
+ safe_asterisk.conf.sample - has been provided. Settings that were
+ previously modified in safe_asterisk can be set there instead.
+ (closes issue ASTERISK-21965) Reported by: Jeremy Kister patches:
+ safe_asterisk.patch uploaded by jkister (License 6232)
+
+ * /, main/http.c: Tolerate presence of RFC2965 Cookie2 header by
+ ignoring it This patch modifies parsing of cookies in Asterisk's
+ http server by doing an explicit comparison of the "Cookie"
+ header instead of looking at the first 6 characters to determine
+ if the header is a cookie header. This avoids parsing "Cookie2"
+ headers and overwriting the previously parsed "Cookie" header.
+ Note that we probably should be appending the cookies in each
+ "Cookie" header to the parsed results; however, while clients can
+ send multiple cookie headers they never really do. While this
+ patch doesn't improve Asterisk's behavior in that regard, it
+ shouldn't make it any worse either. Note that the solution in
+ this patch was pointed out on the issue by the issue reporter,
+ Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
+ Stuart Henderson Tested by: mjordan, Stuart Henderson ........
+ Merged revisions 394899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394900 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
+ realtime scripts with schema for queue_log table This patch
+ updates the realtime SQL scripts with an entry that will create
+ the queue_log table. This brings the PostgreSQL scripts inline
+ with the MySQL scripts, with respect to what tables they will
+ create. (closes issue ASTERISK-21021) Reported by: Eugene
+ patches: queue_log.sql uploaded by varnav (license 6360) ........
+ Merged revisions 394896 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394897 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/iax2/parser.c: Add additional control frame types to the
+ IAX2 parser for debug messages This patch adds some of the more
+ recent control frame types to the IAX2 parser. When IAX2
+ debugging is enabled, it will now show more of the control frame
+ types. (closes issue ASTERISK-22120) Reported by: Birger "WIMPy"
+ Harzenetter patches: iaxcmds.diff uploaded by wimpy
+
+ * configs/iax.conf.sample, /: Document connectedline parameter for
+ chan_iax2 The connectedline parameter for a chan_iax2 peer was
+ undocumented. This patch documents the options in the sample
+ configuration file. (closes issue ASTERISK-21953) Reported by:
+ Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394890 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * CHANGES, main/manager.c: Allow setting allowmultiplelogin on an
+ account basis This patch modifies manager to allow the
+ allowmultiplelogin setting to be set on an account by account
+ basis. When set in the general context, it will act as the
+ default for the defined accounts. Setting it in the account will
+ override the general setting. (closes issue ASTERISK-21324)
+ Reported by: vldmr patches:
+ asterisk-manager-per-user-allowmultiplelogin.patch uploaded by
+ vldmr (License 6487)
+
+2013-07-20 13:25 +0000 [r394858-394870] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_cel.c, CHANGES, main/cel.c, main/asterisk.c,
+ include/asterisk/cel.h: Add CEL local optimization record type
+ This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to
+ represent local channel optimizations. Local channel
+ optimizations were one of several things conveyed by the now
+ defunct BRIDGE_UPDATE event type. This also adds a unit test to
+ test generation of this new CEL event. Review:
+ https://reviewboard.asterisk.org/r/2676/
+
+ * channels/chan_sip.c, include/asterisk/cel.h,
+ apps/app_celgenuserevent.c, apps/app_directed_pickup.c,
+ main/features.c, tests/test_cel.c, CHANGES, apps/app_queue.c,
+ main/cel.c, apps/app_dial.c, main/channel.c,
+ channels/chan_dahdi.c, main/pbx.c, channels/sig_analog.c: Add
+ transfer support to CEL This adds CEL support for blind and
+ attended transfers and call pickup. During the course of adding
+ this functionality I noticed that CONF_ENTER, CONF_EXIT, and
+ BRIDGE_TO_CONF events are particularly useless without a bridge
+ identifier, so I added that as well. This adds tests for blind
+ transfers, several types of attended transfers, and call pickup.
+ The extra field in CEL records now consists of a JSON blob whose
+ fields are defined on a per-event basis. Review:
+ https://reviewboard.asterisk.org/r/2658/ (closes issue
+ ASTERISK-21565)
+
+2013-07-20 01:11 +0000 [r394825-394846] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: Regroup the ao2 search_flags. Moved
+ the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags together into
+ a field and renamed them to OBJ_SEARCH_OBJECT, OBJ_SEARCH_KEY,
+ and OBJ_SEARCH_PARTIAL_KEY respectively. The values were selected
+ to keep existing code compiling and working until the codebase
+ can be changed to stop using these values as bit flags and use
+ them as an enum field. The old names are defined to the new names
+ for backward compatibility.
+
+ * main/channel.c, include/asterisk/audiohook.h, main/audiohook.c:
+ Minor optimizations. * Made ast_audiohook_detach_list() and
+ ast_audiohook_write_list_empty() NULL tolerant. * Made
+ ast_audiohook_detach_list() return void since it is a destructor.
+
+ * bridges/bridge_native_rtp.c, main/bridging.c, main/channel.c,
+ include/asterisk/channel.h: Extract a repeated test into
+ ast_channel_has_audio_frame_or_monitor().
+
+2013-07-19 19:40 +0000 [r394809-394810] Jonathan Rose <jrose@digium.com>
+
+ * res/stasis/control.c, res/stasis_http/resource_channels.c,
+ res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
+ res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json: ARI: MOH start and stop for a
+ channel (issue ASTERISK-21974) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2680/
+
+ * include/asterisk/logger.h, res/stasis_http/resource_channels.c,
+ rest-api/api-docs/playback.json, rest-api/api-docs/channels.json,
+ res/res_stasis_http_bridges.c, res/res_stasis.c,
+ rest-api/api-docs/recordings.json,
+ include/asterisk/core_unreal.h, res/res_stasis_http_playback.c,
+ res/res_stasis_playback.c, channels/chan_bridge_media.c (added),
+ res/stasis/control.c, res/stasis_http/ari_model_validators.c,
+ res/res_stasis_http_channels.c, main/core_unreal.c,
+ include/asterisk/stasis_app.h,
+ res/stasis_http/resource_bridges.c,
+ res/stasis_http/ari_model_validators.h,
+ res/stasis_http/resource_bridges.h,
+ include/asterisk/stasis_app_playback.h,
+ rest-api/api-docs/bridges.json: ARI: Bridge Playback, Bridge
+ Record Adds a new channel driver for creating channels for
+ specific purposes in bridges, primarily to act as either
+ recorders or announcers. Adds ARI commands for playing
+ announcements to ever participant in a bridge as well as for
+ recording a bridge. This patch also includes some
+ documentation/reponse fixes to related ARI models such as
+ playback controls. (closes issue ASTERISK-21592) Reported by:
+ Matt Jordan (closes issue ASTERISK-21593) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2670/
+
+2013-07-19 19:23 +0000 [r394795-394808] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager_bridging.c, include/asterisk/channel.h,
+ main/stasis_channels.c, main/cel.c,
+ apps/confbridge/conf_chan_announce.c, main/manager_channels.c,
+ res/parking/parking_manager.c, main/cdr.c,
+ include/asterisk/stasis_channels.h,
+ apps/confbridge/conf_chan_record.c,
+ apps/confbridge/confbridge_manager.c: Filter channels used as
+ internal mechanisms This adds new flags to the channel tech
+ properties that flag it as different types of implementation
+ detail used exclusively to provide a feature. Examples of
+ channels that would have these flags include the announcement and
+ recording channels used by confbridge which are the only two
+ marked as such by this patch. Review:
+ https://reviewboard.asterisk.org/r/2633/ (closes issue
+ ASTERISK-21873)
+
+ * channels/chan_sip.c: Fix crash when using temporary peers
+ Temporary peers do not have an associated Stasis endpoint and
+ quite a bit of code in chan_sip assumes that all peers have a
+ Stasis endpoint. All endpoint accesses in chan_sip are now
+ wrapped in an endpoint NULL-check.
+
+2013-07-19 18:00 +0000 [r394793] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/stasis_system.h, main/stasis_system.c,
+ main/ccss.c: Convert CCSS manager events to stasis. (closes issue
+ ASTERISK-21473) Review: https://reviewboard.asterisk.org/r/2682/
+
+2013-07-19 17:55 +0000 [r394776-394791] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Made audiohooks, framehooks, and monitor prevent
+ local channel optimization. Audiohooks, framehooks, and monitor
+ represent state on a local channel that will go away if it is
+ optimized out. (closes issue ASTERISK-21954) Reported by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/2685/
+
+ * include/asterisk/channel.h: Fixup doxygen on ast_hangup().
+
+2013-07-18 19:25 +0000 [r394759] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_sip_sdp_rtp.c, channels/chan_gulp.c,
+ res/res_sip_caller_id.c, res/res_sip.c, res/res_sip_session.c,
+ res/res_sip/sip_global_headers.c (added),
+ res/res_sip/config_system.c (added),
+ res/res_sip_one_touch_record_info.c, res/res_sip_mwi.c,
+ res/res_sip_pubsub.c, res/res_sip/config_transport.c,
+ res/res_sip/sip_configuration.c, res/res_sip_refer.c,
+ include/asterisk/res_sip.h, res/res_sip/config_global.c (added),
+ res/res_sip/include/res_sip_private.h, res/res_sip.exports.in:
+ Add a bunch of options from sip.conf to res_sip.conf For a
+ complete list of the options added, see the review linked at the
+ bottom of this commit message. (closes issue ASTERISK-21506)
+ reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2671
+
+2013-07-18 18:05 +0000 [r394744] David M. Lee <dlee@digium.com>
+
+ * res/res_http_websocket.c: Fixed null dereference when WebSocket
+ subprotocol isn't specified
+
+2013-07-18 16:49 +0000 [r394731] Jonathan Rose <jrose@digium.com>
+
+ * main/bridging_roles.c, bridges/bridge_holding.c,
+ apps/app_bridgewait.c: bridge_holding/app_bridgewait: Add new
+ entertainment options This patch adds more entertainment options
+ to holding bridges and the bridge_wait application. Also, holding
+ bridges will now use music on hold as the default entertainment
+ option instead of none. The parameters for app_bridgewait have
+ changed to (role, options) from the previous (options) and the
+ options themselves have changed as well (entertainment options
+ are now contained in an enumerator, role specification is handled
+ by the role parameter, etc) (closes issue ASTERISK-21923)
+ Reported by: Matthew Jordan Review:
+ https://reviewboard.asterisk.org/r/2679/
+
+2013-07-18 16:03 +0000 [r394715] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, res/stasis/control.c,
+ res/stasis_http/resource_channels.c,
+ include/asterisk/stasis_app.h, include/asterisk/channel.h,
+ res/res_mutestream.c: ARI: Add support for suppressing media
+ streams. Also convert res_mutestream to use the core feature
+ behind this. (closes issue ASTERISK-21618) Review:
+ https://reviewboard.asterisk.org/r/2652/
+
+2013-07-18 14:50 +0000 [r394701] Matthew Jordan <mjordan@digium.com>
+
+ * main/http.c: Tweak debug statements This patch does two things:
+ 1. It moves the debug statement that shows the HTTP sub-protocols
+ being compared after the string length calculation such that it
+ shows the correct string length in the output 2. It adds some
+ additional debug that displays when it matches on a sub-protocol
+ and when it fails
+
+2013-07-18 14:08 +0000 [r394686] David M. Lee <dlee@digium.com>
+
+ * main/stasis_cache.c: Fix caching topic shutdown assertions The
+ recent changes to update stasis_cache_topics directly from the
+ publisher thread uncovered a race condition, which was causing
+ asserts in the /stasis/core tests. If the caching topic's
+ subscription is the last reference to the caching topic, it will
+ destroy the caching topic after the final message has been
+ processed. When dispatching to a different thread, this usually
+ gave the unsubscribe enough time to finish before destruction
+ happened. Now, however, it consistently destroys before
+ unsubscription is complete. This patch adds an extra reference to
+ the caching topic, to hold it for the duration of the
+ unsubscription. This patch also removes an extra unref that was
+ happening when the final message was received by the caching
+ topic. It was put there because of an extra ref that was put into
+ the caching topic's constructor. Both have been removed, which
+ makes the destructor a bit less confusing. Review:
+ https://reviewboard.asterisk.org/r/2675/
+
+2013-07-18 12:54 +0000 [r394642] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, res/res_agi.c: Properly indicate failure to open an audio
+ stream in res_agi If there is an error streaming an audio file,
+ the current return status makes it difficult for an AGI script to
+ determine that there was an error with the audio file. This
+ patches changes the result to return -1 and the function returns
+ RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
+ parts of res_agi, this would appear to be the proper way to
+ handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
+ Wainer Tested by: Ariel Wainer Patches:
+ asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2625/
+ ........ Merged revisions 394640 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394641 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-07-17 22:30 +0000 [r394600-394623] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, addons/chan_mobile.c,
+ tests/test_cdr.c, tests/test_stasis_endpoints.c,
+ apps/app_voicemail.c, main/channel.c, main/dial.c,
+ apps/app_meetme.c, tests/test_app.c, main/features.c,
+ tests/test_voicemail_api.c, tests/test_cel.c: Change ast_hangup()
+ to return void and be NULL safe. Since ast_hangup() is
+ effectively a channel destructor, it should be a void function. *
+ Make the few silly callers checking the return value no longer do
+ so. Only the CDR and CEL unit tests checked the return value. *
+ Make all callers take advantage of the NULL safe change and
+ remove the NULL check before the call.
+
+ * main/features.c: Remove some completed and no longer relevant
+ BUGBUG notes.
+
+2013-07-17 18:26 +0000 [r394583] Jonathan Rose <jrose@digium.com>
+
+ * apps/confbridge/conf_chan_announce.c: app_confbridge: Eliminate a
+ reference leak for confbridge announcer channels
+
+2013-07-17 17:49 +0000 [r394552-394567] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Left over spacing issues of review 726.
+
+ * channels/chan_dahdi.c: handle DAHDI_EVENT_REMOVED on a pri
+ D-Channel When a DAHDI device is removed at run-time it sends the
+ event DAHDI_EVENT_REMOVED on each channel. This is intended to
+ signal the userspace program to close the respective file handle,
+ as the driver of the device will need all of them closed to
+ properly clean-up. This event has long since been handled in
+ chan_dahdi (chan_zap at the time). However the event that is sent
+ on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This
+ commit adds handling for closing the file descriptor (and
+ shutting down the span, while we're at it). It also adds a CLI
+ command 'pri destroy span <N>' to destroy the span and its DAHDI
+ channels. Review: https://reviewboard.asterisk.org/r/726/
+
+2013-07-16 22:33 +0000 [r394530-394531] Matthew Jordan <mjordan@digium.com>
+
+ * CHANGES, apps/app_confbridge.c: Add 'kick all' capability to
+ ConfBridge CLI command This patch adds the ability to kick all
+ users out of a conference from the ConfBridge kick CLI command.
+ It is invoked by passing 'all' as the channel parameter to the
+ CLI command, i.e., "confbridge kick <conf> all". Note that this
+ patch was modified slightly to conform to trunk. (closes issue
+ ASTERISK-21827) Reported by: dorianlogan patches:
+ kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
+
+ * main/cel.c: Re-order handlers in CEL to ensure that HANGUP events
+ happen after APP_END When a channel is hungup, both an APP_END
+ event and a HANGUP event can be fired. To ensure that HANGUP
+ events occur after APP_END events, the method callbacks for the
+ APP_END event should be processed prior to the callbacks for the
+ HANGUP event.
+
+2013-07-16 21:44 +0000 [r394513] David M. Lee <dlee@digium.com>
+
+ * res/stasis_http/ari_websockets.c: Debug logging to help with
+ WebSocket connection problems
+
+2013-07-16 20:00 +0000 [r394489] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_gulp.c: chan_gulp: Fix gulp_indicate() handling of
+ AST_CONTROL_PVT_CAUSE_CODE.
+
+2013-07-16 19:13 +0000 [r394473] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_sip_session.c: Prevent crash from trying to end a session
+ in an invalid way. This ensures that code that was only meant to
+ be run on a reinvite failure only runs on a reinvite failure.
+ (closes issue ASTERISK-22061) reported by Rusty Newton
+
+2013-07-16 18:49 +0000 [r394470-394471] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_sip.c: Remove some dead code
+ dealing with old bridging method.
+
+ * bridges/bridge_simple.c: Simplify bridge_simple chan join code.
+
+2013-07-16 18:22 +0000 [r394469] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Re-order cleanup This patch attempts to fix some
+ possible race conditions in shutdown of the CDR engine. It: *
+ Adds a cleanup handler to only unsubscribe and join on stasis
+ messages during graceful shutdown. The cleanup handler should
+ execute before the regular atexit handler, as we want to
+ unsubscribe for any further messages before dispatching the CDRs.
+ * The CDRs are now locked when we dispatch them on shutdown.
+
+2013-07-16 15:30 +0000 [r394442] David M. Lee <dlee@digium.com>
+
+ * res/res_http_websocket.c: Fixed null dereference when WebSocket
+ protocol is omitted
+
+2013-07-15 23:20 +0000 [r394417] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_agent.c (removed), configs/queues.conf.sample,
+ include/asterisk/bridging.h, UPGRADE.txt, main/stasis_channels.c,
+ CHANGES, main/bridging.c, apps/app_agent_pool.c (added),
+ configs/agents.conf.sample, include/asterisk/config_options.h,
+ include/asterisk/stasis_channels.h: Replace chan_agent with
+ app_agent_pool. The ill conceived chan_agent is no more. It is
+ now replaced by app_agent_pool. Agents login using the
+ AgentLogin() application as before. The AgentLogin() application
+ no longer does any authentication. Authentication is now the
+ responsibility of the dialplan. (Besides, the authentication done
+ by chan_agent did not match what the voice prompts asked for.)
+ Sample extensions.conf [login] ; Sample agent 1001 login ; Set
+ COLP for in between calls so the agent does not see the last
+ caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent
+ Waiting" <1001>) ; Give the agent DTMF transfer and disconnect
+ features when connected to a caller. same =>
+ n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same
+ => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup()
+ [caller] ; Sample caller direct connect to agent 1001 exten =>
+ 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is
+ ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through
+ a Queue to agent 1001 exten => 900,1,Queue(agent_q) same =>
+ n,Hangup() Sample queues.conf [agent_q] member =>
+ Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation
+ overview: 1) Logged in agents wait for callers in an agents
+ holding bridge. 2) Caller requests an agent using AgentRequest()
+ 3) A basic bridge is created, the agent is notified, and caller
+ joins the basic bridge to wait for the agent. 4) The agent is
+ either automatically connected to the caller or must ack the call
+ to connect. 5) The agent is moved from the agents holding bridge
+ to the basic bridge. 6) The agent and caller talk. 7) The
+ connection is ended by either party. 8) The agent goes back to
+ the agents holding bridge. To avoid some locking issues with the
+ agent holding bridge, I needed to make some changes to the after
+ bridge callback support. The after bridge callback is now a list
+ of requested callbacks with the last to be added the only active
+ callback. The after bridge callback for failed callbacks will
+ always happen in the channel thread when the channel leaves the
+ bridging system or is destroyed. (closes issue ASTERISK-21554)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2657/
+
+2013-07-15 22:05 +0000 [r394402] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stasis_channels.h: Remove misleading
+ documentation for channel snapshot creation.
+
+2013-07-15 21:22 +0000 [r394397] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http.c: Document the ari.conf allowed_origins
+ setting
+
+2013-07-15 13:43 +0000 [r394370] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_session.c, include/asterisk/res_sip_session.h: Remove
+ some callbacks and functions which are not needed.
+
+2013-07-14 02:41 +0000 [r394278-394346] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c, /: Provide error message for QUEUE_MEMBER when
+ member is not in queue When QUEUE_MEMBER is used and the member
+ specified is not in the queue, Asterisk provides an ERROR message
+ that indicates that the option specified is not valid. This patch
+ now properly displays an ERROR message that the member is not in
+ the queue if an interface is specified. (closes issue
+ ASTERISK-21980) Reported by: Avraam David ........ Merged
+ revisions 394345 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/dns.c: Remove redundant code in dns.c Peter J Philipp
+ pointed out that there are two checks that ensure that len is not
+ less than 0. If len is less than 0, the function returns. Having
+ both of them is clearly redundant. This removes the second and
+ attempts to clarify (slightly) the error condition. (closes issue
+ ASTERISK-21772) Reported by: Peter J Philipp
+
+ * /, funcs/func_strings.c: Clarify documentation for function
+ PASSTHRU It is not apparent to the average user that the PASSTHRU
+ function should not be passed as ${PASSTHRU(string)} but just as
+ PASSTHRU(string) to functions which take a variable name and not
+ its contents. This patch clarifies the behavior in the
+ documentation and provides an example. (closes issue
+ ASTERISK-21717) Reported by: Richard Miller patches:
+ func_strings.diff uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 394302 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394303 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/bridging.c, main/cdr.c: Fix FRACK message from external
+ redirects; handle outbound channels better This patch does the
+ following: * It simplifies the Dial handling in CDRs. As a rule,
+ the caller in a dial relationship is always the Party A. There
+ was some logic present in the handling of the dial message that
+ could, conceivably, pick the caller as Party A for the beginning
+ of the dial and the peer as Party A for the end of the dial. This
+ shouldn't have happened if the code in the bridging framework was
+ doing its job; however, that was broken and it led to the FRACK.
+ As it is, this code was overly ocmplex and not needed: the
+ caller, if present, should always be Party A. Period. * It
+ properly checks to see if a channel will continue on in the
+ dialplan. ast_check_hangup - much like cake at the end - is a
+ lie. It will tell you that you are hungup when you are not. Do
+ not believe it. I would make this function tell the truth, but
+ I'm nervous that we've been depending on it sitting on its throne
+ of lies for far too long, and it would probably break lots of
+ things. So I'm just checking the "internal" soft hangup flags,
+ like everyone else. (closes issue ASTERISK-22060) Reported by:
+ Mark Michelson (issue ASTERISK-21831) Reported by: Matt Jordan
+
+ * channels/chan_sip.c: Pretty up a debug message if the
+ referred-by-uri isn't available Instead of formatting a NULL
+ pointer into a "%s" format string (which is usually not a good
+ thing to do), we instead print "Unknown".
+
+2013-07-12 22:35 +0000 [r394263] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Fix a longstanding issue with MFC-R2
+ configuration that prevented users from mixing different variants
+ or general MFC-R2 settings within the same E1 line. Most users do
+ not have a problem with this since MFC-R2 lines are usually
+ fractional E1s, or the whole E1 has the same country variant and
+ R2 settings. In Venezuela however is common to have inbound
+ MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
+ properly parses the chan_dahdi.conf file to generate a new openr2
+ context every time a new channel => section is found and the
+ configuration was changed. (closes issue ASTERISK-21117) Reported
+ by: Rafael Angulo Related Elastix issue:
+ http://bugs.elastix.org/view.php?id=1612 ........ Merged
+ revisions 394106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 394173 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-07-12 21:42 +0000 [r394249] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, main/channel_internal_api.c,
+ include/asterisk/channel.h, main/bridging.c: Add support to the
+ bridging core for performing COLP updates when channels join a 2
+ party bridge. (closes issue ASTERISK-21829) Review:
+ https://reviewboard.asterisk.org/r/2636/
+
+2013-07-12 21:01 +0000 [r394232] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridging_basic.c: Prevent potential race condition in
+ multiparty basic bridges. For more details about the race
+ condition see the linked review at the bottom of this commit
+ (closes issue ASTERISK-21882) Reported by Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2663
+
+2013-07-12 19:35 +0000 [r394216] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Fix a compiler warning.
+
+2013-07-12 18:23 +0000 [r394203] David M. Lee <dlee@digium.com>
+
+ * tests/test_json.c: Fixed intermittent crash when loading
+ test_json.so The JSON test attempted an overly clever use of
+ RAII_VAR to run code at the beginning and end of each test, in
+ order to validate that no JSON objects were leaked during the
+ test. The problem is that the validation code would run during
+ the initial load, when the tests were initialized. This happens
+ during startup, when other parts of the system might actively be
+ allocating and freeing JSON objects. This patch changes the
+ RAII_VAR to use the new ast_test_register_{init,cleanup}
+ functions to run the validations properly. (closes issue
+ ASTERISK-21978) Review: https://reviewboard.asterisk.org/r/2669/
+
+2013-07-12 17:52 +0000 [r394189] Jason Parker <jparker@digium.com>
+
+ * res/stasis_http/internal.h, res/stasis_http/config.c,
+ res/stasis_http/cli.c, res/res_stasis_http.c: ARI: Add support
+ for Cross-Origin Resource Sharing (CORS), origin headers This
+ rejects requests from any unknown origins. (closes issue
+ ASTERISK-21278) Review: https://reviewboard.asterisk.org/r/2667/
+
+2013-07-11 21:01 +0000 [r394158] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridging_technology.h: Fix bridge tech write
+ callback parameter name.
+
+2013-07-11 20:59 +0000 [r394156] David M. Lee <dlee@digium.com>
+
+ * channels/chan_skinny.c: Fixed chan_skinny for systems were
+ pthread_t isn't an int. I'm looking at you, OS X.
+
+2013-07-11 20:17 +0000 [r394147] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Refactor and cleanup of skinny session
+ handling. Major changes are to pull all packet reading functions
+ into skinny_session and move timeout handling to scheduling
+ arrangements. Thread cancelling is now undertaken directly rather
+ than waiting for the read to timeout (cleanup is popped on thread
+ cancel). Also added some keepalive timings in debugging messages.
+ Keepalive timeout has been increased from 1.1 by keepalive to 3
+ times keepalive. This seems to align (after keepalives stabilise)
+ with when devices reset after not receiving keepalives. Probably
+ needs more work, especially around the first and/or second
+ keepalives that vary significantly by device and firmware
+ version. Review: https://reviewboard.asterisk.org/r/2611/
+
+2013-07-11 16:23 +0000 [r394103] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_exten_state.c: Tweak the subscription failure warning
+ message to include endpoint name and context.
+
+2013-07-11 15:37 +0000 [r394037-394089] David M. Lee <dlee@digium.com>
+
+ * tests/test_cel.c: Correct test_cel cleanup. When I corrected the
+ CEL test crash in r394037, I didn't quite pay attention to how
+ the globals and locals were being shuffled around in the cleanup
+ callback. I removed the nulling of the global variables, which
+ caused them to be double cleaned. This patch puts the global
+ nulling code back (since the vars are cleaned up by RAII_VARs),
+ and removes the explicit ao2_cleanup() (since they were no-ops,
+ because the variables had just been nulled).
+
+ * res/stasis_http/config.c, configs/ari.conf.sample,
+ res/res_stasis_http.c: Change ARI user config to use a type field
+ When I initially wrote the configuration support for ARI users, I
+ determined the section type by a category prefix (i.e.,
+ [user-admin]). This is neither idiomatic Asterisk configuration,
+ nor is it really that user friendly. This patch replaces the
+ category prefix with a type field in the section, which is much
+ cleaner. Review: https://reviewboard.asterisk.org/r/2664/
+
+ * res/stasis_http/config.c: Apply defaults to ari.conf's general
+ section
+
+ * tests/test_voicemail_api.c: test_voicemail_api: fix warning found
+ by gcc-4.8 The voicemail_api test had code like strncmp(a, b,
+ sizeof(a)), but a was a char pointer, instead of a literal or
+ char array. This meant that sizeof was the size of the pointer,
+ not the length of the string. Since the string is in a
+ stringfield and should be null terminated, I just changed it to a
+ plain strcmp.
+
+ * tests/test_cel.c: Fixed some CEL test crashes
+
+2013-07-10 22:26 +0000 [r394024] Kevin Harwell <kharwell@digium.com>
+
+ * contrib/scripts/sip_to_res_sip/astconfigparser.py (added),
+ contrib/scripts/sip_to_res_sip/astdicts.py (added),
+ contrib/scripts/sip_to_res_sip/sip_to_res_sip.py (added),
+ contrib/scripts/sip_to_res_sip (added): PSJIP - sip.conf to
+ res_sip.conf script ** This script is in no way finished. Started
+ the initial "cut" at converting a sip.conf file to a res_sip.conf
+ file. Hopefully the bulk of the framework is in place and only a
+ few minor adjustments need to be made when an option mapping is
+ added that "doesn't fit". This script and supporting files should
+ be executable against python version 2.5. An OrderedDict class
+ (backported from a newer version of python) is included. A
+ MultiOrderedDict class is implemented so options, when added,
+ should be able to be added in order and allowed to have multiple
+ values. Currently the scripts supports the majority of endpoint
+ options found in res_sip.conf. Support has also been added for
+ Aor(s) and the ACL/security sections. Inside the
+ sip_to_res_sip.py file one can see a list of options that still
+ need to be mapped. Also items that still need to be done:
+ templates, includes, parsing '=>' delimiter. Note that some code
+ is hopefully in place already to support templates (e.g.
+ lookup/retrieving defaults from them). However, the parsing of
+ and adding of the section needs to be done.
+
+2013-07-10 20:02 +0000 [r394004] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_outbound_registration.c: Handle outbound registration
+ failures that do not occur as a result of a real response.
+ (closes issue ASTERISK-22064) Reported by: Rusty Newton
+
+2013-07-10 17:13 +0000 [r393968-393987] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http_channels.c, rest-api/api-docs/channels.json:
+ Document the 400 error response for originate
+
+ * res/stasis_http/ari_model_validators.c,
+ res/res_stasis_http_channels.c, rest-api/api-docs/channels.json,
+ res/stasis_http/ari_model_validators.h,
+ res/res_stasis_http_asterisk.c, rest-api/api-docs/asterisk.json:
+ Corrected api-docs for channel variables
+
+2013-07-10 01:56 +0000 [r393930] Russell Bryant <russell@russellbryant.com>
+
+ * /, apps/app_meetme.c, configs/sla.conf.sample: astobj2-ify the
+ SLA code The SLA code within app_meetme was written before
+ asotbj2 had been merged into Asterisk. Worse, support for reloads
+ did not exist at first and was added later as a bolt-on feature.
+ I knew at the time that reloading was not safe at all while SLA
+ was in use, so the reload would be queued up to execute when the
+ system was idle. Unfortunately, this approach was still prone to
+ errors beyond the fact that this was the only place in Asterisk
+ where configuration was not reloaded instantly when requested.
+ This patch converts various SLA objects to be reference counted
+ objects using astobj2. This allows reloads to be processed while
+ the system is in use. The code ensures that the objects will not
+ disappear while one of the other threads is using them. However,
+ they will be immediately removed from the global trunk and
+ station containers so no new calls will use them if removed from
+ configuration. Review: https://reviewboard.asterisk.org/r/2581/
+ ........ Merged revisions 393928 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 393929 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-07-09 21:40 +0000 [r393919] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/lock.h: Make SCOPED_LOCK use RAII_VAR. This
+ fixes an issue with requiring SCOPED_LOCK to be the last variable
+ declaration and removes duplicate code in the process. Review:
+ https://reviewboard.asterisk.org/r/2665/
+
+2013-07-09 21:06 +0000 [r393910] Richard Mudgett <rmudgett@digium.com>
+
+ * main/xmldoc.c: Fix printf NULL string (null) substituion for NULL
+ config framework default.
+
+2013-07-09 20:07 +0000 [r393897] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_gulp.c: Use correct function for getting bridged
+ peer when doing direct media checks. (closes issue
+ ASTERISK-21947) reported by Matt Jordan
+
+2013-07-09 19:38 +0000 [r393896] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/stasis_channels.h, include/asterisk/manager.h:
+ Fix some stasis doxygen comments.
+
+2013-07-09 11:05 +0000 [r393857-393870] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_outbound_registration.c: Ensure all pjsip_regc_*
+ access occurs within a pjlib thread. (closes issue
+ ASTERISK-22054) Reported by: Rusty Newton
+
+ * res/res_sip/config_auth.c: Tweak log message slightly.
+
+ * res/res_sip/config_auth.c: Treat the authentication object as
+ invalid if digest configuration is chosen and the digest is not
+ of the correct length. (closes issue ASTERISK-22003) Reported by:
+ Rusty Newton
+
+2013-07-08 20:31 +0000 [r393834-393843] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_recording.c: Oh menuconfig, why do you hate
+ margins?
+
+ * res/stasis_http/ari_websockets.c: Better structure for the
+ WebSocket validation failure message
+
+2013-07-08 19:53 +0000 [r393831-393833] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip/config_transport.c: Ensure that a valid bind host is
+ specified for transports. (closes issue ASTERISK-22017) Reported
+ by: Rusty Newton
+
+ * include/asterisk/channel.h, main/stasis_channels.c,
+ main/bridging.c, main/manager_channels.c, main/cli.c,
+ main/channel.c, build_tools/cflags-devmode.xml, main/pbx.c,
+ include/asterisk/stasis_channels.h, main/manager.c,
+ main/channel_internal_api.c, res/res_agi.c,
+ main/manager_bridging.c: Refactor operations to access the stasis
+ cache instead of objects directly when retrieving information.
+ (closes issue ASTERISK-21883) Review:
+ https://reviewboard.asterisk.org/r/2645/
+
+2013-07-08 16:04 +0000 [r393816] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http.c: res_stasis_http doesn't depend on
+ res_stasis any more
+
+2013-07-08 15:59 +0000 [r393815] Jonathan Rose <jrose@digium.com>
+
+ * res/parking/parking_controller.c, main/bridging.c,
+ res/parking/parking_bridge.c, res/parking/res_parking.h:
+ res_parking: Apply ringing role option on swap with a channel
+ that rings (closes issue ASTERISK-21877) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2656/
+
+2013-07-08 15:11 +0000 [r393807] Joshua Colp <jcolp@digium.com>
+
+ * res/stasis/control.c: Fix building.
+
+2013-07-08 14:46 +0000 [r393804-393806] Jason Parker <jparker@digium.com>
+
+ * res/stasis_http/resource_asterisk.c,
+ res/res_stasis_http_asterisk.c, res/stasis/control.c,
+ res/stasis_http/resource_asterisk.h,
+ rest-api/api-docs/asterisk.json,
+ res/stasis_http/resource_channels.c,
+ res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
+ res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json: ARI: Add support for
+ getting/setting channel and global variables. This allows for
+ reading and writing of functions on channels. (closes issue
+ ASTERISK-21868) Review: https://reviewboard.asterisk.org/r/2641/
+
+ * main/manager_system.c (added), res/res_stun_monitor.c,
+ main/file.c, main/sounds_index.c,
+ include/asterisk/stasis_system.h (added), channels/chan_iax2.c,
+ include/asterisk/manager.h, main/asterisk.c, include/asterisk.h,
+ main/stasis_system.c (added), main/manager.c,
+ channels/chan_sip.c: Move channel driver Registry manager events
+ to core. This also shuffles the stasis system topic and related
+ handling. (closes issue ASTERISK-21488) Review:
+ https://reviewboard.asterisk.org/r/2631/
+
+2013-07-08 14:26 +0000 [r393801] Matthew Jordan <mjordan@digium.com>
+
+ * main/bridging.c, include/asterisk/core_unreal.h,
+ include/asterisk/core_local.h, include/asterisk/bridging.h,
+ main/core_unreal.c, main/core_local.c, CHANGES: Create Local
+ channel messages on the Stasis message bus and produce AMI events
+ This patch does the following: * It adds a virtual table of
+ callbacks to core_unreal. These callbacks can be supplied by
+ concrete implementations of "unreal" channel drivers, which lets
+ the unreal channel driver call specific functionality when it
+ performs some action. Currently, this is done to notify
+ implementations when an optimization operation has begun, and
+ when an optimization operation has succeeded. * It adds
+ Stasis-Core messages for Local channel bridging and Local channel
+ optimization. Local channel optimization is now two events: a
+ Begin and an End. Some consumers of Stasis-Core may want to know
+ when an operation is beginning so that they can 'prepare' their
+ information; others will be more concerned about when the
+ operation has completed, so that they can 'fix up' information.
+ Stasis-Core allows for both, as does AMI. Review:
+ https://reviewboard.asterisk.org/r/2552
+
+2013-07-08 13:57 +0000 [r393793] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_sip_caller_id.c: Fix some broken logic in sending
+ outbound caller ID. * trust_id_outbound was required even when
+ the caller ID was not marked private. This is against intentions
+ and documentation. * We now check both name and number privacy
+ instead of checking name privacy twice.
+
+2013-07-07 21:29 +0000 [r393777-393785] Matthew Jordan <mjordan@digium.com>
+
+ * main/channel.c: In a channel destructor dispose of items that
+ raise Stasis message properly This patch reorders certain actions
+ that may raise Stasis messages in the channel destructor such
+ that they occur before the Stasis cache is cleared. Once the
+ Stasis cache is cleared, its rather a bad idea to be trying to
+ publish information about a channel. (closes issue
+ ASTERISK-22001) Reported by: Jonathan Rose
+
+ * CHANGES, main/cel.c, main/manager_channels.c, main/cdr.c,
+ main/channel.c, main/pbx.c, include/asterisk/stasis_channels.h,
+ main/channel_internal_api.c, include/asterisk/cdr.h,
+ include/asterisk/channel.h, main/stasis_channels.c: Handle hangup
+ logic in the Stasis message bus and consumers of Stasis messages
+ This patch does the following: * It adds a new soft hangup flag
+ AST_SOFTHANGUP_HANGUP_EXEC that is set when a channel is
+ executing dialplan hangup logic, i.e., the 'h' extension or a
+ hangup handler. Stasis messages now also convey the soft hangup
+ flag so consumers of the messages can know when a channel is
+ executing said hangup logic. * It adds a new channel flag,
+ AST_FLAG_DEAD, which is set when a channel is well and truly
+ dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs, and
+ other consumers of Stasis have been updated to look for this flag
+ to know when the channel should by lying six feet under. * The
+ CDR engine has been updated to better handle a channel entering
+ and leaving a bridge. Previously, a new CDR was automatically
+ created when a channel left a bridge and put into the 'Pending'
+ state; however, this way of handling CDRs made it difficult for
+ the 'endbeforehexten' logic to work correctly - there was always
+ a new CDR waiting in the hangup logic and, even if 'ended',
+ wouldn't be the CDR people wanted to inspect in the hangup
+ routine. This patch completely removes the Pending state and
+ instead defers creation of the new CDR until it gets a new
+ message that requires a new CDR.
+
+2013-07-05 22:08 +0000 [r393749-393768] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http.c: ARI: return a 503 if Asterisk isn't fully
+ booted
+
+ * res/stasis_http/ari_websockets.c: Print error details when set
+ nonblock fails
+
+ * res/stasis_http/ari_model_validators.h,
+ res/stasis_http/resource_events.c, res/res_stasis_http_events.c,
+ rest-api/api-docs/events.json,
+ res/stasis_http/ari_model_validators.c: Document MissingParams
+ error message for /ari/events
+
+2013-07-05 17:33 +0000 [r393740] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_gulp.c, channels/chan_jingle.c, main/json.c,
+ main/manager.c, channels/chan_skinny.c, channels/chan_motif.c,
+ channels/chan_h323.c, include/asterisk/rtp_engine.h,
+ main/asterisk.c, channels/chan_mgcp.c, channels/chan_unistim.c,
+ res/res_rtp_asterisk.c, channels/chan_multicast_rtp.c,
+ main/rtp_engine.c, channels/chan_sip.c, include/asterisk/cdr.h,
+ include/asterisk/channel.h, channels/chan_gtalk.c,
+ include/asterisk/json.h: Refactor RTCP events over to Stasis;
+ associate with channels This patch does the following: * It
+ merges Jaco Kroon's patch from ASTERISK-20754, which provides
+ channel information in the RTCP events. Because Stasis provides a
+ cache, Jaco's patch was modified to pass the channel uniqueid to
+ the RTP layer as opposed to a pointer to the channel. This has
+ the following benefits: (1) It keeps the RTP engine 'clean' of
+ references back to channels (2) It prevents circular dependencies
+ and other potential ref counting issues * The RTP engine now
+ allows any RTP implementation to raise RTCP messages.
+ Potentially, other implementations (such as res_rtp_multicast)
+ could also raise RTCP information. The engine provides structs to
+ represent RTCP headers and RTCP SR/RR reports. * Some general
+ refactoring in res_rtp_asterisk was done to try and tame the RTCP
+ code. It isn't perfect - that's *way* beyond the scope of this
+ work - but it does feel marginally better. * A few random bugs
+ were fixed in the RTCP statistics. (Example: performing an
+ assignment of a = a is probably not correct) * We now raise RTCP
+ events for each SR/RR sent/received. Previously we wouldn't raise
+ an event when we sent a RR report. Note that this work will be of
+ use to others who want to monitor call quality or build modules
+ that report call quality statistics. Since the events are now
+ moving across the Stasis message bus, this is far easier to
+ accomplish. It is also a first step (though by no means the last
+ step) towards getting Olle's pinefrog work incorporated. Again:
+ note that the patch by Jaco Kroon was modified slightly for this
+ work; however, he did all of the hard work in finding the right
+ places to set the channel in the RTP engine across the channel
+ drivers. Much thanks goes to Jaco for his hard work here. Review:
+ https://reviewboard.asterisk.org/r/2603/ (closes issue
+ ASTERISK-20574) Reported by: Jaco Kroon patches:
+ asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
+ (closes issue ASTERISK-21471) Reported by: Matt Jordan
+
+2013-07-05 14:54 +0000 [r393729] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: OneTouchRecord: Add function defined earlier:
+ ast_bridge_features_do()
+
+2013-07-05 03:08 +0000 [r393716] Matthew Jordan <mjordan@digium.com>
+
+ * main/stasis_channels.c, include/asterisk/stasis_channels.h:
+ Remove parkinglot from the channel snapshot Legacy channel
+ drivers often include the ability to set a default parking lot on
+ an endpoint basis; when channels are created for that endpoint,
+ they inherit the parkinglot option. Parking used to use this
+ option more frequently; while it is still supported, other
+ options (such as using channel variables or creation of a custom
+ parkinglot) are supported. More importantly, conveying the
+ parkinglot information through a channel snapshot isn't terribly
+ useful - it is rarely (if ever) changed on a channel and some
+ consumers of channel snapshots, such as ARI, will never use the
+ information. (closes issue ASTERISK-21968) Reported by: Matt
+ Jordan
+
+2013-07-04 18:46 +0000 [r393704] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/parking.h, main/bridging.c,
+ res/parking/parking_manager.c, res/parking/parking_ui.c,
+ main/parking.c, res/parking/parking_controller.c, UPGRADE.txt,
+ res/parking/parking_applications.c, include/asterisk/channel.h,
+ main/cel.c, CHANGES, res/parking/parking_bridge_features.c,
+ res/parking/parking_bridge.c, main/channel.c,
+ res/parking/res_parking.h, bridges/bridge_builtin_features.c,
+ main/features.c: res_parking: Replace Parker snapshots with
+ ParkerDialString This process also involved a large amount of
+ rework regarding how to redial the Parker when a channel leaves a
+ parking lot due to timeout. An attended transfer channel variable
+ has been added to attended transfers to extensions that will
+ eventually park (but haven't at the time of transfer) as well.
+ This resolves one of the two BUGBUG comments remaining in
+ res_parking. (issues ASTERISK-21877) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2638/
+
+2013-07-04 13:37 +0000 [r393675-393687] David M. Lee <dlee@digium.com>
+
+ * res/res_ari_model.c: Fix int width problem for 32-bit... again
+
+ * tests/test_ari_model.c: Fix int width problem for 32-bit
+
+ * main/Makefile, main/utils.c, main/crypt.c (added): Fix utils
+ directory breakage.
+
+2013-07-03 23:59 +0000 [r393600-393633] Richard Mudgett <rmudgett@digium.com>
+
+ * main/config_options.c: Add BUGBUG note for ASTERISK-22009
+
+ * main/bridging.c, apps/app_agent_pool.c (removed),
+ configs/agents.conf.sample, include/asterisk/config_options.h,
+ include/asterisk/stasis_channels.h, channels/chan_agent.c
+ (added), configs/queues.conf.sample, include/asterisk/bridging.h,
+ UPGRADE.txt, main/config_options.c, main/stasis_channels.c,
+ CHANGES: Revert accidental overcommit.
+
+ * include/asterisk/stasis_channels.h, channels/chan_agent.c
+ (removed), configs/queues.conf.sample,
+ include/asterisk/bridging.h, UPGRADE.txt, main/config_options.c,
+ main/stasis_channels.c, CHANGES, main/bridging.c,
+ apps/app_agent_pool.c (added), configs/agents.conf.sample,
+ include/asterisk/config_options.h: Add BUGBUG note for
+ ASTERISK-22009
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault reloading
+ chan_dahdi when round robin is used. * Clear round_robin[] in
+ dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
+ Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 393627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 393628 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h: OneTouchRecord: Make so
+ Monitor/MixMonitor can be toggled/started/stopped. The
+ OneTouchRecord feature has historically been a toggle. This patch
+ adds the ability to make the OneTouchRecord hook optionally
+ start/stop recording only. If OneTouchRecord is already doing
+ what is requested then only the invoker hears the courtesy tone
+ and/or start/stop recording message. The new feature is written
+ so we could easily add explicit start/stop recording DTMF hooks
+ for Monitor and MixMonitor. The majority of the changes in
+ bridge_builtin_features.c is a refactoring of the OneTouchRecord
+ code (Monitor and MixMonitor versions) so it is easy to direct
+ the toggle/start/stop functionality. Review:
+ https://reviewboard.asterisk.org/r/2655/
+
+ * main/bridging.c: Move when bridge channel enter is published so
+ it does not interrupt the thought of some lines of code.
+
+ * main/stasis_config.c: Fix some indentation in stasis_config.c.
+
+2013-07-03 22:04 +0000 [r393589-393599] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Fix some bugs in CDRs; add some CLI commands to help
+ debugging This patch fixes a few minor bugs and one major one:
+ the CDR by bridge container was less than helpful. The mechanism
+ previously used to try and find all of the CDRs in a particular
+ bridge ended up missing CDRs, resulting in incorrect records.
+ When looking up CDRs in a bridge, we now just bite the bullet and
+ do a selection across all existing CDRs.
+
+ * main/stasis_config.c: Let Stasis load itself with default values
+ While a Stasis configuration file is nice, it shouldn't be
+ mandatory. We can carry on with default values.
+
+2013-07-03 20:41 +0000 [r393586] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridging.c: Publish a bridge enter before pulling on a
+ push-and-swap operation. Prior to this patch, the order of
+ procedures on a bridge push was * Add new bridge channel to
+ bridge's array. * Pull the swap channel out of the bridge *
+ Publish a bridge enter event. The problem is that when the swap
+ channel was pulled from the bridge, a bridge leave event would be
+ published. The bridge snapshot published during the bridge leave
+ showed the new channel that had been added to the bridge, but
+ there had been no bridge enter event for that channel. The fix
+ provided here was to change the order a bit * Add new bridge
+ channel to bridge's array. * Publish bridge enter event. * Pull
+ the swap channel out of the bridge. This makes it so that the
+ bridge snapshots during the stasis events are accurate.
+
+2013-07-03 19:46 +0000 [r393528-393576] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http_endpoints.c, res/res_stasis_http_events.c,
+ rest-api-templates/ari_model_validators.c.mustache,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ rest-api-templates/ari_model_validators.h.mustache,
+ res/stasis_http/ari_model_validators.c,
+ res/res_stasis_http_channels.c, res/res_stasis_http_sounds.c,
+ res/res_stasis_http_bridges.c, res/res_stasis_http_recordings.c,
+ res/stasis_http/ari_model_validators.h: Fix load errors related
+ to the new ari_model_validators. The Asterisk strategy of loading
+ modules with RTLD_LAZY to extract metadata from the module works
+ well enough, until you try to take the address of a function. If
+ a module takes the address of a function, that function needs to
+ be resolved at load time. That kinda defeats RTLD_LAZY. This
+ patch adds some ari_validator_{id}_fn() wrapper functions for
+ safely getting the function pointer from a different module.
+
+ * res/res_ari_model.c: Violating the margins to make menuconfig
+ happy
+
+ * res/res_stasis_recording.exports.in (added), Makefile,
+ include/asterisk/file.h, include/asterisk/paths.h,
+ main/channel.c, include/asterisk/app.h,
+ res/stasis_http/resource_channels.c, tests/test_utils.c,
+ apps/app_minivm.c, main/file.c,
+ res/stasis_http/resource_recordings.c, main/app.c,
+ res/res_stasis_recording.c (added),
+ rest-api-templates/swagger_model.py,
+ rest-api/api-docs/channels.json,
+ res/stasis_http/resource_channels.h,
+ res/res_stasis_http_bridges.c, rest-api/api-docs/recordings.json,
+ res/stasis_http/resource_recordings.h, main/asterisk.c,
+ rest-api-templates/asterisk_processor.py, apps/app_voicemail.c,
+ include/asterisk/utils.h, res/res_stasis_playback.c,
+ include/asterisk/stasis_app_recording.h (added),
+ res/res_stasis_http_channels.c, main/utils.c,
+ include/asterisk/channel.h, res/res_stasis_http_recordings.c: ARI
+ - channel recording support This patch is the first step in
+ adding recording support to the Asterisk REST Interface.
+ Recordings are stored in /var/spool/recording. Since recordings
+ may be destructive (overwriting existing files), the API rejects
+ attempts to escape the recording directory (avoiding issues if
+ someone attempts to record to ../../lib/sounds/greeting, for
+ example). (closes issue ASTERISK-21594) (closes issue
+ ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/
+
+ * include/asterisk/stasis.h, configs/stasis_core.conf.sample
+ (removed), main/asterisk.c, main/stasis.c, main/stasis_config.c
+ (added), configs/stasis.conf.sample (added): Configuration for
+ Stasis threadpool The appropriate settings for the Stasis
+ threadpool is very system specific, depending upon both workload
+ and system configuration. This patch adds a stasis.conf file
+ which can be used to configure the key attributes of the
+ threadpool for the Stasis message bus. (closes issue
+ ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/
+
+ * main/utils.c, res/stasis_http/cli.c (added), res/Makefile,
+ configs/ari.conf.sample (added), makeopts.in,
+ res/res_stasis_http.c, res/stasis_http/internal.h (added),
+ configs/stasis_http.conf.sample (removed), main/Makefile,
+ res/stasis_http/config.c (added), main/http.c: No message for rev
+ 393530 found
+
+ * res/res_stasis_json_endpoints.exports.in (removed),
+ res/res_stasis_json_channels.c (removed),
+ rest-api/api-docs/events.json,
+ rest-api-templates/make_ari_stubs.py (added),
+ res/res_stasis_json_recordings.c (removed),
+ res/stasis_http/ari_model_validators.c (added),
+ rest-api-templates/api.wiki.mustache (added),
+ res/res_stasis_json_sounds.exports.in (removed),
+ res/res_stasis_json_endpoints.c (removed), res/Makefile,
+ res/res_stasis_json_events.c (removed), include/asterisk/json.h,
+ res/stasis_http/ari_model_validators.h (added),
+ res/res_stasis_json_sounds.c (removed),
+ res/res_stasis_json_asterisk.exports.in (removed),
+ rest-api/api-docs/channels.json, main/stasis_channels.c,
+ rest-api-templates/res_stasis_json_resource.c.mustache (removed),
+ rest-api/api-docs/recordings.json,
+ rest-api-templates/asterisk_processor.py,
+ res/res_stasis_json_events.exports.in (removed),
+ include/asterisk/stasis_http.h,
+ rest-api-templates/event_function_decl.mustache (removed),
+ rest-api/api-docs/sounds.json, rest-api/api-docs/bridges.json,
+ res/res_stasis_http_endpoints.c, Makefile, main/json.c,
+ rest-api/api-docs/asterisk.json, rest-api/api-docs/playback.json,
+ rest-api-templates/swagger_model.py, res/res_stasis_http.c,
+ res/res_stasis.c, doc/rest-api (added),
+ rest-api-templates/make_stasis_http_stubs.py (removed),
+ rest-api-templates/ari_model_validators.c.mustache (added),
+ res/res_stasis_http_asterisk.c,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ main/stasis_endpoints.c, tests/test_res_stasis.c,
+ res/res_stasis_json_bridges.exports.in (removed),
+ res/res_ari_model.c (added),
+ res/res_stasis_json_playback.exports.in (removed),
+ main/stasis_bridging.c, res/res_stasis_json_bridges.c (removed),
+ res/stasis_http/ari_websockets.c,
+ rest-api-templates/stasis_json_resource.h.mustache (removed),
+ res/stasis_http/resource_recordings.c,
+ res/res_stasis_json_asterisk.c (removed), res/stasis_json
+ (removed), res/stasis_http/resource_recordings.h,
+ rest-api/api-docs/endpoints.json, res/res_stasis_http_playback.c,
+ rest-api-templates/ari_model_validators.h.mustache (added),
+ res/res_stasis_http_channels.c, tests/test_stasis_channels.c,
+ res/res_ari_model.exports.in (added),
+ res/res_stasis_http_recordings.c,
+ rest-api-templates/res_stasis_json_resource.exports.mustache
+ (removed), res/res_stasis_http_events.c, tests/test_ari_model.c
+ (added), rest-api-templates/models.wiki.mustache (added),
+ rest-api-templates/transform.py, res/res_stasis_http_sounds.c,
+ res/res_stasis_json_channels.exports.in (removed),
+ res/res_stasis_http_bridges.c,
+ res/res_stasis_json_recordings.exports.in (removed),
+ res/res_stasis_json_playback.c (removed): No message for rev
+ 393529 found
+
+ * res/res_stasis_http_channels.c, include/asterisk/stasis_http.h,
+ configure.ac, res/res_stasis_http_recordings.c,
+ res/res_stasis_http_events.c, configure,
+ res/res_stasis_http_sounds.c, res/stasis_http/resource_events.c,
+ rest-api-templates/stasis_http_resource.h.mustache,
+ res/res_stasis_http_asterisk.c,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ res/res_http_websocket.exports.in,
+ res/stasis_http/resource_events.h, res/res_stasis_websocket.c
+ (removed), include/asterisk/autoconfig.h.in,
+ rest-api-templates/rest_handler.mustache, res/Makefile,
+ res/res_http_websocket.c,
+ rest-api-templates/param_parsing.mustache (added),
+ res/res_stasis_http_endpoints.c, include/asterisk/http.h,
+ res/res_stasis_http.exports.in, tests/test_utils.c,
+ res/stasis_http/ari_websockets.c (added),
+ rest-api-templates/stasis_http_resource.c.mustache,
+ rest-api-templates/swagger_model.py,
+ res/res_stasis_http_bridges.c, tests/test_stasis_http.c,
+ res/res_stasis_http.c, rest-api-templates/asterisk_processor.py,
+ include/asterisk/utils.h, res/res_stasis_http_playback.c,
+ rest-api/api-docs/events.json: No message for rev 393528 found
+
+2013-07-02 22:01 +0000 [r393508] Jason Parker <jparker@digium.com>
+
+ * main/manager.c, CHANGES: Add a SystemName field to all AMI
+ events. This only gets sent out if configured in asterisk.conf
+ (closes issue ASTERISK-21494)
+
+2013-07-02 21:19 +0000 [r393485-393500] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_mixmonitor.c: MixMonitor: Minor code cleanup.
+
+ * apps/app_mixmonitor.c: MixMonitor: Make
+ start_mixmonitor_callback() options parameter NULL tolerant. *
+ Removed some unnecessary code in start_mixmonitor_callback().
+
+ * apps/app_mixmonitor.c: MixMonitor: Don't use ast_strdupa() in a
+ loop.
+
+ * apps/app_mixmonitor.c: MixMonitor: Update XML documentation and
+ CLI "mixmonitor {start|stop|list}" help.
+
+ * apps/app_mixmonitor.c: MixMonitor: Fix refleak in
+ manager_stop_mixmonitor() if could not stop monitoring.
+
+ * apps/app_mixmonitor.c: MixMonitor: Remove some unnecessary
+ channel locking.
+
+ * apps/app_mixmonitor.c: Fix MixMonitor b option. The option had
+ not been converted to use the replacement for
+ ast_bridged_channel(). One touch mixmonitor now records files
+ again.
+
+ * channels/chan_gtalk.c: Fix chan_gtalk.c compile error.
+
+2013-07-02 20:34 +0000 [r393484] David M. Lee <dlee@digium.com>
+
+ * res/res_sip_notify.c: Add pjproject dependency to res_sip_notify
+
+2013-07-02 18:28 +0000 [r393463] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stasis_bridging.h: Remove unused blind transfer
+ publication structure. I ended up using a bridge blob, so this
+ structure was unused. Keeping it in the header would just cause
+ confusion.
+
+2013-07-02 17:20 +0000 [r393442-393449] Kevin Harwell <kharwell@digium.com>
+
+ * main/aoc.c, main/manager.c: Stasis - Refactor AOC Events
+ Refactored the AMI events in AOC onto Stasis-Core. The
+ ast_aoc_manager_event function now publishes a channel snapshot,
+ along with a JSON blob describing the advice of charge. A
+ "to_ami" handler has also been added that converts the channel
+ snapshot and AOC event data back into the appropriate data
+ structure for use with AMI. (closes issue ASTERISK-21472)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2643/
+
+ * res/res_sip/config_security.c (added), res/res_sip_acl.c,
+ res/res_sip.c, res/res_sip/sip_configuration.c,
+ include/asterisk/res_sip.h, res/res_sip/sip_distributor.c,
+ res/res_sip/config_auth.c, res/res_sip.exports.in,
+ res/res_sip_outbound_authenticator_digest.c,
+ res/res_sip_authenticator_digest.c: New SIP Channel driver:
+ Always Auth Reject If no matching endpoint is found for the
+ incoming request Asterisk will respond with a 401 Unauthorized
+ (rejecting the request), but will first challenge if no
+ authorization creditials are given. Changes also included moving
+ ACL options into a new global 'security' configuration section in
+ res_sip.conf. (closes issue ASTERISK-21433) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2554/
+
+2013-07-02 16:11 +0000 [r393410-393429] Kinsey Moore <kmoore@digium.com>
+
+ * main/stasis_bridging.c: Fix transfer AMI event parameter naming
+
+ * main/cel.c, include/asterisk/cel.h, tests/test_cel.c (added): Add
+ CEL unit tests and do some cleanup This adds several unit tests
+ for CEL functionality and provides the requisite framework for
+ creating additional unit tests. This also cleans up some
+ reference leaks that were occurring in Stasis-Core message
+ callback code. Review: https://reviewboard.asterisk.org/r/2646/
+
+2013-07-02 10:16 +0000 [r393396] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * /, channels/chan_unistim.c: Fix issue with inability to cancell
+ call transfer made by on-sceen menus. Reported by: Igor Olhovskiy
+ ........ Merged revisions 393395 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-07-02 08:23 +0000 [r393383] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/ast_tls_cert: ast_tls_cert: don't recreate
+ generated files Don't regenrate cat.cfg, ca.crt and ca.key if
+ they were already created on a previous run. (closes issue
+ ASTERISK-21932)
+
+2013-07-01 21:28 +0000 [r393364] Kevin Harwell <kharwell@digium.com>
+
+ * include/asterisk/res_sip.h,
+ res/res_sip/include/res_sip_private.h, res/res_sip/sip_options.c,
+ res/res_sip.exports.in, res/res_sip_notify.c (added),
+ res/res_sip/sip_configuration.c: New SIP Channel Driver - Add
+ CLI/AMI initiated NOTIFY requests Added the ability to send
+ unsolicited NOTIFY requests to a particular endpoint with a
+ configured payload. Added both CLI and AMI support. For a given
+ endpoint, this module will iterate over all its contacts sending
+ the appropriate NOTIFY request to each. (closes issue
+ ASTERISK-21436) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2623/
+
+2013-07-01 21:24 +0000 [r393361] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c, main/manager.c, include/asterisk/pbx.h: Prevent crash
+ during synchronous AMI origination by ref bumping returned
+ channel The originate APIs allow callers to provide a pointer to
+ a channel that will point to the originated channel if the
+ function call succeeds. This is used by AMI to provide channel
+ information when the originate is performed synchronously.
+ Unfortunately, if the originate fails in certain ways, the
+ outbound channel is already disposed of during the dialing
+ itself. This results in the channel being improperly dereferenced
+ by the internal originate function in pbx.c. This patch ref bumps
+ the channel to prevent this from occurring. Callers must now
+ unlock and unref the channel (which is more in line with general
+ channel management guidelines anyway). This only affects manager,
+ as it is the only consumer of this API function that actually
+ passes in a channel pointer. Review:
+ https://reviewboard.asterisk.org/r/2617/
+
+2013-07-01 18:56 +0000 [r393326-393332] Jason Parker <jparker@digium.com>
+
+ * res/stasis/control.c, res/stasis_http/resource_channels.c,
+ include/asterisk/stasis_app.h: ARI: Implement channel
+ hold/unhold. This puts the channel on hold (rather than queueing
+ a frame from the channel). (closes issue ASTERISK-21619) Review:
+ https://reviewboard.asterisk.org/r/2647/
+
+ * res/stasis/control.c, res/stasis_http/resource_channels.c,
+ res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
+ res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json: ARI: Implement channel dial.
+ This creates a new outbound channel, and bridges it to a channel
+ already in the Stasis application. (closes issue ASTERISK-21620)
+ Review: https://reviewboard.asterisk.org/r/2634/
+
+2013-07-01 16:01 +0000 [r393309] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/features_config.h, include/asterisk/mixmonitor.h
+ (added), include/asterisk/channel.h, CHANGES,
+ main/features_config.c, apps/app_mixmonitor.c,
+ configs/features.conf.sample, main/mixmonitor.c (added),
+ bridges/bridge_builtin_features.c: bridge_features: Support One
+ touch Monitor/MixMonitor In addition to porting those features,
+ they now enjoy greater feature parity with one another.
+ Specifically, AutoMixMon now has a start and stop message that
+ can be specified with TOUCH_MIXMONITOR_MESSAGE_START and
+ TOUCH_MIXMONITOR_MESSAGE_STOP. (closes issue ASTERISK-21553)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2620/
+
+2013-07-01 13:16 +0000 [r393284] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_agi.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h, channels/chan_sip.c,
+ apps/app_meetme.c, include/asterisk/stasis.h, main/core_local.c,
+ include/asterisk/json.h, channels/chan_gtalk.c,
+ channels/sig_pri.c, channels/chan_iax2.c, apps/app_queue.c,
+ CHANGES, main/json.c, channels/chan_dahdi.c,
+ channels/sig_analog.c: Refactor extraneous channel events This
+ change removes JitterBufStats, ChannelReload, and ChannelUpdate
+ and refactors the following events to travel over Stasis-Core: *
+ LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm
+ * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone *
+ SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/
+ (closes issue ASTERISK-21476)
+
+2013-06-29 13:47 +0000 [r393262-393264] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_pubsub.c: Nothing to see here, move along.
+
+ * res/res_sip_pubsub.c, include/asterisk/res_sip_pubsub.h,
+ res/res_sip_pubsub.exports.in: Implement the defined PUBLISH ESC
+ API within res_sip_pubsub. (closes issue ASTERISK-21452) Review:
+ https://reviewboard.asterisk.org/r/2630/
+
+2013-06-29 00:31 +0000 [r393219-393241] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridging.h, main/bridging.c: Tweak after bridge
+ callback reason to string strings.
+
+ * main/bridging.c: Fix after bridge callback datastore data memory
+ leak.
+
+ * main/datastore.c: This is no longer needed.
+
+ * main/bridging.c: Promote local channel optimizing debug messages
+ to verbose 3 messages.
+
+2013-06-28 19:22 +0000 [r393190-393197] Jonathan Rose <jrose@digium.com>
+
+ * res/parking/parking_ui.c, res/parking/res_parking.h,
+ res/res_parking.c, res/parking/parking_applications.c, CHANGES:
+ res_parking: Dynamic Parking Lots (closes issue ASTERISK-21644)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2615/
+
+ * include/asterisk/features.h, main/features.c: features: call
+ pickup stasis refactoring (issue ASTERISK-21544) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/2588/
+
+2013-06-28 19:05 +0000 [r393184] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridging_features.h: Fix overlapping enum
+ ast_bridge_feature_flags. Things may no longer behave in an
+ unexpected fashion. Local channel optimization to holding bridges
+ will work again.
+
+2013-06-28 18:42 +0000 [r393182] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, bridges/bridge_builtin_features.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ main/stasis_bridging.c, res/res_sip_refer.c,
+ include/asterisk/bridging.h, main/manager_bridging.c,
+ channels/chan_iax2.c, include/asterisk/stasis_bridging.h,
+ main/bridging.c: Add stasis publications for blind and attended
+ transfers. This creates stasis messages that are sent during a
+ blind or attended transfer. The stasis messages also are
+ converted to AMI events. Review:
+ https://reviewboard.asterisk.org/r/2619 (closes issue
+ ASTERISK-21337) Reported by Matt Jordan
+
+2013-06-28 17:31 +0000 [r393164] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_cdr.c, main/cdr.c: Handle an originated channel being
+ sent into a non-empty bridge Originated channels are a bit odd -
+ they are technically a dialed channel (thus the party B or peer)
+ but, since there is no caller, they are treated as the party A.
+ When entering into a bridge that already contains participants,
+ the CDR engine - if the CDR record is in the Dial state -
+ attempts to match the person entering the bridge with an existing
+ participant. The idea is that if you dialed someone and the
+ person you dialed is already in the bridge, you don't need a new
+ CDR record, the existing CDR record describes the relationship.
+ Unfortunately, for an originated channel, there is no Party B. If
+ no one was in the bridge this didn't cause any issues; however,
+ if participants were in the bridge the CDR engine would attempt
+ to match a non-existant Party B on the channel's CDR record and
+ explode. This patch fixes that, and a unit test has been added to
+ cover this case.
+
+2013-06-28 16:23 +0000 [r393144] Jason Parker <jparker@digium.com>
+
+ * res/res_stasis_http_channels.c,
+ res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json,
+ res/stasis_http/resource_channels.c: Change ARI originate to also
+ allow dialing an exten/context/priority. The old way didn't make
+ much sense, so some of the fields were repurposed. (closes issue
+ ASTERISK-21658) Review: https://reviewboard.asterisk.org/r/2626/
+
+2013-06-28 15:50 +0000 [r393130] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c, include/asterisk/cdr.h, include/asterisk/parking.h,
+ main/asterisk.c, main/bridging.c: Better handle parking in CDRs
+ Parking typically occurs when a channel is transferred to a
+ parking extension. When this occurs, the channel never actually
+ hits the dialplan if the extension it was transferred to was a
+ "parking extension", that is, the extension in the first priority
+ calls the Park application. Instead, the channel is immediately
+ sent into the holding bridge acting as the parking bridge. This
+ is problematic. Because we never go out to the dialplan, the CDRs
+ won't transition properly and the application field will not be
+ set to "Park". CDRs typically swallow holding bridges, so the CDR
+ itself won't even be generated. This patch handles this by
+ pulling out the holding bridge handling into its own CDR state.
+ CDRs now have an explicit parking state that accounts for this
+ specific subclass of the holding bridge. In addition, we handle
+ the parking stasis message to set application specific data on
+ the CDR such that the last known application for the CDR properly
+ reflects "Park". This is a bit sad since we're working around the
+ odd internal implementation of parking that exists in Asterisk
+ (and that we had to maintain in order to continue to meet some
+ odd use cases of parking), but at least the code to handle that
+ is where it belongs: in CDRs as opposed to sprinkled liberally
+ throughout the codebase. This patch also properly clears the
+ OUTBOUND channel flag from a channel when it leaves a bridge, and
+ tweaks up dialing handling to properly compare the correct CDR
+ with the channel calling/being dialed.
+
+2013-06-28 15:36 +0000 [r393128] Jason Parker <jparker@digium.com>
+
+ * res/stasis_http/resource_channels.c: Change some 500 errors to
+ 400.
+
+2013-06-28 02:14 +0000 [r393083-393100] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http.c: Removed stray apostrophe. Apparently the
+ pluralization of an acronym does not use an apostophe, according
+ to most modern style guides. I feel like I've been living a lie
+ this whole time.
+
+ * res/res_stasis_http.c: Removed the automatic 302 redirects for
+ ARI URL's that end with a slash. There were some problems
+ redirecting RESTful API requests; notably the client would change
+ the request method to GET on the redirected requests. After some
+ looking into, I decided that a 404 would be simpler and have more
+ consistent behavior.
+
+2013-06-27 21:01 +0000 [r393034-393066] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Change the name of some local variables in
+ bridging.c to reflect what they really mean.
+
+ * main/config_options.c, include/asterisk/config_options.h: Add
+ config framework non-empty string validation requirement option.
+ Add config framework OPT_CHAR_ARRAY_T and OPT_STRINGFIELD_T
+ non-empty requirement option. There are cases were you don't want
+ a config option string to be empty. To require the option string
+ to be non-empty, just set the aco_option_register() flags
+ parameter to non-zero. * Updated some config framework enum
+ aco_option_type comments.
+
+2013-06-26 20:59 +0000 [r393005] Jonathan Rose <jrose@digium.com>
+
+ * main/bridging.c, funcs/func_channel.c,
+ include/asterisk/bridging.h: func_channel: Read/Write
+ after_bridge_goto option Allows reading and setting of a
+ channel's after_bridge_goto datastore (closes issue
+ ASTERISK-21875) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2628/
+
+2013-06-26 19:29 +0000 [r392987] Jason Parker <jparker@digium.com>
+
+ * res/stasis/control.c, res/stasis_http/resource_channels.c,
+ res/res_stasis_http_channels.c, include/asterisk/stasis_app.h,
+ res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json: ARI: Add support for continuing
+ to a different location in dialplan. This allows going elsewhere
+ in the dialplan, so that the location can be specified after
+ exiting the Stasis application. (closes issue ASTERISK-21870)
+ Review: https://reviewboard.asterisk.org/r/2644/
+
+2013-06-26 19:15 +0000 [r392933-392972] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_parking.c: Remove some redundant parking config error
+ messages.
+
+ * main/bridging.c: Fix several problems with
+ ast_bridge_add_channel(). * Fix locking problems.
+ ast_bridge_move() locks two bridges. To do that, deadlock
+ avoidance must be done. Called bridge_move_locked() instead. *
+ Fix inconsistency in the bridge dissolve check callers. The
+ original caller has already removed the channel from the bridge.
+ The new caller has not removed the channel from the bridge.
+ Reverted bridge_dissolve_check() and added
+ bridge_dissolve_check_stolen() to be used by the new caller on
+ the original bridge after the channel is moved to the new bridge.
+ * Fix memory leak of features if the added channel was already in
+ a bridge. * Fix incorrect call to ast_bridge_impart(). * Renamed
+ bridge_chan to yanked_chan.
+
+ * apps/confbridge/conf_chan_announce.c, channels/chan_sip.c,
+ include/asterisk/bridging.h: Fix incorrect calls to
+ ast_bridge_impart(). There was a misunderstanding about
+ ast_bridge_impart()'s handling of the imparted channel's
+ reference. The channel reference is passed by the caller unless
+ ast_bridge_impart() returns an error. * Fixed a memory leak in
+ conf_announce_channel_push() if the impart failed.
+
+ * main/features.c: AMI Bridge action: Get channel xfer config after
+ we have found the second channel.
+
+2013-06-25 22:28 +0000 [r392915] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c, res/parking/parking_controller.c,
+ res/parking/parking_applications.c, CHANGES, main/bridging.c,
+ res/parking/parking_bridge_features.c,
+ res/parking/parking_manager.c, include/asterisk/features.h,
+ res/parking/parking_bridge.c, res/parking/res_parking.h:
+ res_parking: Add Parking manager action to the new parking system
+ (closes issue ASTERISK-21641) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2573/
+
+2013-06-25 20:25 +0000 [r392898] Jason Parker <jparker@digium.com>
+
+ * Makefile: Fix typo with XML docs.
+
+2013-06-25 19:22 +0000 [r392864-392879] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/sorcery.h: Add a note about being ready to
+ accept observer invocations before adding an observer.
+
+ * res/res_sip/sip_options.c: Move where the sorcery observer is
+ added for qualify to guarantee the sched_qualifies container
+ exists.
+
+2013-06-25 13:03 +0000 [r392829] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c, main/cel.c, apps/app_dial.c,
+ include/asterisk/stasis_channels.h, include/asterisk/cel.h,
+ apps/app_celgenuserevent.c, main/stasis_channels.c: CEL
+ refactoring cleanup This change removes AST_CEL_BRIDGE_UPDATE
+ since it should no longer be used because masquerade situations
+ are now accounted for in other ways. This also refactors usage of
+ AST_CEL_FORWARD to be produced by a Dial message which has been
+ extended with a "forward" field. (closes issue ASTERISK-21566)
+ Review: https://reviewboard.asterisk.org/r/2635/
+
+2013-06-25 01:12 +0000 [r392797-392812] Matthew Jordan <mjordan@digium.com>
+
+ * main/http.c, main/config_options.c, main/named_acl.c,
+ res/res_calendar.c, /, channels/chan_motif.c: Fix memory/ref
+ counting leaks in a variety of locations This patch fixes the
+ following memory leaks: * http.c: The structure containing the
+ addresses to bind to was not being deallocated when no longer
+ used * named_acl.c: The global configuration information was not
+ disposed of * config_options.c: An invalid read was occurring for
+ certain option types. * res_calendar.c: The loaded calendars on
+ module unload were not being properly disposed of. *
+ chan_motif.c: The format capabilities needed to be disposed of on
+ module unload. In addition, this now specifies the default
+ options for the maxpayloads and maxicecandidates in such a way
+ that it doesn't cause the invalid read in config_options.c to
+ occur. (issue ASTERISK-21906) Reported by: John Hardin patches:
+ http.patch uploaded by jhardin (license 6512) named_acl.patch
+ uploaded by jhardin (license 6512) config_options.patch uploaded
+ by jhardin (license 6512) res_calendar.patch uploaded by jhardin
+ (license 6512) chan_motif.patch uploaded by jhardin (license
+ 6512) ........ Merged revisions 392810 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/parking.c, main/devicestate.c, main/cel.c,
+ main/presencestate.c, main/sorcery.c,
+ res/parking/parking_bridge.c, main/cdr.c, main/manager.c: Fix a
+ variety of memory leaks This patch addresses the following
+ memory/ref counting leaks: * main/devicestate.c - unsubscribe and
+ join our devicestate message subscription * main/cel.c - clean up
+ the datastore and config objects on exist * main/parking.c -
+ cleanup memory leak of retriever snapshot on message payload
+ destruction * res/parking/parking_bridge.c - cleanup memory leak
+ of retrieve snapshot on message payload destruction *
+ main/presencestate.c - unsubscribe and join the caching topic on
+ exit * manager.c - properly unregister the manager action
+ "BlindTransfer" * sorcery.c - shutdown the threadpool on exit and
+ dispose of any wizards (issue ASTERISK-21906) Reported by: John
+ Hardin patches: cel.patch uploaded by jhardin (license #6512)
+ devicestate.patch uploaded by jhardin (license #6512)
+ manager.patch uploaded by jardin (license #6512)
+ presencestate.patch uploaded by jhardin (license #6512)
+ retriever-channel-snapshot.patch uploaded by jhardin (license
+ #6512) sorcery.patch uploaded by jhardin (license #6512)
+
+2013-06-24 22:05 +0000 [r392778-392779] David M. Lee <dlee@digium.com>
+
+ * tests/test_endpoints.c, tests/test_stasis_endpoints.c: Few more
+ menuselect fixes missed in r392777
+
+ * rest-api-templates/res_stasis_json_resource.c.mustache,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ res/stasis_json/resource_sounds.h: Fixed templates so that the
+ changes from r392777 won't be overwritten the next time we run
+ the generators.
+
+2013-06-24 21:40 +0000 [r392777] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_stasis_http_sounds.c, res/res_statsd.c,
+ res/res_stasis_http_bridges.c, res/res_stasis_json_asterisk.c,
+ res/res_stasis_test.c, res/res_stasis_json_playback.c,
+ res/res_stasis_http.c, res/res_stasis.c, apps/app_stasis.c,
+ res/res_stasis_http_asterisk.c, res/res_stasis_json_channels.c,
+ res/res_stasis_http_playback.c, res/res_stasis_playback.c,
+ res/res_stasis_websocket.c, res/res_stasis_json_recordings.c,
+ res/res_stasis_http_channels.c, res/res_stasis_json_endpoints.c,
+ res/res_stasis_json_events.c, res/res_stasis_http_recordings.c,
+ res/res_stasis_answer.c, res/res_chan_stats.c,
+ res/res_stasis_http_endpoints.c, res/res_stasis_http_events.c,
+ res/res_stasis_json_sounds.c, res/res_stasis_bridge_add.c,
+ res/res_stasis_json_bridges.c: Fix menuselect display for stasis
+ modules. The menuselect parser is very simple. It looks for
+ AST_MODULE_INFO and uses any quoted string on that line as the
+ module summary display.
+
+2013-06-24 19:28 +0000 [r392729-392747] Mark Michelson <mmichelson@digium.com>
+
+ * /: Remove stray properties from merge.
+
+ * /, main/features_config.c, doc/appdocsxml.dtd: Add documentation
+ for features configuration. Review:
+ https://reviewboard.asterisk.org/r/2616 (closes issue
+ ASTERISK-21542) Reported by Matt Jordan
+
+2013-06-24 13:49 +0000 [r392700] Kinsey Moore <kmoore@digium.com>
+
+ * main/asterisk.c, main/media_index.c (added),
+ include/asterisk/file.h, include/asterisk/http.h,
+ include/asterisk/sounds_index.h (added),
+ res/stasis_http/resource_sounds.c, include/asterisk/media_index.h
+ (added), main/file.c, main/http.c, include/asterisk/format.h,
+ rest-api/api-docs/sounds.json, include/asterisk/_private.h,
+ main/sounds_index.c (added), res/res_stasis_http.c: Index
+ installed sounds and implement ARI sounds queries This adds
+ support for stasis/sounds and stasis/sounds/{ID} queries via the
+ Asterisk RESTful Interface (ARI, formerly Stasis-HTTP). The
+ following changes have been made to accomplish this: * A modular
+ indexer was created for local media. * A new function to get an
+ ast_format associated with a file extension was added. *
+ Modifications were made to the built-in HTTP server so that URI
+ decoding could be deferred to the URI handler when necessary. *
+ The Stasis-HTTP sounds JSON documentation was modified to handle
+ cases where multiple languages are installed in different
+ formats. * Register and Unregister events for formats were added
+ to the system topic. (closes issue ASTERISK-21584) (closes issue
+ ASTERISK-21585) Review: https://reviewboard.asterisk.org/r/2507/
+
+2013-06-23 19:19 +0000 [r392676] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c: Properly pack the parameters into ast_json_pack
+ when sending a send fax message This patch properly packs the
+ parameters into the send fax message so that it actually work.
+ Missing a ',' between two string fields can be difficult to
+ debug, particularly when the actual packing succeeds.
+ Interestingly enough, this didn't actually crash until the JSON
+ blob we deref'd and disposed of. Since that happened in a
+ different thread, it was pretty tough to track down.
+
+2013-06-23 18:59 +0000 [r392627-392667] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_outbound_registration.c,
+ res/res_sip_endpoint_identifier_ip.c, res/res_sip_acl.c: Add some
+ more missing ast_sorcery_generic_alloc conversions.
+
+ * tests/test_sorcery_astdb.c, tests/test_sorcery_realtime.c: Add
+ missing ast_sorcery_generic_alloc conversions.
+
+ * main/manager_endpoints.c: Fix a bug where messages were getting
+ duplicated on AMI. This was caused by forwarding all endpoint
+ messages to manager which includes channel messages that are
+ related to the endpoint. This change causes only the PeerStatus
+ messages to be forwarded to manager thus eliminating the
+ duplicate channel messages.
+
+2013-06-22 22:42 +0000 [r392607] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c: Properly extract channel variables for the
+ SendFAX/ReceiveFAX Stasis messages By the time something extracts
+ the pointers from ast_json_pack, the channels will already be
+ disposed of. This patch properly pulls the information out of the
+ variables and packs them into the JSON blob.
+
+2013-06-22 14:26 +0000 [r392565-392586] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/sorcery.h, res/res_sip/config_auth.c,
+ res/res_sip/sip_options.c, res/res_sip/location.c,
+ tests/test_sorcery.c, main/sorcery.c,
+ res/res_sip/config_domain_aliases.c,
+ res/res_sip/config_transport.c, res/res_sip/sip_configuration.c:
+ Make sorcery details opaque and add extended fields. Sorcery
+ specific object information is now opaque and allocated with the
+ object. This means that modules do not need to be recompiled if
+ the sorcery specific part is changed. It also means that sorcery
+ can store additional information on objects and ensure it is
+ freed or the reference count decreased when the object goes away.
+ To facilitate the above a generic sorcery allocator function has
+ been added which also ensures that allocated objects do not have
+ a lock. Extended fields have been added thanks to all of the
+ above which allows specific fields to be marked as extended, and
+ thus simply stored as-is within the object. Type safety is *NOT*
+ enforced on these fields. A consumer of them has to query and
+ ultimately perform their own safety check. What does this mean?
+ Extra modules can extend already defined structures without
+ having to modify them. Tests have also been included to verify
+ extended field functionality. Review:
+ https://reviewboard.asterisk.org/r/2585/
+
+ * res/res_sip_pubsub.exports.in, channels/sip/include/sip.h,
+ include/asterisk/sdp_srtp.h (added), channels/sip/sdp_crypto.c
+ (removed), main/pbx.c, channels/sip/srtp.c (removed),
+ res/res_sip_transport_websocket.c (added), channels/chan_sip.c,
+ res/res_sip_registrar.c, res/res_sip/sip_distributor.c,
+ include/asterisk/res_sip_session.h,
+ include/asterisk/res_sip_exten_state.h (added),
+ res/res_sip/security_events.c (added),
+ res/res_sip_registrar_expire.c (added), res/res_sip.c,
+ res/res_sip_pidf.c (added), include/asterisk/res_sip_pubsub.h,
+ channels/sip/include/sdp_crypto.h (removed),
+ res/res_sip/location.c, res/res_sip_outbound_registration.c,
+ channels/sip/include/srtp.h (removed),
+ res/res_sip_endpoint_identifier_anonymous.c (added),
+ res/res_sip_one_touch_record_info.c (added),
+ res/res_sip_pubsub.c, res/res_sip/config_transport.c,
+ configs/res_sip.conf.sample, res/res_sip/sip_configuration.c,
+ res/res_sip_diversion.c (added), res/res_sip_refer.c (added),
+ include/asterisk/res_sip.h, res/res_sip_dtmf_info.c,
+ main/sdp_srtp.c (added), res/res_sip/include/res_sip_private.h,
+ res/res_sip.exports.in, res/res_sip_exten_state.exports.in
+ (added), res/res_sip_session.exports.in, res/res_sip_sdp_rtp.c,
+ res/res_sip_messaging.c (added), res/res_sip_caller_id.c,
+ channels/chan_gulp.c, res/res_sip_session.c,
+ res/res_sip_exten_state.c (added), res/res_sip/sip_options.c:
+ Merge in current pimp_my_sip work, including: 1. Security events
+ 2. Websocket support 3. Diversion header + redirecting support 4.
+ An anonymous endpoint identifier 5. Inbound extension state
+ subscription support 6. PIDF notify generation 7. One touch
+ recording support (special thanks Sean Bright!) 8. Blind and
+ attended transfer support 9. Automatic inbound registration
+ expiration 10. SRTP support 11. Media offer control dialplan
+ function 12. Connected line support 13. SendText() support 14.
+ Qualify support 15. Inband DTMF detection 16. Call and pickup
+ groups 17. Messaging support Thanks everyone! Side note: I'm
+ reminded of the song "How Far We've Come" by Matchbox Twenty.
+
+2013-06-22 13:58 +0000 [r392564] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c: Fix a deadlock and possible crash in res_fax This
+ patch fixes two bugs. (1) It unlocks the channel in the framehook
+ handlers before attempting to grab the peer from the bridge. The
+ locking order for the bridging framework is bridge first, then
+ channel - having the channel locked while attempting to obtain
+ the bridge lock causes a locking inversion and a deadlock. This
+ patch bumps the channel ref count prior to releasing the lock in
+ the framehook to avoid lifetime issues. Note that this does
+ expose a subtle problem in framehooks; that is, something could
+ modify the framehook list while we are executing, causing issues
+ in the framehook list traversal that the callback executes in.
+ Fixing this is a much larger problem that is beyond the scope of
+ this patch - (a) we already unlock the channel in this particular
+ framehook and we haven't run into a problem yet (as modifying the
+ framehook list when a channel is about to perform a fax gateway
+ would be a very odd operation) and (b) migrating to an ao2
+ container of framehooks would be more invasive at this point. See
+ the referenced ASTERISK issue for more information. (2) Directly
+ packing channel variables into a JSON object turned out to be
+ unsafe. A condition existed where the strings in the JSON blob
+ were no longer safe to be accessed if the channel object itself
+ was disposed of. (issue ASTERISK-21951)
+
+2013-06-22 12:40 +0000 [r392538] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/manager.h,
+ channels/chan_gulp.c, main/stasis_endpoints.c, res/res_sip.c,
+ main/manager.c, channels/chan_sip.c, channels/chan_skinny.c,
+ res/res_sip/sip_configuration.c, include/asterisk/res_sip.h,
+ main/manager_endpoints.c (added),
+ include/asterisk/stasis_endpoints.h: Migrate PeerStatus events to
+ stasis, add stasis endpoints, and add chan_pjsip device state.
+ (closes issue ASTERISK-21489) (closes issue ASTERISK-21503)
+ Review: https://reviewboard.asterisk.org/r/2601/
+
+2013-06-21 22:39 +0000 [r392514] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, include/asterisk/bridging_technology.h,
+ bridges/bridge_holding.c, include/asterisk/bridging.h,
+ bridges/bridge_simple.c, bridges/bridge_softmix.c,
+ bridges/bridge_native_rtp.c: Extract a useful routine from the
+ softmix bridge technology. * Extract a useful routine from the
+ softmix bridge technology for other technologies. Make other
+ technologies use it if they can. * Made native and 1-1 bridges
+ write to all parties if the bridge channel writing the frame into
+ the bridge is NULL. Softmix will also do the same for frame types
+ that make sense. * Tweak the bridge write routine return value
+ meaning and adjust the bridge technologies to match.
+
+2013-06-21 21:22 +0000 [r392489] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_gulp.c: Add BUGBUG for broken direct media in
+ chan_gulp (issue ASTERISK-21947)
+
+2013-06-21 18:54 +0000 [r392464] Jason Parker <jparker@digium.com>
+
+ * rest-api/api-docs/channels.json: Fix typo.
+
+2013-06-21 18:10 +0000 [r392437] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Add channel optimization interaction with frame
+ hooks BUGBUG comments.
+
+2013-06-21 18:05 +0000 [r392436] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_unistim.c: Change chan_unistim to use core transfer
+ API. Review: https://reviewboard.asterisk.org/r/2553 (closes
+ issue ASTERISK-21527) Reported by Matt Jordan
+
+2013-06-21 17:48 +0000 [r392435] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/bridging.h, main/features.c,
+ bridges/bridge_softmix.c, main/bridging.c,
+ include/asterisk/bridging_technology.h: Change several bridge
+ functions to return error status. The bridge frame queue
+ functions need to return an error status if the frame failed to
+ be queued because of an error condition. The main calls that
+ needed to return the status are:
+ ast_bridge_channel_queue_action_data() and
+ ast_bridge_channel_write_action_data(). The other return changes
+ are ripple effects.
+
+2013-06-21 14:21 +0000 [r392409] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/scripts/autosupport: Update autosupport script This patch
+ updates the autosupport script to collect all information
+ available to the Asterisk CLI command "digium_phones". It also
+ makes minor improvements in options handling. (closes issue
+ AST-1163) Reported by: Trey Blancher patches:
+ 390347_autosupport.diff uploaded by tblancher (License 5821)
+ 390348_autosupport.diff uploaded by tblancher (License 5821)
+
+2013-06-20 21:13 +0000 [r392364] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_session.c: Add a log message for when an incoming
+ session is rejected due to the extension not being found.
+
+2013-06-20 17:21 +0000 [r392335] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, res/parking/parking_bridge_features.c,
+ apps/confbridge/conf_config_parser.c,
+ include/asterisk/bridging_features.h, main/features.c: Fix
+ potential bridge hook resource leak if the hook install fails.
+
+2013-06-20 16:29 +0000 [r392318] Mark Michelson <mmichelson@digium.com>
+
+ * main/threadpool.c: Fix threadpool rapid growth problem. When a
+ threadpool is set to autoincrement its threadcount, an issue may
+ arise when multiple tasks are queued at once into the threadpool.
+ Since threads start active, each new task would result in
+ autoincrementing the thread count. So if all threads were active,
+ and a thread's autoincrement value were 5, then 3 new tasks would
+ result in 15 threads being created even though the initial
+ autoincrement was sufficient to handle the number of tasks. This
+ change introduces three behavior changes: 1) New threads in the
+ threadpool start idle instead of active. 2) When a threadpool
+ autoincrements, one thread is activated after the growth. 3) When
+ a threadpool's size is incremented manually, all added threads
+ are activated. For a more detailed explanation about the changes,
+ please see the Review Board link at the bottom of this commit.
+ Review: https://reviewboard.asterisk.org/r/2629
+
+2013-06-19 22:52 +0000 [r392279] David M. Lee <dlee@digium.com>
+
+ * Makefile, main/Makefile: Fix build problem on OS X Mountain Lion
+ (10.8) For about forever, our build flags for OS X have been
+ slightly off, but good enough to build and run. Apparently they
+ aren't good enough any more. Previously, we would compile with
+ macosx-version-min unset and link with it set. This combination,
+ using GCC 4.8, on Mountain Lion, would create a bad executable
+ ("Illegal Instruction: 4", or something like that) This patch
+ consistently sets macosx-version-min for both compiling and
+ linking, which makes everything happy enough to build and run.
+
+2013-06-19 12:55 +0000 [r392241] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/cel.h, main/cel.c: Pull CEL linkedid
+ manipulation into cel.c This finishes moving all CEL linkedid
+ tracking entirely within cel.c since that is now possible with
+ channel snapshots. This also removes another CEL linkedid
+ manipulation function from cel.h that has already been
+ internalized and is neither called nor available to link against.
+ Review: https://reviewboard.asterisk.org/r/2632/
+
+2013-06-19 01:28 +0000 [r392190-392214] Matthew Jordan <mjordan@digium.com>
+
+ * funcs/func_cdr.c: Handle variable substitution in dummy variables
+ When func_cdr is used for variable substitution, there is no
+ channel name and hence no run-time information available for CDR
+ variable substitution. In that case, the correct thing to do is
+ to use the CDR object on the channel passed to the function. This
+ patch checks to see if the channel passed in has a name - if not,
+ it uses ast_cdr_format_var instead of ast_cdr_get_var. This
+ allows CDR backends to continue to use variable substitution in
+ order to resolve ast_cdr object properties.
+
+ * tests/test_substitution.c: Fix the test_substitution test In
+ r391947, the CDR function was modified such that it will return a
+ value for the start,answer, and end times if asked. That time
+ will just be 0 if it hasn't happened yet.
+
+2013-06-18 19:31 +0000 [r392139-392166] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, include/asterisk/bridging.h: Bridging: Fix crash
+ on destruction of a partially constructed bridge. * Promoted some
+ bridge construction debug messages to warnings.
+
+ * main/bridging.c: Add some safety cleanup for a failed push into a
+ bridge.
+
+ * main/bridging_basic.c: Remove stub comment on function that is
+ not a stub.
+
+2013-06-18 14:30 +0000 [r392116] Kinsey Moore <kmoore@digium.com>
+
+ * main/stasis_bridging.c, include/asterisk/stasis_bridging.h,
+ rest-api/api-docs/bridges.json: Fix bridge snapshot conversion to
+ JSON This makes ast_bridge_snapshot_to_json conform to the
+ swagger Bridge model by adding the two fields it required.
+ Review: https://reviewboard.asterisk.org/r/2583/
+
+2013-06-17 18:58 +0000 [r392076] David M. Lee <dlee@digium.com>
+
+ * funcs/func_cdr.c, main/cdr.c: Fix build warnings related to
+ printf/scanf of tv_usec. The type of tv_usec is suseconds_t. On
+ Linux, this is usually a long int, but the specification is
+ actually pretty lax on what it might actually be. And, sadly,
+ there's no printf/scanf width specifier for suseconds_t. So it
+ could bit an int or a long, but there's not a great way to tell
+ which it is. This patch fixes scanf by reading into a long
+ temporary variable that's then stored into the tv_usec. It fixes
+ printf by casting the tv_usec to a long first. This patch also
+ adds some missing width specifiers for some debug statements,
+ which would cause ".000001" to be displayed at ".1".
+
+2013-06-17 18:37 +0000 [r392053-392073] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_vpb.cc: chan_vpb: Fix compile error
+ and __ast_channel_alloc() prototype const inconsistency.
+
+ * channels/chan_misdn.c: chan_misdn: Fix compile error after CDR
+ merge.
+
+2013-06-17 16:59 +0000 [r392032] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/app.h: Fix a build warning with stasis messages.
+
+2013-06-17 14:40 +0000 [r392004-392005] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager_channels.c: Prevent sending a NewExten event after a
+ Hangup during a stack restore When a channel is originated, its
+ application is typically set to AppDial2, indicating that it was
+ a dialed channel through the Dial API. Asterisk during an
+ originate will perform a stack execute to direct the outgoing
+ channel to a particular place in the dialplan or application.
+ When the stack returns, the previous application (AppDial2) is
+ restored. Unfortunately, in the case of an originated channel,
+ the stack restore happens after hangup. A stasis message is sent
+ notifying everyone that the application was restored, and this
+ causes a NewExten event to go out after the Hangup event,
+ violating the basic contract consumers have of the channel
+ lifetime. While we could preclude the message from going out,
+ restoring the channel's state before it executed the next higher
+ frame in the stack has to occur, and other places in the code
+ depend on this behavior. Since we know that channel hung up (it's
+ a ZOMBIE!), this patch simply checks to see if the channel has
+ been zombified before sending a NewExten event. Note that this
+ will fix a number of bouncing tests in the Test Suite. Go tests.
+
+ * CHANGES: Restore bad merge on CHANGES The patch for CDRs moved
+ around a lot of content in CHANGES to try and organize the areas
+ that were affected. This missed some changes that went in with a
+ merge and removed some updates - this patch adds them back in.
+
+2013-06-17 12:28 +0000 [r391982] Joshua Colp <jcolp@digium.com>
+
+ * main/cdr.c: Fix build warning (which is transmogrified into an
+ error) with my compiler due to uninitialized variable.
+
+2013-06-17 03:31 +0000 [r391947-391964] Matthew Jordan <mjordan@digium.com>
+
+ * addons/cdr_mysql.c: Make cdr_mysql compile again by not directly
+ setting the run-time CDR object A stray ast_cdr_setvar was missed
+ in cdr_mysql (silly addons). This has now been refactored to not
+ set the property, as the property would have been set on a
+ run-time object that was already dispatched to the backend. The
+ module simply remembers the value it wanted to set and writes it
+ to MySQL later in the processing.
+
+ * main/manager.c, apps/app_osplookup.c, main/features.c,
+ apps/app_dumpchan.c, main/manager_channels.c, main/bridging.c,
+ cdr/cdr_custom.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
+ apps/app_dial.c, main/stasis_cache.c, cdr/cdr_syslog.c,
+ cel/cel_tds.c, channels/chan_agent.c, apps/app_disa.c,
+ apps/app_queue.c, CHANGES, res/res_monitor.c, apps/app_forkcdr.c,
+ include/asterisk/stasis_channels.h, main/test.c,
+ channels/chan_h323.c, main/asterisk.c, channels/chan_unistim.c,
+ addons/chan_ooh323.c, include/asterisk/cel.h,
+ apps/app_authenticate.c, cdr/cdr_pgsql.c, apps/app_followme.c,
+ channels/chan_iax2.c, res/res_config_sqlite.c, main/stasis.c,
+ cdr/cdr_csv.c, main/cli.c, main/dial.c, channels/chan_skinny.c,
+ cel/cel_manager.c, res/res_agi.c, main/stasis_channels.c,
+ cdr/cdr_odbc.c, tests/test_cdr.c (added), main/bridging_basic.c,
+ main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c,
+ UPGRADE.txt, include/asterisk/cdr.h, include/asterisk/channel.h,
+ res/res_stasis_answer.c, main/cel.c, cdr/cdr_tds.c,
+ funcs/func_channel.c, funcs/func_cdr.c,
+ include/asterisk/bridging.h, addons/cdr_mysql.c,
+ funcs/func_callerid.c, apps/app_cdr.c, include/asterisk/time.h,
+ cel/cel_radius.c, include/asterisk/stasis_internal.h (added),
+ include/asterisk/channel_internal.h, main/utils.c,
+ cdr/cdr_adaptive_odbc.c, cdr/cdr_radius.c, main/channel.c,
+ main/cdr.c, include/asterisk/test.h, channels/chan_dahdi.c:
+ Update Asterisk's CDRs for the new bridging framework This patch
+ is the initial push to update Asterisk's CDR engine for the new
+ bridging framework. This patch guts the existing CDR engine and
+ builds the new on top of messages coming across Stasis. As
+ changes in channel state and bridge state are detected, CDRs are
+ built and dispatched accordingly. This fundamentally changes CDRs
+ in a few ways. (1) CDRs are now *very* reflective of the actual
+ state of channels and bridges. This means CDRs track well with
+ what an actual channel is doing - which is useful in transfer
+ scenarios (which were previously difficult to pin down). It does,
+ however, mean that CDRs cannot be 'fooled'. Previous behavior in
+ Asterisk allowed for CDR applications, channels, and other
+ properties to be spoofed in parts of the code - this no longer
+ works. (2) CDRs have defined behavior in multi-party scenarios.
+ This behavior will not be what everyone wants, but it is a
+ defined behavior and as such, it is predictable. (3) The CDR
+ manipulation functions and applications have been overhauled.
+ Major changes have been made to ResetCDR and ForkCDR in
+ particular. Many of the options for these two applications no
+ longer made any sense with the new framework and the (slightly)
+ more immutable nature of CDRs. There are a plethora of other
+ changes. For a full description of CDR behavior, see the CDR
+ specification on the Asterisk wiki. (closes issue ASTERISK-21196)
+ Review: https://reviewboard.asterisk.org/r/2486/
+
+2013-06-14 23:26 +0000 [r391921] Mark Michelson <mmichelson@digium.com>
+
+ * main/app.c: Fix regression in MWI stasis handling. In revision
+ 389733, mwi state allocation was placed into its own function
+ instead of performing the allocation in-line when required. The
+ issue was that in ast_publish_mwi_state_full(), the local
+ variable "uniqueid" was no longer being set, but it was still
+ being used as the topic for MWI. This meant that all MWI
+ publications ended up being published to the "" (empty string)
+ mailbox topic. Thus MWI subscriptions for specific mailboxes were
+ never notified of mailbox state changes. This change fixes the
+ issue by removing the local uniqueid variable from
+ ast_publish_mwi_state_full() and instead referencing the
+ mwi_state->uniqueid field since it has been properly set. (closes
+ issue ASTERISK-21913) Reported by Malcolm Davenport
+
+2013-06-14 21:57 +0000 [r391902] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_registrar.c: Ensure that the number of added contacts
+ never goes below 0. This can happen when a REGISTER request is
+ removing a contact. (closes issue ASTERISK-21911) Reported by:
+ mdavenport
+
+2013-06-14 18:50 +0000 [r391855-391856] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/stasis_bridging.h,
+ rest-api/api-docs/bridges.json, main/stasis_bridging.c: Revert
+ parts of r391855 that were not ready to go in to trunk
+
+ * main/stasis_bridging.c, main/cel.c,
+ include/asterisk/stasis_bridging.h,
+ rest-api/api-docs/bridges.json: Fix two more possible crashes in
+ CEL These are locations that should return valid snapshots, but
+ need to be handled if not.
+
+2013-06-14 16:32 +0000 [r391828] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_mixmonitor.c: app_mixmonitor: Fix crashes caused by
+ unloading app_mixmonitor Unloading app_mixmonitor while active
+ mixmonitors were running would cause a segfault. This patch fixes
+ that by making it impossible to unload app_mixmonitor while
+ mixmonitors are active. Review:
+ https://reviewboard.asterisk.org/r/2624/ ........ Merged
+ revisions 391778 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391794 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-14 16:12 +0000 [r391776-391777] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c: Fix a crash in CEL bridge snapshot handling Properly
+ search for bridge association structures so that they are found
+ when expected and handle cases where they don't exist.
+
+ * main/bridging.c: Publish bridge snapshots more often Bridge
+ snapshot events were missing some important transitions that were
+ noticed in subsequent snapshots. Snapshots will now be published
+ on all bridge reconfigurations.
+
+2013-06-13 21:53 +0000 [r391732] Matthew Jordan <mjordan@digium.com>
+
+ * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c,
+ utils/refcounter.c: Make the utils directory compile... again.
+ Utils is a source folder that lies, eventually all developers
+ will cry, "I know I must maintain it, But really with this last
+ commit I can kiss my software ethics good-bye."
+
+2013-06-13 19:04 +0000 [r391701] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c, apps/confbridge/conf_config_parser.c, /,
+ apps/confbridge/include/confbridge.h: app_confbridge: Fix memory
+ leak on reload. The config framework options should not be
+ registered multiple times. Instead the configuration just needs
+ to be reprocessed by the config framework. ........ Merged
+ revisions 391700 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-13 18:26 +0000 [r391699] Mark Michelson <mmichelson@digium.com>
+
+ * main/features_config.c: Just return outright on a reload since we
+ have already processed configuration.
+
+2013-06-13 18:20 +0000 [r391689] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c: Ensure that Asterisk still starts up when cel.conf is
+ missing
+
+2013-06-13 18:17 +0000 [r391676] Mark Michelson <mmichelson@digium.com>
+
+ * main/features_config.c: Fix memory leak in features_config.c The
+ options should not be registered multiple times. Instead, the
+ configuration just needs to be reprocessed by the config
+ framework. This also exposed that we were not properly telling
+ the config framework to treat the configuration processing with
+ the "reload" semantics when a reload occurred. Both of these
+ errors are fixed now. Thanks to Richard Mudgett for discovering
+ the leak.
+
+2013-06-13 18:14 +0000 [r391675] Matthew Jordan <mjordan@digium.com>
+
+ * main/json.c, main/manager.c, include/asterisk/json.h: Blow away
+ usage of libjansson's foreach macro While very handy, this macro
+ didn't occur until a later version of libjansson. We'd prefer to
+ be compatible with older versions still - as such, iteration over
+ key/value pairs in a JSON object have to be done with a little
+ bit more manual work.
+
+2013-06-13 13:46 +0000 [r391622-391643] Kinsey Moore <kmoore@digium.com>
+
+ * main/cel.c, include/asterisk/parking.h, main/asterisk.c,
+ res/parking/parking_manager.c, main/parking.c,
+ include/asterisk/cel.h, main/features.c,
+ include/asterisk/_private.h: Refactor CEL bridge events on top of
+ Stasis-Core This pulls bridge-related CEL event triggers out of
+ the code in which they were residing and pulls them into cel.c
+ where they are now triggered by changes in bridge snapshots. To
+ get access to the Stasis-Core parking topic in cel.c, the
+ Stasis-Core portions of parking init have been pulled into core
+ Asterisk init. This also adds a new CEL event
+ (AST_CEL_BRIDGE_TO_CONF) that indicates a two-party bridge has
+ transitioned to a multi-party conference. The reverse cannot
+ occur in CEL terms even though it may occur in actuality and two
+ party bridges which receive a AST_CEL_BRIDGE_TO_CONF will be
+ treated as multi-party conferences for the duration of the
+ bridge. Review: https://reviewboard.asterisk.org/r/2563/ (closes
+ issue ASTERISK-21564)
+
+ * main/channel.c, include/asterisk/config_options.h, main/pbx.c,
+ include/asterisk/stasis_channels.h, main/stasis_bridging.c,
+ main/config_options.c, main/stasis_channels.c,
+ include/asterisk/strings.h, main/cel.c,
+ include/asterisk/stasis_bridging.h, main/asterisk.c: Refactor CEL
+ channel events on top of Stasis-Core This uses the channel state
+ change events from Stasis-Core to determine when channel-related
+ CEL events should be raised. Those refactored in this patch are:
+ * AST_CEL_CHANNEL_START * AST_CEL_ANSWER * AST_CEL_APP_START *
+ AST_CEL_APP_END * AST_CEL_HANGUP * AST_CEL_CHANNEL_END Retirement
+ of Linked IDs is also refactored. CEL configuration has been
+ refactored to use the config framework. Note: Some HANGUP events
+ are not generated correctly because the bridge layer does not
+ propagate hangupcause/hangupsource information yet. Review:
+ https://reviewboard.asterisk.org/r/2544/ (closes issue
+ ASTERISK-21563)
+
+2013-06-13 11:02 +0000 [r391596] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/channel.h, include/asterisk/stasis_endpoints.h,
+ main/endpoints.c, res/stasis_http/resource_endpoints.c,
+ main/stasis_cache.c, main/stasis_endpoints.c,
+ main/channel_internal_api.c, include/asterisk/stasis.h: Add
+ support for requiring that all queued messages on a caching topic
+ have been handled before retrieving from the cache and also
+ change adding channels to an endpoint to be an immediate
+ operation. Review: https://reviewboard.asterisk.org/r/2599/
+
+2013-06-12 21:08 +0000 [r391561] David M. Lee <dlee@digium.com>
+
+ * res/res_http_websocket.c, /: Fix segfault for certain invalid
+ WebSocket input. The WebSocket code would allocate, on the stack,
+ a string large enough to hold a key provided by the client, and
+ the WEBSOCKET_GUID. If the key is NULL, this causes a segfault.
+ If the key is too large, it could overflow the stack. This patch
+ checks the key for NULL and checks the length of the key to avoid
+ stack smashing nastiness. (closes issue ASTERISK-21825) Reported
+ by: Alfred Farrugia Tested by: Alfred Farrugia, David M. Lee
+ Patches: issueA21825_check_if_key_is_sent.patch uploaded by
+ Walter Doekes (license 5674) ........ Merged revisions 391560
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-12 02:29 +0000 [r391479-391521] Matthew Jordan <mjordan@digium.com>
+
+ * main/endpoints.c, main/loader.c, main/format.c, /: Fix memory
+ leak while loading modules, adding formats, and destroying
+ endpoints This patch fixes three memory leaks * When we load a
+ module with the LOAD_PRIORITY flag, we remove its entry from the
+ load order list. Unfortunately, we don't free the memory
+ associated with entry in the list. This patch corrects that and
+ properly frees the memory for the module in the list. * When
+ adding a custom format (such as SILK or CELT), the routine for
+ adding the format was leaking a reference. RAII_VAR cleans this
+ up properly. * We now de-ref the channel_snapshot appropriately
+ when an endpoint is disposed of ........ Merged revisions 391489
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 391507 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/stasis_channels.c, bridges/bridge_native_rtp.c: Fix memory
+ leaks in stasis_channels and bridge_native_rtp This patch fixes
+ two memory leaks: * A memory leak in packing channels into a
+ multi-channel blob payload when publishing dial messages. The
+ multi-channel blob payload does not steal the references - this
+ approach was chosen because it works well with the RAII_VAR
+ macro. Unfortunately, this does mean that you actually have to
+ use the RAII_VAR macro (or manually deref it yourself) * RTP
+ instances returned as a result of one of the glue operations are
+ ref counted and have to be de-ref'd appropriately. We now do
+ that, as saying that we should do it and then not would be silly.
+
+2013-06-11 22:57 +0000 [r391455] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridging.c: Remove incorrect comment about local channel
+ optimization occurring when performing an attended transfer on an
+ entire bridge.
+
+2013-06-11 22:21 +0000 [r391430-391453] Jonathan Rose <jrose@digium.com>
+
+ * bridges/bridge_native_rtp.c, include/asterisk/framehook.h,
+ main/framehook.c: bridge_native_rtp: Fix native bridge tech being
+ incompatible when it should be. When checking compatability for
+ the native RTP bridge technology there is a race condition
+ between clearing framehooks that are destroyed when leaving
+ certain bridges with certain technologies (such as
+ bridge_native_rtp) and joining bridges with the bridge_native_rtp
+ technology. Yes, that means a channel in a native RTP bridge
+ could move to another native RTP bridge and be considered
+ incompatible with the new native RTP bridge causing it to revert
+ to a simple bridge technology0. This fixes that bug by ignoring
+ framehooks that have been marked for destruction when checking
+ for compatibility with the bridge_native_rtp technology.
+
+ * bridges/bridge_native_rtp.c: bridge_native_rtp: Fix possible
+ segfaults on leaves/joins native_rtp_bridge_get can return any
+ result from the ast_rtp_glue_result enumerator and the join/leave
+ functions for bridge_native_rtp seem to assume that if the result
+ wasn't local that it was remote. Meanwhile forbid can be returned
+ by that function which can mean certain glue pointers are NULL.
+ Then when the join/leave functions try to use members of that
+ pointer, boom. Segfault.
+
+2013-06-11 15:46 +0000 [r391403] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/stasis.h, main/stasis_channels.c,
+ tests/test_stasis.c, main/manager_channels.c, main/manager.c,
+ main/stasis_message.c, main/parking.c,
+ tests/test_stasis_channels.c: Add vtable and methods for to_json
+ and to_ami for Stasis messages When a Stasis message type is
+ defined in a loadable module, handling those messages for AMI and
+ res_stasis events can be cumbersome. This patch adds a vtable to
+ stasis_message_type, with to_ami and to_json virtual functions.
+ These allow messages to be handled abstractly without putting
+ module-specific code in core. As an example, the VarSet AMI event
+ was refactored to use the to_ami virtual function. (closes issue
+ ASTERISK-21817) Review: https://reviewboard.asterisk.org/r/2579/
+
+2013-06-11 10:24 +0000 [r391380] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix issue with no sound in both way
+ in case of previous call to chan_unistim phone was canceled.
+ (related to ASTERISK-20183) ........ Merged revisions 391379 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-11 08:13 +0000 [r391335] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_iax2.c, /: IAX2: Transfer Reject: Lock bridgecallno
+ before touching it, refactor 1). When touching the bridgecallno,
+ we need to lock it. 2). Remove magic number '0' and replace with
+ TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2613/ ........ Merged
+ revisions 391333 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391334 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-10 22:38 +0000 [r391314] Matthew Jordan <mjordan@digium.com>
+
+ * main/loader.c: Make the reload stasis message bump the ref count
+ of its sub-object JSON objects are reference stealing. Hence, if
+ you've RAII_VAR'd some subobject and want to pack it into another
+ JSON object, you have to bump the reference count. Using the 'O'
+ option during the pack will bump the reference count for you.
+
+2013-06-10 21:04 +0000 [r391297] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Change chan_skinny to use core transfer
+ API. Changes for both attended and blind transfers in chan_skinny
+ to use the new transfer API instead of masquerade. (closes issue
+ ASTERISK-21526) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2557/
+
+2013-06-10 16:03 +0000 [r391271] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_agi.c: Add AGI command arguments to AsyncAGI event This
+ makes the AGI AsyncAGI event put provided AGI command arguments
+ in the event's environment. (closes issue ASTERISK-21304)
+ Patch-By: Dirk Wendland
+
+2013-06-10 15:32 +0000 [r391269] Mark Michelson <mmichelson@digium.com>
+
+ * main/features_config.c: Temporary fix for people using sample
+ features.conf from previous Asterisk versions. People who use the
+ features.conf.sample file from Asterisk 11 and before in trunk
+ were given a rude awakening when features configuration changes
+ were made. Because it uses the config framework and the config
+ framework is strict about what is accepted and what isn't, people
+ that had parking options configured found that Asterisk no longer
+ started. This is because parking options are currently handled in
+ res_parking.conf instead of features.conf. This fix seeks to
+ create a temporary band-aid fix for the problem, but having
+ parking options from the general section be passed to a handler
+ that will simply print that the option is no longer supported.
+ This will not cause Asterisk to exit. The fix only applies to
+ options in the general section. There are two main reasons for
+ this: 1) The sample features.conf file only has parking options
+ in the general section. There are no configured parking lots.
+ Therefore it's not quite as "urgent" to get the parking lot
+ parsing fixed. 2) The plan is to move parking configuration back
+ from res_parking.conf to features.conf. When that happens, the
+ parking lots will also be addressed at that time.
+
+2013-06-10 14:36 +0000 [r391245] Matthew Jordan <mjordan@digium.com>
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Add
+ announce-to-first-user option for app_queue In r386792, the
+ ability to play prompts to the first caller in a call queue was
+ added. While this is arguably a bug fix for those who expect the
+ first caller to continue receiving prompts while the agent is
+ dialed, it has the side effect of preventing the first caller
+ from hearing the agent immediately upon bridging. This may not be
+ a problem for those who really want this option, but for those
+ who didn't care whether or not the first caller in queue heard
+ their position, it was an issue. This patch disables the ability
+ for the first caller in the queue to hear prompts and adds a new
+ option, announce-to-first-user, to queues.conf. Those who the
+ behavior can enable it by setting this value to True. Note that
+ if we ever implement the ability to have the prompts be stopped
+ upon bridging, this option can be removed. (closes issue
+ ASTERISK-21782) Reported by: Remi Quezada ........ Merged
+ revisions 391215 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391241 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-10 13:07 +0000 [r391199] Kinsey Moore <kmoore@digium.com>
+
+ * rest-api/api-docs/events.json, res/stasis/control.c,
+ res/stasis/app.c, res/res_stasis_bridge_add.exports.in (added),
+ include/asterisk/stasis_app.h,
+ res/stasis_http/resource_bridges.c, res/stasis/app.h,
+ res/res_stasis_json_events.c, include/asterisk/stasis_bridging.h,
+ rest-api/api-docs/bridges.json,
+ res/stasis_http/resource_bridges.h, res/res_stasis_bridge_add.c
+ (added), main/stasis_bridging.c,
+ res/stasis_json/resource_events.h, res/res_stasis.c,
+ res/res_stasis_json_events.exports.in: Stasis-HTTP: Flesh out
+ bridge-related capabilities This adds support for Stasis
+ applications to receive bridge-related messages when the
+ application shows interest in a given bridge. To supplement this
+ work and test it, this also adds support for the following
+ bridge-related Stasis-HTTP functionality: * GET stasis/bridges *
+ GET stasis/bridges/{bridgeId} * POST stasis/bridges * DELETE
+ stasis/bridges/{bridgeId} * POST
+ stasis/bridges/{bridgeId}/addChannel * POST
+ stasis/bridges/{bridgeId}/removeChannel Review:
+ https://reviewboard.asterisk.org/r/2572/ (closes issue
+ ASTERISK-21711) (closes issue ASTERISK-21621) (closes issue
+ ASTERISK-21622) (closes issue ASTERISK-21623) (closes issue
+ ASTERISK-21624) (closes issue ASTERISK-21625) (closes issue
+ ASTERISK-21626)
+
+2013-06-10 09:33 +0000 [r391064-391154] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+ unlock bridgecallno ........ Merged revisions 391143 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391148 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: fix bad edit after conflict resolution
+ ........ Merged revisions 391107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391111 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: IAX2: refactor nativebridge transfer
+ remove triple checking of iaxs[fr->callno]->transferring reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2602/ ........ Merged
+ revisions 391065 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391084 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: IAX2: fix race condition with
+ nativebridge transfers. 1). When touching the bridgecallno, we
+ need to lock it. 2). stop_stuff() which calls
+ iax2_destroy_helper() Assumes the lock on the pvt is already
+ held, when iax2_destroy_helper() is called. Thus we need to lock
+ the bridgecallno pvt before we call
+ stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating
+ the state of 'callno->transferring' of the current leg, we can't
+ change it to READY unless the bridgecallno is locked. Why, if we
+ are interrupted by the other call leg before 'transferring =
+ TRANSFER_RELEASED', the interrupt will find that it is READY and
+ that the bridgecallno is also READY so Releases the legs. (closes
+ issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2594/ ........ Merged
+ revisions 391062 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 391063 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-06-09 21:11 +0000 [r391012-391040] Matthew Jordan <mjordan@digium.com>
+
+ * main/app.c: Clean up MWI topic pool before message type
+ destruction Topics need to be disposed of prior to the message
+ types that are published on them. This includes topic pools. This
+ prevents an assertion from being raised on shutdown.
+
+ * main/manager.c: Only initialize manager_bridging during startup
+ This moves the initialization call behind the protection against
+ reloads. We don't want to re-add message router routes during
+ reloads.
+
+ * main/backtrace.c (added), main/logger.c, include/asterisk/lock.h,
+ main/astmm.c, utils/extconf.c, main/astobj2.c,
+ include/asterisk/backtrace.h (added), include/asterisk/logger.h:
+ Add backtrace generation to MALLOC_DEBUG memory corruption
+ reports This patch allows astmm to access the backtrace
+ generation code in Asterisk. When memory is allocated, a
+ backtrace is created and stored with the memory region that
+ tracks the allocation. If a memory corruption is detected, the
+ backtrace is printed to the astmm log. The backtrace will make
+ use of the BETTER_BACKTRACES build option if available. As a
+ result, this patch moves the backtrace generation code into its
+ own file and uses the non-wrapped versions of the C library
+ memory allocation routines. This allows the memory allocation
+ code to safely use the backtrace generation routines without
+ infinitely recursing. Review:
+ https://reviewboard.asterisk.org/r/2567
+
+2013-06-08 06:31 +0000 [r390940-390991] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, include/asterisk/bridging_technology.h: Add more
+ support for native bridging. * Added a start technology callback
+ that technologies can use to start bridging operations. It is
+ expected that native bridges will find this useful. * Factored
+ out bridge_channel_complete_join().
+
+ * main/bridging.c, include/asterisk/bridging_technology.h,
+ bridges/bridge_softmix.c: Fix a crash when a bridge switches from
+ the softmix bridge technology to another. A three party bridge
+ uses the softmix bridging technology. This technology has a
+ dedicated thread used to perform the analog mixing. When one of
+ these parties leaves the bridge, the bridge technology is changed
+ from the softmix technology to a two-party mixing technology.
+ Changing technologies is done by removing channels from the old
+ technology and adding them to the new technology. Since the
+ remaining channels do not leave the bridge, the softmix mixing
+ thread could continue to process all channels in the bridge. If
+ the bridge code is not able to start destruction of the softmix
+ technology before the softmix mixing thread wakes up, a crash
+ happens. * Added a stop technology callback that technologies can
+ use to request any helper threads to stop in preparation for
+ being destroyed. (closes issue AST-1156) Reported by: John
+ Bigelow
+
+ * include/asterisk/bridging_technology.h: Update some doxygen
+ comments.
+
+ * bridges/bridge_softmix.c: The bridge uniqueid is available for
+ softmix destructor.
+
+ * bridges/bridge_softmix.c: Add some bridge identifiers to some
+ softmix messages.
+
+2013-06-07 20:51 +0000 [r390920] Jonathan Rose <jrose@digium.com>
+
+ * res/parking/parking_devicestate.c (added): res_parking: Add
+ parking_devicestate.c left out from previous commit (issue
+ ASTERISK-21645) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2545/
+
+2013-06-07 19:51 +0000 [r390885-390901] Jason Parker <jparker@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c,
+ main/manager.c: Make app_queue AMI events more consistent. Give
+ Join/Leave more useful names. This also removes the
+ eventwhencalled and eventmemberstatus configuration options.
+ These events can just be filtered via manager.conf blacklists.
+ (closes issue ASTERISK-21469) Review:
+ https://reviewboard.asterisk.org/r/2586/
+
+ * res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json,
+ res/stasis_json/resource_channels.h,
+ res/stasis_http/resource_channels.c,
+ res/res_stasis_http_channels.c: Implement ARI POST to /channels,
+ to originate a call. (closes issue ASTERISK-21617) Review:
+ https://reviewboard.asterisk.org/r/2597/
+
+2013-06-07 16:22 +0000 [r390864] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_devicestate.c: Ensure that all unit tests compile with
+ the cache clear rework in place
+
+2013-06-07 16:07 +0000 [r390848-390849] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c, res/parking/parking_controller.c,
+ include/asterisk/pbx.h, CHANGES,
+ res/parking/parking_bridge_features.c,
+ res/parking/parking_bridge.c, main/pbx.c,
+ res/parking/res_parking.h, res/res_parking.c: res_parking:
+ Automatically generate extensions, hints, etc. (closes issue
+ ASTERISK-21645) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2545/
+
+ * include/asterisk/manager.h, main/manager.c, apps/app_meetme.c,
+ apps/confbridge/confbridge_manager.c: app_meetme: Refactor
+ manager events to use stasis (closes issue ASTERISK-21467)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2564/
+
+2013-06-07 12:56 +0000 [r390830] Kinsey Moore <kmoore@digium.com>
+
+ * main/bridging.c, main/channel.c, main/stasis_cache.c,
+ include/asterisk/stasis.h, main/stasis_channels.c,
+ main/endpoints.c, tests/test_stasis.c: Rework stasis cache clear
+ events Stasis cache clear message payloads now consist of a
+ stasis_message representative of the message to be cleared from
+ the cache. This allows multiple parallel caches to coexist and be
+ cleared properly by the same cache clear message even when keyed
+ on different fields. This change fixes a bug where multiple cache
+ clears could be posted for channels. The cache clear is now
+ produced in the destructor instead of ast_hangup. Additionally,
+ dummy channels are no longer capable of producing channel
+ snapshots. Review: https://reviewboard.asterisk.org/r/2596
+
+2013-06-07 01:06 +0000 [r390803-390804] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, channels/sig_analog.c, channels/sig_pri.c:
+ Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the
+ common transfer functions. (closes issue ASTERISK-21523) Reported
+ by: Matt Jordan (closes issue ASTERISK-21524) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2600/
+
+ * main/features_config.c: Tweak applicationmap and featuregroup
+ config containers. * Change applicationmap and featuregroup to
+ replace duplicate config items rather than reject them. * Remove
+ some unneeded warning messages when getting the applicationmap
+ allows duplicates from DYNAMIC_FEATURES.
+
+2013-06-06 23:32 +0000 [r390787] Mark Michelson <mmichelson@digium.com>
+
+ * main/features_config.c: Conditionally reject duplicate entries in
+ applicationmap containers. When reading from a config file, it's
+ important to reject duplicates. Otherwise, featuregroups will
+ have ambiguity when pointing to applicationmap items. However,
+ when constructing the channel's current applicationmap, we don't
+ care about duplicate names since it's the DTMF that identifies a
+ feature, not the name.
+
+2013-06-06 22:46 +0000 [r390771] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, configs/skinny.conf.sample,
+ CHANGES, main/bridging.c, configs/iax.conf.sample,
+ configs/chan_dahdi.conf.sample,
+ bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h,
+ include/asterisk/bridging.h, main/features.c: Reimplement
+ bridging and DTMF features related channel variables in the
+ bridging core. * The channel variable
+ ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel driver
+ specific. If the channel variable is set on the transferrer
+ channel, the sound will be played to the target of an attended
+ transfer. * The channel variable BRIDGEPEER becomes a comma
+ separated list of peers in a multi-party bridge. The BRIDGEPEER
+ value can have a maximum of 10 peers listed. Any more peers in
+ the bridge will not be included in the list. BRIDGEPEER is not
+ valid in holding bridges like parking since those channels do not
+ talk to each other even though they are in a bridge. * The
+ channel variable BRIDGEPVTCALLID is only valid for two party
+ bridges and will contain a value if the BRIDGEPEER's channel
+ driver supports it. * The channel variable DYNAMIC_PEERNAME is
+ redundant with BRIDGEPEER and is removed. The more useful
+ DYNAMIC_WHO_ACTIVATED gives the channel name that activated the
+ dynamic feature. * The channel variables DYNAMIC_FEATURENAME and
+ DYNAMIC_WHO_ACTIVATED are set only on the channel executing the
+ dynamic feature. Executing a dynamic feature on the bridge peer
+ in a multi-party bridge will execute it on all peers of the
+ activating channel. (closes issue ASTERISK-21555) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/2582/
+
+2013-06-06 21:40 +0000 [r390751] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/channel.h, main/features_config.c (added),
+ include/asterisk/features.h, channels/chan_dahdi.c,
+ channels/chan_misdn.c, channels/sig_analog.c, main/manager.c,
+ bridges/bridge_builtin_features.c, main/features.c,
+ channels/sip/include/sip.h, main/bridging.c,
+ channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
+ channels/chan_sip.c, include/asterisk/features_config.h (added):
+ Refactor the features configuration scheme. Features
+ configuration is handled in its own API in features_config.h and
+ features_config.c. This way, features configuration is accessible
+ to anything that needs it. In addition, features configuration
+ has been altered to be more channel-oriented. Most callers of
+ features API code will be supplying a channel so that the
+ individual channel's settings will be acquired rather than the
+ global setting. Missing from this commit is XML documentation for
+ the features configuration. That will be handled in a separate
+ commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue
+ ASTERISK-21542)
+
+2013-06-06 20:50 +0000 [r390733-390734] Richard Mudgett <rmudgett@digium.com>
+
+ * main/stasis_message_router.c: Fix compiler warning.
+
+ * main/bridging.c, main/features.c, apps/app_bridgewait.c: * Fix a
+ couple missed hook installs that need
+ AST_BRIDGE_HOOK_REMOVE_ON_PULL. * Rename some hook flag
+ parameters to remove_flags.
+
+2013-06-06 20:37 +0000 [r390730] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_agi.c: Fix documentation generation Regression from
+ r390701
+
+2013-06-06 20:32 +0000 [r390729] Jason Parker <jparker@digium.com>
+
+ * /: Remove props that people will yell at me for. I'm sorry I
+ broke automerge. :(
+
+2013-06-06 20:30 +0000 [r390728] Kinsey Moore <kmoore@digium.com>
+
+ * res/parking/parking_manager.c: Fix documentation that was in
+ review during the great suffix/prefix swap
+
+2013-06-06 19:51 +0000 [r390698-390701] Jason Parker <jparker@digium.com>
+
+ * CHANGES, /, res/res_agi.c: Split AGI manager events, to remove
+ SubEvent field. This moves them to stasis, in the process.
+ (closes issue ASTERISK-21470) Review:
+ https://reviewboard.asterisk.org/r/2587/
+
+ * main/stasis_message_router.c,
+ include/asterisk/stasis_message_router.h: Convert message_router
+ routes to ao2. Add support for removal. Review:
+ https://reviewboard.asterisk.org/r/2591/
+
+2013-06-06 18:21 +0000 [r390669] Jonathan Rose <jrose@digium.com>
+
+ * main/bridging.c: Parking: Enable code responsible for
+ intercepting park exten transfers
+
+2013-06-06 01:52 +0000 [r390612-390639] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add a BUGBUG note.
+
+ * main/bridging.c: Misc core external attended transfer fixes. *
+ Fix external attended transfer bridge move/swap method. One of
+ the transferrer channels was not kicked out of the bridge. * Fix
+ several off-nominal extended attended transfer paths. Mainly the
+ channels involved needed to be hung up or kicked out of the
+ bridge.
+
+ * main/core_local.c: Make local channels use ast_channel_move()
+ instead of the inlined version.
+
+2013-06-05 21:14 +0000 [r390584-390585] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/stasis.h: Corrected comment on stasis_cache_get
+
+ * main/manager_channels.c: Fixed refcounting problems with chanspy
+ AMI support. The ast_multi_channel_blob_get_channel function does
+ not bump the refcount on the channel snapshot that it returns.
+ This is typical for Stasis message payloads, since being
+ immutable means that the object won't get unreffed out from
+ underneath you. The manager code for chanspy was unreffing the
+ snapshots it got out of the multi-channel blob, which was one
+ unref too many.
+
+2013-06-05 19:19 +0000 [r390510-390550] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridging.c, res/parking/parking_bridge_features.c,
+ main/bridging_basic.c, include/asterisk/bridging_features.h,
+ main/features.c, bridges/bridge_builtin_interval_features.c:
+ Remove remaining traces of remove_on_pull from hooks and hook
+ APIs.
+
+ * include/asterisk/bridging_features.h: Give the
+ AST_BRIDGE_HOOK_REMOVE_ON_PULL a legitimate value.
+
+ * main/bridging.c, include/asterisk/bridging_features.h: Change the
+ remove_on_pull flag on ast_bridge_hook to be a set of flags. This
+ change is used to make bridge hook removal more generic. This
+ way, depending on the circumstance, the appropriate bridge hooks
+ may be removed.
+
+2013-06-05 14:50 +0000 [r390473] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Publish the channel state snapshot *before*
+ calling device state so a device state producer can use an up to
+ date snapshot.
+
+2013-06-05 14:47 +0000 [r390472] David M. Lee <dlee@digium.com>
+
+ * main/channel_internal_api.c: Fixed a consistency problem with
+ channel snapshot and endpoint state. When channels are added to
+ an endpoint, the code originally posted a channel snapshot to the
+ endoint's topic directly. Turns out, this is a bad idea. This
+ causes the endpoint to see an inconsistent view of the channel,
+ since it will later receive in-flight messages with old channel
+ snapshots. This patch instead just publishes channel state
+ immediately after setting up the forward to the endpoint's topic.
+ This gives the endpoints a consistent view of the channel's
+ state.
+
+2013-06-04 22:55 +0000 [r390439-390440] Richard Mudgett <rmudgett@digium.com>
+
+ * bridges/bridge_native_rtp.c: Add BUGBUG comment.
+
+ * bridges/bridge_native_rtp.c: Simple lock, assignment, unlock
+ sandwich optimization.
+
+2013-06-04 15:55 +0000 [r390352-390398] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/manager.h: Corrected the docs on
+ ast_manager_event_blob_create
+
+ * configure, include/asterisk/autoconfig.h.in, main/Makefile,
+ configure.ac, makeopts.in: Correct autoconf script for finding
+ UUID support. The library that provides UUID support varies
+ greatly from system to system. On most Linux distros, it's in
+ libuuid. On OpenBSD, it's in libe2fs-uuid. On OS X, it is in
+ libsystem. This patch plays hide-and-seek with UUID support,
+ looking for it in the three places we know about. It also
+ corrects the Makefile so that it uses the configured library name
+ and include path. (closes issue ASTERISK-21816) Reported by: Brad
+ Latus (snuffy) Tested by: Brad Latus (snuffy)
+
+2013-05-31 19:00 +0000 [r390317] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_userevent.c, main/stasis_channels.c, main/pbx.c:
+ Refactor code and fix a reference leak Refactor some channel blob
+ publishing code to use ast_channel_publish_blob now that it is
+ available and fix a JSON reference leak that was occurring during
+ varset publishing.
+
+2013-05-31 16:15 +0000 [r390289-390291] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, main/channel.c, main/manager.c,
+ main/channel_internal_api.c: Remove ast_channel_bridge() and
+ associated code called only by it. * Added some more BUGBUG
+ notes.
+
+ * include/asterisk/stasis_channels.h,
+ bridges/bridge_builtin_features.c, include/asterisk/bridging.h,
+ main/stasis_channels.c, main/bridging.c, main/channel.c: Fixup
+ hold/unhold with attended and blind transfers. * DTMF attended
+ and blind transfers have hold/unhold behavior restored. *
+ External attended and blind transfers unhold the transfered party
+ when the transfer is initiated. * Made prohibit blind
+ transferring a bridge marked as masquerade only. (ConfBridge
+ bridges) * Made running an application or playing a file inside a
+ bridge post the hold/unhold messages if MOH is requested. Review:
+ https://reviewboard.asterisk.org/r/2574/
+
+2013-05-31 14:36 +0000 [r390268] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/manager.h, main/asterisk.c, main/manager.c:
+ Replace ast_manager_publish_message() with a more useful version.
+ It's much easier to just create a blob of the message. Convert
+ some AMI events to use it. Review:
+ https://reviewboard.asterisk.org/r/2577/
+
+2013-05-31 12:41 +0000 [r390249-390250] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c, include/asterisk/stasis_bridging.h,
+ apps/confbridge/include/confbridge.h, main/stasis_bridging.c,
+ apps/confbridge/confbridge_manager.c: Remove remnant of snapshot
+ blob JSON types Remove usage of the once-mandatory snapshot blob
+ type field, refactor confbridge stasis messages accordingly, and
+ remove ast_bridge_blob_json_type(). Review:
+ https://reviewboard.asterisk.org/r/2575/
+
+ * include/asterisk/stasis_channels.h, main/stasis_channels.c: Add
+ snapshot cache that indexes by channel name This adds a new
+ channel snapshot cache in parallel to the existing cache; the
+ difference being that it indexes the channel snapshots by channel
+ name instead of channel uniqueid. Review:
+ https://reviewboard.asterisk.org/r/2576
+
+2013-05-31 10:42 +0000 [r390230] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Multiple revisions 390228-390229
+ ........ r390228 | may | 2013-05-31 14:19:52 +0400 (Fri, 31 May
+ 2013) | 14 lines reject call attempts when gatekeeper is
+ configured but not registered (closes issue ASTERISK-21800)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+ Tested by: Dmitry Melekhov ........ Merged revisions 390181 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 390223 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ r390229
+ | may | 2013-05-31 14:34:20 +0400 (Fri, 31 May 2013) | 4 lines
+ remove unnecessary declarations (issue ASTERISK-21800) ........
+ Merged revisions 390228-390229 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-31 07:57 +0000 [r390180] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * Makefile: Let find do its own globbing. Previously a stray .c
+ file would cause xmldocs to not get built.
+
+2013-05-30 19:23 +0000 [r390122-390154] David M. Lee <dlee@digium.com>
+
+ * main/app.c: Missed a line from a bad merge in r390122
+
+ * include/asterisk/security_events.h, main/asterisk.c,
+ main/bridging.c, main/stasis_cache.c, include/asterisk.h,
+ main/security_events.c, include/asterisk/stasis.h,
+ main/devicestate.c, main/named_acl.c,
+ include/asterisk/stasis_bridging.h, main/presencestate.c,
+ main/stasis.c, main/channel.c,
+ include/asterisk/stasis_channels.h, main/stasis_bridging.c,
+ main/test.c, main/app.c, main/stasis_channels.c: Avoid
+ unnecessary cleanups during immediate shutdown This patch
+ addresses issues during immediate shutdowns, where modules are
+ not unloaded, but Asterisk atexit handlers are run. In the
+ typical case, this usually isn't a big deal. But the introduction
+ of the Stasis message bus makes it much more likely for
+ asynchronous activity to be happening off in some thread during
+ shutdown. During an immediate shutdown, Asterisk skips unloading
+ modules. But while it is processing the atexit handlers, there is
+ a window of time where some of the core message types have been
+ cleaned up, but the message bus is still running. Specifically,
+ it's still running module subscriptions that might be using the
+ core message types. If a message is received by that subscription
+ in that window, it will attempt to use a message type that has
+ been cleaned up. To solve this problem, this patch introduces
+ ast_register_cleanup(). This function operates identically to
+ ast_register_atexit(), except that cleanup calls are not invoked
+ on an immediate shutdown. All of the core message type and topic
+ cleanup was moved from atexit handlers to cleanup handlers. This
+ ensures that core type and topic cleanup only happens if the
+ modules that used them are first unloaded. This patch also
+ changes the ast_assert() when accessing a cleaned up or
+ uninitialized message type to an error log message. Message type
+ functions are actually NULL safe across the board, so the assert
+ was a bit heavy handed. Especially for anyone with DO_CRASH
+ enabled. Review: https://reviewboard.asterisk.org/r/2562/
+
+2013-05-29 20:24 +0000 [r390068] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix segfault when dealing with chan_agent
+ channels. Check the returned bridged pointer for NULL to avoid a
+ crash. It looks like chan_agent is returning a NULL pointer when
+ it probably should be returning a pointer to the channel the
+ Agent channel is pretending to be. (closes issue ASTERISK-21793)
+ Reported by: Rodrigo P. Telles Patches:
+ jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Rodrigo P. Telles ........ Merged revisions
+ 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 390047 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-29 19:54 +0000 [r390042] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Remove unused RAII vars.
+
+2013-05-29 03:22 +0000 [r389990] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c: Pack the right number of items into the status and
+ receive fax blobs The code was still attempting to pack an
+ additional item into the blobs that didn't exist. Crashes ensued.
+ This patch modifies the publishing of these messages so that the
+ correct number of items are packed in the JSON.
+
+2013-05-29 02:26 +0000 [r389974] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_monitor.c, include/asterisk/stasis_channels.h,
+ res/res_fax.c, apps/app_fax.c, main/stasis_channels.c,
+ res/res_musiconhold.c: Resolve a merge conflict When
+ ast_channel_cached_blob_create was merged,
+ ast_channel_blob_create_from_cache was partially removed in an
+ unresolved merge conflict. This restores
+ ast_channel_blob_create_from_cache and refactors usage of
+ ast_channel_cached_blob_create (requires an ast_channel) to use
+ ast_channel_blob_create_from_cache (requires a channel uniqueid)
+ instead.
+
+2013-05-28 17:47 +0000 [r389897] Jonathan Rose <jrose@digium.com>
+
+ * /, main/slinfactory.c: Fix a memory copying bug in slinfactory
+ which was causing mixmonitor issues. Reported by: Michael Walton
+ Tested by: Jonathan Rose Patches:
+ slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton
+ (license 6502) (closes issue ASTERISK-21799) ........ Merged
+ revisions 389895 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 389896 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-28 15:54 +0000 [r389848-389870] Mark Michelson <mmichelson@digium.com>
+
+ * main/bridging.c: Add missing NULL check to acquire_bridge()
+ function.
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Add attended
+ transfer support for chan_sip.c This now uses the core API for
+ performing attended transfers. Review
+ https://reviewboard.asterisk.org/r/2513 (Closes issue
+ ASTERISK-21520) reported by Matt Jordan
+
+ * include/asterisk/channel.h, CHANGES, main/bridging.c,
+ channels/chan_mgcp.c, main/channel.c, main/pbx.c,
+ bridges/bridge_builtin_features.c, channels/chan_sip.c,
+ apps/confbridge/confbridge_manager.c,
+ include/asterisk/bridging.h, main/features.c: Adds support for a
+ core attended transfer function plus adds some hiding of
+ masquerades. The attended transfer API call can complete the
+ attended transfer in a number of ways depending on the current
+ bridged states of the channels involved. The hiding of
+ masquerades is done in some bridging-related functions, such as
+ the manager Bridge action and the Bridge dialplan application. In
+ addition, call pickup was edited to "move" a channel rather than
+ masquerade it. Review: https://reviewboard.asterisk.org/r/2511
+ (closes issue ASTERISK-21334) Reported by Matt Jordan (closes
+ issue Asterisk-21336) Reported by Matt Jordan
+
+2013-05-27 01:33 +0000 [r389770-389827] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax.c, res/res_fax_spandsp.c: Fix some more fax test
+ errors due to needing the peer in a bridge In r389799, a number
+ of fax errors in gateway mode were fixed by using the appropriate
+ function to get a channel's peer while in a bridge. This patch
+ does two things: (1) It uses the same function in res_fax_spandsp
+ while starting the fax gateway. Without this, the fax gateway
+ will not actually start up, as res_fax_spandsp also must inspect
+ the channel's peer in a two-party bridge (2) It refactors some
+ ao2 objects in sendfax_exec to use RAII_VAR. This was reverted in
+ r389799 as some off nominal paths were getting hit without the
+ fix in (1) that indicated an ao2 object issue; this turned out to
+ be a red herring (which is an odd phrase)
+
+ * main/stasis_endpoints.c: Initialize the message type before the
+ topic Caching topics will during initialization attempt to
+ reference their message type. The message type therefore has to
+ be initialized prior to the topic to prevent the dreaded
+ assertion.
+
+ * res/res_fax.c: Fix a few fax gateway failures Fax gateway
+ requires knowledge of a channel's peer in a bridge. This patch
+ now uses the supported mechanisms to get this information. This
+ is acceptable for a few reasons: * Fax gateway can only ever work
+ in a 2-party bridge * Fax gateway cannot work when not in a
+ bridge * Fax gateway cannot work without knowledge of the
+ capabilities of both channels in the fax operation (it is, after
+ all, a gateway)
+
+ * main/devicestate.c, main/asterisk.c, res/res_fax.c: Fix a variety
+ of memory corruption/assertion errors * Initialize a Stasis-Core
+ message type prior to initializing a caching topic. The caching
+ topic will attempt to use the message type. * Don't attempt to
+ publish Stasis-Core messages from remote console connections.
+ They aren't the main process; they shouldn't attempt to behave as
+ it (they also don't have the infrastructure to do so) * Don't
+ treat a JSON object as an ao2 object (whoops) * In asterisk.c,
+ ref bump the JSON even package that is distributed with the event
+ meta data. The callers assume that they own the reference, and
+ the packing routine steals references.
+
+ * main/asterisk.c: Restore initialization of security topics During
+ a merge the security topic initialization got blown away. This
+ patch restores it.
+
+2013-05-24 21:23 +0000 [r389746-389748] Jason Parker <jparker@digium.com>
+
+ * /: grr, props.
+
+ * main/manager_channels.c, channels/chan_mgcp.c,
+ channels/chan_unistim.c, /, channels/chan_sip.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ channels/chan_iax2.c, CHANGES, res/res_sip_sdp_rtp.c,
+ main/channel.c, channels/chan_dahdi.c,
+ include/asterisk/stasis_channels.h, channels/sig_analog.c,
+ channels/chan_misdn.c, channels/chan_skinny.c,
+ channels/chan_motif.c, channels/chan_h323.c,
+ main/stasis_channels.c: Split Hold event into Hold/Unhold, and
+ move it into core. (closes issue ASTERISK-21487) Review:
+ https://reviewboard.asterisk.org/r/2565/
+
+2013-05-24 21:01 +0000 [r389738] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_stasis.c: Remove a junk define BLOB_HANDLER_BUCKETS is a
+ remnant of using "type" fields in JSON/snapshot blobs and is no
+ longer used.
+
+2013-05-24 20:44 +0000 [r389680-389733] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/_private.h, res/res_xmpp.c, CHANGES,
+ channels/chan_iax2.c, res/res_jabber.c, res/res_monitor.c,
+ main/cli.c, main/cdr.c, main/json.c, main/manager.c,
+ channels/chan_skinny.c, main/app.c, main/stasis_channels.c,
+ res/parking/parking_manager.c, main/asterisk.c,
+ channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c,
+ apps/app_fax.c, include/asterisk/json.h, res/res_musiconhold.c,
+ include/asterisk/manager.h, channels/sig_pri.c, main/enum.c,
+ main/loader.c, include/asterisk/app.h, channels/chan_dahdi.c,
+ include/asterisk/stasis_channels.h, apps/app_minivm.c,
+ apps/app_chanspy.c, main/manager_channels.c, res/res_sip_mwi.c,
+ main/manager_mwi.c (added), apps/app_voicemail.c, main/dnsmgr.c,
+ channels/chan_sip.c, res/res_fax.c: Migrate a large number of AMI
+ events over to Stasis-Core This patch moves a number of AMI
+ events over to the Stasis-Core message bus. This includes: *
+ ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop *
+ FullyBooted/Reload * All Voicemail/MWI related events In
+ addition, it adds some Stasis-Core and AMI support for generic
+ AMI messages, refactors the message router in AMI to use a single
+ router with topic forwarding for the topics that AMI cares about,
+ and refactors MWI message types and topics to be more name
+ compliant. Review: https://reviewboard.asterisk.org/r/2532
+ (closes issue ASTERISK-21462)
+
+ * /, main/logger.c: Print all logger messages on shutdown When
+ Asterisk shuts down and shuts down the loggin gsubsystem, any
+ messages currently in flight will not get logged. This patch
+ prevents the loop writing messages from breaking out prematurely,
+ such that all of the messages are logged. (closes issue
+ ASTERISK-21716) Reported by: Corey Farrell patches:
+ logger-process-all-messages.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 389676 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 389677 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-24 10:23 +0000 [r389663] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix several problems caused by
+ multiple line usage with i2004 phones. Reported by: Daniel
+ Bohling, MihaiMircea (closes issue ASTERISK-21061) (closes issue
+ ASTERISK-21120) ........ Merged revisions 389661 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-23 21:46 +0000 [r389639] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_playback.c, res/stasis_http/resource_channels.c,
+ include/asterisk/stasis_http.h, res/res_stasis_http.c:
+ stasis-http: Provide a response body for 201 created responses
+
+2013-05-23 21:11 +0000 [r389618-389623] Jonathan Rose <jrose@digium.com>
+
+ * res/parking/parking_bridge.c: res_parking: Add a verbose message
+ when a channel is parked
+
+ * res/parking/parking_bridge.c: res_parking: Fix some simple bugs
+ Both of them are covered in the dynamic parking review on
+ https://reviewboard.asterisk.org/r/2550 - Remove unref against
+ parking lot that the bridge did on dissolve since the reference
+ wasn't taken in the first place. On a swap, reapply bridge roles
+ in order to get music on hold and such playing on the channel
+ that swaps into the bridge.
+
+2013-05-23 20:25 +0000 [r389609] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_session.c: Fix a crash due to the INVITE session
+ being destroyed before the session. This change ensures that the
+ INVITE session remains valid for the lifetime of the session
+ object itself by increasing the session count on the dialog that
+ the INVITE session is allocated from. Once this reaches zero
+ (normally as a result of decrementing it within the session
+ destructor) the dialog, and INVITE session, are destroyed.
+
+2013-05-23 20:21 +0000 [r389587-389603] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/channel.h, res/stasis_http/resource_channels.h,
+ rest-api/api-docs/channels.json,
+ include/asterisk/stasis_app_playback.h,
+ res/stasis_http/resource_playback.c, include/asterisk/app.h,
+ res/res_stasis_playback.c, res/stasis/control.c,
+ res/stasis_http/resource_channels.c,
+ rest-api/api-docs/playback.json, res/res_stasis_http_channels.c,
+ include/asterisk/stasis_app.h, main/app.c: This patch adds
+ support for controlling a playback operation from the Asterisk
+ REST interface. This adds the /playback/{playbackId}/control
+ resource, which may be POSTed to to pause, unpause, reverse,
+ forward or restart the media playback. Attempts to control a
+ playback that is not currently playing will either return a 404
+ Not Found (because the playback object no longer exists) or a 409
+ Conflict (because the playback object is still in the queue to be
+ played). This patch also adds skipms and offsetms parameters to
+ the /channels/{channelId}/play resource. (closes issue
+ ASTERISK-21587) Review: https://reviewboard.asterisk.org/r/2559
+
+ * res/stasis_http/resource_channels.h, main/stasis_channels.c,
+ rest-api/api-docs/channels.json,
+ res/res_stasis_playback.exports.in (added),
+ res/res_stasis_http.c, res/stasis_json/resource_events.h,
+ res/res_stasis_json_events.exports.in, res/res_stasis_playback.c
+ (added), rest-api/api-docs/events.json, res/stasis/control.c,
+ main/channel_internal_api.c, include/asterisk/stasis_http.h,
+ res/res_stasis_http_channels.c, res/res_stasis_json_events.c,
+ include/asterisk/stasis_app_playback.h (added),
+ res/stasis_http/resource_playback.c, include/asterisk/app.h,
+ include/asterisk/stasis_channels.h,
+ res/stasis_json/resource_channels.h,
+ res/stasis_http/resource_channels.c: This patch implements the
+ REST API's for POST /channels/{channelId}/play and GET
+ /playback/{playbackId}. This allows an external application to
+ initiate playback of a sound on a channel while the channel is in
+ the Stasis application. /play commands are issued asynchronously,
+ and return immediately with the URL of the associated /playback
+ resource. Playback commands queue up, playing in succession. The
+ /playback resource shows the state of a playback operation as
+ enqueued, playing or complete. (Although the operation will only
+ be in the 'complete' state for a very short time, since it is
+ almost immediately freed up). (closes issue ASTERISK-21283)
+ (closes issue ASTERISK-21586) Review:
+ https://reviewboard.asterisk.org/r/2531/
+
+2013-05-23 18:40 +0000 [r389569] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix inverted test preventing DTMF disconnect
+ from working.
+
+2013-05-23 18:39 +0000 [r389551-389568] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_sdp_rtp.c: Fix a bug where the DTMF mode was not set
+ on newly created RTP instances in the res_sip_sdp_rtp module.
+
+ * res/res_sip_sdp_rtp.c: Fix a bug with applying the end result of
+ the codec negotiation to the Asterisk channel.
+
+ * res/res_sip_session.c: Fix a bug where the codec order as
+ configured was not being obeyed.
+
+2013-05-22 19:15 +0000 [r389519] David M. Lee <dlee@digium.com>
+
+ * main/app.c: Fixed startup race condition which caused occasional
+ stasis_mwi_state_type assertions. The caching topic (which refers
+ to the message type) was created before the message type. If the
+ initial subscription message gets processed before the type can
+ be initialized, the assertion about using an uninitialized type
+ fires.
+
+2013-05-22 18:20 +0000 [r389492-389505] Jason Parker <jparker@digium.com>
+
+ * /: Remove bad props, before anybody notices.
+
+ * include/asterisk/dial.h, apps/app_followme.c, apps/app_queue.c,
+ apps/app_dial.c, main/dial.c, /: Add dial events to app_queue and
+ app_followme. Also fixes an issue in app_dial, where the channels
+ were swapped on dial events. (closes issue ASTERISK-21551)
+ (closes issue ASTERISK-21550) Review:
+ https://reviewboard.asterisk.org/r/2549/
+
+2013-05-21 22:49 +0000 [r389454] David M. Lee <dlee@digium.com>
+
+ * main/stasis_bridging.c: Fix destruction order assert for
+ stasis_bridging
+
+2013-05-21 21:08 +0000 [r389426] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Conditional out more app_queue logging that
+ needs to be reworked. Fixes crash because app_queue was
+ unconditionally freeing a datastore that was still on a channel.
+
+2013-05-21 18:45 +0000 [r389402] Matthew Jordan <mjordan@digium.com>
+
+ * apps/confbridge/confbridge_manager.c, apps/app_confbridge.c:
+ Raise the ConfBridgeMute/Unmute events when a CLI or AMI action
+ triggers the change New in 12 are the ConfBridgeMute/Unmute
+ events, which are triggered when a user changes their mute/unmute
+ state. This was typically triggered when a user hit a DTMF key
+ that triggered the mute/unmute menu handler. Forgotten in this is
+ when an AMI action or CLI command triggers the mute/unmute. This
+ patch now raises the events in those situations as well. (closes
+ issue ASTERISK-21802) Reported by: Birger "WIMPy" Harzenetter
+
+2013-05-21 18:00 +0000 [r389378] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_parkandannounce.c (removed), main/bridging_basic.c
+ (added), channels/chan_mgcp.c,
+ res/res_stasis_json_events.exports.in, channels/chan_sip.c,
+ main/channel_internal_api.c, apps/confbridge/conf_chan_record.c
+ (added), main/core_unreal.c (added),
+ res/parking/parking_controller.c, UPGRADE.txt,
+ funcs/func_jitterbuffer.c, include/asterisk/_private.h,
+ main/core_local.c (added), apps/app_queue.c,
+ include/asterisk/framehook.h,
+ res/parking/parking_bridge_features.c, channels/chan_jingle.c,
+ res/res_parking.c (added), apps/app_channelredirect.c,
+ apps/confbridge/confbridge_manager.c (added),
+ include/asterisk/bridging.h, main/abstract_jb.c,
+ channels/chan_h323.c, main/config_options.c,
+ bridges/bridge_native_rtp.c (added),
+ res/stasis_json/resource_events.h,
+ apps/confbridge/conf_chan_announce.c (added),
+ res/parking/parking_manager.c, addons/chan_ooh323.c,
+ main/frame.c, channels/chan_local.c (removed), main/rtp_engine.c,
+ bridges/bridge_holding.c (added), configs/res_parking.conf.sample
+ (added), bridges/bridge_softmix.c, apps/app_bridgewait.c (added),
+ res/Makefile, res/res_stasis_json_events.c, channels/chan_iax2.c,
+ bridges/bridge_multiplexed.c (removed), channels/chan_gulp.c,
+ apps/confbridge/conf_config_parser.c, main/cli.c,
+ channels/chan_dahdi.c, include/asterisk/bridging_technology.h,
+ channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
+ include/asterisk/bridging_roles.h (added),
+ rest-api-templates/res_stasis_json_resource.c.mustache,
+ include/asterisk/parking.h (added), include/asterisk/ccss.h,
+ main/manager_channels.c, main/bridging.c,
+ res/parking/parking_ui.c, apps/app_dial.c, main/pbx.c,
+ main/strings.c, channels/chan_bridge.c (removed),
+ channels/chan_agent.c, bridges/bridge_simple.c,
+ res/parking/parking_applications.c, include/asterisk/channel.h,
+ CHANGES, include/asterisk/manager.h,
+ include/asterisk/stasis_bridging.h (added),
+ configs/features.conf.sample, res/parking/res_parking.h,
+ channels/chan_misdn.c, funcs/func_channel.c, res/parking (added),
+ include/asterisk/bridging_basic.h (added), apps/app_mixmonitor.c,
+ apps/app_chanspy.c, main/asterisk.c,
+ include/asterisk/core_unreal.h (added), channels/chan_unistim.c,
+ main/bridging_roles.c (added), rest-api/api-docs/events.json,
+ channels/chan_vpb.cc, main/parking.c (added),
+ include/asterisk/core_local.h (added), apps/app_followme.c,
+ res/parking/parking_bridge.c, include/asterisk/abstract_jb.h,
+ main/channel.c, include/asterisk/config_options.h,
+ main/manager.c, main/stasis_bridging.c (added),
+ bridges/bridge_builtin_features.c,
+ apps/confbridge/include/confbridge.h,
+ include/asterisk/bridging_features.h, funcs/func_frame_trace.c,
+ channels/chan_motif.c, bridges/bridge_builtin_interval_features.c
+ (added), main/manager_bridging.c (added),
+ rest-api-templates/stasis_json_resource.h.mustache,
+ apps/app_confbridge.c, include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h: Merge in the bridge_construction branch
+ to make the system use the Bridging API. Breaks many things until
+ they can be reworked. A partial list: chan_agent chan_dahdi,
+ chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF
+ attended transfers Protocol attended transfers
+
+2013-05-21 14:17 +0000 [r389343] David M. Lee <dlee@digium.com>
+
+ * apps/app_userevent.c, main/stasis_channels.c: Fixed some extra
+ field assertion when the event WebSocket is connected
+
+2013-05-20 19:24 +0000 [r389306] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c: Set the AST_CDR_FLAG_ORIGINATED flag on originated
+ channel's CDRs This may alleviate some of the CDR woes with
+ originated channels, as CDRs do like to know when a channel was
+ originated. Eventually this will get converted to be a channel
+ flag, so its location is still good to know post the great CDR
+ shakeup of 2013.
+
+2013-05-20 18:03 +0000 [r389247-389251] Richard Mudgett <rmudgett@digium.com>
+
+ * res/ael/ael.tab.h, formats/format_h264.c,
+ tests/test_stasis_http.c, res/res_curl.c, tests/test_json.c,
+ utils/refcounter.c, tests/test_dlinklists.c, pbx/pbx_lua.c,
+ res/stasis_http/resource_bridges.c, main/cel.c,
+ res/stasis_http/resource_bridges.h,
+ res/res_stasis_http_endpoints.c,
+ res/stasis_http/resource_asterisk.c, funcs/func_realtime.c,
+ main/json.c, main/stasis_message_router.c,
+ res/stasis_http/resource_asterisk.h, funcs/func_channel.c,
+ utils/ael_main.c, codecs/codec_dahdi.c, funcs/func_iconv.c,
+ res/res_stasis_http.c, tests/test_stasis.c, res/res_stasis.c,
+ res/res_stasis_http_asterisk.c, cel/cel_pgsql.c,
+ cel/cel_radius.c, funcs/func_rand.c, res/res_stasis_websocket.c,
+ tests/test_gosub.c, tests/test_res_stasis.c, res/ael/pval.c,
+ tests/test_time.c, cel/cel_custom.c,
+ res/stasis_http/resource_channels.c,
+ res/stasis_http/resource_recordings.c,
+ res/stasis_http/resource_channels.h,
+ res/stasis_http/resource_endpoints.c,
+ res/stasis_http/resource_recordings.h,
+ res/stasis_http/resource_events.c, main/manager_channels.c,
+ res/res_clioriginate.c, res/stasis_http/resource_endpoints.h,
+ main/stasis_cache.c, funcs/func_version.c,
+ tests/test_astobj2_thrash.c, res/stasis_http/resource_events.h,
+ cel/cel_tds.c, tests/test_xml_escape.c,
+ res/res_stasis_http_channels.c, res/res_ael_share.c,
+ res/res_stasis_http_recordings.c, res/res_srtp.c,
+ res/res_stasis_http_events.c, main/hashtab.c, res/ael/ael_lex.c,
+ cel/cel_sqlite3_custom.c, main/event.c,
+ res/res_stasis_http_bridges.c, contrib/utils/eagi_proxy.c,
+ main/udptl.c, funcs/func_dialgroup.c, main/sha1.c,
+ main/threadstorage.c, tests/test_hashtab_thrash.c,
+ res/res_pktccops.c, main/stasis.c, cel/cel_odbc.c,
+ main/stasis_message.c, res/res_smdi.c, res/ael/ael.tab.c,
+ cel/cel_manager.c, funcs/func_odbc.c: Fixup svn:keywords in all
+ *.c and *.h files.
+
+ * include/asterisk/stringfields.h, channels/sip/sdp_crypto.c,
+ channels/sip/include/dialog.h, include/asterisk/res_srtp.h,
+ channels/sip/srtp.c, include/asterisk/cel.h,
+ include/asterisk/stasis_http.h, include/asterisk/stasis_app.h,
+ include/asterisk/stasis.h, apps/app_morsecode.c,
+ apps/app_waituntil.c, include/asterisk/json.h,
+ include/asterisk/stasis_message_router.h,
+ include/asterisk/hashtab.h,
+ channels/sip/include/dialplan_functions.h,
+ include/asterisk/paths.h, include/asterisk/event.h,
+ apps/app_setcallerid.c, include/asterisk/event_defs.h,
+ channels/sip/include/globals.h, apps/app_celgenuserevent.c,
+ channels/sip/dialplan_functions.c, include/asterisk/pktccops.h,
+ channels/sip/include/sdp_crypto.h,
+ include/asterisk/ael_structs.h, include/asterisk/udptl.h,
+ channels/sip/include/srtp.h, include/asterisk/frame_defs.h,
+ apps/app_stasis.c, include/asterisk/sha1.h,
+ include/asterisk/smdi.h: Fixup svn:keywords in all *.c and *.h
+ files.
+
+2013-05-20 17:44 +0000 [r389246] Jason Parker <jparker@digium.com>
+
+ * /: Add doxygen.log to svn:ignore property. ........ Merged
+ revisions 389244 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 389245 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-20 14:21 +0000 [r389217] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_stasis_answer.exports.in (added): Add missing exports
+ file This exposes stasis_app_control_answer and allows
+ res_stasis_http_channels to load properly.
+
+2013-05-20 14:02 +0000 [r389204] Joshua Colp <jcolp@digium.com>
+
+ * main/sorcery.c: In Sorcery pass the name of the object being
+ allocated to the allocator.
+
+2013-05-20 13:45 +0000 [r389202] Kinsey Moore <kmoore@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Add documentation for
+ record_file_append When this option was added, it was noted in
+ CHANGES, but was missing the XML documentation that this patch
+ adds. (closes issue ASTERISK-21780) Patch-by: Brad Latus (snuffy)
+
+2013-05-19 20:52 +0000 [r389180] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.h, addons/chan_ooh323.c: add
+ ast_publish_channel_state according new event framework
+
+2013-05-19 19:45 +0000 [r389164] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add transfer softkey to ringout state to
+ enable blond transfers. (closes issue ASTERISK-21327) Reported
+ by: wedhorn Tested by: myself Patches: skinny-blindxfer01.diff
+ uploaded by wedhorn (license 5019)
+
+2013-05-19 17:45 +0000 [r389148] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_sip_outbound_registration.c,
+ res/res_sip_endpoint_identifier_ip.c, res/res_sip_acl.c,
+ res/res_sip.c: Add base XML documentation for res_sip Thanks to
+ Brad Latus, this patch adds a significant amount much-needed
+ documentation to res_sip. It should cover all existing
+ configuration options currently in Asterisk trunk. Patch-by: Brad
+ Latus (snuffy) Review: https://reviewboard.asterisk.org/r/2471/
+
+2013-05-19 02:21 +0000 [r389116-389132] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Don't hold the outgoing lock for a prolonged period
+ of time as it may block the originator.
+
+ * main/pbx.c: If the caller of the originate API calls wants the
+ channel ensure it has been requested and dialed.
+
+2013-05-18 23:20 +0000 [r389097] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Add call
+ forward no answer to skinny and cleanup general callfwd handling.
+ CallforwardNoAnswer uses a sched to determine when to forward the
+ call. Defaults to 20secs but configurable in skinny.conf. Adds
+ dialType to each subchannel structure to be used to differentiate
+ between normal dials that result in a call being placed (default)
+ and other uses for the skinny_dialer (such as cfwd digit
+ collection). Restructured all cfwd handling to use this new
+ arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn
+ Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by
+ wedhorn (license 5019)
+
+2013-05-18 22:49 +0000 [r389053-389085] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Fix a bug where synchronous origination (oddly enough
+ triggered by doing an async manager Originate) would not work
+ properly.
+
+ * include/asterisk/dial.h, main/manager_channels.c, main/dial.c,
+ main/pbx.c: Move origination to use the dialing API and send
+ Stasis messages on dial begin and end. (closes issue
+ ASTERISK-21549) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2512/
+
+2013-05-17 21:10 +0000 [r389011] David M. Lee <dlee@digium.com>
+
+ * res/res_chan_stats.c, main/stasis.c, main/manager.c,
+ funcs/func_presencestate.c, main/stasis_message_router.c,
+ main/app.c, main/stasis_channels.c, res/res_stasis.c,
+ main/manager_channels.c, apps/app_voicemail.c,
+ main/stasis_cache.c, main/pbx.c, main/stasis_endpoints.c,
+ channels/chan_sip.c, include/asterisk/stasis.h,
+ main/devicestate.c, res/res_jabber.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/endpoints.c,
+ include/asterisk/stasis_message_router.h: Fix shutdown assertions
+ in stasis-core In r388005, macros were introduced to consistently
+ define message types. This added an assert if a message type was
+ used either before it was initialized or after it had been
+ cleaned up. It turns out that this assertion fires during
+ shutdown. This actually exposed a hidden shutdown ordering
+ problem. Since unsubscribing is asynchronous, it's possible that
+ the message types used by the subscription could be freed before
+ the final message of the subscription was processed. This patch
+ adds stasis_subscription_join(), which blocks until the last
+ message has been processed by the subscription. Since joining was
+ most commonly done right after an unsubscribe, a
+ stasis_unsubscribe_and_join() convenience function was also
+ added. Similar functions were also added to the
+ stasis_caching_topic and stasis_message_router, since they wrap
+ subscriptions and have similar problems. Other code in trunk was
+ refactored to join() where appropriate, or at least verify that
+ the subscription was complete before being destroyed. Review:
+ https://reviewboard.asterisk.org/r/2540
+
+2013-05-17 20:24 +0000 [r389009] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_iax2.c: Remove Character Limit On "inkeys" For IAX2
+ Currently, the buffer for processing "inkeys" is limited to 256
+ characters. If the user has many keys and the names of those key
+ files are long, the 256 character limit is not enough. * Change
+ inkeys buffer to be dynamic (closes issue ASTERISK-21398)
+ Reported by: Pavel Kopchyk Tested by: Pavel Kopchyk, Michael L.
+ Young Patches: asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
+ by Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2501/
+
+2013-05-17 17:43 +0000 [r388976] Matthew Jordan <mjordan@digium.com>
+
+ * main/stasis_channels.c, apps/app_dial.c, main/channel.c,
+ main/dial.c, include/asterisk/stasis_channels.h: Publish the
+ outbound channel's application/data when dialing This patch does
+ two things: * It fixes a bug where the outbound channel's
+ application/data set by the dialing API/app_dial is not
+ communicated until the channel is hung up. If that happens, AMI
+ would incorrectly send a NewExten event immediately after a
+ Hangup. This isn't really AMI's fault, as the dialing APIs never
+ communicated the 'helpful' app/data on the outbound channel until
+ it was hungup. * It makes public sending a stasis message about a
+ change in channel state. This is useful enough that - for now at
+ least - it should be public. If operations on a channel go to
+ being more coarse-grained, this function could be made private
+ again. Review: https://reviewboard.asterisk.org/r/2548 Note that
+ this problem was found and reported by Matt DiMeo.
+
+2013-05-17 17:36 +0000 [r388975] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/json.h, main/named_acl.c, CHANGES,
+ channels/chan_iax2.c, tests/test_security_events.c,
+ res/res_sip.c, main/json.c, main/manager.c,
+ channels/sip/include/config_parser.h, res/res_sip_nat.c,
+ channels/sip/dialplan_functions.c, include/asterisk/netsock2.h,
+ res/res_sip_outbound_registration.c,
+ channels/sip/config_parser.c, include/asterisk/security_events.h,
+ channels/sip/include/sip.h,
+ include/asterisk/security_events_defs.h, main/asterisk.c,
+ res/res_security_log.c, include/asterisk/acl.h,
+ res/res_sip/config_transport.c, channels/chan_sip.c,
+ main/security_events.c, channels/sip/security_events.c,
+ include/asterisk/res_sip.h: Stasis: Update security events to use
+ Stasis Also moves ACL messages to the security topic and gets rid
+ of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2496/
+
+2013-05-15 21:13 +0000 [r388896] David M. Lee <dlee@digium.com>
+
+ * res/stasis/app.c, res/stasis/app.h: Fixed inverted logic in
+ app_add_channel(). Also added some missing doc comments for
+ stasis/app.h.
+
+2013-05-15 15:58 +0000 [r388840] Kevin Harwell <kharwell@digium.com>
+
+ * /, main/lock.c: Fix for segfault in __ast_rwlock_destroy with
+ DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
+ causes a segfault while trying to access a possible NULL t->track
+ object. A NULL check has been added before trying to access the
+ memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
+ Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 388838 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388839 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-15 15:03 +0000 [r388818] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c, /: Fix VM snapshot handling for combined
+ INBOX. The snapshot API contains an option that allow for
+ combining of new and old messages within a single snapshot. New
+ messages, however, include options beyond just 'INBOX' - it also
+ includes the Urgent folder. A previous patch that combined INBOX
+ and Urgent accidentally impacted snapshots that attempted to gain
+ messages from just the Old folder. This patch fixes the snapshot
+ gathering such that the API returns the appropriate messages for
+ the folder selected, with and without the combine option. This
+ should make it more clear about what's happening. Review:
+ https://reviewboard.asterisk.org/r/2539/ ........ Merged
+ revisions 388816 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-15 12:42 +0000 [r388770] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_srtp.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Use srtp_shutdown when available This allows the
+ SRTP library to be shut down properly when the functionality is
+ offered by libsrtp. Review:
+ https://reviewboard.asterisk.org/r/2538/ (closes issue
+ ASTERISK-21719) ........ Merged revisions 388768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388769 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-15 02:37 +0000 [r388729-388751] David M. Lee <dlee@digium.com>
+
+ * main/app.c, main/devicestate.c, main/named_acl.c,
+ res/res_stasis_test.c, main/asterisk.c, main/presencestate.c,
+ main/stasis.c, main/stasis_cache.c, main/stasis_endpoints.c,
+ include/asterisk/stasis.h, main/test.c: Refactored the rest of
+ the message types to use the STASIS_MESSAGE_TYPE_* macros.
+
+ * include/asterisk/module.h, include/asterisk/stasis_app.h,
+ include/asterisk/stasis_app_impl.h (added), res/Makefile,
+ res/res_stasis_answer.c (added), res/res_stasis.c,
+ apps/app_stasis.c, res/stasis (added): Break res_stasis into
+ smaller files. When implementing playback for stasis-http, the
+ monolithicedness of res_stasis really started to get in my way.
+ This patch breaks the major components of res_stasis.c into
+ individual files. * res/stasis/app.c - Stasis application
+ tracking * res/stasis/control.c - Channel control objects *
+ res/stasis/command.c - Channel command object This refactoring
+ also allows res_stasis applications to be loaded as independent
+ modules, such as the new res_stasis_answer module. The bulk of
+ this patch is simply moving code from one file to another,
+ adjusting names and adding accessors as necessary. Review:
+ https://reviewboard.asterisk.org/r/2530/
+
+2013-05-14 19:03 +0000 [r388701] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c, /: Make ao2 global
+ objects not always use the debug version of the ao2_ref() calls.
+ The debug versions of ao2_ref() should only be used if REF_DEBUG
+ is enabled so nothing is written to /tmp/refs unexpectedly.
+ (closes issue ASTERISK-21785) Reported by: abelbeck Patches:
+ jira_asterisk_21785_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: abelbeck ........ Merged revisions 388700
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-14 12:47 +0000 [r388668] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_stasis_json_playback.exports.in (added),
+ res/stasis_http/resource_playback.h,
+ res/res_stasis_json_bridges.c (added),
+ rest-api-templates/stasis_json_resource.h.mustache (added),
+ res/stasis_http/resource_channels.h,
+ res/res_stasis_json_asterisk.c (added),
+ rest-api-templates/res_stasis_json_resource.c.mustache (added),
+ res/stasis_json (added), res/stasis_http/resource_recordings.h,
+ res/res_stasis_json_events.exports.in (added),
+ res/stasis_http/resource_endpoints.h,
+ res/stasis_http/resource_events.h,
+ res/stasis_http/resource_sounds.h,
+ rest-api-templates/res_stasis_json_resource.exports.mustache
+ (added), res/stasis_http/resource_bridges.h,
+ res/stasis_json/resource_playback.h (added),
+ res/res_stasis_http_events.c,
+ res/stasis_http/resource_asterisk.h,
+ res/stasis_json/resource_channels.h (added),
+ res/res_stasis_json_channels.exports.in (added),
+ res/stasis_json/resource_recordings.h (added),
+ res/stasis_json/resource_endpoints.h (added),
+ res/res_stasis_json_playback.c (added),
+ res/res_stasis_json_recordings.exports.in (added),
+ res/stasis_json/resource_events.h (added), res/res_stasis.c,
+ rest-api-templates/make_stasis_http_stubs.py,
+ rest-api-templates/stasis_http_resource.h.mustache,
+ res/res_stasis_json_endpoints.exports.in (added),
+ res/res_stasis_json_channels.c (added),
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ res/stasis_json/resource_sounds.h (added),
+ tests/test_res_stasis.c, res/res_stasis_json_recordings.c
+ (added), res/res_stasis_json_sounds.exports.in (added),
+ res/stasis_json/resource_bridges.h (added),
+ res/res_stasis_json_endpoints.c (added),
+ res/res_stasis_json_events.c (added),
+ res/res_stasis_json_bridges.exports.in (added),
+ res/stasis_json/resource_asterisk.h (added),
+ res/res_stasis_json_sounds.c (added),
+ res/res_stasis_json_asterisk.exports.in (added): Move JSON event
+ generators into separate modules This moves the JSON event
+ generators out of the Stasis-HTTP modules and into standalone
+ JSON-related counterparts so that Stasis-HTTP and res_stasis can
+ depend on them without creating dependency cycles. This also
+ provides a future location for Swagger Model validator functions
+ once the generators for that code are written. Review:
+ https://reviewboard.asterisk.org/r/2534/
+
+2013-05-13 21:21 +0000 [r388602-388617] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/logger.c, /: Fix Missing CALL-ID When Logging Through Syslog
+ The CALL-ID (ie [C-00000074]) is missing when logging to syslog.
+ This was just an oversight when this feature was added. * Add
+ CALL-IDs when using syslog (closes issue ASTERISK-21430) Reported
+ by: Nikola Ciprich Tested by: Nikola Ciprich, Michael L. Young
+ Patches: asterisk-21430-syslog-callid_trunk.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2526/ ........ Merged
+ revisions 388605 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Fix Crash Caused By One-way Audio With
+ auto_* NAT Settings Fix The prior code committed, r385473, failed
+ to take into consideration that not all outgoing calls will be to
+ a peer. My fault. This patch does the following: * Check if there
+ is a related peer involved. If there is, check and set NAT
+ settings according to the peer's settings. * Fix a problem with
+ realtime peers. If the global setting has auto_force_rport set
+ and we issued a "sip reload" while a peer is still registered,
+ the peer's flags for NAT are reset to off. When this happens, we
+ were always setting the contact address of the peer to that of
+ the full contact info that we had. (closes issue ASTERISK-21374)
+ Reported by: jmls Tested by: Michael L. Young Patches:
+ asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2524/
+ ........ Merged revisions 388601 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-13 20:37 +0000 [r388598] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_srtp.c, /: Revert r388529 for now Adding the cleanup
+ function needs some deeper thought since it apparently doesn't
+ exist for all variants of libsrtp. ........ Merged revisions
+ 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 388597 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-13 19:29 +0000 [r388579] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: pbx: Fix lack of cleanup on macrolock and
+ context_table (closes issue ASTERISK-21723) Reported by: Corey
+ Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
+ Farrell (license 5909) ........ Merged revisions 388532 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388578 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-13 18:10 +0000 [r388531] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_srtp.c: Close libsrtp properly Ensure that libsrtp is
+ shutdown properly when res_srtp is unloaded. (closes issue
+ ASTERISK-21719) Reported by: Corey Farrell Patches:
+ res_srtp-library-shutdown.patch uploaded by Corey Farrell
+ ........ Merged revisions 388529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388530 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-13 17:20 +0000 [r388526] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_gulp.c: chan_gulp: Minor readability Improvements
+ to chan_gulp (closes issue ASTERISK-21670) Reported by: Snuffy
+ Review: https://reviewboard.asterisk.org/r/2473/ Patches:
+ gulp-coding-guide.diff uploaded by snuffy (license 5024)
+
+2013-05-13 14:28 +0000 [r388479] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /: Fix SendText AMI action to never return
+ non-zero. AMI actions must never return non-zero unless they
+ intend to close the AMI connection. (Which is almost never.)
+ (closes issue ASTERISK-21779) Reported by: Paul Goldbaum ........
+ Merged revisions 388477 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388478 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-10 22:12 +0000 [r388427] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, /: Allow mISDN to send PROGRESS
+ messsage. * Made isdn_msg_parser.c build a progress message with
+ the mandatory progress indicator IE. (The mISDNuser NT state
+ machine rejected sending the incomplete message.) Note: The
+ associated mISDN and mISDNuser patches respectively are viewable
+ here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
+ http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
+ issue AST-1153) Reported by: Guenther Kelleter Patches:
+ progress-chan_misdn.diff (license #6372) patch uploaded by
+ Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
+ uploaded by Guenther Kelleter progress-misdnuser.diff (license
+ #6372) mISDNuser patch uploaded by Guenther Kelleter ........
+ Merged revisions 388425 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388426 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-10 20:50 +0000 [r388380] Mark Michelson <mmichelson@digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added
+ an io context without removing it. This caused a memory leak when
+ the module was unloaded. (closes ASTERISK-21718) Reported by
+ Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
+ Corey Farrell (License #5909) ........ Merged revisions 388376
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 388378 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-10 20:28 +0000 [r388375] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_config_odbc.c: Fix Finding Extensions With Patterns Using
+ ODBC Realtime After the merge of support for the realtime sorcery
+ module, extensions that contained a pattern were not being found
+ through odbc realtime. It was tracked down to this one line that
+ was advancing to the next variable list before it should have
+ been. The removal of this one line fixes this. Tested this fix on
+ my machine. Received confirmation that this is the right fix from
+ file on IRC.
+
+2013-05-10 17:12 +0000 [r388318-388350] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_http_recordings.c,
+ res/res_stasis_http_endpoints.c, main/loader.c,
+ res/res_stasis_http_events.c, res/res_stasis_http_sounds.c,
+ res/res_stasis_http_bridges.c, res/res_stasis_http.c,
+ res/res_stasis.c, apps/app_stasis.c,
+ res/res_stasis_http_asterisk.c,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ res/res_stasis_http_playback.c, res/res_stasis_websocket.c,
+ tests/test_res_stasis.c, res/res_stasis_http_channels.c,
+ include/asterisk/stasis_app.h: Address unload order issues for
+ res_stasis* modules I've noticed when doing a graceful shutdown
+ that the res_stasis_http.so module gets unloaded before the
+ modules that use it, which causes some asserts during their
+ unload. While r386928 was a quick hack to get it to not assert
+ and die, this patch increases the use counts on res_stasis.so and
+ res_stasis_http.so properly. It's a bigger change than I
+ expected, hence the review instead of just committing it. Review:
+ https://reviewboard.asterisk.org/r/2489/
+
+ * include/asterisk/stasis.h: Avoided __ast names for the private
+ variables created by the STASIS_MESSAGE_TYPE_*() macros.
+
+2013-05-10 13:13 +0000 [r388275] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_userevent.c,
+ rest-api-templates/event_function_decl.mustache (added),
+ res/stasis_http/resource_sounds.h, CHANGES,
+ res/res_stasis_http_events.c, include/asterisk/stasis_channels.h,
+ main/stasis_channels.c, rest-api-templates/swagger_model.py,
+ res/res_stasis.c, main/manager_channels.c,
+ rest-api-templates/stasis_http_resource.h.mustache,
+ res/stasis_http/resource_recordings.h,
+ rest-api-templates/asterisk_processor.py,
+ rest-api-templates/res_stasis_http_resource.c.mustache,
+ res/stasis_http/resource_endpoints.h,
+ rest-api/api-docs/events.json, res/stasis_http/resource_events.h,
+ res/res_stasis_websocket.c: Add channel events for res_stasis
+ apps This change adds a framework in res_stasis for handling
+ events from channel topics. JSON event generation and validation
+ code is created from event documentation in
+ rest-api/api-docs/events.json to assist in JSON event generation,
+ ensure consistency, and ensure that accurate documentation is
+ available for ALL events that are received by res_stasis
+ applications. The userevent application has been refactored along
+ with the code that handles userevent channel blob events to pass
+ the headers as key/value pairs in the JSON blob. As a
+ side-effect, app_userevent now handles duplicate keys by
+ overwriting the previous value. Review:
+ https://reviewboard.asterisk.org/r/2428/ (closes issue
+ ASTERISK-21180) Patch-By: Kinsey Moore <kmoore@digium.com>
+
+2013-05-10 11:47 +0000 [r388254] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_sip.c: Fix copy/paste error in
+ one-touch-recording implementation. ........ Merged revisions
+ 388253 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-09 14:41 +0000 [r388175] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_userevent.c: Don't expect to pack three tuples when you
+ only have two
+
+2013-05-09 04:11 +0000 [r388110-388113] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix The Payload Being Set On CN
+ Packets And Do Not Set Marker Bit When we send out a CN packet
+ (for instance, in the case of using rtpkeepalives), we are not
+ setting the payload code properly. Also, we are setting the
+ marker bit when we shouldn't be according to RFC 3389, section 4.
+ AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we
+ should be using ast_rtp_codecs_payload_code() rather than
+ ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
+ appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
+ * Remove the setting of the marker bit * Fix the debug message by
+ incrementing the seqno after the debug message is set in order to
+ display the correct seqno that was sent out (closes issue
+ ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
+ Katzmann, Michael L. Young Patches:
+ asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2500/ ........ Merged
+ revisions 388111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 388112 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_queue.c: Fix Segfault In app_queue When
+ "persistentmembers" Is Enabled And Using Realtime When the
+ "ignorebusy" setting was deprecated, we added some code to allow
+ us to be compatible with older setups that are still using the
+ "ignorebusy" setting instead of "ringinuse". We set a char
+ *variable with the column name to use, which helps the realtime
+ functions to use the correct column in their SQL queries. When
+ "persistentmembers" is enabled, we are not setting this variable
+ before the realtime functions were called to load members. This
+ results in the variable being NULL and therefore causing a
+ segfault when loading members during the module's process of
+ loading. The solution was to move the code that sets that
+ variable to be before these realtime functions are called during
+ the loading of the module. (closes issue ASTERISK-21738) Reported
+ by: JoshE Tested by: JoshE Patches:
+ asterisk-21738-rt-ringinuse-field-not-set.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2499/ ........ Merged
+ revisions 388108 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-08 22:00 +0000 [r388014-388075] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_websocket.c: Fixed MODFLAG for
+ res_stasis_websocket
+
+ * build_tools/cflags.xml, include/asterisk/inline_api.h: Add
+ development flag to disable the inline API. A GCC bug[1] can, in
+ some cases, pop up an unsuppressible pedwarn when using a static
+ inline standard library function from a non-static inline
+ function. This normally doesn't show up, but can occur if you're
+ running an upgrade version of GCC (such as GCC 4.8 on OS X, which
+ normally runs GCC 4.2). [1]:
+ http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
+
+ * main/enum.c, main/srv.c: Removed #if checks for crazy old
+ versions of OS X. The <arpa/nameser_compat.h> was introduced way
+ back in OS X Panther, which itself was end-of-lifed back in 2007.
+ We can assume that any OS X machine we build on will need that
+ header file :-) Why bother removing it? The flag we're checking
+ (__APPLE_CC__) is actually Apple's build number. Self-compiled
+ versions of GCC (such as installing the latest version of GCC
+ from homebrew) sets the value to 0, making it useless for this
+ sort of compile flaggery.
+
+ * tests/test_stasis_endpoints.c: Fixed set-but-not-used warning
+ caught by newer GCC
+
+2013-05-08 18:36 +0000 [r388008] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_directory.c: Don't perform a realtime lookup with a NULL
+ keyword Previously, a call to ast_load_realtime_multientry could
+ get away with passing a NULL parameter to the function, even
+ though it really isn't supposed to do that. After the change over
+ to using ast_variable instead of variadic arguments, the realtime
+ engine gets unhappy if you do this. This was always an unintended
+ function call in app_directory anyway - now, we just don't call
+ into the realtime function calls if we don't have anything to
+ query on.
+
+2013-05-08 18:34 +0000 [r388005] David M. Lee <dlee@digium.com>
+
+ * main/channel.c, include/asterisk/stasis_channels.h,
+ tests/test_stasis_channels.c, apps/app_userevent.c,
+ include/asterisk/stasis.h, main/stasis_channels.c,
+ res/res_stasis.c, main/manager_channels.c: Remove required type
+ field from channel blobs When we first introduced the channel
+ blob types, the JSON blobs were self identifying by a required
+ "type" field in the JSON object itself. This, as it turns out,
+ was a bad idea. When we introduced the message router, it was
+ useless for routing based on the JSON type. And messages had two
+ type fields to check: the stasis_message_type() of the message
+ itself, plus the type field in the JSON blob (but only if it was
+ a blob message). This patch corrects that mistake by removing the
+ required type field from JSON blobs, and introducing first class
+ stasis_message_type objects for the actual message type. Since we
+ now will have a proliferation of message types, I introduced a
+ few macros to help reduce the amount of boilerplate necessary to
+ set them up. Review: https://reviewboard.asterisk.org/r/2509
+
+2013-05-08 16:58 +0000 [r387974] Richard Mudgett <rmudgett@digium.com>
+
+ * utils: Add version.c to list of ignored files in the utils
+ directory.
+
+2013-05-08 13:39 +0000 [r387932] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_test.exports.in (added), tests/test_endpoints.c
+ (added), include/asterisk/stasis_endpoints.h (added),
+ res/res_stasis_test.c (added),
+ res/stasis_http/resource_endpoints.c, channels/sip/include/sip.h,
+ main/asterisk.c, rest-api/api-docs/endpoints.json,
+ res/stasis_http/resource_endpoints.h, main/stasis_cache.c,
+ main/stasis_endpoints.c (added), channels/chan_sip.c,
+ include/asterisk/endpoints.h (added), include/asterisk/astobj2.h,
+ main/channel_internal_api.c, include/asterisk/stasis_test.h
+ (added), include/asterisk/stasis.h, main/endpoints.c (added),
+ main/astobj2.c, res/res_stasis_http_endpoints.c,
+ tests/test_stasis_endpoints.c (added): Initial support for
+ endpoints. An endpoint is an external device/system that may
+ offer/accept channels to/from Asterisk. While this is a very
+ useful concept for end users, it is surprisingly not a core
+ concept within Asterisk itself. This patch defines ast_endpoint
+ as a separate object, which channel drivers may use to expose
+ their concept of an endpoint. As the channel driver creates
+ channels, it can use ast_endpoint_add_channel() to associate
+ channels to the endpoint. This updated the endpoint
+ appropriately, and forwards all of the channel's events to the
+ endpoint's topic. In order to avoid excessive locking on the
+ endpoint object itself, the mutable state is not accessible via
+ getters. Instead, you can create a snapshot using
+ ast_endpoint_snapshot_create() to get a consistent snapshot of
+ the internal state. This patch also includes a set of topics and
+ messages associated with endpoints, and implementations of the
+ endpoint-related RESTful API. chan_sip was updated to create
+ endpoints with SIP peers, but the state of the endpoints is not
+ updated with the state of the peer. Along for the ride in this
+ patch is a Stasis test API. This is a stasis_message_sink object,
+ which can be subscribed to a Stasis topic. It has functions for
+ blocking while waiting for conditions in the message sink to be
+ fulfilled. (closes issue ASTERISK-21421) Review:
+ https://reviewboard.asterisk.org/r/2492/
+
+2013-05-08 07:21 +0000 [r387885] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing
+ up and fail to be sent out after retries fail RFC6665 4.2.2: ...
+ after a failed State NOTIFY transaction remove the subscription
+ The problem is that the State Notify requests rely on the 200OK
+ reponse for pacing control and to not confuse the notify
+ susbsystem. The issue is, the pendinginvite isn't cleared if a
+ response isn't received, thus further notify's are never sent.
+ The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
+ subscription after failure. (closes issue ASTERISK-21677)
+ Reported by: Dan Martens Tested by: alecdavis alecdavis (license
+ 585) Review https://reviewboard.asterisk.org/r/2475/ ........
+ Merged revisions 387875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387880 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-07 18:32 +0000 [r387803-387825] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/lock.h: Fixed up \example marker in lock.h
+ Doxygen comment. The \example tags marks an entire file as an
+ example, not a code snippet.
+
+ * res/res_config_pgsql.c, main/manager.c, /: Minor fixups to
+ Doxygen comments. The \example tags marks an entire file as an
+ example, not a code snippet. ........ Merged revisions 387823
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk/json.h: Better explained the depths of reference
+ stealing.
+
+2013-05-07 17:53 +0000 [r387802] Jason Parker <jparker@digium.com>
+
+ * include/asterisk.h: Fix build breakage, from LOW_MEMORY fix.
+
+2013-05-06 17:15 +0000 [r387740-387741] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: Update ao2_destructor_fn doxygen.
+
+ * channels/chan_dahdi.c: Make a log NOTICE more explicit that the
+ event comes from DAHDI and not PRI.
+
+2013-05-06 17:01 +0000 [r387738] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c: Fix building with LOW_MEMORY defined.
+
+2013-05-06 15:58 +0000 [r387690] Russell Bryant <russell@russellbryant.com>
+
+ * /, apps/app_meetme.c: Make SLA reload more paranoid. Reload
+ support was originally not included for SLA. It was added later,
+ but in a fairly non-traditional way. It basically sets a flag
+ indicating that a reload is pending, and then waits for a time
+ where it thinks everything SLA related is idle and unused, and
+ *then* executes the reload. It does this because the reload
+ process is destructive. It starts by throwing everything away and
+ starting over. There are a number of problems with this approach.
+ One of them is that the check to see if anything in use was
+ incomplete. This patch makes it more complete and thus less
+ likely for a crash to occur during reload processing. However,
+ this approach still has problems so some much more significant
+ reworking of this code will need to come in as a next step. Patch
+ credit and testing by CoreDial, LLC. ........ Merged revisions
+ 387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 387689 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-06 13:04 +0000 [r387662] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/sorcery.h, res/res_sorcery_astdb.c,
+ tests/test_sorcery.c, main/sorcery.c: Add support for observers
+ and JSON objectset creation to sorcery. This change adds the
+ ability for modules to add themselves as observers to sorcery
+ object types. Observers can be notified when objects are created,
+ updated, or deleted as well as when the object type is loaded or
+ reloaded. Observer notifications are done using a thread pool in
+ a serialized fashion so the caller of the sorcery API calls is
+ minimally impacted. This also adds the ability to create JSON
+ changesets of a sorcery object. Tests are also present to confirm
+ all of the above functionality. Review:
+ https://reviewboard.asterisk.org/r/2477/
+
+2013-05-04 16:00 +0000 [r387630-387633] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk.h, main/asterisk.c: Clean up documentation;
+ prevent ref leak on exit This patch: * Cleans up some doxygen *
+ Prevents leaking the system level Stasis topics and messages on
+ exit (users of valgrind will be happier)
+
+ * funcs/func_global.c: Migrate SHARED's use of the VarSet AMI event
+ to Stasis-Core This patch removes the direct call to AMI from the
+ SHARED function and instead call Stasis-Core. Stasis-Core
+ delivers the notification that a shared variable has changed on a
+ channel to all interested consumers. (issue ASTERISK-21462)
+
+2013-05-03 18:03 +0000 [r387594] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_iax2.c, main/asterisk.c, include/asterisk.h,
+ channels/chan_sip.c, res/res_stun_monitor.c, main/event.c:
+ Stasis: Convert network change events into network change stasis
+ messages (issue ASTERISK-21103) Review:
+ https://reviewboard.asterisk.org/r/2490/
+
+2013-05-03 11:35 +0000 [r387545] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip_sdp_rtp.c, channels/chan_gulp.c: Use the configured
+ formats for Gulp sessions if there are no joint formats between
+ requested formats and configured formats. (closes issue
+ ASTERISK-21756)
+
+2013-05-02 20:59 +0000 [r387519] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_stack.c, build_tools/post_process_documentation.py:
+ Migrate AMI VarSet events raised by GoSub local variables This
+ patch moves VarSet events for local variables raised by GoSub
+ over to Stasis-Core. It also tweaks up the post-processing
+ documentation scripts to not combine parameters if both
+ parameters are already documented. (issue ASTERISK-21462)
+
+2013-05-02 19:06 +0000 [r387482] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Remove the ABI compatability ast_channel_alloc().
+ It is no longer needed.
+
+2013-05-02 17:15 +0000 [r387423] Matthew Jordan <mjordan@digium.com>
+
+ * utils/Makefile, /: Update utils Makefile to handle r387294 Alec's
+ patch that added the Asterisk version to 'core show locks'
+ angered the items in utils, as they exist somewhat outside of the
+ Asterisk build system. Some day, this Makefile should get nuked
+ from high orbit, but for now, include version.c in its list of
+ stuff to pile in. ........ Merged revisions 387421 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387422 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-02 16:39 +0000 [r387420] Jonathan Rose <jrose@digium.com>
+
+ * main/event.c, include/asterisk/event_defs.h: Putting all event
+ defs and names back for now due to res_corosync dependency
+
+2013-05-02 08:24 +0000 [r387296-387369] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Session-Expires: Set timer to correctly expire at (~2/3) of the
+ interval when not the refresher RFC 4028 Section 10 if the side
+ not performing refreshes does not receive a session refresh
+ request before the session expiration, it SHOULD send a BYE to
+ terminate the session, slightly before the session expiration.
+ The minimum of 32 seconds and one third of the session interval
+ is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
+ Session-Expires interval, or if the remote device was the
+ refresher, asterisk would timeout at interval end. Now, when not
+ refresher, timeout as per RFC noted above. (closes issue
+ ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2488/ ........ Merged
+ revisions 387344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387345 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
+ response when it's a RE-INVITE when asterisk is the refresher.
+ RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
+ Session-Expires header field in a response, even if none were
+ present in the request." What changed After ASTERISK-20787,
+ inbound calls to asterisk with no Session-Expires in the INVITE
+ are now are offered a Session-Expires (1800 asterisk default) in
+ the response, with asterisk as the refresher. Symptom: After 900
+ seconds (asterisk default refresher period 1800), asterisk
+ RE-INVITEs the device, the device may respond with a much lower
+ Session-Expires (180 in our case) value that it is now using.
+ Asterisk ignores this response, as it's deemed both an INBOUND
+ CALL, and a RE-INVITE. After 180 seconds the device times out and
+ sends BYE (hangs up), asterisk is still working with the
+ refresher period of 1800 as it ignored the 'Session Expires: 180'
+ in the previous 200OK response. Fix: handle_response_invite()
+ when 200OK, remove check for outbound and reinvite. (closes issue
+ ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2463/ ........ Merged
+ revisions 387312 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387319 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_dahdi.c, /: chan_dahdi: fix lower bound check with
+ -ve integer conversion from a float Lower bound of a 16bit signed
+ int is -32768 not -32767 (closes issue ASTERISK-21744) Reported
+ by: alecdavis Tested by: alecdavis alecdavis (license 585)
+ ........ Merged revisions 387297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387298 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/utils.c: Add Asterisk Version to core show locks Assist
+ with reporting 'core show locks' when submitting bug reports.
+ Example below: =========================== == SVN-branch-1.8-...
+ == Currently Held Locks =========================== (closes issue
+ ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) ........ Merged revisions 387294 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387295 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-01 21:55 +0000 [r387260-387261] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: Simplify
+ chan_local.c:manager_optimize_away() using ao2_find().
+
+ * channels/chan_local.c: Cleanup chan_local.c:local_new(). * Remove
+ t and ama local variables. There is no way they could be anything
+ other than default because p->owner can only be NULL at this
+ point. * Rename tmp and tmp2 to owner and chan respectively. *
+ Remove redundant initialization of channel context, exten,
+ priority.
+
+2013-05-01 21:18 +0000 [r387220] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Clear the DTMF sending digit tracking
+ on off nominal paths In certain situations, when the RTP engine
+ goes to send a DTMF end digit it may be in a situation where the
+ remote address is no longer available, or the digit that was
+ supposed to be sent is invalid. In such cases, we need to clear
+ the RTP counters appropriately. Otherwise, when the RTP source is
+ set again, we'll continue to think that we're in the middle of
+ sending a DTMF digit, which can confuse the remote party
+ (signficantly). (closes issue ASTERISK-21522) Reported by: Corey
+ Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
+ Farrell (License 5909) ........ Merged revisions 387213 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387216 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-01 21:09 +0000 [r387181-387212] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: Trivial changes. Comments, parentheses,
+ spelling, wording.
+
+ * channels/chan_local.c: Make chan_local locals container an
+ explicit list container. Pretending that chan_local locals
+ container can have more than one bucket is silly. The container
+ has no key to help search.
+
+ * channels/chan_local.c: Whitespace changes.
+
+ * main/loader.c: Make mod_load_cmp() not as klunky. There is a
+ reason the heap comparison functions like qsort(), and other
+ comparison functions specify <0, >0, and =0 for the return
+ values.
+
+ * channels/chan_unistim.c: Remove some unnecessary calls to
+ ast_bridged_channel() in chan_unistim.c
+
+ * channels/chan_mgcp.c: Remove some unnecessary calls to
+ ast_bridged_channel() in chan_mgcp.c
+
+ * channels/chan_skinny.c: Remove some unnecessary calls to
+ ast_bridged_channel() in chan_skinny.c
+
+ * channels/chan_iax2.c: Remove some unnecessary calls to
+ ast_bridged_channel() in chan_iax2.c
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Remove some
+ unnecessary calls to ast_bridged_channel() in
+ chan_dahdi.c/sig_analog.c
+
+2013-05-01 18:38 +0000 [r387135] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Prevent crash in 'sip show peers' when
+ the number of peers on a system is large When you have lots of
+ SIP peers (according to the issue reporter, around 3500), the
+ 'sip show peers' CLI command or AMI action can crash due to a
+ poorly placed string duplication that occurs on the stack. This
+ patch refactors the command to not allocate the string on the
+ stack, and handles the formatting of a single peer in a separate
+ function call. (closes issue ASTERISK-21466) Reported by:
+ Guillaume Knispel patches:
+ fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
+ uploaded by gknispel (License 6492) ........ Merged revisions
+ 387134 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-05-01 17:15 +0000 [r387108] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Move some annoying chan_dahdi debug
+ messages to level 5.
+
+2013-04-30 22:50 +0000 [r387039] Matthew Jordan <mjordan@digium.com>
+
+ * main/features.c, /: Fix CDR not being created during an
+ externally initiated blind transfer Way back when in the dark
+ days of Asterisk 1.8.9, blind transferring a call in a context
+ that included the 'h' extension would inadvertently execute the
+ hangup code logic on the transferred channel. This was a "bad
+ thing". The fix was to properly check for the softhangup flags on
+ the channel and only execute the 'h' extension logic (and, in
+ later versions, hangup handler logic) if the channel was well and
+ truly dead (Jim). Unfortunately, CDRs are fickle. Setting the
+ softhangup flag when we detected that the channel was leaving the
+ bridge (but not to die) caused some crucial snippet of CDR code,
+ lying in ambush in the middle of the bridging code, to not get
+ executed. This had the effect of blowing away one of the CDRs
+ that is typically created during a blind transfer. While we live
+ and die by the adage "don't touch CDRs in release branches", this
+ was our bad. The attached patch restores the CDR behavior, and
+ still manages to not run the 'h' extension during a blind
+ transfer (at least not when it's supposed to). Thanks to Steve
+ Davies for diagnosing this and providing a fix. Review:
+ https://reviewboard.asterisk.org/r/2476 (closes issue
+ ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
+ Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
+ one47 (License 5012) ........ Merged revisions 387036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 387038 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-30 22:37 +0000 [r387035-387037] Jonathan Rose <jrose@digium.com>
+
+ * main/event.c, include/asterisk/json.h, channels/chan_iax2.c,
+ main/named_acl.c, include/asterisk/acl.h, main/json.c,
+ main/manager.c, channels/chan_sip.c,
+ include/asterisk/event_defs.h: Stasis Core: Refactor ACL Change
+ events to go out over the stasis core msg bus (issue
+ ASTERISK-21103) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2481/
+
+ * main/event.c, /: Add forgotten event types to event_names array
+ ........ Merged revisions 387030 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-30 18:12 +0000 [r386990] Jason Parker <jparker@digium.com>
+
+ * channels/chan_gulp.c: Fix a log message.
+
+2013-04-30 13:48 +0000 [r386931] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Use the proper lower bound when
+ doing saturation arithmetic. 16 bit signed integers have a range
+ of [-32768, 32768). The existing code was using the interval
+ (-32768, 32768) instead. This patch fixes that. Review:
+ https://reviewboard.asterisk.org/r/2479/ ........ Merged
+ revisions 386929 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386930 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-30 13:37 +0000 [r386928] David M. Lee <dlee@digium.com>
+
+ * tests/test_stasis_http.c, res/res_stasis_http.c: Just a couple of
+ Stasis-HTTP nitpick fixes. * Fixed crash when res_stasis_http is
+ unloaded before the implementation modules. * Cleaned up test
+ initialization for test_stasis_http.so.
+
+2013-04-29 23:36 +0000 [r386879] Rusty Newton <rnewton@digium.com>
+
+ * sounds/Makefile, /: Modifying sounds/Makefile to pull down 1.4.24
+ core sounds 1.4.24 core sounds includes a full set of Italian
+ prompts for core sounds and a fix for the missing voicemail
+ prompts in the Russian language. (closes issue ASTERISK-19431)
+ (closes issue ASTERISK-19721) ........ Merged revisions 386877
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 386878 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-29 13:38 +0000 [r386793-386841] Olle Johansson <oej@edvina.net>
+
+ * CHANGES, apps/app_queue.c, /: Play periodic prompts for first
+ call in a call queue Review:
+ https://reviewboard.asterisk.org/r/2263/ ........ Merged
+ revisions 386792 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386794 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk/doxygen/commits.h: Change pointer to existing
+ wiki page instead of non-existing page
+
+2013-04-28 03:32 +0000 [r386774] Kinsey Moore <kmoore@digium.com>
+
+ * rest-api-templates/swagger_model.py: Fix spelling error in python
+ doc
+
+2013-04-27 19:03 +0000 [r386731-386760] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sip.c: Tweak res_sip priority so it gets loaded first
+ before all other SIP stuff.
+
+ * res/res_config_sqlite.c: Update res_config_sqlite to use the
+ ast_variable lists.
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c,
+ include/asterisk/config.h, CHANGES, res/res_config_ldap.c,
+ main/config.c, tests/test_sorcery_realtime.c (added),
+ main/sorcery.c, res/res_sorcery_realtime.c (added),
+ addons/res_config_mysql.c, res/res_config_sqlite3.c,
+ res/res_config_curl.c: Add support for a realtime sorcery module.
+ This change does the following: 1. Adds the sorcery realtime
+ module 2. Adds unit tests for the sorcery realtime module 3.
+ Changes the realtime core to use an ast_variable list instead of
+ variadic arguments 4. Changes all realtime drivers to accept an
+ ast_variable list Review:
+ https://reviewboard.asterisk.org/r/2424/
+
+2013-04-26 21:52 +0000 [r386685-386686] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_sip_nat.c, res/res_sip_registrar.c,
+ res/res_sip_dtmf_info.c,
+ res/res_sip_outbound_authenticator_digest.c,
+ res/res_sip_rfc3326.c, res/res_sip_outbound_registration.c,
+ res/res_sip_endpoint_identifier_ip.c,
+ res/res_sip_endpoint_identifier_constant.c, res/res_sip_mwi.c,
+ res/res_sip_acl.c, res/res_sip_logger.c,
+ res/res_sip_endpoint_identifier_user.c, res/res_sip_pubsub.c: Add
+ missing module dependencies to various res_sip* modules This
+ patch updates the various res_sip modules with their proper
+ menuselect options and proper dependencies, such that Asterisk
+ still has a snowball's chance in hell of compiling without
+ pjproject. Much thanks to snuffy(-home|-work) for making
+ everyone's life easier with this patch. Review:
+ https://reviewboard.asterisk.org/r/2472/ (closes issue
+ ASTERISK-21669) Reported by: snuffy patches: xml-depends.diff
+ uploaded by snuffy (license 5024)
+
+ * /, main/config.c: Clean up memory leak in config file on off
+ nominal paths when glob is allowed If a system allows for its
+ usage, Asterisk will use glob to help parse Asterisk .conf files.
+ The config file loading routine was leaking the memory allocated
+ by the glob() routine when the config file was in an unmodified
+ or invalid state. This patch properly calls globfree in those off
+ nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
+ Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 386672 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386677 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-26 21:31 +0000 [r386684] David M. Lee <dlee@digium.com>
+
+ * main/loader.c: By popular demand, putting the
+ about-to-load-module printf back. But now it only prints during
+ the initial startup, and prints at verbose 1 level.
+
+2013-04-26 21:27 +0000 [r386676] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/features.c: Clean up resources in features on exit This
+ patch cleans up two things features: * It properly unregisters
+ the CLI commands that features registered * It cancels and
+ performs a pthread_join on the created parking thread. This not
+ only properly joins a non-detached thread, but also prevents
+ disposing of the parking lots prior to the parking thread
+ completely exiting. (closes issue ASTERISK-21407) Reported by:
+ Corey Farrell patches: features_shutdown-r2.patch uploaded by
+ Corey Farrell (License 5909) ........ Merged revisions 386641
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 386642 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-26 21:00 +0000 [r386640] David M. Lee <dlee@digium.com>
+
+ * main/loader.c: Removing stray printf from r386540
+
+2013-04-26 20:32 +0000 [r386638] Mark Michelson <mmichelson@digium.com>
+
+ * main/uuid.c: Add an \extref doxygen pointer for libuuid. Thanks
+ to Olle Johansson for suggesting this.
+
+2013-04-26 20:05 +0000 [r386623-386624] David M. Lee <dlee@digium.com>
+
+ * res/res_chan_stats.c (added), res/res_statsd.exports.in (added),
+ configs/statsd.conf.sample (added), include/asterisk/utils.h,
+ include/asterisk/statsd.h (added), res/res_statsd.c (added):
+ Example of how to use the Stasis message bus In order to get
+ people familiar with the Stasis message bus, it would be useful
+ to have something of a tutorial. Since I'm not clever enough to
+ think of some cool integration we could do with Twitter, I
+ settled for something that might actually be useful. This patch
+ adds a res_statsd.so module, which implements a basic statsd[1]
+ client. Statsd is a very simple statistics gathering server,
+ which can publish its results to a backend graphing engine, like
+ Graphite[2]. There are several different Statsd server
+ implementations[3], so you can pick what works best for your
+ environment. The actual example of how to use the Stasis message
+ bus is in res_chan_stats.so. This module demonstrates how to use
+ subscriptions and the message router by monitoring messages and
+ posting channels stats to the statsd server. A wiki page walking
+ through res_chan_stats.so is forthcoming. [1]:
+ https://github.com/etsy/statsd/ [2]:
+ http://graphite.readthedocs.org/en/latest/ [3]:
+ http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/
+ Review: https://reviewboard.asterisk.org/r/2460/
+
+ * res/res_sip: Ignore *.[oi] files in res/res_sip
+
+2013-04-25 21:32 +0000 [r386577] Joshua Colp <jcolp@digium.com>
+
+ * configs/res_sip.conf.sample: Don't bind to anything in the sample
+ configuration so we don't clash with chan_sip on a "make samples"
+ right now.
+
+2013-04-25 18:28 +0000 [r386540-386541] Mark Michelson <mmichelson@digium.com>
+
+ * /: REmove automerge properties.
+
+ * res/res_sip_acl.c (added), configs/res_sip.conf.sample (added),
+ res/res_sip/sip_configuration.c, /, res/res_sip_dtmf_info.c
+ (added), res/res_sip/include/res_sip_private.h,
+ include/asterisk/res_sip_session.h (added), main/threadpool.c,
+ res/res_sip_endpoint_identifier_ip.c (added),
+ tests/test_sorcery.c, res/res_sip_sdp_rtp.c (added),
+ res/res_sip/sip_outbound_auth.c, main/loader.c,
+ res/res_sip_caller_id.c (added),
+ res/res_sip_endpoint_identifier_user.c (added),
+ res/res_sip/include, res/res_sip_nat.c (added), configure,
+ res/res_sip_session.c (added), res/res_sip/sip_options.c,
+ include/asterisk/sorcery.h, res/res_sip_pubsub.exports.in
+ (added), res/res_sip_rfc3326.c (added), res/res_sip/location.c,
+ res/res_sip_endpoint_identifier_constant.c (added),
+ res/res_sip_mwi.c (added), res/res_sip (added),
+ res/res_sip_pubsub.c (added), res/res_sorcery_config.c,
+ res/res_sip/config_transport.c, include/asterisk/threadpool.h,
+ res/res_sip_registrar.c (added),
+ include/asterisk/autoconfig.h.in, include/asterisk/res_sip.h
+ (added), res/res_sip/sip_distributor.c,
+ res/res_sip/config_auth.c, res/res_sip.exports.in (added),
+ res/Makefile, res/res_sip_authenticator_digest.c (added),
+ main/taskprocessor.c, res/res_sip_session.exports.in (added),
+ channels/Makefile, main/astobj2.c,
+ res/res_sip/config_domain_aliases.c, channels/chan_gulp.c
+ (added), res/res_sip_logger.c (added), res/res_sip.c (added),
+ include/asterisk/res_sip_pubsub.h (added),
+ res/res_sip_outbound_authenticator_digest.c (added),
+ res/res_sip_outbound_registration.c (added), main/sorcery.c:
+ Merge the pimp_my_sip branch into trunk. The pimp_my_sip branch
+ is being merged at this point because it offers basic
+ functionality, and from an API standpoint, things are complete.
+ SIP work is *not* feature-complete; however, with the completion
+ of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
+ been created, and thus it is possible for developers to attempt
+ to create new SIP work. API documentation can be found in the
+ doxygen in the code, but usability documentation is still
+ lacking.
+
+2013-04-25 03:04 +0000 [r386485-386487] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Displaying Symmetric RTP Global
+ Setting * Use comedia_string() to display correctly the symmetric
+ rtp setting when running "sip show settings" ........ Merged
+ revisions 386486 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Change Case On Forcerport For Consistency
+ * Change "ForcerPort" to "Forcerport" to match everywhere else it
+ is displayed ........ Merged revisions 386483 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386484 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-24 21:47 +0000 [r386461-386462] David M. Lee <dlee@digium.com>
+
+ * res/stasis_http/resource_endpoints.h,
+ res/stasis_http/resource_events.h,
+ res/stasis_http/resource_asterisk.h,
+ res/stasis_http/resource_playback.h,
+ res/stasis_http/resource_channels.h,
+ res/stasis_http/resource_sounds.h,
+ res/stasis_http/resource_bridges.h,
+ res/stasis_http/resource_recordings.h,
+ rest-api-templates/stasis_http_resource.h.mustache: Document JSON
+ models in resource_*.h
+
+ * rest-api-templates/swagger_model.py: Oops. Mustache doesn't like
+ dictionaries
+
+2013-04-23 20:18 +0000 [r386375] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_config_parser.c, apps/app_confbridge.c:
+ confbridge: Make search the conference bridges container using
+ OBJ_KEY. * Make confbridge config parsing user profile, bridge
+ profile, and menu container hash/cmp functions correctly check
+ the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags. * Made
+ confbridge load_module()/unload_module() free all resources on
+ failure conditions.
+
+2013-04-23 18:57 +0000 [r386352] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_stasis.c: Fix some bad whitespace This crept in with the
+ RESTful HTTP interface merge.
+
+2013-04-22 16:44 +0000 [r386289] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix crash when AMI redirect action redirects
+ two channels out of a bridge. The two party bridging loops were
+ changing the bridge peer pointers without the channel locks held.
+ Thus when ast_channel_massquerade() tested and used the pointer
+ there is a small window of opportunity for the pointers to become
+ NULL even though the masquerade code has the channels locked.
+ (closes issue ASTERISK-21356) Reported by: William luke Patches:
+ jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: William luke ........ Merged revisions 386256
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 386286 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-22 16:22 +0000 [r386266] Andrew Latham <lathama@gmail.com>
+
+ * include/asterisk/srv.h: Doxygen - Markup Guidelines Expand on a
+ commit by OEJ to use the Coding-Guidelines (issue ASTERISK-20259)
+
+2013-04-22 14:58 +0000 [r386232] David M. Lee <dlee@digium.com>
+
+ * res/Makefile, include/asterisk/json.h,
+ res/res_stasis_http_recordings.c (added), res/stasis_http.make
+ (added), tests/test_strings.c, res/res_stasis_http_endpoints.c
+ (added), res/res_stasis_http_events.c (added),
+ include/asterisk/http.h, Makefile, main/json.c,
+ res/res_stasis_http.exports.in (added), rest-api-templates
+ (added), res/stasis_http/resource_channels.c,
+ res/res_stasis_http_sounds.c (added), rest-api (added),
+ main/http.c, res/res_stasis_http_bridges.c (added),
+ tests/test_stasis_http.c (added), include/asterisk/strings.h,
+ res/res_stasis_http.c (added), tests/test_stasis.c,
+ res/res_stasis.c, res/res_stasis_http_asterisk.c (added),
+ res/res_stasis_http_playback.c (added), res/stasis_http (added),
+ configs/stasis_http.conf.sample (added),
+ include/asterisk/stasis_http.h (added),
+ res/res_stasis_http_channels.c (added),
+ include/asterisk/stasis_app.h: This patch adds a RESTful HTTP
+ interface to Asterisk. The API itself is documented using
+ Swagger, a lightweight mechanism for documenting RESTful API's
+ using JSON. This allows us to use swagger-ui to provide
+ executable documentation for the API, generate client bindings in
+ different languages, and generate a lot of the boilerplate code
+ for implementing the RESTful bindings. The API docs live in the
+ rest-api/ directory. The RESTful bindings are generated from the
+ Swagger API docs using a set of Mustache templates. The code
+ generator is written in Python, and uses Pystache. Pystache has
+ no dependencies, and be installed easily using pip. Code
+ generation code lives in rest-api-templates/. The generated code
+ reduces a lot of boilerplate when it comes to handling HTTP
+ requests. It also helps us have greater consistency in the REST
+ API. (closes issue ASTERISK-20891) Review:
+ https://reviewboard.asterisk.org/r/2376/
+
+2013-04-22 12:45 +0000 [r386211] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/srv.h: Fix mistake in Doxygen. Doxygen is only
+ *ONE* comment that applies to the NEXT piece of code.
+
+2013-04-22 01:05 +0000 [r386190] Russell Bryant <russell@russellbryant.com>
+
+ * apps/app_meetme.c: sla: remove redundant locking. sla.lock was
+ already locked in the only place that sla_check_reload() was
+ called. Remove the redundant locking of sla.lock done in this
+ function. Less recursive locking is A Good Thing.
+
+2013-04-19 22:27 +0000 [r386160] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_timing_pthread.c: Prevent res_timing_pthread from
+ blocking callers There were several reports of deadlock when
+ using res_timing_pthread. Backtraces indicated that one thread
+ was blocked waiting for the write to the pipe to complete and
+ this thread held the container lock for the timers. Therefore any
+ thread that wanted to create a new timer or read an existing
+ timer would block waiting for either the timer lock or the
+ container lock and deadlock ensued. This patch changes the way
+ the pipe is used to eliminate this source of deadlocks: 1) The
+ pipe is placed in non-blocking mode so that it would never block
+ even if the following changes someone fail... 2) Instead of
+ writing bytes into the pipe for each "tick" that's fired the pipe
+ now has two states--signaled and unsignaled. If signaled, the
+ pipe is hot and any pollers of the read side filedescriptor will
+ be woken up. If unsigned the pipe is idle. This eliminates even
+ the chance of filling up the pipe and reduces the potential
+ overhead of calling unnecessary writes. 3) Since we're tracking
+ the signaled / unsignaled state, we can eliminate the exta poll
+ system call for every firing because we know that there is data
+ to be read. (closes issue ASTERISK-21389) Reported by: Matt
+ Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
+ 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
+ uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
+ Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
+ Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
+ by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
+ isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
+ https://reviewboard.asterisk.org/r/2441/ ........ Merged
+ revisions 386109 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386159 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-19 05:20 +0000 [r386019-386054] David M. Lee <dlee@digium.com>
+
+ * main/cli.c, /: cli.c: Properly initialize debug_modules and
+ verbose_modules. This avoids some lock errors on the core set
+ {debug,verbose} commands. ........ Merged revisions 386049 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 386051 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk/http_websocket.h, res/res_http_websocket.c:
+ Allow WebSocket connections on more URL's This patch adds the
+ concept of ast_websocket_server to res_http_websocket, allowing
+ WebSocket connections on URL's more more than /ws. The existing
+ funcitons for managing the WebSocket subprotocols on /ws still
+ work, so this patch should be completely backward compatible.
+ (closes issue ASTERISK-21279) Review:
+ https://reviewboard.asterisk.org/r/2453/
+
+ * main/message.c, /: Fix lock errors on startup. In messages.c,
+ there are several places in the code where we create a
+ tmp_tech_holder and pass that into an ao2_find call.
+ Unfortunately, we weren't initializing the rwlock on the
+ tmp_tech_holder, which the hash function was locking. It's
+ apparently harmless, but still not the best code. This patch
+ extracts all that copy/pasted code into two functions,
+ msg_find_by_tech and msg_find_by_tech_name, which properly
+ initialize and destroy the rwlock on the tmp_tech_holder. Review:
+ https://reviewboard.asterisk.org/r/2454/ ........ Merged
+ revisions 386006 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-16 23:44 +0000 [r385939] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, res/res_xmpp.c, res/res_jabber.c: res_xmpp and res_jabber need
+ to search 'cachable' in the attrib section of the received IE,
+ not data. (issue ASTERISK-20175) (closes issue ASTERISK-21429)
+ (closes issue ASTERISK-21069) (closes issue ASTERISK-21164)
+ Reported by: alecdavis Tested by: alecdavis alecdavis (license
+ 585) Review https://reviewboard.asterisk.org/r/2452/
+
+2013-04-16 17:50 +0000 [r385860-385886] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_corosync.c: Allow res_corosync to build
+ ast_enable_distributed_devstate is no longer applicable to how
+ the distributed device state system works and is no longer
+ necessary.
+
+ * funcs/func_presencestate.c, include/asterisk/presencestate.h,
+ main/presencestate.c, main/pbx.c: Move presence state
+ distribution to Stasis-core Convert presence state events to
+ Stasis-core messages and remove redundant serializers where
+ possible. Review: https://reviewboard.asterisk.org/r/2410/
+ (closes issue ASTERISK-21102) Patch-by: Kinsey Moore
+ <kmoore@digium.com>
+
+ * include/asterisk/xmpp.h, tests/test_devicestate.c,
+ main/devicestate.c, res/res_xmpp.c, apps/app_queue.c,
+ res/res_jabber.c, main/asterisk.c,
+ include/asterisk/devicestate.h, main/pbx.c, main/ccss.c: Move
+ device state distribution to Stasis-core In the move from
+ Asterisk's event system to Stasis, this makes distributed device
+ state aggregation always-on, removes unnecessary task processors
+ where possible, and collapses aggregate and non-aggregate states
+ into a single cache for ease of retrieval. This also removes an
+ intermediary step in device state aggregation. Review:
+ https://reviewboard.asterisk.org/r/2389/ (closes issue
+ ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com>
+
+2013-04-16 14:09 +0000 [r385835] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/stasis_channels.h: Fixed a typo
+
+2013-04-15 17:26 +0000 [r385782] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Don't unnecessarily rebuild things on every run of
+ 'make'. Review: https://reviewboard.asterisk.org/r/2449/ ........
+ Merged revisions 385745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385768 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-15 16:47 +0000 [r385718-385743] David M. Lee <dlee@digium.com>
+
+ * res/res_stasis_websocket.c: Avoid unused variable warning when
+ not in devmode
+
+ * apps/stasis_json.c (removed), main/stasis_channels.c,
+ tests/test_app_stasis.c (removed), res/res_stasis.c (added),
+ main/manager_channels.c, apps/app_stasis.c, tests/test_json.c,
+ res/res_stasis_websocket.c, tests/test_res_stasis.c (added),
+ tests/test_stasis_channels.c, include/asterisk/app_stasis.h
+ (removed), include/asterisk/stasis_app.h (added),
+ include/asterisk/json.h, main/json.c,
+ include/asterisk/stasis_channels.h, res/res_stasis.exports.in
+ (added), apps/Makefile, apps/app_stasis.exports.in (removed):
+ Moved core logic from app_stasis to res_stasis After some
+ discussion on asterisk-dev, it was decided that the bulk of the
+ logic in app_stasis actually belongs in a resource module instead
+ of the application module. This patch does that, leaves the app
+ specific stuff in app_stasis, and fixes up everything else to be
+ consistent with that change. * Renamed test_app_stasis to
+ test_res_stasis * Renamed app_stasis.h to stasis_app.h * This is
+ still stasis application support, even though it's no longer in
+ an app_ module. The name should never have been tied to the type
+ of module, anyways. * Now that json isn't a resource module
+ anymore, moved the ast_channel_snapshot_to_json function to
+ main/stasis_channels.c, where it makes more sense. Review:
+ https://reviewboard.asterisk.org/r/2430/
+
+ * main/channel.c, include/asterisk/cli.h,
+ include/asterisk/strings.h, apps/app_stasis.c,
+ main/manager_channels.c: DTMF events are now published on a
+ channel's stasis_topic. AMI was refactored to use these events
+ rather than producing the events directly in channel.c. Finally,
+ the code was added to app_stasis to produce DTMF events on the
+ WebSocket. The AMI events are completely backward compatible,
+ including sending events on transmitted DTMF, and sending DTMF
+ start events. The Stasis-HTTP events are somewhat simplified.
+ Since DTMF start and DTMF send events are generally less useful,
+ Stasis-HTTP will only send events on received DTMF end. (closes
+ issue ASTERISK-21282) (closes issue ASTERISK-21359) Review:
+ https://reviewboard.asterisk.org/r/2439
+
+ * BSDmakefile, contrib/realtime/mysql/voicemail_data.sql,
+ build_tools/sha1sum-sh, res/res_mutestream.c,
+ configs/res_curl.conf.sample, tests/test_func_file.c,
+ res/res_rtp_multicast.c, include/asterisk/select.h,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/bridging_features.h, tests/test_locale.c,
+ doc/Makefile, tests/test_poll.c,
+ contrib/realtime/mysql/musiconhold.sql, res/res_timing_kqueue.c,
+ contrib/realtime/mysql/queue_log.sql,
+ channels/sip/include/security_events.h, channels/sig_ss7.c,
+ channels/chan_multicast_rtp.c, channels/sig_ss7.h, /,
+ tests/test_expr.c, apps/app_saycounted.c,
+ channels/sip/security_events.c,
+ contrib/realtime/mysql/voicemail_messages.sql: Fix the
+ svn:keywords property on several files. Normally I think keyword
+ expansion is silly, but the one time it would have been good, it
+ didn't work because the property had quotes in it. This patch
+ fixes obviously busted svn:keywords properties. ........ Merged
+ revisions 385683 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385689 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-14 03:01 +0000 [r385635-385638] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_multicast.c, /: Calculate the timestamp for outbound
+ RTP if we don't have timing information This patch calculates the
+ timestamp for outbound RTP when we don't have timing information.
+ This uses the same approach in res_rtp_asterisk. Thanks to both
+ Pietro and Tzafrir for providing patches. (closes issue
+ ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
+ Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
+ by tzafrir (License 5035) rtp-timestamp.patch uploaded by
+ pbertera (License 5943) ........ Merged revisions 385636 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385637 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_alsa.c: Don't attempt to create a voice frame on
+ a read error Prior to this patch, a read error in snd_pcm_readi
+ would still be treated as a nominal result when constructing a
+ voice frame from the expected data. Since the value returned is
+ negative, as opposed to the number of samples read, this could
+ result in a crash. With this patch, we now return a null frame
+ when a read error is detected. Note that the patch on
+ ASTERISK-21329 was modified slightly for this commit, in that we
+ bail immediately on detecting the read error, rather than
+ bypassing the construction of the voice frame. (closes issue
+ ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
+ chan_alsa.diff uploaded by kawasaki (License 6489) ........
+ Merged revisions 385633 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385634 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-12 22:38 +0000 [r385595] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/app_queue.c: Fix Manager Segfault When app_queue Is
+ Unloaded When app_queue is unloaded, some manager commands are
+ not being unregistered which result in a segfault. This patch
+ corrects this. (closes issue ASTERISK-21397) Reported by: Peter
+ Katzmann, Corey Farrell Tested by: Corey Farrell Patches:
+ asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
+ Young (license 5026)
+ asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2444/
+ ........ Merged revisions 385593 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385594 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-12 22:26 +0000 [r385585] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/codec_resample.c: Allow codec_resample to be unloaded
+ Ensure that trans_size is correct to prevent uninitialized
+ entries from preventing reload. (closes issue ASTERISK-21401)
+ Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+ codec_resample-unload.patch uploaded by Corey Farrell ........
+ Merged revisions 385582 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-12 22:22 +0000 [r385573] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_voicemail.c, /: Fix app_voicemail Segfault And A Few
+ Memory Leaks The original report was that app_voicemail would
+ crash. This was caused by ast_config_load() returning
+ CONFIG_STATUS_FILEINVALID but no checks being performed for that
+ return status. After adding the initial patch to fix this issue,
+ Jaco Kroon (jkroon) added some fixes to memory leaks he had
+ discovered. During review, Walter Doekes (wdoekes) suggested
+ adding a helper function in order to determine if we had a valid
+ configuration or not. This patch does the following: * Creates a
+ helper function to check if the configuration is valid * Adds
+ calls to the new helper function where appropiate * Fixes memory
+ leaks where the code returned without running
+ ast_config_destroy() on the configuration that was loaded (closes
+ issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
+ Kroon, Michael L. Young Patches:
+ asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
+ (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2443/ ........ Merged
+ revisions 385551 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385557 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-12 21:48 +0000 [r385548] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/sorcery.h: Fix documentation.
+
+2013-04-12 21:11 +0000 [r385522] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/manager.h, main/manager_channels.c: Expose
+ channel snapshot manager blob generation These functions are
+ already used in one branch (jrose's parking branch) and will soon
+ be used in other branches as well.
+
+2013-04-12 15:06 +0000 [r385474] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix One-Way Audio With auto_* NAT
+ Settings When SIP Calls Initiated By PBX When we reload Asterisk
+ or chan_sip, the flags force_rport and comedia that are turned on
+ and off when using the auto_force_rport and auto_comedia nat
+ settings go back to the default setting off. These flags are
+ turned on when needed or off when not needed at the time that a
+ peer registers, re-registers or initiates a call. This would
+ apply even when only the default global setting
+ "nat=auto_force_rport" is being used, which in this case would
+ only affect the force_rport flag. Everything is good except for
+ the following: The nat setting is set to auto_force_rport and
+ auto_comedia. We reload Asterisk and the peer's registration has
+ not expired. We load in the settings for the peer which turns
+ force_rport and comedia back to off. Since the peer has not
+ re-registered or placed a call yet, those flags remain off. We
+ then initiate a call to the peer from the PBX. The force_rport
+ and comedia flags stay off. If NAT is involved, we end up with
+ one-way audio since we never checked to see if the peer is behind
+ NAT or not. This patch does the following: * Moves the checking
+ of whether a peer is behind NAT into its own function * Create a
+ function to set the peer's NAT flags if they are using the auto_*
+ NAT settings * Adds calls in sip_request_call() to these new
+ functions in order to setup the dialog according to the peer's
+ settings (closes issue ASTERISK-21374) Reported by: Michael L.
+ Young Tested by: Michael L. Young Patches:
+ asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2421/
+ ........ Merged revisions 385473 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-12 08:52 +0000 [r385406-385431] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_iax2.c, /: IAX2 defer_full_frames fail to get sent
+ Ensure iax2_process_thread is signalled when a deferred frame is
+ queued to it. (closes issue ASTERISK-18827) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2426/ ........ Merged
+ revisions 385429 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385430 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_iax2.c, /: IAX2, prevent network thread starting
+ before all helper threads are ready On startup, it's possible for
+ a frame to arrive before the processing threads were ready. In
+ iax2_process_thread() the first pass through falls into
+ ast_cond_wait, should a frame arrive before we are at
+ ast_cond_wait, the signal will be ignored. The result
+ iax2_process_thread stays at ast_cond_wait forever, with deferred
+ frames being queued. Fix: When creating initial idle
+ iax2_process_threads, wait for init_cond to be signalled after
+ each thread is started. (issue ASTERISK-18827) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2427/ ........ Merged
+ revisions 385402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385403 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-11 16:53 +0000 [r385277-385314] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/cli_aliases.conf.sample: Fix 'pri intense debug span'
+ alias. ........ Merged revisions 385313 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/features.c: Eliminated dial_features_destroy() since it is
+ equivalent to ast_free_ptr()
+
+ * main/manager.c, main/features.c: * Fix unlocked accesses to
+ feature_list. The feature_list is now also protected by the
+ features_lock. * Made all calls to ast_find_call_feature() have
+ the features_lock held. * Fixed set_config_flags() to actually
+ use find_group() to look for feature groups in DYNAMIC_FEATURES.
+ The code originally assumed all feature groups were listed in
+ DYNAMIC_FEATURES. * Make everyone use ast_rdlock_call_features(),
+ ast_unlock_call_features(), and new ast_wrlock_call_features()
+ instead of directly calling the rwlock API on features_lock.
+
+2013-04-10 15:34 +0000 [r385236] David M. Lee <dlee@digium.com>
+
+ * main/stasis_channels.c: Fixed manager channelvars support. For
+ the events that have been ported to Stasis, this was broken in
+ r384910, when a couple of lines of code was lost in a merge.
+
+2013-04-10 14:26 +0000 [r385174-385202] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_config_ldap.c, /: Use LDAP memory management functions
+ instead of Asterisk's When MALLOC_DEBUG is enabled with
+ res_config_ldap, issues (munmap_chunk: invalid pointer errors)
+ can occur as the memory is being allocated with Asterisk's
+ wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
+ library's wrappers. This patch uses the LDAP library's wrappers
+ where appropriate, so that compiling with MALLOC_DEBUG doesn't
+ cause more problems than it solves. Note that the patch listed
+ below was modified slightly for this commit to account for some
+ additional memory allocation/deallocations. (closes issue
+ ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
+ patches: issue18789-1.8-r316873.patch uploaded by seanbright
+ (License 5060) ........ Merged revisions 385190 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385199 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Fix crash in chan_sip when a core
+ initiated op occurs at the same time as a BYE When a BYE request
+ is processed in chan_sip, the current SIP dialog is detached from
+ its associated Asterisk channel structure. The tech_pvt pointer
+ in the channel object is set to NULL, and the dialog persists for
+ an RFC mandated period of time to handle re-transmits. While this
+ process occurs, the channel is locked (which is good).
+ Unfortunately, operations that are initiated externally have no
+ way of knowing that the channel they've just obtained (which is
+ still valid) and that they are attempting to lock is about to
+ have its tech_pvt pointer removed. By the time they obtain the
+ channel lock and call the channel technology callback, the
+ tech_pvt is NULL. This patch adds a few checks to some channel
+ callbacks that make sure the tech_pvt isn't NULL before using it.
+ Prime offenders were the DTMF digit callbacks, which would crash
+ if AMI initiated a DTMF on the channel at the same time as a BYE
+ was received from the UA. This patch also adds checks on
+ sip_transfer (as AMI can also cause a callback into this
+ function), as well as sip_indicate (as lots of things can queue
+ an indication onto a channel). Review:
+ https://reviewboard.asterisk.org/r/2434/ (closes issue
+ ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions
+ 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 385173 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-09 19:58 +0000 [r385142] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Rename struct feature_ds to struct
+ feature_datastore. Because "struct feature_ds *feature_ds" is not
+ a good thing.
+
+2013-04-09 18:22 +0000 [r385116] David M. Lee <dlee@digium.com>
+
+ * apps/app_stasis.c: Backported app_stasis fix from stasis-http
+ branch. The hash and compare functions for the control container
+ was reusing the wrong ones, causing some problems. I fixed it,
+ but in the wrong branch. Oh well, it happens.
+
+2013-04-09 06:16 +0000 [r385088] Russell Bryant <russell@russellbryant.com>
+
+ * main/features.c, CHANGES: Add inheritance support to
+ FEATURE()/FEATUREMAP(). The settings saved on the channel for
+ FEATURE()/FEATUREMAP() were only for that channel. This patch
+ adds the ability to have these settings inherited to child
+ channels if you set FEATURE(inherit)=yes. Closes issue
+ ASTERISK-21306. Review: https://reviewboard.asterisk.org/r/2415/
+
+2013-04-08 23:38 +0000 [r385049] Rusty Newton <rnewton@digium.com>
+
+ * /, configs/extconfig.conf.sample: Modified the list of keys for
+ the driver backends for sake of sample clarity Added a line
+ showing the mapping of "mysql" to res_config_mysql available in
+ add-ons. We used "mysql" as an example driver key in the sample,
+ but didn't show what module it mapped too. Also added a subtitle
+ above the list of keys for driver backends. ........ Merged
+ revisions 385047 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 385048 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-08 18:24 +0000 [r384989] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * build_tools/make_version, build_tools/make_buildopts_h,
+ build_tools/make_linker_version_script, Makefile,
+ build_tools/mkpkgconfig: Clean up Makefile "warning" clutter when
+ makeopts doesn't exist. Review:
+ https://reviewboard.asterisk.org/r/2304
+
+2013-04-08 15:38 +0000 [r384910-384942] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_stasis_websocket.c, res/res_http_websocket.c: Don't
+ attempt a websocket protocol removal if res_http_websocket isn't
+ there This patch sets the protocols container provided by
+ res_http_websocket to NULL when the module gets unloaded and adds
+ the necessary checks when adding/ removing a websocket protocol.
+ This prevents some FRACKing on an invalid pointer to the disposed
+ container if a module that uses res_http_websocket is unloaded
+ after it.
+
+ * include/asterisk/channel.h, CHANGES, main/channel.c, main/dial.c,
+ include/asterisk/stasis_channels.h (added), main/features.c,
+ apps/stasis_json.c, pbx/pbx_realtime.c, main/stasis_channels.c
+ (added), apps/app_stasis.c, main/manager_channels.c,
+ apps/app_dial.c, main/pbx.c, main/channel_internal_api.c,
+ tests/test_stasis_channels.c (added),
+ include/asterisk/app_stasis.h, apps/app_userevent.c: Add
+ multi-channel Stasis messages; refactor Dial AMI events to Stasis
+ This patch does the following: * A new Stasis payload has been
+ defined for multi-channel messages. This payload can store
+ multiple ast_channel_snapshot objects along with a single JSON
+ blob. The payload object itself is opaque; the snapshots are
+ stored in a container keyed by roles. APIs have been provided to
+ query for and retrieve the snapshots from the payload object. *
+ The Dial AMI events have been refactored onto Stasis. This
+ includes dial messages in app_dial, as well as the core dialing
+ framework. The AMI events have been modified to send out a
+ DialBegin/DialEnd events, as opposed to the subevent type that
+ was previously used. * Stasis messages, types, and other objects
+ related to channels have been placed in their own file,
+ stasis_channels. Unit tests for some of these objects/messages
+ have also been written.
+
+2013-04-08 13:27 +0000 [r384879] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/json.h, include/asterisk/localtime.h,
+ tests/test_app_stasis.c (added), include/asterisk/frame.h,
+ apps/app_stasis.c (added), tests/test_json.c, main/json.c,
+ res/res_stasis_websocket.c (added), main/frame.c, apps/Makefile,
+ tests/test_abstract_jb.c, apps/app_stasis.exports.in (added),
+ apps/stasis_json.c (added), include/asterisk/app_stasis.h
+ (added): Stasis application WebSocket support This is the API
+ that binds the Stasis dialplan application to external Stasis
+ applications. It also adds the beginnings of WebSocket
+ application support. This module registers a dialplan function
+ named Stasis, which is used to put a channel into the named
+ Stasis app. As a channel enters and leaves the Stasis diaplan
+ application, the Stasis app receives a 'stasis-start' and
+ 'stasis-end' events. Stasis apps register themselves using the
+ stasis_app_register and stasis_app_unregister functions. Messages
+ are sent to an application using stasis_app_send. Finally, Stasis
+ apps control channels through the use of the stasis_app_control
+ object, and the family of stasis_app_control_* functions. Other
+ changes along for the ride are: * An ast_frame_dtor function
+ that's RAII_VAR safe * Some common JSON encoders for name/number,
+ timeval, and context/extension/priority Review:
+ https://reviewboard.asterisk.org/r/2361/
+
+2013-04-06 16:00 +0000 [r384857] Joshua Colp <jcolp@digium.com>
+
+ * tests/test_sorcery_astdb.c (added), res/res_sorcery_astdb.c
+ (added): Add a res_sorcery_astdb module which uses the astdb to
+ persist objects. Review: https://reviewboard.asterisk.org/r/2420/
+
+2013-04-05 20:41 +0000 [r384828] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c, UPGRADE-11.txt: Fix For Not Overriding
+ The Default Settings In chan_sip The initial report was that the
+ "nat" setting in the [general] section was not having any effect
+ in overriding the default setting. Upon confirming that this was
+ happening and looking into what was causing this, it was
+ discovered that other default settings would not be overriden as
+ well. This patch works similar to what occurs in build_peer(). We
+ create a temporary ast_flags structure and using a mask, we
+ override the default settings with whatever is set in the
+ [general] section. In the bug report, the reporter who helped to
+ test this patch noted that the directmedia settings were being
+ overriden properly as well as the nat settings. This issue is
+ also present in Asterisk 1.8 and a separate patch will be applied
+ to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina
+ Tested by: Alexandre Vezina, Michael L. Young Patches:
+ asterisk-21225-handle-options-default-prob_v4.diff Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2385/ ........ Merged
+ revisions 384827 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-04 18:15 +0000 [r384696-384760] Richard Mudgett <rmudgett@digium.com>
+
+ * main/event.c: Separate some event struct definitions from
+ instantiation.
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ UPGRADE.txt: chan_dahdi: Change inband_on_proceeding option
+ default to no/disabled. (issue ASTERISK-21151)
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /: chan_dahdi: Add
+ inband_on_proceeding compatibility option. The new
+ inband_on_proceeding option causes Asterisk to assume inband
+ audio may be present when a PROCEEDING message is received. Q.931
+ Section 5.1.2 says the network cannot assume that the CPE side
+ has attached to the B channel at this time without explicitly
+ sending the progress indicator ie informing the CPE side to
+ attach to the B channel for audio. However, some non-compliant
+ ISDN switches send a PROCEEDING without the progress indicator ie
+ indicating inband audio is available and assume that the CPE
+ device has connected the media path for listening to ringback and
+ other messages. ASTERISK-17834 which causes this issue was
+ dealing with a non-compliant network switch. (closes issue
+ ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
+ ........ Merged revisions 384685 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384689 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-03 17:17 +0000 [r384642] Matthew Jordan <mjordan@digium.com>
+
+ * /, funcs/func_channel.c: Update documentation for CHANNEL
+ function Document that you can read/write the 'accountcode' and
+ 'amaflags' on a channel. ........ Merged revisions 384640 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384641 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-03 16:01 +0000 [r384616] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: astobj2: Fix rbtree duplicate handling.
+ OBJ_PARTIAL_KEY searching a rbtree did not find all possible
+ matches if the container did not accept duplicates. Added
+ matching node bias to indicate which matching node is being
+ searched for: first, last, any.
+
+2013-04-02 17:35 +0000 [r384546] David M. Lee <dlee@digium.com>
+
+ * Makefile, /: Fixed spurious rebuilds of func_version.
+ func_version.so was being rebuilt every time, because build.h was
+ changing every build, because of the cleantest dependency that
+ was added in r384410 to fix parallel make bugs. Now build.h will
+ only be created if it does not exist, which was the original
+ behavior of the Makefile. ........ Merged revisions 384544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384545 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-02 12:18 +0000 [r384518] Joshua Colp <jcolp@digium.com>
+
+ * main/sorcery.c: Pass the object type name to the configuration
+ framework.
+
+2013-04-02 11:40 +0000 [r384514] Matthew Jordan <mjordan@digium.com>
+
+ * main/xmldoc.c, include/asterisk/app.h: Make things work again
+ Sorry folks. ',' are still greater than '|'. Thanks for playing
+ along :-)
+
+2013-04-01 20:10 +0000 [r384488] David M. Lee <dlee@digium.com>
+
+ * contrib/scripts/install_prereq: install_prereq: Build jansson
+ from source, when necessary When r383579 was committed, it made
+ Jansson a required dependency. While libjansson-dev and
+ jansson-devel are available on recent distros, some older (but
+ still supported) distros don't have it. There's a pull request[1]
+ to get it into repoforge, but that still doesn't help everyone.
+ (And helps no one until the pull request is merged and packages
+ are built). This patch adds Jansson install from source to the
+ install_unpackaged() function. There are a few gotcha's, which
+ makes this change not completely trivial. * Since Jansson may be
+ installed by a package, don't install from source if a package
+ installation can be found * libresample may also be installed via
+ package, so I added a similar check to that. * Since Jansson
+ installs into /usr/local, this patch also adds /usr/local/lib to
+ /etc/ld.so.conf.d so that the library can be found. * The
+ alternative was to install into /usr, but then it gets
+ complicated having to deal with EL's /usr/lib{32,64} shenanigans.
+ [1]: https://github.com/repoforge/rpms/pull/250 Review:
+ https://reviewboard.asterisk.org/r/2414/
+
+2013-04-01 14:44 +0000 [r384452] Matthew Jordan <mjordan@digium.com>
+
+ * main/xmldoc.c, include/asterisk/app.h: Make appropriate items
+ parse using '|' instead of ',' This patch fixes a bug introduced
+ in r76703, wherein Asterisk could only parse arguments in the
+ so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
+ syntax of NoOp,foo|bar is now parsed correctly.
+
+2013-04-01 14:10 +0000 [r384416] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_voicemail.c: Remove silly use of strncmp. ........
+ Merged revisions 384414 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-04-01 13:37 +0000 [r384412-384413] David M. Lee <dlee@digium.com>
+
+ * main/stasis.c, tests/test_stasis.c: stasis: Fixed message
+ ordering issues when forwarding This patch fixes an issue of
+ message ordering that occurs when multiple topics are forwarded
+ to an aggregator topic (such as ast_channel_topic_all()). It is
+ (very reasonably) expected that the rules governing message
+ dispatch order still apply, so long as the messages start from
+ the same thread, and are received by the same subscription.
+ Because the existing code had an additional layer of dispatching
+ via the Stasis thread pool for forwards, those promises couldn't
+ be kept. Forwarding subscriptions no longer have their own
+ mailbox, and now dispatch directly from the forwarding topic's
+ stasis_publish() call. This means that the topic's lock is held
+ for the duration of not only a message's dispatch, but the
+ dispatch of all the forwards. This shouldn't be a problem right
+ now, but if an aggregator topic had many subscribers, it could
+ become a problem. But I figure we can write more clever code when
+ the time comes, if necessary. Review:
+ https://reviewboard.asterisk.org/r/2419/
+
+ * Makefile, /: Fix parallel make problems. Occasionally, make -j
+ would fail due to missing includes, or other unusual errors. This
+ was due to the 'cleantest' target, which was designed to force a
+ make clean when some change in the code would cause the typical
+ depedency checking to fail. Several targets in the main Makefile
+ did not depend upon cleantest, hence would run in parallel to it.
+ By adding the dependency, make -j runs happily now. Review:
+ https://reviewboard.asterisk.org/r/2418/ ........ Merged
+ revisions 384410 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384411 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-30 05:15 +0000 [r384389-384390] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c: Properly format an intmax_t value
+
+ * apps/app_voicemail.c, include/asterisk/test.h, main/manager.c,
+ main/test.c: Convert TestEvent AMI events over to Stasis Core
+ This patch migrates the TestEvent AMI events to first be
+ dispatched over the Stasis-Core message bus. This helps to
+ preserve the ordering of the events with other events in the AMI
+ system, such as the various channel related events.
+
+2013-03-29 16:37 +0000 [r384327] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: app_voicemail: Add blank argument to
+ externnotify if no context argument At least one call to
+ run_externnotify provides a NULL context parameter and because
+ the snprintf statement doesn't account for a NULL context
+ parameter, it simply writes '(null)' to the arguments string
+ instead. This patch makes it write two quotes back to back for
+ that argument instead in the event of a NULL context. (closes
+ issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
+ modified from patch-20130306 uploaded by Karsten Wemheuer
+ (License 5930) ........ Merged revisions 384325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384326 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-28 23:59 +0000 [r384302] Richard Mudgett <rmudgett@digium.com>
+
+ * main/sorcery.c, main/stasis.c, main/uuid.c,
+ res/res_calendar_exchange.c, res/res_sorcery_config.c,
+ include/asterisk/uuid.h, tests/test_uuid.c: Add uuid wrapper API
+ call ast_uuid_generate_str(). * Updated test_uuid.c to test the
+ new API call. * Made system use the new API call to eliminate
+ "10's of lines" where used. * Fixed untested ast_strdup() return
+ in stasis_subscribe() by eliminating the need for it. struct
+ stasis_subscription now contains the uniqueid[] string. * Fixed
+ some issues in exchangecal_write_event(): Create uid with enough
+ space for a UUID string to avoid a realloc. Fix off by one error
+ if the calendar event provided a UUID string. There is no need to
+ check for NULL before calling ast_free().
+
+2013-03-28 15:45 +0000 [r384219-384261] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager_channels.c, main/stasis.c, apps/app_voicemail.c,
+ main/channel.c, main/pbx.c, main/stasis_cache.c,
+ include/asterisk/stasis.h, main/app.c, pbx/pbx_realtime.c,
+ include/asterisk/channel.h, tests/test_stasis.c: Break the world.
+ Stasis message type accessors should now all be named correctly.
+
+ * channels/chan_mgcp.c, channels/chan_unistim.c,
+ channels/chan_dahdi.c, include/asterisk/app.h,
+ channels/chan_sip.c, channels/chan_skinny.c, main/app.c,
+ res/res_xmpp.c, channels/chan_iax2.c, channels/sig_pri.c,
+ res/res_jabber.c: Convert MWI state message type to the new
+ stasis naming convention
+
+2013-03-27 21:52 +0000 [r384201] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/stasis.h, include/asterisk/channel.h,
+ include/asterisk/app.h: Added a doxygen group for Stasis messages
+ and topics
+
+2013-03-27 19:52 +0000 [r384164] Kinsey Moore <kmoore@digium.com>
+
+ * main/format_pref.c, /, channels/chan_sip.c: Address uninitialized
+ conditional that valgrind found ........ Merged revisions 384162
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 384163 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-27 18:52 +0000 [r384120] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/http.c: Fix a file descriptor leak in off nominal path
+ While looking at the security vulnerability in ASTERISK-20967,
+ Walter noticed a file descriptor leak and some other issues in
+ off nominal code paths. This patch corrects them. Note that this
+ patch is not related to the vulnerability in ASTERISK-20967, but
+ the patch was placed on that issue. (closes issue ASTERISK-20967)
+ Reported by: wdoekes patches:
+ issueA20967_file_leak_and_unused_wkspace.patch uploaded by
+ wdoekes (License 5674) ........ Merged revisions 384118 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384119 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-27 17:07 +0000 [r384050] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_rtp_asterisk.c: Fix white noise on SRTP decryption
+ When res_rtp_asterisk.c was altered to avoid attempting to apply
+ unprotect algorithms to non-audio RTP packets, the test used was
+ incorrect. This caused the audio packets to not be decrypted and
+ resulted in loud white noise on the other endpoint (or both
+ endpoints depending on the call legs involved). The test now
+ properly checks the version field in the RTP header to ensure
+ that RTP and RTCP are decrypted while other types of packets are
+ not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
+ Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
+ uploaded by Kinsey Moore ........ Merged revisions 384048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 384049 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-27 15:27 +0000 [r383975-384019] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/security_events.c,
+ channels/sip/include/sip.h: AST-2013-003: Prevent username
+ disclosure in SIP channel driver When authenticating a SIP
+ request with alwaysauthreject enabled, allowguest disabled, and
+ autocreatepeer disabled, Asterisk discloses whether a user exists
+ for INVITE, SUBSCRIBE, and REGISTER transactions in multiple
+ ways. The information is disclosed when: * A "407 Proxy
+ Authentication Required" response is sent instead of a "401
+ Unauthorized" response * The presence or absence of additional
+ tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)")
+ * A "401 Unauthorized" response is sent instead of "403
+ Forbidden" response after a retransmission * Retransmission are
+ sent when a matching peer did not exist, but not when a matching
+ peer did exist. This patch resolves these various vectors by
+ ensuring that the responses sent in all scenarios is the same,
+ regardless of the presence of a matching peer. This issue was
+ reported by Walter Doekes, OSSO B.V. A substantial portion of the
+ testing and the solution to this problem was done by Walter as
+ well - a huge thanks to his tireless efforts in finding all the
+ ways in which this setting didn't work, providing automated
+ tests, and working with Kinsey on getting this fixed. (closes
+ issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes,
+ kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes
+ (License 6273, 5674) ........ Merged revisions 384003 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/http.c, /: AST-2013-002: Prevent denial of service in HTTP
+ server AST-2012-014, fixed in January of this year, contained a
+ fix for Asterisk's HTTP server for a remotely-triggered crash.
+ While the fix put in place fixed the possibility for the crash to
+ be triggered, a denial of service vector still exists with that
+ solution if an attacker sends one or more HTTP POST requests with
+ very large Content-Length values. This patch resolves this by
+ capping the Content-Length at 1024 bytes. Any attempt to send an
+ HTTP POST with Content-Length greater than this cap will not
+ result in any memory allocation. The POST will be responded to
+ with an HTTP 413 "Request Entity Too Large" response. This issue
+ was reported by Christoph Hebeisen of TELUS Security Labs (closes
+ issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
+ AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-10.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-11.diff uploaded by mmichelson (License 5049)
+ ........ Merged revisions 383978 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_format_attr_h264.c: AST-2013-001: Prevent buffer
+ overflow through H.264 format negotiation The format attribute
+ resource for H.264 video performs an unsafe read against a media
+ attribute when parsing the SDP. The value passed in with the
+ format attribute is not checked for its length when parsed into a
+ fixed length buffer. This patch resolves the vulnerability by
+ only reading as many characters from the SDP value as will fit
+ into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf
+ Harnhammar patches: h264_overflow_security_patch.diff uploaded by
+ jrose (License 6182) ........ Merged revisions 383973 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-27 07:24 +0000 [r383948] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix skinny encall button to not blind
+ xfer. The softbutton endcall should not turn a transfer into a
+ blind transfer but hangup the exten being called and leave the
+ original call on hold. This does that. (closes issue
+ ASTERISK-21321) Reported by: wedhorn Tested by: snuffy, myself
+ Patches: skinny-xferendcall01.diff uploaded by wedhorn (license
+ 5019)
+
+2013-03-26 23:34 +0000 [r383925] Joshua Colp <jcolp@digium.com>
+
+ * main/sorcery.c: Remove the noop handler from sorcery so it does
+ not produce an empty value.
+
+2013-03-26 02:30 +0000 [r383841-383879] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Resolve deadlock between SIP registration
+ and channel based functions In r373424, several reentrancy
+ problems in chan_sip were addressed. As a result, the SIP channel
+ driver is now properly locking the channel driver private
+ information in certain operations that it wasn't previously. This
+ exposed two latent problems either in register_verify or by
+ functions called by register_verify. This includes: * Holding the
+ private lock while calling sip_send_mwi_to_peer. This can create
+ a new sip_pvt via sip_alloc, which will obtain the channel
+ container lock. This is a locking inversion, as any channel
+ related lock must be obtained prior to obtaining the SIP channel
+ technology private lock. Note that this issue was already fixed
+ in Asterisk 11. * Holding the private lock while calling
+ sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
+ sip_poke_peer can create a new SIP private, causing the same
+ locking inversion. Note that this locking inversion typically
+ occured when CLI commands were run while a SIP REGISTER request
+ was being processed, as many CLI commands (such as 'sip show
+ channels', 'core show channels', etc.) have to obtain the channel
+ container lock. (issue ASTERISK-21068) Reported by: Nicolas
+ Bouliane (issue ASTERISK-20550) Reported by: David Brillert
+ (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
+ ASTERISK-21296) Reported by: Gabriel Birke ........ Merged
+ revisions 383863 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383878 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cdr.c, /: Resolve deadlock between pending CDR and batch CDR
+ locks r375757 attempted to resolve a race condition between
+ multiple submissions of CDRs while in batch mode from attempting
+ to destroy the scheduled batch submission by extending the batch
+ CDR lock. Unfortunately, this causes a deadlock between the
+ pending CDR lock and the batch CDR lock. This patch resolves the
+ intent of r375757 by simply providing a new lock that protects
+ the scheduling of the batches. The original batch CDR lock is
+ kept to protect manipulation of the batch CDR settings, but has
+ been placed such that it is not held when the pending lock is
+ held. Thanks to Chase Venters for providing lock analysis on the
+ issue. (issue ASTERISK-21162) Reported by: Chase Venters ........
+ Merged revisions 383839 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383840 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-26 01:46 +0000 [r383837-383838] Russell Bryant <russell@russellbryant.com>
+
+ * channels/chan_skinny.c: Suppress compiler warning. This code
+ caused a compiler warning when --enable-dev-mode was not used.
+ The warning was that this variable was set but not used. That was
+ indeed the case as the only place this is used is as an argument
+ to SKINNY_DEBUG which is compiled out when not in dev mode.
+
+ * /, apps/app_meetme.c: Fix multi-station answer race condition.
+ When an SLA trunk is ringing (inbound call on the trunk) Asterisk
+ will make outbound calls to the stations that have that trunk. If
+ more than one station answers the call at the same time, all
+ channels other than the first one to answer are left in a bad
+ state. The channel gets leaked, is not connected to anything, and
+ there's no way to get rid of it. We now properly clean up these
+ losing channels by hanging up on them. Since they lost the race,
+ as we process their answer, there is no ringing trunk for them to
+ answer. ........ Merged revisions 383835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383836 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-25 23:25 +0000 [r383799] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, /: Set the CALLERID(dnid-num-plan) for
+ incoming ISDN calls. The CALLEDTON channel variable is set for
+ incoming ISDN calls to the lower 7 bits of the Q.931
+ type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
+ should have the same value. (closes issue ASTERISK-21248)
+ Reported by: rmudgett ........ Merged revisions 383796 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383798 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-25 20:15 +0000 [r383753-383754] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager_channels.c: Fix typo
+
+ * main/stasis.c: Fix missing ' ' around '='
+
+2013-03-25 19:28 +0000 [r383726-383747] David M. Lee <dlee@digium.com>
+
+ * contrib/scripts/install_prereq: install_prereq: removed some
+ out-of-date comments
+
+ * contrib/scripts/install_prereq: install_prereq: Adding
+ jansson-devel to RH packages
+
+ * main/channel_internal_api.c, include/asterisk/channel.h, CHANGES,
+ main/manager_channels.c, main/channel.c, main/manager.c: Move
+ NewCallerid, HangupRequest and SoftHangupRequest to Stasis
+ HangupRequest and SoftHangupRequest are now ast_channel_blob
+ Stasis messages, with the cause code as an optional field in the
+ blob. NewCallerid now simply watches for changes in the callerid
+ information in channel snapshots, and creates the AMI event
+ appropriately. Since the original NewCallerid event honored the
+ channelvars setting in manager.conf, the channel variables
+ configured there had to become a part of the channel snapshot.
+ These are now a part of every snapshot based event, making the
+ configuration description "every time a channel-oriented event is
+ emitted" less of a lie. There a a few other changes wrapped up in
+ here as well. * When ast_channel_topic() is given NULL for a
+ channel, it returns the ast_channel_topic_all() topic instead of
+ NULL. This can clean up a lot of NULL checking we're doing
+ currently. * The fields Cause and Cause-txt were removed from the
+ base channel information and put only on the Hangup events, since
+ those fields are meaningless outside of a Hangup event. * Removed
+ the pipe-delimiter processing of the channelvars field, since
+ that's been deprecated forever. (closes issue ASTERISK-21096)
+ Review: https://reviewboard.asterisk.org/r/2405/
+
+2013-03-25 12:38 +0000 [r383669] Sean Bright <sean@malleable.com>
+
+ * res/res_config_curl.c, /: Properly delimit post data in
+ res_config_curl. ........ Merged revisions 383667 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383668 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-22 20:51 +0000 [r383633] David M. Lee <dlee@digium.com>
+
+ * main/json.c, main/Makefile: Fixed another issue from r383579.
+ Core modules don't honor <depend> flags in MODULEINFO, which
+ broke jansson if specified --with-jansson to configure.
+
+2013-03-22 20:43 +0000 [r383632] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/app_mixmonitor.c: Fix StopMixMonitor Hanging Up When
+ Unable To Stop MixMonitor On A Channel A regression was
+ accidentally introduced when allowing an optional ID to be used
+ when calling StopMixMonitor. When we are unable to stop
+ MixMonitor on a channel, -1 is being returned which triggers the
+ hangup of the channel. This patch restores the prior behavior by
+ returning 0 whether we were successful or not. It also allows the
+ call from the manager to use the return code when the action
+ fails. (closes issue ASTERISK-21294) Reported by: daroz Tested
+ by: daroz Patches: asterisk-21294-stop_mixmonitor_hangingup.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2404/ ........ Merged
+ revisions 383631 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-22 19:26 +0000 [r383579-383611] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/json.h, main/asterisk.c, main/json.c: Corrected
+ some module issues introduced by r383579. When I moved res_json.c
+ to json.c, I left the MODULE_INFO stuff in there, which was
+ interesting if you ran module show. I also forgot to call what
+ was in module_load() from asterisk main().
+
+ * main/manager_channels.c (added), tests/test_json.c,
+ res/res_json.c (removed), main/pbx.c,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ apps/app_userevent.c, include/asterisk/channel.h, CHANGES,
+ include/asterisk/manager.h, main/channel.c, main/json.c (added),
+ main/manager.c, configure, res/res_json.exports.in (removed),
+ pbx/pbx_realtime.c: Move more channel events to Stasis; move
+ res_json.c to main/json.c. This patch started out simply as
+ fixing the bouncing tests introduced in r382685, but required
+ some other changes to give it a decent implementation. To fix the
+ bouncing tests, the UserEvent and Newexten AMI events needed to
+ be refactored to dispatch via Stasis. Dispatching directly to AMI
+ resulted in those events sometimes getting ahead of the
+ associated Newchannel events, which would understandably confuse
+ anyone. I found that instead of creating a zillion different
+ message types and structures associated with them, it would be
+ preferable to define a message type that has a channel snapshot
+ and a blob of structured data with a small bit of additional
+ information. The JSON object model provides a very nice way of
+ representing structured data, so I went with that. * Move JSON
+ support from res_json.c to main/json.c * Made libjansson-dev a
+ required dependency * Added an ast_channel_blob message type,
+ which has a channel snapshot and JSON blob of data. * Changed
+ UserEvent and Newexten events so that they are dispatched via
+ ast_channel_blob messages on the channel's topic. * Got rid of
+ the ast_channel_varset message; used ast_channel_blob instead. *
+ Extracted the manager functions converting Stasis channel events
+ to AMI events into manager_channel.c. (issue ASTERISK-21096)
+ Review: https://reviewboard.asterisk.org/r/2381/
+
+2013-03-22 06:32 +0000 [r383560] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix skinny voicemail indication issues.
+ Unsubscribe from MWI stasis event on channel reload. (closes
+ issue ASTERISK-21216) Reported by: wedhorn Tested by: snuffy,
+ myself Patches: skinny-mwiind02.diff uploaded by snuffy (license
+ 5024)
+
+2013-03-21 20:09 +0000 [r383541] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/stasis.h: Corrected doc error for Stasis. I
+ guess the mutex isn't necessary. Thanks, rmudgett!
+
+2013-03-21 17:41 +0000 [r383519] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: Fix astobj2 doxygen comment.
+
+2013-03-20 20:27 +0000 [r383458-383462] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_curl.c, /: Have func_curl log a warning when a curl
+ request fails. Review: https://reviewboard.asterisk.org/r/2403/
+ ........ Merged revisions 383460 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383461 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * funcs/func_curl.c, /: Minor cleanup in func_curl near hashcompat
+ code. Review: https://reviewboard.asterisk.org/r/2402/ ........
+ Merged revisions 383457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-20 16:01 +0000 [r383422] Kinsey Moore <kmoore@digium.com>
+
+ * main/stasis.c: Resolve a race condition in Stasis Because of the
+ way that topics were handled when publishing, it was possible to
+ dispatch a message to a subscription after that subscription had
+ been unsubscribed such that the dispatched message arrived at the
+ callback after the callback had received its final message. In
+ callbacks that cleaned up user data, this would often cause a
+ segfault. This has been resolved by locking the topic during the
+ entirety of dispatch. To prevent long publishing and topic
+ locking times, forwarding subscriptions have been made to be
+ standard subscriptions instead of mailboxless subscriptions which
+ were dispatched at publishing time.
+
+2013-03-20 14:52 +0000 [r383405] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sorcery_memory.c, include/asterisk/sorcery.h,
+ tests/test_sorcery.c, main/sorcery.c: Pass the sorcery instance
+ to wizards for CUD operations as well as retrieve.
+
+2013-03-19 19:07 +0000 [r383377] Kinsey Moore <kmoore@digium.com>
+
+ * main/stasis_message_router.c: Fix lock destruction/unlock
+ inversion When using scoped locks, the unref of an AO2 object
+ should happen after the unlock occurs which requires usage of
+ scoped refs.
+
+2013-03-19 16:00 +0000 [r383343] David M. Lee <dlee@digium.com>
+
+ * /, codecs/Makefile: Multiple revisions 383341-383342 ........
+ r383341 | dlee | 2013-03-19 10:57:29 -0500 (Tue, 19 Mar 2013) | 5
+ lines Removed codecs/g722/*.i on make clean ........ Merged
+ revisions 383340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r383342 | dlee | 2013-03-19 10:58:33 -0500 (Tue, 19 Mar 2013) | 1
+ line Remove codecs/speex/*.i on make clean ........ Merged
+ revisions 383341-383342 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-16 16:00 +0000 [r383284-383287] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_mgcp.c, res/res_jabber.c: Make sure things
+ compile...
+
+ * channels/sig_pri.h, main/channel.c, include/asterisk/app.h,
+ channels/chan_dahdi.c, channels/chan_skinny.c,
+ include/asterisk/xmpp.h, apps/app_minivm.c, main/app.c,
+ channels/sip/include/sip.h, main/asterisk.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, channels/chan_sip.c,
+ include/asterisk/stasis.h, res/res_xmpp.c, channels/sig_pri.c,
+ channels/chan_iax2.c, res/res_jabber.c, main/stasis.c: Transition
+ MWI to Stasis-core Remove MWI's dependency on the event system by
+ moving it to Stasis-core. This also introduces forwarding topic
+ pools in Stasis-core which aggregate many dynamically allocated
+ topics into a single primary topic. Review:
+ https://reviewboard.asterisk.org/r/2368/ (closes issue
+ ASTERISK-21097) Patch-by: Kinsey Moore
+
+2013-03-16 15:40 +0000 [r383267-383283] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c, CHANGES: Add support for using XMPP buddy state
+ via device state. This change allows you to use XMPP buddy state
+ in places where device state can be used be used, such as
+ dialplan hints. If at least one resource is available the buddy
+ is considered available. Now your phone can reflect their IM
+ status too!
+
+ * res/res_xmpp.c, /: Fix a bug where resources were not found due
+ to hashing on the priority itself. ........ Merged revisions
+ 383266 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-15 17:35 +0000 [r383225-383242] David M. Lee <dlee@digium.com>
+
+ * main/stasis.c, main/stasis_cache.c, main/stasis_message_router.c
+ (added), main/stasis_message.c,
+ include/asterisk/stasis_message_router.h (added),
+ tests/test_stasis.c: A simplistic router for stasis_message's.
+ Often times, when subscribing to a topic, one wants to handle
+ different message types differently. While one could cascade
+ if/else statements through the subscription handler, it is much
+ cleaner to specify a different callback for each message type.
+ The stasis_message_router is here to help! A
+ stasis_message_router is constructed for a particular
+ stasis_topic, which is subscribes to. Call
+ stasis_message_router_unsubscribe() to cancel that subscription.
+ Once constructed, routes can be added using
+ stasis_message_router_add() (or
+ stasis_message_router_set_default() for any messages not handled
+ by other routes). There may be only one route per
+ stasis_message_type. The route's callback is invoked just as if
+ it were a callback for a subscription; but it only gets called
+ for messages of the specified type. (issue ASTERISK-20887)
+ Review: https://reviewboard.asterisk.org/r/2390/
+
+ * configs/stasis_core.conf.sample (added): Sample config file for
+ stasis-core. (issue ASTERISK-20887)
+
+2013-03-15 13:04 +0000 [r383167-383169] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c, main/channel_internal_api.c, tests/test_stasis.c:
+ Take advantage of the fact that stasis_unsubscribe now returns
+ NULL
+
+ * main/stasis.c, main/stasis_cache.c, include/asterisk/stasis.h:
+ Make stasis unsubscription functions return NULL Unsubscribing
+ things in Asterisk seems to very commonly follow with NULLing out
+ the variable that was unsubscribed. This change makes that a bit
+ simpler.
+
+ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
+ main/http.c: tcptls: Prevent unsupported options from being set
+ AMI, HTTP, and chan_sip all support TLS in some way, but none of
+ them support all the options that Asterisk's TLS core is capable
+ of interpreting. This prevents consumers of the TLS/SSL layer
+ from setting TLS/SSL options that they do not support. This also
+ gets tlsverifyclient closer to a working state by requesting the
+ client certificate when tlsverifyclient is set. Currently, there
+ is no consumer of main/tcptls.c in Asterisk that supports this
+ feature and so it can not be properly tested. Review:
+ https://reviewboard.asterisk.org/r/2370/ Reported-by: John
+ Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........
+ Merged revisions 383165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383166 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-15 01:38 +0000 [r383122-383126] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: When a session timer expires during a
+ T.38 call, re-invite with correct SDP When a session timer
+ expires during a dialog that has re-negotiated to T.38 and
+ Asterisk is the refresher, Asterisk will send a re-INVITE with an
+ SDP containing audio media only. This causes some hilarity with
+ the poor fax session under weigh. This patch corrects that by
+ sending T.38 parameters if we are in the middle of a T.38
+ session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal
+ patches:
+ dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
+ uploaded by nbansal (License 6418) ........ Merged revisions
+ 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 383125 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * pbx/pbx_spool.c, /: Fix processing of call files when using
+ KQueue on OS X In certain situations, call files are not
+ processed when using KQueue with pbx_spool. Asterisk was sending
+ an invalid timeout value when the spool directory is empty,
+ causing the call to kevent to error immediately. This can create
+ a tight loop, increasing the CPU load on the system. (closes
+ issue ASTERISK-21176) Reported by: Carlton O'Riley patches:
+ kqueue_osx.patch uploaded by coriley (License 6473) ........
+ Merged revisions 383120 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383121 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-14 16:57 +0000 [r383063] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /: Fix whitespace in AST_EXT_LIB_CHECK
+ macro. ........ Merged revisions 383061 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 383062 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-13 14:39 +0000 [r383008] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c: Always set the RTP instance data in the
+ RTP engine Not informing the RTP engine of the instance data
+ creates shrapnel.
+
+2013-03-12 22:43 +0000 [r382989] Andrew Latham <lathama@gmail.com>
+
+ * res/res_config_ldap.c: Update Doxygen Push some cleanups upstream
+ before testing another ticket. (issue ASTERISK-20259)
+
+2013-03-12 21:19 +0000 [r382941-382954] Michael L. Young <elgueromexicano@gmail.com>
+
+ * addons/res_config_mysql.c, /: Fix Sorting Order For Parking Lots
+ Stored In Static Realtime When retrieving the parking lots from a
+ MySQL database table, the current order is "filename, cat_metric
+ desc, var_metric asc, category". If there are multiple parking
+ lots with the same cat_metric but different categories,
+ everything is being sorted on cat_metric first resulting in
+ errors when loading the parking lots. This patch fixes the
+ problem by sorting on the category field first, then the
+ cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
+ Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
+ (license 5026) ........ Merged revisions 382942 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382943 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * contrib/realtime/postgresql/realtime.sql,
+ contrib/realtime/mysql/sippeers.sql, /: Update Contributed
+ Realtime Schema Files - IPv6 Addresses This commit updates some
+ fields in the contributed realtime schema files to handle IPv6
+ addresses. (closes issue ASTERISK-21173) Reported by: Torrey
+ Searle Patches: realtime_sql.patch Torrey Searle (license 5334)
+ asterisk-21173-update-ip-fields.diff Michael L. Young (license
+ 5026) ........ Merged revisions 382939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382940 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-12 20:07 +0000 [r382924] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_xmpp.c: Fix a crash when res_xmpp is configured using
+ a username without a domain. (closes issue ASTERISK-21156)
+ Reported by: amsoft2001 ........ Merged revisions 382923 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-12 19:08 +0000 [r382900] Jason Parker <jparker@digium.com>
+
+ * res/res_rtp_asterisk.c, build_tools/menuselect-deps.in,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/Makefile, CHANGES, makeopts.in, res/pjproject (removed):
+ Switch to using external pjproject libraries. ICE/STUN/TURN
+ support in res_rtp_asterisk is also now optional.
+
+2013-03-12 16:30 +0000 [r382852] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Include the Username field in SIP
+ Registry events when Status is registered In ASTERISK-17888, the
+ AMI Registry event during SIP registrations was supposed to
+ include the Username field. Somehow, one of the events was
+ missed. This patch corrects that - the Username field should be
+ included in all AMI Registry events involving SIP registrations.
+ (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by:
+ Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov
+ (license 6479) ........ Merged revisions 382847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382848 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-12 08:55 +0000 [r382828] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix core dump on CLI usage Fix issue
+ with 'unistim show info' CLI command when device connected not
+ configured ........ Merged revisions 382827 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-11 15:22 +0000 [r382787] Kevin Harwell <kharwell@digium.com>
+
+ * channels/chan_sip.c, CHANGES, channels/sip/include/sip.h: Added
+ an option to disallow music on hold Added an option
+ "discard_remote_hold_retrieval" (default "no") that if set does
+ not trigger the music on hold event. This essentially stops
+ telling the peer to start music on hold. (issue ABE-2899)
+ Reported by: Denis Alberto Martinez Review:
+ https://reviewboard.asterisk.org/r/2336/
+
+2013-03-09 00:21 +0000 [r382764] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c,
+ apps/confbridge/conf_state_empty.c, apps/confbridge/conf_state.c,
+ apps/confbridge/conf_config_parser.c,
+ apps/confbridge/conf_state_single.c,
+ apps/confbridge/conf_state_inactive.c,
+ apps/confbridge/conf_state_single_marked.c,
+ apps/confbridge/include/confbridge.h,
+ apps/confbridge/include/conf_state.h,
+ apps/confbridge/conf_state_multi.c, apps/app_confbridge.c:
+ confbridge: Rename items for clarity and consistency. struct
+ conference_bridge_user -> struct confbridge_user struct
+ conference_bridge -> struct confbridge_conference struct
+ conference_state -> struct confbridge_state struct
+ conference_bridge_user *conference_bridge_user -> struct
+ confbridge_user *user struct conference_bridge_user *cbu ->
+ struct confbridge_user *user struct conference_bridge
+ *conference_bridge -> struct confbridge_conference *conference
+ The names are now generally shorter, consistently used, and don't
+ conflict with the struct names. This patch handles the renaming
+ part of the issue. (issue ASTERISK-20776) Reported by: rmudgett
+
+2013-03-08 20:26 +0000 [r382746] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Update the via header when
+ relaying SMS MESSAGE Prior to this change, certain conditions for
+ sending the message would result in an address of '(null)' being
+ used in the via header of the SIP message because a NULl value of
+ pvt->ourip was used when initially generating the via header.
+ This is fixed by adding a call to build_via when the address is
+ set before sending the message. (closes issue ASTERISK-21148)
+ Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch
+ uploaded by Zhi Cheng (license 6475) ........ Merged revisions
+ 382739 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-08 16:59 +0000 [r382721-382724] David M. Lee <dlee@digium.com>
+
+ * main/stasis_cache.c, include/asterisk/stasis.h: Stasis
+ documentation updates. (issue ASTERISK-20887) (issue
+ ASTERISK-20959)
+
+ * main/stasis.c, main/channel.c, main/channel_internal_api.c:
+ Ensure dummy channels get a stasis topic. Fixes test failure
+ introduced in r382685. (issue ASTERISK-20887) (issue
+ ASTERISK-20959)
+
+2013-03-08 16:00 +0000 [r382705] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_stasis.c, main/stasis_cache.c,
+ include/asterisk/stasis.h: Add message dump capability to stasis
+ cache layer The cache dump mechanism allows the developer to
+ retreive multiple items of a given type (or of all types) from
+ the cache residing in a stasis caching topic in addition to the
+ existing single-item cache retreival mechanism. This also adds to
+ the caching unit tests to ensure that the new cache dump
+ mechanism is functioning properly. Review:
+ https://reviewboard.asterisk.org/r/2367/ (issue ASTERISK-21097)
+
+2013-03-08 15:15 +0000 [r382685] David M. Lee <dlee@digium.com>
+
+ * main/stasis_message.c (added), main/manager.c,
+ main/asterisk.exports.in, include/asterisk/channel_internal.h,
+ main/channel_internal_api.c, include/asterisk/stasis.h (added),
+ include/asterisk/channel.h, tests/test_stasis.c (added),
+ main/asterisk.c, main/stasis.c (added), main/channel.c,
+ main/stasis_cache.c (added), main/pbx.c: This patch adds a new
+ message bus API to Asterisk. For the initial use of this bus, I
+ took some work kmoore did creating channel snapshots. So rather
+ than create AMI events directly in the channel code, this patch
+ generates Stasis events, which manager.c uses to then publish the
+ AMI event. This message bus provides a generic publish/subscribe
+ mechanism within Asterisk. This message bus is: - Loosely
+ coupled; new message types can be added in seperate modules. -
+ Easy to use; publishing and subscribing are straightforward
+ operations. In addition to basic publish/subscribe, the patch
+ also provides mechanisms for message forwarding, and for message
+ caching. (issue ASTERISK-20887) (closes issue ASTERISK-20959)
+ Review: https://reviewboard.asterisk.org/r/2339/
+
+2013-03-08 04:11 +0000 [r382670-382671] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Remove unused function After r382670,
+ get_ip_and_port_from_sdp was no longer used.
+
+ * channels/chan_sip.c: Don't reset the RTP address on a glare
+ re-INVITE Originally, way back in r201583, we added the alternate
+ RTP address so that the RTP engine would expect to receive audio
+ from a new source when a glare re-INVITE occurred. In r382589, we
+ remove the alternate RTP source, as the 'secret' probation mode
+ allows for switching to a new RTP source when a previous source
+ stops sending RTP. At the time, it seemed appropriate to set the
+ RTP source based on the information in the glared re-INVITE.
+ Unfortunately, that doesn't work so well - in a glared re-INVITE
+ that occurs with no SDP - such as in a connected line update that
+ glances - we'll set the RTP source to an invalid address. In
+ subsequent re-INVITE requests from this Asterisk instance, we'll
+ then send an invalid media address, which will result in the
+ remote side sending a 488. Whoops. There isn't any need to reset
+ the RTP source - if we're using strictrtp, we'll simply
+ synchronize to a new source when we stop getting packets from the
+ old one. If we aren't using strictrtp, then again there shouldn't
+ be a problem. Note that the Asterisk Test Suite's connectedline
+ test caught this error.
+
+2013-03-07 21:55 +0000 [r382648] David M. Lee <dlee@digium.com>
+
+ * main/threadpool.c: Changing log level of "Not changing threadpool
+ size" from notice to debug.
+
+2013-03-07 21:14 +0000 [r382636] Jason Parker <jparker@digium.com>
+
+ * res/res_sorcery_config.c, res/res_sorcery_memory.c: Load sorcery
+ modules earlier, so they can actually be used.
+
+2013-03-07 19:14 +0000 [r382621] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_voicemail.c: Let vm_mailbox_snapshot combine "Urgent"
+ when no folder is specified r381835 fixed a bug in
+ vm_mailbox_snapshot where combining INBOX and Old forgot that
+ Urgent also "counts" as new messages. This fixed the problem when
+ any of the three folders was specified and the combine option was
+ used. It missed the case where the folder isn't specified and we
+ build a snapshot of all folders. This patch corrects that.
+ ........ Merged revisions 382617 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-07 16:48 +0000 [r382600-382604] Kinsey Moore <kmoore@digium.com>
+
+ * main/xmldoc.c: Fix a memory leak in xmldoc Another instance of
+ attribute retrieval not being freed properly.
+
+ * main/xmldoc.c: Resolve more memory leaks in xmldoc Many places
+ that allocated to pull out an attribute are now freed properly.
+
+2013-03-07 15:48 +0000 [r382589] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c: Add a 'secret'
+ probation strictrtp mode to handle delayed changes in RTP source
+ Often, Asterisk may realize that a change in the source of an RTP
+ stream is about to occur and ask that the RTP engine reset it's
+ lock on the current RTP source. In certain scenarios, it may take
+ awhile for the new remote system to send RTP packets, while the
+ old remote system may continue providing RTP during that time
+ period. This causes Asterisk to re-lock onto the old source,
+ thereby rejecting the new source when the old source stops
+ sending RTP and the new source begins. This patch prevents that
+ by having a constant secondary, 'secret' probation mode enabled
+ when an RTP source has been chosen. RTP packets from other
+ sources are always considered, but never chosen unless the
+ current RTP source stops sending RTP. Review:
+ https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124)
+ Reported by: John Bigelow Tested by: John Bigelow (closes issue
+ AST-1125) Reported by: John Bigelow Tested by: John Bigelow
+ ........ Merged revisions 382573 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-07 15:36 +0000 [r382489-382587] Kinsey Moore <kmoore@digium.com>
+
+ * main/xmldoc.c: Fix minor memory leak in xmldoc Strings retrieved
+ via ast_xml_get_text() must be freed with ast_xml_free_text().
+
+ * /, main/logger.c: Ensure that logmsgs are freed properly Messages
+ sent while the logger thread is shutting down will now have their
+ associated callid freed properly. ........ Merged revisions
+ 382574 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/threadpool.c: Fix ref leak in threadpool.c If
+ ast_threadpool_set_size with a size equal to the current size, a
+ reference to a set_size_data structure would be leaked.
+
+ * main/threadpool.c: Resolve a ref leak in threadpool.c Ownership
+ of the listener reference is not transferred because the listener
+ is reffed when placed into the taskprocessor. Ensure that the
+ listener is dereffed properly.
+
+2013-03-05 13:14 +0000 [r382440] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c,
+ configs/res_ldap.conf.sample,
+ contrib/realtime/postgresql/realtime.sql,
+ configs/sip.conf.sample, CHANGES,
+ contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, channels/sip/include/sip.h,
+ CREDITS: Add RFC 3327 Path header support to chan_sip This patch
+ adds support for RFC 3327 "Path" headers. This can be enabled in
+ sip.conf using the 'supportpath' setting, either on a global
+ basis or on a peer basis. This setting enables Asterisk to route
+ outgoing out-of-dialog requests via a set of proxies by using a
+ pre-loaded route-set defined by the Path headers in the REGISTER
+ request. This patch also adds Realtime support for dynamically
+ updating the Path information for a peer. A huge thank-you to
+ Klaus Darillion and Olle E Johansson for their efforts in writing
+ this patch. Review: https://reviewboard.asterisk.org/r/2235/
+ Review: https://reviewboard.asterisk.org/r/991/ (closes issue
+ ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej,
+ mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000
+ (License 5054) oolong-path-support-trunk in team branch by oej
+ (License 5267)
+
+2013-03-05 03:53 +0000 [r382411] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix several unreleased mutex locks
+ that cause problem with processing calls Reported by: Daniel
+ Bohling Tested by: Daniel Bohling (Closes issue ASTERISK-21119)
+ ........ Merged revisions 382409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382410 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-04 21:15 +0000 [r382392] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/format_cap.h, main/bridging.c: Fixup some bridge
+ and format capabilities comments and whitespace.
+
+2013-03-04 21:14 +0000 [r382391] Jason Parker <jparker@digium.com>
+
+ * /, main/event.c: Fix comparison of presence state in event
+ subsystem. Several new IEs were not given types (or names),
+ causing the comparison function to improperly succeed. This adds
+ those. (closes issue AST-1128) ........ Merged revisions 382390
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-04 20:18 +0000 [r382386] Kevin Harwell <kharwell@digium.com>
+
+ * /, apps/app_confbridge.c: Confbridge CLI new record file name
+ check. This fix checks to make sure that if a confbridge record
+ start command is issued from the CLI it will always use the file
+ name given on the CLI even if it changes between start/stop
+ records for a conference. Previously it had been reusing the same
+ file between start/stops even if a new filename was given. (issue
+ AST-1088) Reported by: John Bigelow ........ Merged revisions
+ 382385 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-03-01 18:01 +0000 [r382340] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/sorcery.h, tests/test_sorcery.c, main/sorcery.c:
+ Add support for registering a sorcery handler which supports
+ multiple fields using a regex. Review:
+ https://reviewboard.asterisk.org/r/2332/
+
+2013-03-01 04:32 +0000 [r382323] Michael L. Young <elgueromexicano@gmail.com>
+
+ * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
+ contrib/realtime/postgresql/realtime.sql, CHANGES: Fix / Clean Up
+ Some Items To Handle The New auto_* NAT Options The original
+ report had to do with a realtime peer behind NAT being pruned and
+ the peer's private address being used instead of its external
+ address. Upon debugging, it was discovered that this was being
+ caused by the addition of the auto_force_rport and auto_comedia
+ settings. This patch does the following: * Adds a missing note to
+ the CHANGES file indicating that the default global nat setting
+ is auto_force_rport * Constify the 'req' parameter for
+ check_via() * Add calls to check_via() in a couple of places in
+ order for the auto_* settings to do their job in attempting to
+ determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT
+ and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use
+ where it was needed * Moves the copying of peer flags up in
+ build_peer() to before they are used; this fixes the realtime
+ prune issue * Update the contrib/realtime schemas to allow the
+ nat column to handle the different nat setting combinations we
+ have This patch received a review and "Ship It!" on the issue
+ itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested
+ by: JoshE, Michael L. Young Patches:
+ asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young
+ (license 5026) ........ Merged revisions 382322 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-28 21:59 +0000 [r382297-382299] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, /: While the ICE negotiation is occurring
+ leave strictrtp in an open state, media can and will come from
+ different places. ........ Merged revisions 382298 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_rtp_asterisk.c, /: Fix a bug with ICE and strictrtp where
+ media could get dropped. If the end result of the ICE negotiation
+ resulted in the path for media changing it was possible for the
+ strictrtp code to discard the RTP packets. This change causes
+ strictrtp to enter learning mode once again when the ICE
+ negotiation has completed successfully. ........ Merged revisions
+ 382296 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-28 21:31 +0000 [r382294-382295] Richard Mudgett <rmudgett@digium.com>
+
+ * main/threadpool.c: threadpool: Make ast_threadpool_push() return
+ -1 if shutting_down
+
+ * main/threadpool.c, include/asterisk/threadpool.h: threadpool:
+ Whitespace and comment corrections.
+
+2013-02-28 21:21 +0000 [r382292] Jason Parker <jparker@digium.com>
+
+ * res/res_rtp_asterisk.c, include/asterisk.h: Don't undefine
+ bzero()/bcopy(). This was causing build failures against external
+ libraries that happened to use them, unless silly hacks were
+ added to the modules that used those headers. Review:
+ https://reviewboard.asterisk.org/r/2359/
+
+2013-02-28 17:17 +0000 [r382232-382236] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
+ attempting to set caller ID A deadlock can occur in chan_iax2
+ when it attempts to set the caller ID, as it already holds the
+ iax2 private lock and improperly fails to obtain the channel lock
+ before calling ast_set_callerid. By not safely obtaining the
+ channel lock, a locking inversion can take place, causing a
+ deadlock. This patch solves this by calling the required deadlock
+ avoidance functions that obtain the channel lock before setting
+ the caller ID. Thanks to Pavel for fixing my syntax errors and
+ testing this patch out. (closes issue ASTERISK-21128) Reported
+ by: Pavel Troller Tested by: Pavel Troller patches:
+ ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
+ ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
+ (license 6302) ........ Merged revisions 382233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382234 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_meetme.c, CHANGES: Let channels joining a MeetMe
+ conference opt out of the denoiser For some channel drivers,
+ specifically those that have a varying rate in the number of
+ audio samples, the audio quality for a MeetMe conference can be
+ exceedingly poor. This is due to a unilateral application of the
+ DENOISE function in func_speex to channels joining the
+ conference. The denoiser function in the speex library is
+ initialized with the number of audio samples in each sample that
+ will be provided to it. If the number of audio samples changes,
+ the denoiser has to be thrown away and re-initialized. While this
+ could be worked around by removing func_speex, that doesn't help
+ if you actually use the denoiser with other channels on the
+ system. This patches does the following: * Checks for the
+ presence of func_speex as opposed to codec_speex when determining
+ if the DENOISE function is present (which is where the function
+ is actually implemented) * Adds an option to MeetMe 'n' that
+ causes the denoiser to not be applied to a channel when it joins.
+ This keeps the current behavior the default, but let's users
+ disable the denoiser if it causes problems on their system.
+ Review: https://reviewboard.asterisk.org/r/2358 (closes issue
+ AST-1062) Reported by: Thomas Arimont ........ Merged revisions
+ 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 382230 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-27 20:31 +0000 [r382203-382204] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_skinny.c: More places to eliminate the cast to argv
+ but were not giving warnings.
+
+ * channels/chan_skinny.c: Fix compiler warning by eliminating the
+ need for a cast.
+
+2013-02-27 16:19 +0000 [r382182] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Relax dialog checking in
+ get_sip_pvt_byid_locked so it works when the dialog is forked.
+ (closes issue ASTERISK-20638) Reported by: eelcob Patches:
+ pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
+ 6442) ........ Merged revisions 382171 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382174 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-26 20:05 +0000 [r382113] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: Consider linux-gnuspe as linux-gnu *
+ The powerpcspe Linux port uses linux-gnuspe as the OS string. *
+ Our build system shouldn't really care for that, so just call it
+ linux-gnu. * Original report: Roland Stigge ,
+ http://bugs.debian.org/701505 Review:
+ https://reviewboard.asterisk.org/r/2357/ ........ Merged
+ revisions 382110 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382111 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-26 19:36 +0000 [r382109] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Correct RPID parsing for unquoted
+ display-name. Parsing Remote-Party-ID will now succeed if
+ display-name is of the *(token LWS) kind and not just the
+ quoted-string kind. Review:
+ https://reviewboard.asterisk.org/r/2341/ ........ Merged
+ revisions 382107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382108 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-26 19:29 +0000 [r382106] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, main/Makefile: Remove unneeded linux-gnueabi* As of r380522
+ the configure scripts converts the value of linux-gnueabi* of
+ OSARCH to "linux-gnu". So no point in testing for those values.
+ ........ Merged revisions 382087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382096 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-26 15:52 +0000 [r382067-382070] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_confbridge.c: Clean up ConfBridge commands to account
+ for wait_marked users When ConfBridge was refactored to better
+ handle the concept of marked, wait_marked, and normal users
+ co-existing in a conference (thereby implementing a state machine
+ for the conference), the wait_marked users were put into their
+ own list of conference participants, separate from the active
+ users. This list is used for wait_marked users when they are
+ waiting in a conference but no marked user has joined; normal
+ users may have joined at this point however. There are several
+ AMI/CLI commands that affect conference users that were not
+ checking the wait_marked users list: * CLI/AMI commands that
+ mute/unmute a participant. In this case, wait_marked users have
+ to remain in their particular state and should not be affected -
+ however, the commands would return "Channel not found" as opposed
+ to the appropriate error condition. * CLI/AMI commands that kick
+ a participant. An admin should always be able to kick a
+ participant out of the conference. This patch fixes both sets of
+ commands, and cleans up the CLI commands slightly by allowing
+ them to complete a participant name (this was supposed to have
+ been added, but the function call was commented out and wasn't
+ implemented). Review: https://reviewboard.asterisk.org/r/2346/
+ (closes issue AST-1114) Reported by: John Bigelow Tested by: John
+ Bigelow ........ Merged revisions 382068 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample, /: Ensure that the default
+ bridge/user profiles are always available ConfBridge and Page
+ require that there always be a default bridge and user profile
+ available. While properties of the default profiles can be
+ overriden in the configuration file, removing them can create
+ situations where neither application can function properly. This
+ patch ensures that if an administrator removes the profiles from
+ the confbridge.conf configuration file, the profiles are added
+ upon load. Documentation clarifying this has been added to the
+ confbridge.conf.sample file. Review:
+ https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115)
+ Reported by: John Bigelow Tested by: John Bigelow ........ Merged
+ revisions 382066 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-25 12:51 +0000 [r382023] Matthew Jordan <mjordan@digium.com>
+
+ * addons/res_config_mysql.c, /: Clean up use of va_end/va_args in
+ res_config_mysql There were several problems using variadic
+ argument macros in res_config_mysql. * Improper use of va_end.
+ Multiple calls to va_end were possible resulting in an unbalanced
+ matching of va_start/va_end. * Calls to va_arg after a possible
+ encounter of a SENTINEL value. This patch corrects those errors.
+ (closes issue ASTERISK-19451) Reported by: wdoekes patches:
+ ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
+ ........ Merged revisions 382021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 382022 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-25 07:09 +0000 [r382007-382008] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: More called details fixup for skinny.
+ Basically sets the callerid and callername to the first device
+ talked to for the purposes of putting the the calls made log on
+ the device. Does not affect the device displaying who the device
+ is currently talking to. Also some minor changes to use
+ sub->exten in lieu of l->lastnumberdialed. (closes issue
+ ASTERISK-21095) Reported by: wedhorn Tested by: snuffy, myself
+ Patches: skinny-calllogsoutbound03.diff uploaded by wedhorn
+ (license 5019)
+
+ * channels/chan_skinny.c: Add prinotify messages to skinny. Adds
+ both fixed and variable prinotify messages and clearprinotify
+ messages to skinny. Also adds cli function for pushing messages
+ to devices. i Initial code by snuffy, expanded by myself to
+ include fixed messages. (closes issue ASTERISK-21091) Reported
+ by: snuffy Tested by: snuffy, myself Patches:
+ skinny-prinotify02.diff uploaded by wedhorn (license 5019)
+
+2013-02-24 23:01 +0000 [r381918-381977] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_jingle.c, /: Set the sin_family on the bind address
+ socket during initialization Somehow, chan_jingle has managed to
+ operate for years without setting the sin_family on its bindaddr
+ socket. This patch properly sets the field during initial module
+ load to AF_INET. Note that the patch on the issue was modified
+ slightly to change the initialization of the socket from
+ allocation of a chan_jingle private to the module initialization,
+ as the bindaddr object (which is static) only needs to have the
+ address set once. (closes issue ASTERISK-19341) Reported by:
+ andre valentin patches: 0105-chan_jingle.patch uploaded by
+ avalentin (License 6064) ........ Merged revisions 381975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381976 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /: Don't display the AMI ALL class authorization
+ for users if they don't have it When converting AMI class
+ authorizations to a string representation, the method always
+ appends the ALL class authorization. This is especially important
+ for events, as they should always communicate that class
+ authorization - even if the event itself does not specify ALL as
+ a class authorization for itself. (Events have always assumed
+ that the ALL class authorization is implied when they are raised)
+ Unfortunately, this did mean that specifying a user with
+ restricted class authorizations would show up in the 'manager
+ show user' CLI command as having the ALL class authorization.
+ Rather then modifying the existing string manipulation function,
+ this patch adds a function that will only return a string if the
+ field being compared explicitly matches class authorization field
+ it is being compared against. This prevents ALL from being
+ returned unless it is actually specified for the user. (closes
+ issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged
+ revisions 381939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381943 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_parkandannounce.c, /: Make ParkAndAnnounce return to
+ priority + 1 when return context is not defined The
+ ParkAndAnnounce application documentation for the optional
+ return_context parameter states the following: return_context The
+ goto-style label to jump the call back into after timeout.
+ Default 'priority+1'. Unfortunately, the application was sending
+ the channel back into the dialplan at 'priority', which is the
+ ParkAndAnnounce application call. This causes an infinite loop of
+ the channel constantly being parked, announced, timed out,
+ parked, announced, timed out... while fun, especially for those
+ callers you wish to drive to the end of madness, this was not the
+ intent of the application. (closes issue ASTERISK-20113) Reported
+ by: serginuez patches: app_parkandannounce.diff uploaded by
+ serginuez (License 6405) ........ Merged revisions 381916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381917 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-22 19:40 +0000 [r381894] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, res/res_agi.c: Fix FastAGI To Properly Check For A Connection
+ When IPv6 support was added to FastAGI, the intent was to have
+ the ability to check all addresses resolved for a host since we
+ might receive an IPv4 address and an IPv6 address. The problem
+ with the current code, is that, since we are doing O_NONBLOCK, we
+ get EINPROGRESS when calling ast_connect() but are ignoring this
+ instead of handling it. We break out of the loop and continue on.
+ When we later call ast_poll(), it succeeds but we never check if
+ we have a connection or not on the socket level. We then attempt
+ to send data to the host address that we think is setup and it
+ fails. We then check the errno and see that we have "connection
+ refused" and then return with agi failed. This patch does the
+ following: * Handles EINPROGRESS by creating the function
+ handle_connection() - ast_poll() was moved into this function -
+ This function checks the results of the connection on the socket
+ level after calling ast_poll() * Continues to the next address if
+ the above fails to create a connection * Once all addresses
+ resolved are tried and we still are unable to establish a
+ connection, then we return that the FastAGI call failed (closes
+ issue ASTERISK-21065) Reported by: Jeremy Kister Tested by:
+ Jeremy Kister, Michael L. Young Patches:
+ asterisk-21065_poll_correctly_v4.diff Michael L. Young (license
+ 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........
+ Merged revisions 381893 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-22 15:51 +0000 [r381881] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c, /: app_dial: Honor the 'c' flag when the calling
+ party hangs up Apparently this feature became broken in 11,
+ probably as a result of the Hangup Cause project. (closes issue
+ ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch
+ uploaded by Heiko Wundram (license 5822) ........ Merged
+ revisions 381880 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-22 01:52 +0000 [r381869] Matthew Jordan <mjordan@digium.com>
+
+ * configure.ac, /, configure: Properly detect launchd Asterisk was
+ a little too pro-active in claiming that it found launchd. On
+ systems without launchd - such as FreeBSD - this resulted in
+ certain items in Asterisk that conflict with launchd to not be
+ selectable, such as res_timing_kqueue. (closes issue
+ ASTERISK-20749) Reported by: Oleg Baranov ........ Merged
+ revisions 381847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381848 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-19 19:47 +0000 [r381792] Kevin Harwell <kharwell@digium.com>
+
+ * main/features.c: Write the correct callid to the data1 field in
+ queue_log for transfer events. The incorrect callid was being
+ written to the "data1" field in queue_log table for transfer
+ events. The callid of the queue was being written instead of the
+ transfer target's callid. This now gets the correct "transfer to"
+ number and places that in the "data1" field of the queue_log
+ table when a transfer event is triggered. (closes issue
+ ASTERISK-19960) Reported by: vladimir shmagin ........ Merged
+ revisions 381770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381791 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-19 17:17 +0000 [r381749] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_motif.c, include/asterisk/module.h,
+ res/snmp/agent.c, main/loader.c, main/cli.c: Add The Status Of A
+ Module To The Output Of "CLI> module show" When a module's
+ configuration is not loadable, we still load the module but it is
+ not in a running state. When trying to troubleshoot, let's say,
+ why chan_motif is ignoring inbound XMPP traffic, there is no way
+ to indicate that a loaded module is not currently running.
+ (closes issue ASTERISK-21108) Reported by: Rusty Newton Tested
+ by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2331/
+
+2013-02-19 16:23 +0000 [r381729-381741] Kevin Harwell <kharwell@digium.com>
+
+ * apps/app_confbridge.c: Confbridge channels staying active when
+ all participants leave. If you started/stopped recording of a
+ conference multiple times channels would remain active even when
+ all participants left the conference. This was due to the fact
+ that a reference to the confbridge was being added every time a
+ start record command was issued, but when the recording was
+ stopped there was no matching de-reference thus keeping the
+ conference alive. Made sure only a single reference is added for
+ the record thread no matter how many times recording is
+ started/stopped. A de-reference is issued upon thread ending.
+ Note, this issue is being fixed under AST-1088 since it relates
+ to it and should have been corrected along with those
+ modifications. (issue AST-1088) Reported by: John Bigelow
+ ........ Merged revisions 381737 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_config_parser.c,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
+ CHANGES: Added Confbridge record_file_append option. Currently,
+ if one starts, stops, and then starts a recording again for a
+ conference the recorded data is appended to the file originally
+ created on the first record start. An option record_file_append
+ has been added that defaults to "yes", but when set to "no" will
+ force creation of a new file between every record start/stop.
+ (issue AST-1088) Reported by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/374/
+
+2013-02-19 06:54 +0000 [r381717-381718] Damien Wedhorn <voip@facts.com.au>
+
+ * configs/skinny.conf.sample, channels/chan_skinny.c: Add
+ serviceURL stuff to skinny. Patch adds all the packet and
+ structure stuff to skinny to enable setting service URLs in
+ skinny, such as corporate directories. This stuff is only
+ relevant during load/unload as when activated. Also some minor
+ changes removing duplicated counting of addons and speedials in
+ handle_skinny_show_devices. Review:
+ https://reviewboard.asterisk.org/r/2321/
+
+ * channels/chan_skinny.c: Fixup skinny CLI completion. Auto
+ complete for skinny debug allows multiple options and negation,
+ also add debug all option. Usage example: 'skinny debug all
+ -packets' (each can be autocompleted including -packet). Change
+ show device to use device name. Remove the duplicate ast_strdup's
+ from place calling device complete return immediately from
+ complete devicename and complete linename so that multiple
+ options are displayed on the CLI if more than one option
+ available. Review: https://reviewboard.asterisk.org/r/2333/
+
+2013-02-18 22:23 +0000 [r381703] Kevin Harwell <kharwell@digium.com>
+
+ * /, apps/app_confbridge.c: Fixed Confbridge file recording
+ deadlock and appending. A deadlock occurred after
+ starting/stopping and then restarting a confbridge recording.
+ Upon starting a recording a record thread is created that holds a
+ lock until just before exiting. Stopping the recording does not
+ stop/exit the thread or release the lock. The thread waits until
+ recording begins again. Starting a stopped recording signals the
+ thread to continue and start recording again. However restarting
+ the recording also created another record thread resulting in a
+ deadlock. The fix was to make sure the record thread was only
+ created once. Also it was noted that filenames for the recordings
+ were being concatenated for each start/stop. This was fixed by
+ creating a new file for each conference session and appending the
+ actual recorded data within the file (e.g. passing the 'a' option
+ to MixMonitor). (issue AST-1088) Reported by: John Bigelow
+ Review: http://reviewboard.digium.internal/r/374/ ........ Merged
+ revisions 381702 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-18 20:31 +0000 [r381670] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, configs/sip.conf.sample: Remove "registertrying" and add
+ "rtp_engine" from/to sip.conf.sample The "registertrying" option
+ was removed in r343220. The "rtp_engine" option was added in
+ r186078 but erroneously named "engine" in the sample. Note that
+ there is no global sip setting for a different engine. ........
+ Merged revisions 381668 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381669 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-18 19:48 +0000 [r381656] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_presencestate.c, /: PRESENCE_STATE: Provide better
+ documentation for the 'e' option. Notes that the 'e' option
+ actually decodes data when used as a write function such as with
+ the SET application while it encodes data when used to read.
+ Review: https://reviewboard.asterisk.org/r/2335/ ........ Merged
+ revisions 381655 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-18 19:12 +0000 [r381644] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c: confbridge: Add flags column to CLI
+ "confbridge list <conference>" * Added the following flags to the
+ CLI "confbridge list <conference>" output: A - The user is an
+ admin M - The user is a marked user W - The user must wait for a
+ marked user to join E - The user will be kicked after the last
+ marked user leaves the conference w - The user is waiting for a
+ marked user to join * Added the following header to the AMI
+ ConfbridgeList events: WaitMarked, EndMarked, and Waiting.
+ (closes issue AST-1101) Reported by: John Bigelow Patches:
+ confbridge-show-admin3.txt (license #5091) patch uploaded by John
+ Bigelow Modified
+
+2013-02-16 20:44 +0000 [r381628] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c: confbridge: Rename i iterator variables to
+ iter.
+
+2013-02-16 16:28 +0000 [r381615] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Don't send presencestate information if
+ the state is invalid Previously, presencestate information was
+ sent whenever the state was not NOT_SET. When r381594 actually
+ returned INVALID presence state in all the places it was supposed
+ to, it caused chan_sip to start adding presence state information
+ to NOTIFY requests that it previously would not have added.
+ chan_sip shouldn't be adding presence state information when the
+ provider is in an invalid state; users can't set the state to
+ invalid and an invalid state always implies that the provider is
+ in an error condition. (issue AST-1084) ........ Merged revisions
+ 381613 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-16 16:24 +0000 [r381614] Joshua Colp <jcolp@digium.com>
+
+ * main/sorcery.c, res/res_sorcery_config.c,
+ res/res_sorcery_memory.c, include/asterisk/sorcery.h,
+ tests/test_sorcery.c: Add support for retrieving multiple objects
+ from sorcery using a regex on their id. Review:
+ https://reviewboard.asterisk.org/r/2329/
+
+2013-02-15 23:29 +0000 [r381595] Matthew Jordan <mjordan@digium.com>
+
+ * main/presencestate.c, funcs/func_presencestate.c, main/manager.c,
+ /: Fix crash in PresenceState AMI action when specifying an
+ invalid provider This patch fixes a crash in Asterisk that could
+ be caused by using the PresenceState AMI action while providing
+ an invalid provider. This patch also adds some additional
+ warnings when a user attempts to provide the PresenceState action
+ with invalid data, and removes some NOTICE statements that were
+ still lurking in the code from testing. (closes issue AST-1084)
+ Reported by: John Bigelow Tested by: John Bigelow ........ Merged
+ revisions 381594 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-15 18:51 +0000 [r381568] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix a crash that occurred when a BYE was
+ received on a replaced dialog. Reference counting for the channel
+ and its tech_pvt got messed up at some point between 1.8 and 11.
+ The result was that if a BYE for a dialog that had been replaced
+ (via an INVITE with Replaces) was received, Asterisk would crash
+ due to trying to access data on a channel that was no longer
+ there. The fix I introduced is to remove code that both unrefs
+ the sip_pvt and sets the channel's tech_pvt to NULL when an
+ INVITE with Replaces is handled. This way when a BYE is received,
+ the tech_pvt will be non-NULL and so the BYE can be processed and
+ not cause a crash. (closes issue ASTERISK-20929) reported by
+ Kristopher Lalletti patches: ASTERISK-20929.patch uploaded by
+ Mark Michelson (License #5049) ........ Merged revisions 381566
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-15 18:44 +0000 [r381567] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/sorcery.h, main/config_options.c,
+ main/sorcery.c: Disable strict XML documentation config checking;
+ fix crash caused by sorcery This patch does two things: 1. It
+ disables (temporarily) strict XML documentation checking for
+ module configurations. We should re-enable it before making any
+ release from trunk. 2. Pass the module flag AST_MODULE through
+ sorcery. This means several of the API calls are now macros and
+ will do this automatically for you. The config framework needs
+ the module that objects are registering to so it can properly
+ construct the documentation. (This was already a required field,
+ but sorcery was getting by without it)
+
+2013-02-15 17:38 +0000 [r381557] Kevin Harwell <kharwell@digium.com>
+
+ * include/asterisk/logger.h, main/autoservice.c, main/logger.c:
+ Stopped spamming of debug messages during attended transfer.
+ While autoservice is running and servicing a channel the callid
+ is being stored and removed in the thread's local storage for
+ each iteration of the thread loop. If debug was set to a
+ sufficient level the log file would be spammed with callid thread
+ local storage debug messages. Added a new function that checks to
+ see if the callid to be stored is different than what is already
+ contained (if anything). If it is different then store/replace
+ and log, otherwise just leave as is. Also made it so all logging
+ of debug messages pertaining to the callid thread storage outputs
+ only when TEST_FRAMEWORK is defined. (issue ASTERISK-21014)
+ (closes issue ASTERISK-21014) Report by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/2324/
+
+2013-02-15 17:33 +0000 [r381556] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Use video and text crypto
+ attributes to append RTP profiles to SDP Some bad copy/pasting
+ resulted in using the audio crypto attribute for both text and
+ video RTP. Also the audio crypto isn't set until after these, so
+ it was really just bad all around. (closes ASTERISK-20905)
+ Reported by: Kristopher Lalletti patches:
+ rtp_crypto_video_text.diff uploaded by Jonathan Rose (license
+ 6182) ........ Merged revisions 381553 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-15 15:26 +0000 [r381527-381543] Matthew Jordan <mjordan@digium.com>
+
+ * /: Remove automerge propertrties added in r381527
+
+ * main/config_options.c, doc/appdocsxml.dtd, main/asterisk.c,
+ main/xmldoc.c, main/udptl.c, include/asterisk/xml.h, /,
+ main/xml.c, include/asterisk/_private.h, res/res_xmpp.c,
+ main/named_acl.c, configs/motif.conf.sample,
+ apps/confbridge/conf_config_parser.c, Makefile,
+ include/asterisk/config_options.h, configs/xmpp.conf.sample,
+ apps/app_skel.c, channels/chan_motif.c,
+ include/asterisk/xmldoc.h: Add CLI configuration documentation
+ This patch allows a module to define its configuration in XML in
+ source, such that it can be parsed by the XML documentation
+ engine. Documentation is generated in a two-pass approach: 1. The
+ documentation is first generated from the XML pulled from the
+ source 2. The documentation is then enhanced by the registration
+ of configuration options that use the configuration framework
+ This patch include configuration documentation for the following
+ modules: * chan_motif * res_xmpp * app_confbridge * app_skel *
+ udptl Two new CLI commands have been added: * config show help -
+ show configuration help by module, category, and item * xmldoc
+ dump - dump the in-memory representation of the XML documentation
+ to a new XML file. Review:
+ https://reviewboard.asterisk.org/r/2278 Review:
+ https://reviewboard.asterisk.org/r/2058 patches: on review 2058
+ uploaded by twilson
+
+2013-02-14 19:58 +0000 [r381470-381471] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Remove extraneous stuff from r381470.
+
+ * channels/chan_skinny.c: Add back sending dialnumber to skinny.
+ Don't know why it seemed to work during testing, but it really is
+ needed for protocol v17 (and probably above).
+
+2013-02-14 19:52 +0000 [r381469] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO
+ because it isn't a real hangup. It doesn't hurt to check
+ AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside
+ of a bridge. (issue ASTERISK-20492) ........ Merged revisions
+ 381466 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 381467 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-14 19:25 +0000 [r381465] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Respect callerid presentation in skinny.
+ Fix chan_skinny so that it respects callerID presentation of
+ inbound calls to device and a couple of other minor fixes: 145
+ packet (add OCTAL_FROM amd callerid), and dont send dialednumber
+ message if protocol >= 17. (closes issue ASTERISK-21066) Reported
+ by: snuffy Tested by: snuffy, myself Patches:
+ skinny-respect-clid-restrictions-v2.diff uploaded by snuffy
+ (license 5024)
+
+2013-02-14 18:47 +0000 [r381448] Kinsey Moore <kmoore@digium.com>
+
+ * main/logger.c, include/asterisk/term.h, apps/app_queue.c,
+ main/asterisk.c, main/term.c, main/data.c, main/pbx.c,
+ main/manager.c: Revamp of terminal color codes The core module
+ related to coloring terminal output was old and needed some love.
+ The main thing here was an attempt to get rid of the obscene
+ number of stack-local buffers that were allocated for no other
+ reason than to colorize some output. Instead, this uses a simple
+ trick to allocate several buffers within threadlocal storage,
+ then automatically rotates between them, so that you can make
+ multiple calls to the colorization routine within one function
+ and not need to allocate multiple buffers. Review:
+ https://reviewboard.asterisk.org/r/2241/ Patches: bug.patch
+ uploaded by Tilghman Lesher
+
+2013-02-14 17:06 +0000 [r381398-381427] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Use a shuffling algorithm to find unused
+ IAX2 call numbers. While adding red-black tree containers to
+ astobj2 in r376575, Richard pointed out the way chan_iax2 finds
+ unused call numbers will prevent ao2_container integrity checks
+ at runtime. This patch removes the ao2_container and instead uses
+ fixed sized arrays and a modified Fisher-Yates-Durstenfeld
+ shuffle to maintain the call number list. While the locking
+ semantics are similar to the ao2_container implementation, this
+ implementation should be faster and more memory efficient.
+ Review: https://reviewboard.asterisk.org/r/2288/
+
+ * include/asterisk/doxygen/asterisk-git-howto.h: Update the name of
+ the update_tags utility in the git mirror how-to.
+
+2013-02-14 03:49 +0000 [r381366] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_db.c: Don't throw a spurious error when using
+ DBdeltree The function call ast_db_deltree returns the number of
+ row deleted, or a negative number if it failed. DBdeltree was
+ treating any non-zero return as an error, causing a spurious
+ verbose error message to be displayed. This patch handles the
+ return code of ast_db_deltree correctly. (closes issue
+ ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff
+ uploaded by ianc (License #5955) ........ Merged revisions 381364
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 381365 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-12 21:45 +0000 [r381326] David M. Lee <dlee@digium.com>
+
+ * tests/test_threadpool.c, tests/test_taskprocessor.c,
+ main/threadpool.c, main/taskprocessor.c,
+ include/asterisk/threadpool.h: Add a serializer interface to the
+ threadpool This patch adds the ability to create a serializer
+ from a thread pool. A serializer is a ast_taskprocessor with the
+ same contract as a default taskprocessor (tasks execute serially)
+ except instead of executing out of a dedicated thread, execution
+ occurs in a thread from a ast_threadpool. Think of it as a
+ lightweight thread. While it guarantees that each task will
+ complete before executing the next, there is no guarantee as to
+ which thread from the pool individual tasks will execute. This
+ normally only matters if your code relys on thread specific
+ information, such as thread locals. This patch also fixes a bug
+ in how the 'was_empty' parameter is computed for the push
+ callback, and gets rid of the unused 'shutting_down' field.
+ Review: https://reviewboard.asterisk.org/r/2323/
+
+2013-02-12 20:57 +0000 [r381307] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp_engine.c, /: Do not allow native RTP bridging if
+ packetization of media streams differs. The RTP engine will no
+ longer allow for local and remote native RTP bridges if
+ packetization of streams differs. Allowing native bridging in
+ this scenario has been known to cause FAX failures. (closes
+ ASTERISK-20650) Reported by: Maciej Krajewski Patches:
+ ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
+ Review: https://reviewboard.asterisk.org/r/2319 ........ Merged
+ revisions 381281 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381306 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-12 20:18 +0000 [r381285] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c,
+ channels/sip/security_events.c: Fix some more REF_DEBUG-related
+ build errors When sip_ref_peer and sip_unref_peer were exported
+ to be usable in channels/sip/security_events.c, modifications to
+ those functions when building under REF_DEBUG were not taken into
+ account. This change moves the necessary defines into sip.h to
+ make them accessible to other parts of chan_sip that need them.
+ ........ Merged revisions 381282 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-12 03:31 +0000 [r381256] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_confbridge.c: Adding Some More Manager Events To
+ ConfBridge Currently, ConfBridge does not send manager events for
+ ConfbridgeMute, ConfbridgeUnmute, ConfbridgeStartRecord and
+ ConfbridgeStopRecord. This patch adds these events to the
+ manager. The reporter's patch moves some other events up to the
+ beginning of the file. The patch being committed is based on the
+ patch contributed from the reporter of this issue. I have made a
+ lot of modifications to the patch in order for it to fit in
+ better with what we currently are doing in the code when it comes
+ to manager events. I also made a few changes to the <see-also>
+ elements on some of the events. (closes issue ASTERISK-20827)
+ Reported by: Clint Davis Tested by: Clint Davis, Michael L. Young
+ Patches: 20827.diff uploaded by Clint Davis (license 6453)
+ asterisk-20827-confbridge-events.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2309/
+
+2013-02-11 21:17 +0000 [r381219] Kevin Harwell <kharwell@digium.com>
+
+ * /, apps/app_playback.c: Properly load say.conf upon reload of
+ module app_playback. If say.conf did not exists prior to
+ originally loading module app_playback it would not load on
+ subsequent reloads of the module once it had been created. This
+ occurred because upon reload of the app_playback module it would
+ only load a new configuration if an old one had previously
+ existed. This fix simply removed the association between checking
+ if an old configuration existed and the loading of the new one.
+ (closes issue ASTERISK-20800) Reported by: pgoergler ........
+ Merged revisions 381216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 381217 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-11 21:10 +0000 [r381218] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/astobj2.h: Fix compilation error with REF_DEBUG
+ When the red/black tree work was committed, there was an extra ",
+ " in the REF_DEBUG definition of ao2_container_alloc_rbtree.
+
+2013-02-11 20:39 +0000 [r381214] David M. Lee <dlee@digium.com>
+
+ * tests/test_json.c, res/res_json.c: Minor fixes to res_json and
+ test_json. * Made input checking more consistent with other
+ Asterisk code * Added validation to ast_json_dump_new_file *
+ Fixed tests for ownereship semantics (issue ASTERISK-20887)
+
+2013-02-11 18:54 +0000 [r381195] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix some issues with skinny callid. Add
+ extra string to transmit_callinfo_var, Only set string2 to tonum
+ for outgoing calls and changes to send_callinfo and push_callinfo
+ to not set callid name to last number. (closes issue
+ ASTERISK-21063) Reported by: wedhorn Tested by: snuffy, myself
+ Patches: skinny-callinfoupdate03.diff uploaded by wedhorn
+ (license 5019)
+
+2013-02-11 18:00 +0000 [r381177] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: features: Don't cache a struct ast_app pointer.
+ Caching a struct ast_app pointer is not a good idea because
+ someone could unload the application. After the applicaiton
+ unload the cached ast_app pointer is no longer valid. Only pbx.c
+ can cache the pointer because it knows when the application is
+ unloaded and removes the pointer. * Fixed one-touch Monitor and
+ MixMonitor to not cache the ast_app pointer and not use the silly
+ monitor_ok/mixmonitor_ok/stopmixmonitor_ok flags. * Extracted
+ bridge_check_monitor() from ast_bridge_call() and use propper
+ locking.
+
+2013-02-11 15:11 +0000 [r381160] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_xmpp.c: Fix crash in res_xmpp when deleting pubsub
+ node from CLI An error existed in res_xmpp where it would attempt
+ to delete attributes from a node that itself was also deleted.
+ Per the iksemel documentation, attributes added using iks_insert
+ are copied to the parent node's stack, and will be reclaimed when
+ that node is itself destroyed. (closes issue ASTERISK-20982)
+ Reported by: marcelloceschia patches: delete-node-fix.diff
+ uploaded by marcelloceschia (License 6036) ........ Merged
+ revisions 381159 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-10 14:58 +0000 [r381134] Joshua Colp <jcolp@digium.com>
+
+ * main/sorcery.c, include/asterisk/sorcery.h, tests/test_sorcery.c:
+ Add additional functionality to the Sorcery API. This commit adds
+ native implementation support for copying and diffing objects, as
+ well as the ability to load or reload on a per-object type level.
+ Review: https://reviewboard.asterisk.org/r/2320/
+
+2013-02-09 20:58 +0000 [r381069-381118] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: pbx: Fix regression caused by taking advantage of the
+ function name sort. Taking advantage of the sorted order of the
+ registered functions container requires that they are actually
+ inserted in the expected sort order. * Insert the registered
+ functions into the container in case sensitive position. As a
+ result, only the complete_functions() routine needs to search the
+ entire container because it does a case insensitive search for
+ convenience. Caught by the unit tests.
+
+ * main/pbx.c: pbx: Make function and application containers take
+ advantage of being sorted. * Fixed "core show function" tab
+ completion and token count checking. * Refactored function and
+ application container handling code to reduce redundancy. * Made
+ __ast_pbx_run() return using the defines the caller should
+ expect. Doesn't change the returned values. Just made use the
+ defines.
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
+ Make ast_do_masquerade() a void function.
+
+ * /, apps/app_confbridge.c: app_confbridge: Fix crash from
+ receiving an AMI action after ConfBridge unloaded. Unloading
+ ConfBridge caused the next AMI action received to crash Asterisk.
+ * Add the missing unregister of AMI action
+ ConfbridgeSetSingleVideoSrc when ConfBridge is unloaded. (closes
+ issue ASTERISK-20994) Reported by: Jeremy Kister Patches:
+ jira_asterisk_20994_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Rusty Newton, Jeremy Kister ........ Merged
+ revisions 381067 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-08 17:36 +0000 [r381068] Jonathan Rose <jrose@digium.com>
+
+ * configs/features.conf.sample, main/features.c, CHANGES: Call
+ Parking: Set PARKINGLOT and PARKINGSLOT variables on all parked
+ calls These two variables were previously not being set when
+ comebacktoorigin=yes and the example configs seemed to imply that
+ they should be. Since there is no harm in this and since calls
+ that are sent back to origin are capable of continuing in the
+ dialplan, this seemed like a no-brainer. Also it supports some
+ bridging tests I've been working on.
+
+2013-02-07 17:57 +0000 [r381037] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sorcery_config.c: Fix a bug where a changed configuration
+ file might not be available to all sorcery object types. Since
+ res_sorcery_config used a static name of "res_sorcery_config" to
+ inform the configuration file API that it asked for the
+ configuration file it was possible during a reload for some
+ sorcery object types not to receive the new configuration file.
+ This change introduces a UUID on a per-sorcery config instance
+ basis so that the unchanged state is kept on an instance basis
+ and not for the res_sorcery_config module as a whole.
+
+2013-02-07 15:16 +0000 [r381017] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/stringfields.h, tests/test_stringfields.c: Add
+ aggregate operations for stuctures with string fields Add
+ struct-level comparison and copying of string fields to reduce
+ the complexity of whole-struct comparison and copying when using
+ string fields. The new macros do not take into account
+ non-stringfield data. Review:
+ https://reviewboard.asterisk.org/r/2308/
+
+2013-02-06 20:18 +0000 [r380977] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c: Fixed failing test from r380696. When I
+ added my extensive suite of session timer unit tests, apparently
+ one of them was failing and I never noticed. If neither Min-SE
+ nor Session-Expires is set in the header, it was responding with
+ a Session-Expires of the global maxmimum instead of the
+ configured max for the endpoint. (issue ASTERISK-20787) ........
+ Merged revisions 380973 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380974 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-06 08:44 +0000 [r380925-380943] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix reload skinny with active devices.
+ Patch ensures that d->activeline and l->activesub are moved over
+ to the new device and line so that on callend the appropriate
+ subs can be found to complete hangup before device resets.
+ (closes issue ASTERISK-16610) Reported by: wedhorn Tested by:
+ snuffy, myself Patches: skinny-reloadactive01.diff uploaded by
+ wedhorn (license 5019) ........ Merged revisions 380942 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Reset skinny
+ vmexten and immeddial char on reload. Make skinny reset vmexten
+ and immeddial to '\0' on reload to ensure that it is set to '\0'
+ if the appropriate item is removed/commented in skinny.conf. Also
+ small fix re immeddial char in skinny.conf and add immedial
+ setting to skinny show settings. (closes issue ASTERISK-21037)
+ Reported by: snuffy Tested by: snuffy, myself Patches:
+ immed_dial_fix.diff uploaded by snuffy (license 5024)
+
+2013-02-05 19:11 +0000 [r380855-380896] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c, /, apps/app_page.c: app_page and
+ app_confbridge: Fix custom announcement on entering conference.
+ The Page and ConfBridge custom announcement did not play when
+ users entered the conference. * Fix the
+ CONFBRIDGE(user,announcement) file not getting played. The code
+ to do this got removed accidentally when the ConfBridge code was
+ restructured to be more state machine like. * Fixed
+ play_prompt_to_user() doxygen comments. * Fixed the Page A(x) and
+ n options for the caller. The caller never played the
+ announcement file and totally ignored the n option. The code to
+ do this was lost when the application was converted to use
+ ConfBridge. * Factored out setup_profile_bridge(),
+ setup_profile_paged(), and setup_profile_caller() routines to
+ setup ConfBridge profiles. Made each profile setup routine use
+ the default template if one has not already been setup by
+ dialplan. (closes issue ASTERISK-20990) Reported by: Jeremy
+ Kister Tested by: rmudgett ........ Merged revisions 380894 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
+ error messages on exiting conference. A marked user ending a
+ conference with only end_marked users generates error messages:
+ ERROR[0000][C-00000000]: confbridge/conf_state.c:47
+ conf_invalid_event_fn: Invalid event for confbridge user '' * The
+ MULTI_MARKED state was doing too much when it was kicking out the
+ end_marked users from the conference. The kicked out users will
+ clean up after themselves when they exit the conference. (closes
+ issue ASTERISK-20991) Reported by: Jeremy Kister Tested by:
+ rmudgett ........ Merged revisions 380892 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_page.c: app_page: Fixup application XML documentation
+ typos and inaccuracies. ........ Merged revisions 380869 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_config_parser.c, /: Because the compiler can
+ check types with a struct copy and memcpy() cannot. ........
+ Merged revisions 380856 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/dial.c, /: Separate option_types[] from the struct
+ definition. Updated the option_types[] doxygen comment. ........
+ Merged revisions 380853 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380854 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-04 19:52 +0000 [r380817] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/aconfigure, res/pjproject/build/os-auto.mak.in,
+ Makefile, res/pjproject/aconfigure.ac, /, res/Makefile,
+ res/pjproject/build/common.mak: Fix how we build pjproject. Allow
+ parallel builds, better tolerate failures, build faster. This
+ also stops running dependencies before top-level configure has
+ been run. (closes issue ASTERISK-20815) Review:
+ https://reviewboard.asterisk.org/r/2292/ ........ Merged
+ revisions 380816 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-02-02 01:52 +0000 [r380792] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add variable length displayprompt packet
+ to skinny and use octals. Add new variable length displayprompt
+ packet (0x0145) to skinny. Uses the new packet if the device is
+ reporting protocol versions >= 17. Add the use of octal codes for
+ sending prompts to both the new and old displayprompt messages
+ (also cleaned up soft_key_template_default to use the defined
+ octal codes). Review: https://reviewboard.asterisk.org/r/2294/
+
+2013-02-01 19:35 +0000 [r380774] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/iax2/firmware.c: chan_iax2: Fix compile error if
+ MALLOC_DEBUG enabled. NEVER INCLUDE astmm.h DIRECTLY!!
+
+2013-02-01 06:37 +0000 [r380755] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Adds variable length callinfo packets to
+ skinny. Add packet 0x014A (variable length call info messages) to
+ skinny for newer firmware. Plenty of unknown information but
+ includes the equivalent functionality as the fixed size callinfo
+ packet already included. Only send this packet if protocol
+ reported is >= 17. Review:
+ https://reviewboard.asterisk.org/r/2290/
+
+2013-01-31 22:03 +0000 [r380738] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/pjlib/src/pj/ssl_sock_ossl.c,
+ res/pjproject/pjlib/src/pj/log.c,
+ res/pjproject/pjlib/src/pj/pool_buf.c, /,
+ res/pjproject/pjsip-apps/src/samples/icedemo.c,
+ res/pjproject/pjlib/include/pj/config_site.h,
+ res/pjproject/pjmedia/src/test/test.c: Multiple revisions
+ 380735-380736 ........ r380735 | qwell | 2013-01-31 15:40:09
+ -0600 (Thu, 31 Jan 2013) | 1 line Fix a few compiler warnings.
+ ........ r380736 | qwell | 2013-01-31 15:42:34 -0600 (Thu, 31 Jan
+ 2013) | 1 line Ignore warnings caused by PJ_TODO()s in pjproject.
+ ........ Merged revisions 380735-380736 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-31 20:17 +0000 [r380699] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c: Process session timers, even if
+ Session-Expires header is missing Previously, Asterisk only
+ processed session timer information if both the 'Supported:
+ timer' and 'Session-Expires' headers were present. However, the
+ Session-Expires header is optional. If we were to receive a
+ request with a Min-SE greater than our configured
+ session-expires, we would respond with a 'Session-Expires' header
+ that was too small. This patch cleans the situation up a bit,
+ always processing timer information if the 'Supported: timer'
+ header is present. (closes issue ASTERISK-20787) Reported by:
+ Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/
+ ........ Merged revisions 380696 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380698 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-31 19:52 +0000 [r380695] Sean Bright <sean@malleable.com>
+
+ * channels/iax2/include/firmware.h (added),
+ channels/iax2/include/parser.h, channels/chan_iax2.c,
+ channels/iax2/firmware.c (added): Move IAX firmware related
+ functionality into separate files. This patch is mostly a
+ reorganization of existing code with a few exceptions: * Added
+ doxygen comments to all of the extracted functions. * Split
+ reload_firmware(int unload) into iax_firmware_reload() and
+ iax_firmware_unload() for readability. * Create
+ iax_firmware_traverse() to support the 'iax2 show firmware' CLI
+ command. * Renamed iax_check_version() to
+ iax_firmware_get_version() and change its arguments and return
+ value so that it returns a success/failure value and sets the
+ selected version into an out parameter to avoid confusion with
+ failure and version 0.
+
+2013-01-31 19:04 +0000 [r380674] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/pjmedia/build/Makefile,
+ res/pjproject/build/cc-auto.mak.in, /,
+ res/pjproject/pjlib-util/build/Makefile,
+ res/pjproject/pjlib/build/Makefile,
+ res/pjproject/build/rules.mak,
+ res/pjproject/pjnath/build/Makefile,
+ res/pjproject/pjsip/build/Makefile, res/pjproject/aconfigure,
+ res/pjproject/pjsip-apps/build/Makefile,
+ res/pjproject/aconfigure.ac: Multiple revisions 380671-380673
+ ........ r380671 | qwell | 2013-01-31 12:59:28 -0600 (Thu, 31 Jan
+ 2013) | 4 lines Remove a cross-compile workaround. ar and ranlib
+ can be easily detected with autoconf. ........ r380672 | qwell |
+ 2013-01-31 13:00:38 -0600 (Thu, 31 Jan 2013) | 2 lines Always
+ check for libm, regardless of configure options. ........ r380673
+ | qwell | 2013-01-31 13:03:03 -0600 (Thu, 31 Jan 2013) | 7 lines
+ Add support for parallel builds of pjproject. Also adds proper
+ dependency checking, and direct .a file targets. We don't take
+ advantage of this currently, but we will soon. (issue
+ ASTERISK-20815) ........ Merged revisions 380671-380673 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-31 18:22 +0000 [r380576-380666] Richard Mudgett <rmudgett@digium.com>
+
+ * bridges/bridge_multiplexed.c: bridge_multiplexed: Keep the
+ multiplexed thread until no more bridges use it. * Fixed the
+ potential of losing the multiplexed bridge thread when the last
+ channel leaves and another joins while the multiplexed thread is
+ being shut down. * Refactored and improved the management of the
+ serviced channels array. * Changed the channels count to a
+ bridges count so it only needs to be incremented rather than
+ changed by two.
+
+ * main/frame.c, funcs/func_frame_trace.c: Improve func FRAME_TRACE
+ DTMF digit format.
+
+ * include/asterisk/bridging.h: Eliminate an unused lock in
+ ast_bridge_channel.
+
+ * main/channel.c: Eliminate a use of a C++ keyword as a variable.
+ new to new_frame
+
+ * channels/iax2: Add ignore properties to channels/iax2
+
+ * /, include/asterisk/channel.h: Make CHECK_BLOCKING() debug
+ message more useful. Change the displayed pthread value to hex
+ format so it can be easily matched with CLI core show threads or
+ gdb. ........ Merged revisions 380611 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380612 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix "dahdi show channels
+ group" for groups greater than 31. The variable type used was not
+ large enough to hold a group bit field. ........ Merged revisions
+ 380572 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 380575 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-30 17:49 +0000 [r380460-380522] Matthew Jordan <mjordan@digium.com>
+
+ * configure, configure.ac, /: Support building Asterisk for
+ Raspberry Pi/Raspbian with hard-float support Building Asterisk
+ on Raspbian with hard-float support fails as it uses the string
+ 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'.
+ This patch modifies the configure script for Asterisk such that
+ it will match on any string beginning with 'linux-gnueabi', as
+ opposed to requiring an explicit match. (closes issue
+ ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian
+ Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse
+ (license 6459) linux-gnueabihf-autoconf.patch uploaded by
+ Christian Hesse (license 6459) ........ Merged revisions 380520
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 380521 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Unregister SIP provider API if module
+ load is declined A user in #asterisk ran into a problem where a
+ configuration error prevented the chan_sip module from being
+ loaded. Upon fixing their configuratione error, they could no
+ longer load the chan_sip module. This was because the
+ configuration checking happened after the SIP provider was
+ registered with the Asterisk core, and subsequent attempts to
+ load the SIP module failed as the provider was already
+ registered. Since we want to detect any failure in registering
+ chan_sip as early as possible (as that could be emblematic of a
+ deeper mismatch between module and Asterisk core), this patch
+ does not change the registration location, but does ensure that
+ if a module load is declined, we unregister the module as the SIP
+ api provider. ........ Merged revisions 380480 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Perform case insensitive comparisons for
+ T.38 attributes RFC5347 section 2.5.2 states the following: ...
+ The attribute "T38MaxBitRate" was once incorrectly registered
+ with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
+ T.38 examples and common implementation practice, the form
+ "T38MaxBitRate" SHOULD be generated by implementations conforming
+ to this package. In general, it is RECOMMENDED that
+ implementations of this package accept lowercase, uppercase, and
+ mixed upper/lowercase encodings of all the T.38 attributes. ...
+ Asterisk currently does not perform case insensitive matching on
+ the T.38 attributes. This causes the T38MaxBitRate attribute to
+ be negotiated at 2400 baud instead of 14400 (or whatever value
+ you actually wanted). This patch makes it so that when we compare
+ T.38 attributes, we do so in a case insensitive fashion. Note
+ that while the issue reporter did not directly write the patch,
+ they contributed to it (and would have provided one themselves if
+ the license had gone through a tad faster), and hence get
+ attribution for it. Review:
+ https://reviewboard.asterisk.org/r/2298/ (closes issue
+ ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill
+ patches: -- uploaded by Eric Hill ........ Merged revisions
+ 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 380465 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_calendar_icalendar.c, /: Fix memory leak in
+ res_calendar_icalendar The ICalendar module had a systemic memory
+ leak on each fetch of data from the ICalendar source. The
+ previous fetched data was not being properly disposed. This patch
+ makes it so that before each fetch of data, we dispose of the
+ previously fetched data. (closes issue ASTERISK-21012) Reported
+ by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions
+ 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 380452 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-29 22:58 +0000 [r380433] Sean Bright <sean@malleable.com>
+
+ * channels/iax2/provision.c (added), channels/Makefile,
+ channels/chan_iax2.c, channels/iax2-parser.c (removed),
+ channels/iax2/include/iax2.h (added), channels/iax2-provision.h
+ (removed), channels/iax2/parser.c (added), channels/iax2 (added),
+ channels/iax2-parser.h (removed),
+ channels/iax2/include/provision.h (added), channels/iax2/include
+ (added), channels/iax2/include/parser.h (added), channels/iax2.h
+ (removed), channels/iax2-provision.c (removed): Move the
+ ancillary iax2 source files into a separate sub-directory. This
+ patch just moves the IAX2 source and header files into a separate
+ iax2 sub-directory in the channels directory, similar to how the
+ sip source files are structured. The only thing that was added
+ was an #ifndef to protect provision.h from multiple inclusion.
+
+2013-01-29 20:19 +0000 [r380407] Joshua Colp <jcolp@digium.com>
+
+ * tests/test_sorcery.c, main/sorcery.c: Fix an issue where building
+ with DEBUG_FD_LEAKS enabled would not work due to sorcery using
+ calls called "open" and "close".
+
+2013-01-29 18:02 +0000 [r380386] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_agent.c, /: chan_agent: Prevent multiple channels
+ from logging in as the same agent. Multiple channels logging in
+ as the same agent can result in dead channels waiting for a
+ condition signal that will never come because another channel
+ thread stole it. A symptom is chan_sip repeatedly generating
+ warning messages about rescheduling autodestruction of dialogs
+ with an agent channel owner. * Made only login_exec() (the app
+ AgentLogin) clear the agent_pvt->chan pointer to prevent multiple
+ channels from logging in as the same agent. agent_read(),
+ agent_call(), and agent_set_base_channel() no longer disconnect
+ the agent channel from the agent_pvt. This also eliminates the
+ need to keep checking for agent_pvt->chan being NULL. * Made
+ agent_hangup() not wake up the AgentLogin agent thread until it
+ is done. * Made agent_request() not able to get the agent until
+ he has logged in and any wrapup time has expired. * Made
+ agent_request() use ast_hangup() instead of agent_hangup() to
+ correctly dispose of a channel. * Removed
+ agent_set_base_channel(). Nobody calls it and it is a bad thing
+ in general. * Made only agent_devicestate() determine the current
+ device state of an agent. Note: Agent group device states have
+ never been supported. Review:
+ https://reviewboard.asterisk.org/r/2260/ ........ Merged
+ revisions 380364 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380384 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-29 17:46 +0000 [r380383] David M. Lee <dlee@digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Corrected crypto tag in SDP ANSWER
+ for SRTP. (again) The original fix (r380043) for getting Asterisk
+ to respond with the correct tag overlooked some corner cases, and
+ the fact that the same code is in 1.8. This patch moves the
+ building of the crypto line out of sdp_crypto_process(). Instead,
+ it merely copies the accepted tag. The call to sdp_crypto_offer()
+ will build the crypto line in all cases now, using a tag of "1"
+ in the case of sending offers. (closes issue ASTERISK-20849)
+ Reported by: José Luis Millán Review:
+ https://reviewboard.asterisk.org/r/2295/ ........ Merged
+ revisions 380347 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380350 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-29 17:06 +0000 [r380349] Jonathan Rose <jrose@digium.com>
+
+ * /, main/features.c: call_parking: Make sure fallbacks are used
+ when lacking a flat channel exten A regression was introduced
+ which removed automatic fallback behavior from the PBX. This
+ behavior was used by call parking (or at least documented as how
+ the feature works) in order to select an extension when the flat
+ channel extension wasn't available from the comebackcontext.
+ Parking now handles the fallbacks internally in order to keep
+ behavior matching with how it is documented. (closes issue
+ ASTERISK-20716) Reported by: Chris Gentle Review:
+ https://reviewboard.asterisk.org/r/2296/ ........ Merged
+ revisions 380348 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-29 14:48 +0000 [r380299-380332] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Ensure that a declined media stream is
+ terminated with a '\r\n' In r369028, chan_sip's processing of
+ media streams in an SDP was modified to better handle multiple
+ offered media streams. Part of that change modified how streams
+ were declined. Previously, declined media streams were not
+ handled in an RFC compliant manner; now, we set the port number
+ to 0 in the media stream definition and proceed on with the next
+ media stream. Unfortunately, the formatting of the declined media
+ stream forgot to append a '\r\n' to the end of the media stream.
+ This is normally added to the accepted media streams later on in
+ the processing of the SDP. Since the declined media stream uses a
+ different buffer than the accepted media streams (and is a
+ malloc'd buffer as opposed to a struct ast_str), it's easier to
+ just slap the '\r\n' on the declined media stream buffer rather
+ than attempt to append it later on. So, that's what we do. And
+ now some devices (and probably some providers) will be a bit
+ happier (but probably not terribly happy, since we just rejected
+ something they offered). Review:
+ https://reviewboard.asterisk.org/r/2297/ (closes issue
+ ASTERISK-20908) Reported by: Dennis DeDonatis Tested by: Dennis
+ DeDonatis ........ Merged revisions 380331 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * autoconf/ast_check_pwlib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Update configure script to be
+ compatible with ptlib 2.10.9 With ptlib 2.10.9, the configure
+ script fails due to grep returning multiple matches for the
+ pattern it searches for. This patch updates the pattern matching
+ to return only the actual version for the symbol searched for,
+ PTLIB_VERSION. (closes issue ASTERISK-20980) Reported by: Stefan
+ Reuter patches: ASTERISK-20980-1.patch uploaded by Stefan Reuter
+ (license 5339) ........ Merged revisions 380297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380298 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-28 21:09 +0000 [r380256] Sean Bright <sean@malleable.com>
+
+ * channels/iax2.h, channels/chan_iax2.c, /: Correct the number of
+ available call numbers in IAX2. There is currently an edge case
+ where call number 32768 might be allocated for a call, even
+ though the IAX2 protocol requires call numbers be only 15 bits.
+ This resulted in some unpredictable behavior when call number
+ 32678 is chosen. This patch was mostly written by Richard Mudgett
+ via ReviewBoard. I'm just committing it. Review:
+ https://reviewboard.asterisk.org/r/2293/ ........ Merged
+ revisions 380254 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380255 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-28 01:58 +0000 [r380209-380212] Russell Bryant <russell@russellbryant.com>
+
+ * main/file.c, /: Change cleanup ordering in filestream destructor.
+ This patch came about due to a problem observed where wav files
+ had an empty header. The header is supposed to be updated in
+ wav_close(). It turns out that this was broken when the
+ cache_record_files option from asterisk.conf was enabled. The
+ cleanup code was moving the file to its final destination
+ *before* running the close() method of the file destructor, so
+ the header didn't get updated. Another problem here is that the
+ move was being done before actually closing the FILE *. Finally,
+ the last bug fixed here is that I noticed that wav_close() checks
+ for stream->filename to be non-NULL. In the previous cleanup
+ order, it's checking a pointer to freed memory. This doesn't
+ actually cause anything to break, but it's treading on dangerous
+ waters. Now the free() of stream->filename is happening after the
+ format module's close() method gets called, so it's safer.
+ Review: https://reviewboard.asterisk.org/r/2286/ ........ Merged
+ revisions 380210 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 380211 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/logger.c, CHANGES, configs/logger.conf.sample: Add
+ queue_log_realtime_use_gmt option to logger.conf Add an option
+ that lets you specify that the timestamps going into the realtime
+ queue log should be in GMT instead of local time. Review:
+ https://reviewboard.asterisk.org/r/2287/
+
+2013-01-27 20:33 +0000 [r380194] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/confbridge/conf_config_parser.c, /: Fix Some Configured
+ Conference Bridge Sounds Not Being Set The "sound_only_one" sound
+ was not being set even though it was configured. In looking into
+ this, I found that the "join" and "leave" prompts were not being
+ set either. (closes issue ASTERISK-20898) Reported by: Stephan
+ Tested by: Stephan Patches:
+ asterisk-20898-custom-sounds-ignored.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2289/ ........ Merged
+ revisions 380193 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-27 18:40 +0000 [r380165-380178] Joshua Colp <jcolp@digium.com>
+
+ * tests/test_sorcery.c: Add a unit test which confirms the apply
+ handler callback is called when it should be.
+
+ * main/sorcery.c: Fix a bug where the apply function was not
+ getting called.
+
+2013-01-25 23:23 +0000 [r380142] Richard Mudgett <rmudgett@digium.com>
+
+ * bridges/bridge_multiplexed.c: bridge_multiplexed: Rename
+ variables so they are not the same as the struct name. * Rename
+ multiplexed_thread variables to muxed_thread. It is shorter and
+ my editer tagging works much better. Struct names and variable
+ names have different purposes and therefore should have different
+ names. * Renamed the multiplexed_threads container to
+ muxed_threads for consistency.
+
+2013-01-25 20:46 +0000 [r380121] Jason Parker <jparker@digium.com>
+
+ * res/res_sorcery_config.c, res/res_sorcery_memory.c: Make sorcery
+ modules global, since they are required by other modules that are
+ global.
+
+2013-01-25 20:00 +0000 [r380108-380109] Richard Mudgett <rmudgett@digium.com>
+
+ * bridges/bridge_multiplexed.c, main/bridging.c: Misc bridge code
+ improvements * Made multiplexed_bridge_destroy() check if
+ anything to destroy and cleared bridge_pvt pointer after
+ destruction. * Made multiplexed_add_or_remove() handling of the
+ chans array simpler. * Extracted bridge_channel_poke(). *
+ Simplified bridge_array_remove() handling of the bridge->array[].
+ The array does not have a NULL sentinel pointer. * Made
+ ast_bridge_new() not create a temporary bridge just to see if it
+ can be done. Only need to check if there is an appropriate bridge
+ tech available. * Made ast_bridge_new() clean up on allocation
+ failures. * Made destroy_bridge() free resources in the opposite
+ order of creation.
+
+ * bridges/bridge_multiplexed.c, main/bridging.c,
+ bridges/bridge_simple.c, bridges/bridge_softmix.c: More trivial
+ bridge code cleanup. * Breaking long lines * Word wrapping
+ comment blocks. * Removing redundant initializers. * Debug
+ message wording.
+
+2013-01-25 14:23 +0000 [r380069-380082] Joshua Colp <jcolp@digium.com>
+
+ * res/res_sorcery_config.c: Add a missing '\' to a log message.
+
+ * tests/test_sorcery.c (added), main/asterisk.c, main/sorcery.c
+ (added), res/res_sorcery_config.c (added),
+ configs/test_sorcery.conf.sample (added),
+ res/res_sorcery_memory.c (added), configs/sorcery.conf.sample
+ (added), include/asterisk/sorcery.h (added): Merge the sorcery
+ data access layer API. Sorcery is a unifying data access layer
+ which provides a pluggable mechanism to allow object creation,
+ retrieval, updating, and deletion using different backends (or
+ wizards). This is a fancy way of saying "one interface to rule
+ them all" where them is configuration, realtime, and anything
+ else that comes along. Review:
+ https://reviewboard.asterisk.org/r/2259/
+
+2013-01-25 05:49 +0000 [r380057] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Add force
+ dial keys to skinny. Adds a dial softkey when the device is in
+ DAFD. The softkey is greyed (unusable) until a possible dialplan
+ match is entered. Code includes updating transmit_selectsoftkeys
+ to allow the use of a button mask. Also add option to use # or *
+ as a dial now button. Original patch by snuffy cleaned up by
+ myself. Review: https://reviewboard.asterisk.org/r/2277/
+
+2013-01-24 16:40 +0000 [r380044] David M. Lee <dlee@digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Corrected crypto tag in SDP ANSWER
+ for SRTP. When Asterisk responds with an SDP ANSWER for SRTP, it
+ had the code to correctly fill in the crypto data, which was
+ overwritten by a call to sdp_crypto_offer. Corrected the
+ situation by changing sdp_crypto_offer to not replacing crypto
+ data if it already exists. (closes issue ASTERISK-20849) Reported
+ by: José Luis Millán Tested by: Iñaki Baz Castillo Patches:
+ fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
+ ........ Merged revisions 380043 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-24 04:02 +0000 [r380029] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_confbridge.c: Correct documentation for
+ ConfbridgeList AMI action The documentation for ConfbridgeList
+ states that the Conference field is optional. That's not really
+ the case: if you fail to provide a Conference number, the command
+ will kick back an error. (closes issue AST-1090) Reported by:
+ John Bigelow ........ Merged revisions 380028 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-23 16:50 +0000 [r380004] Kinsey Moore <kmoore@digium.com>
+
+ * contrib/scripts/autosupport: Add support for DPMA to autosupport
+ This adds the ability to get the DPMA version, a listing of the
+ local firmware directory, and indexes of configured remote
+ directories. (closes issue AST-1070) Reported By: Malcolm
+ Davenport Tested By: Kinsey Moore <kmoore@digium.com>
+
+2013-01-23 00:30 +0000 [r379966] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c, /: Attempt to be more helpful when using a bad
+ ao2 object pointer. Put the external obj pointer in the message
+ instead of the internal version. ........ Merged revisions 379963
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 379964 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-22 22:19 +0000 [r379950] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_fax_spandsp.c: res_fax_spandsp: fix t38 transmission
+ bug caused by not returning success This patch fixes the problem,
+ but the issue includes a test which is still being considered for
+ the automated test suite. (issue ASTERISK-20919) Reported by:
+ NITESH BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by
+ NITESH BANSAL (license 6418) ........ Merged revisions 379949
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-22 20:58 +0000 [r379936] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Remove a large block of commented out code
+ from chan_iax2. During the conversion to the newer CLI command
+ structure the old definitions were commented out. I think it's
+ safe to remove them completely now.
+
+2013-01-22 19:29 +0000 [r379912] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c, sounds/Makefile: app_meetme: Use new
+ prompts for administrator menu The old prompts for the
+ administrator menu were inadequate. They didn't mention that the
+ menu had additional options through the 8 key and pressing the 8
+ key wouldn't reveal what those options were. This patch fixes all
+ of that while also organizing code pertaining to each individual
+ menu type which was previously all stored in one gigantic
+ function along with many of the basic conference functions.
+ (closes issue AST-996) Reported by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/360/ ........ Merged
+ revisions 379885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379892 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-22 16:48 +0000 [r379864] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Remove stray property.
+
+2013-01-22 15:16 +0000 [r379828-379830] Matthew Jordan <mjordan@digium.com>
+
+ * funcs/func_frame_trace.c, res/res_agi.c, main/file.c, main/app.c,
+ CHANGES, include/asterisk/frame.h, apps/app_playback.c,
+ apps/app_controlplayback.c, include/asterisk/file.h,
+ main/channel.c: Add ControlPlayback manager action This patch
+ adds the capability for asynchronous manipulation of audio being
+ played back to a channel though a new AMI action
+ "ControlPlayback". The ControlPlayback action supports a number
+ of operations, the availability of which depend on the
+ application being used to send audio to the channel. When the
+ audio playback was initiated using the ControlPlayback
+ application or CONTROL STREAM FILE AGI command, the audio can be
+ paused, stopped, restarted, reversed, or skipped forward. When
+ initiated by other mechanisms (such as the Playback application),
+ the audio can be stopped, reversed, or skipped forward. Review:
+ https://reviewboard.asterisk.org/r/2265/ (closes issue
+ ASTERISK-20882) Reported by: mjordan
+
+ * /, apps/app_meetme.c: Fix station ringback; trunk hangup issues
+ in SLA This patch fixes two bugs: * If an outbound call is made
+ from a SLA phone using SLAStation, then there is no ringtone
+ audible to the phone that originates the call. The indication of
+ the ringing was not being passed to the SLA station; this patch
+ fixes that by passing through the progress indications. * If an
+ SLA station hangs up before the called party answers, then the
+ channel to the called party continues to ring until a timeout
+ occurs. If the called party manages to answer, Asterisk attempts
+ to connect the called party to a non-existant MeetMe room. This
+ patch corrects the behavior by abandoning the call attempt if it
+ detects that the SLA station is no longer in use while attempting
+ to call the called party. Review:
+ https://reviewboard.asterisk.org/r/2275/ (closes issue
+ ASTERISK-20462) Reported by: dkerr patches:
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
+ 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
+ asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) ........ Merged revisions 379825 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379826 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-22 00:36 +0000 [r379809] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_bridge.c, apps/app_confbridge.c: confbridge:
+ Minor fixes playing user counts to the conference. * Generate a
+ warning message if sound files do not exist when trying to play
+ the user count to the conference. Use the new helper routine
+ sound_file_exists() for consistency. * Put the new user into
+ autoservice when playing user counts to the conference. * Check
+ the return value of ast_bridge_impart(). ........ Merged
+ revisions 379808 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-21 20:41 +0000 [r379791] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/scripts/safe_asterisk, main/asterisk.c,
+ contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk, /: Update init.d scripts to
+ handle stderr; readd splash screen for remote consoles When
+ r376428 was commited to re-order start up sequences to be more
+ tolerant of forking with thread primitives, a few items were
+ changed that caused changes in behavior on some distros. This
+ includes: * Not displaying the splash screen on a remote console.
+ * Displaying an error message on stderr when a remote console
+ cannot connect to a running instance of Asterisk. In the first
+ case, the splash screen was re-added (thanks to Michael L.
+ Young). In the second case, the various init.d scripts were
+ modified to pipe stderr to /dev/null, as the error message is
+ useful - if you execute a remote console or a remote console
+ command execution and it fail, it should tell you. Note that the
+ error message was always present, it just failed to be printed
+ prior to r376428. Much thanks to the folks who quickly reported
+ this problem, provided solutions, and promptly tested the various
+ init.d scripts on a variety of distros. (closes issue
+ ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
+ Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
+ asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
+ 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
+ (license 6283) ........ Merged revisions 379760 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379777 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 379790 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-21 20:35 +0000 [r379753-379789] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c, bridges/bridge_builtin_features.c: Better
+ protect bridge_channel state from other threads.
+
+ * main/bridging.c: Extract common bridging code into bridge_stop()
+ and bridge_force_out_all().
+
+ * main/bridging.c, bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h,
+ include/asterisk/bridging.h: Made some bridging API calls void.
+ Some bridging comments updated.
+
+2013-01-21 18:47 +0000 [r379721] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
+ frames When iLBC is being used with a jitter buffer and the jb
+ has to interpolate frames, it generates frames with a null
+ pointer and a non-zero datalen. This is now handled properly.
+ (closes issue ASTERISK-20914) Reported By: John McEleney Patches:
+ ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
+ ........ Merged revisions 379718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379719 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-21 18:45 +0000 [r379703-379720] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Trivial bridge code cleanup.
+
+ * bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h,
+ include/asterisk/bridging.h,
+ include/asterisk/bridging_technology.h: Bridge API comment
+ tweaks.
+
+2013-01-21 07:26 +0000 [r379678] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix device call logging issues in
+ skinny Skinny device call logging (ie missed, place and received
+ calls) has issues because the incorrect sequence of callstates
+ is/can be sent to the device. This patch removes some extra
+ callstate updates driven by forces external to skinny and ensures
+ the needed intermediary callstate messages are sent. (closes
+ issue ASTERISK-20964) Reported by: wedhorn Tested by: snuffy,
+ myself Patches: ast11-skinny-calllog01.diff uploaded by wedhorn
+ (license 5019) ........ Merged revisions 379677 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-21 04:50 +0000 [r379644] Andrew Latham <lathama@gmail.com>
+
+ * contrib/scripts/install_prereq, /: Add LDAP libraries to install
+ script Add LDAP dev package to Debian/Ubuntu install list.
+ Existed in Redhat already. (issue ASTERISK-20886) ........ Merged
+ revisions 379643 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-21 04:17 +0000 [r379610-379612] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_minivm.c: Fix crash in app_minivm when mime encoding
+ string An incorrect string initializations was left in
+ ast_str_encode_mime from the patch that converted string
+ manipulations to use ast_str strings (r191140). The string
+ initialization causes a crash when ast_str_set is called on the
+ string later on in the function. (closes issue ASTERISK-18697)
+ Reported by: Chris Boot patches:
+ minivm-null-pointer-dereference-fix.patch uploaded by bootc
+ (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
+ Tested by: Chris Warr ........ Merged revisions 379608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379609 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /: Re-add merge properties
+
+2013-01-20 03:06 +0000 [r379583] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix issues with skinny sessions Fixes
+ a couple of issues with the way skinny handles sessions by
+ ensuring sessions aren't used after being freed. Some other minor
+ changes. Review: https://reviewboard.asterisk.org/r/2272/
+ ........ Merged revisions 379582 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-19 20:54 +0000 [r379549] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
+ builtin roundf() for systems lacking it. (closes issue
+ ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
+ Reported-by: Ovidiu Sas ........ Merged revisions 379547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379548 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-19 00:19 +0000 [r379518] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/asterisk.c: Fix astcanary startup problem due to wrong
+ pid value from before daemon call When Asterisk forks itself into
+ the background via a call to daemon, it must re-set the pid value
+ of the new process. Otherwise, astcanary gets the pid value of
+ the process before the fork, which prevents it from running.
+ Asterisk eventually starts lowering its priority, as it can no
+ longer communicate with the proverbial canary in the coal mine.
+ This patch ensures that the correct process identifier is used by
+ astcanary. Note that this is getting committed to 10 as a
+ regression fix. (closes issue ASTERISK-20947) Reported by: Jakob
+ Hirsch Tested by: mjordan patches:
+ asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
+ (license 6113) ........ Merged revisions 379509 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 379513 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-18 22:42 +0000 [r379495] David M. Lee <dlee@digium.com>
+
+ * configure.ac, Makefile, configure, main/Makefile: Up the minimum
+ OS X version to 10.6. * This allows us to remove some
+ special-case build logic. * 10.5 is down to less that 8% of the
+ OS X market share. 10.4 is down to under 2%. * Apple is no longer
+ releasing security updates for 10.5 and earlier.
+
+2013-01-18 21:52 +0000 [r379479] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_confbridge.c: Fix regression in Confbridge user count
+ When the restructuring work got committed to Confbridge in
+ r375470 to fix many open issues, it caused a regression in the
+ reported count of users when conference information was requested
+ via CLI or manager. This corrects the user count and user
+ information displayed when listing conference information from
+ the CLI and manager. (closes issue ASTERISK-20938) Reported By:
+ Timo Teras Patches: confbridge-list.patch uploaded by Timo Teras
+ (license 5409) ........ Merged revisions 379478 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-18 21:35 +0000 [r379477] David M. Lee <dlee@digium.com>
+
+ * Makefile, /, configure, main/Makefile, configure.ac,
+ UPGRADE-11.txt, UPGRADE.txt, makeopts.in: Specify the -rpath
+ linker flag when prefix != /usr. This allows Asterisk to start
+ without having to specify the LD_LIBRARY_PATH. This can be
+ disabled by passing --disable-rpath to configure. (closes issue
+ ASTERISK-20407) Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2132/ ........ Merged
+ revisions 379475 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-18 18:25 +0000 [r379461] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_voicemail.c: app_voicemail: Improve msg_id handling
+ app_voicemail will no longer issue error messages when it
+ retrieves an msg_id with a NULL value from realtime and will
+ instead simply populate the msg_id field with a newly generated
+ msg_id. In addition, this patch changes the way msg_ids are
+ generated to eliminate certain causes of duplicate IDs appearing
+ within a single system. In addition, when messages are copied,
+ they will now receive a new msg_id. (closes issue ASTERISK-20717)
+ Reported by: Alec Davis Review:
+ https://reviewboard.asterisk.org/r/2220/ ........ Merged
+ revisions 379460 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-18 15:42 +0000 [r379432] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_threadpool.c (added), tests/test_taskprocessor.c
+ (added), main/threadpool.c (added), main/taskprocessor.c,
+ include/asterisk/threadpool.h (added), /,
+ include/asterisk/taskprocessor.h: Add threadpool support to
+ Asterisk. This commit consists of two parts. Part one changes the
+ taskprocessor API to be less self-contained. Instead, the
+ taskprocessor is now more of a task queue that informs a listener
+ of changes to the queue. The listener then has the responsibility
+ of executing the tasks as it pleases. There is a default listener
+ implementation that functions the same way as "classic"
+ taskprocessors, in that it creates a single thread for tasks to
+ execute in. Old users of taskprocessors have not been altered and
+ still function the same way. Part two introduces the threadpool
+ API. A threadpool is a special type of taskprocessor listener
+ that has multiple threads associated with it. The threadpool also
+ has an optional listener that can adjust the threadpool as
+ conditions change. In addition the threadpool has a set of
+ options that can allow for the threadpool to grow and shrink on
+ its own as tasks are added and executed. Both set of changes
+ contain accompanying unit tests. (closes issue ASTERISK-20691)
+ Reported By: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2242
+
+2013-01-18 05:31 +0000 [r379394] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h: Fix Record-Route parsing
+ for large headers. Record-Route parsing copied the header into a
+ char[256] array, which can be a problem if the header is longer
+ than that. This patch parses the header in place, without the
+ copy, avoiding the issue. In addition to the original patch, I
+ added a unit test for the new get_in_brackets_const function.
+ (closes issue ASTERISK-20837) Reported by: Corey Farrell Patches:
+ chan_sip-build_route-optimized-rev1.patch uploaded by Corey
+ Farrell (license 5909) (with minor changes by dlee) ........
+ Merged revisions 379392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379393 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-17 02:32 +0000 [r379344] Matthew Jordan <mjordan@digium.com>
+
+ * /, addons/chan_mobile.c: Fix issue where chan_mobile fails to
+ bind to first available port Per the bluez API, in order to bind
+ to the first available port, the rc_channel field of the socket
+ addressing structure used to bind the socket should be set to 0.
+ Previously, Asterisk had set the rc_channel field set to 1,
+ causing it to connect to whatever happens to be on port 1. We
+ could probably not explicitly set rc_channel to 0 since we memset
+ the struct earlier, but explicitly setting it will hopefully
+ prevent someone from coming in and setting it to some explicit
+ port in the future. (closes issue ASTERISK-16357) Reported by:
+ challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
+ eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
+ Nikolay Ilduganov (license 6253) ........ Merged revisions 379342
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 379343 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-16 22:51 +0000 [r379312] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Further fix misinformation in the description
+ of manager MailboxStatus command. The description still claimed
+ that it returned the number of messages rather than whether there
+ were messages waiting. ........ Merged revisions 379310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379311 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-16 21:13 +0000 [r379278] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/install_prereq, /: Reduce number of packages
+ install_prereq installs on Debian systems. 'search' will look for
+ any package containing the name provided, so we need to force a
+ more exact search. ........ Merged revisions 379276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379277 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-16 18:09 +0000 [r379231-379233] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Reduce call-id logging resource usage. Since
+ there is no need for the call-id logging ao2 object to have a
+ lock, don't create it with one. ........ Merged revisions 379232
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix compile error. (issue
+ ASTERISK-15456) ........ Merged revisions 379226 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379230 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-16 17:46 +0000 [r379144-379229] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd, /: Let
+ documentation reference links specify which module they're
+ linking to Again, since res_jabber/res_xmpp have duplicate APIs,
+ their documentation ref links have to specify which reference
+ they're referring to. The various documentation parsers can
+ interpret the module attribute however they want in order to
+ construct the appropriate links. ........ Merged revisions 379228
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd: Multiple
+ revisions 379209-379210 ........ r379209 | mjordan | 2013-01-16
+ 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines Add module tags to
+ documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp
+ provide the same APIs (app/func/manager/etc.), the XML
+ documentation for each needs to call out which module is
+ providing the documentation. The module attribute has been added
+ to the various XML fragments for this purpose. ........ r379210 |
+ mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
+ Update the dtd to actually *support* the module attribute in all
+ elements Mea culpa. ........ Merged revisions 379209-379210 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * addons/chan_mobile.c, /: Fix parsing SMSSRC for SMS messages The
+ parser for SMS messages would incorrectly parse out the from
+ number. The parsing would incorrectly start scanning for the from
+ number at the same index as the first double quote ("); this
+ would inadvertently cause it to treat the first double quote as
+ the terminating double quote for the from number as well. The
+ SMSSRC should now populate correctly. (closes issue
+ ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
+ patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
+ issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
+ sms-sender-fix.diff uploaded by roeften (license 5884) ........
+ Merged revisions 379178 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379179 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_misdn.c, /: Set the INVALID_EXTEN channel variable
+ when chan_misdn forces the 'i' extension The chan_misdn channel
+ driver will send a channel with an invalid destination to the 'i'
+ extension itself if said extension can be reached. It forgot,
+ however, to set the INVALID_EXTEN channel variable when it
+ bounces the channel to this extension. Dialplan writers
+ everywhere moaned at yet another inconsistency. This is yet
+ another example of why duplicating logic in multiple places
+ results in bugs that stick around in Jira for just under three
+ years. Yes: ASTERISK-15456 was created on January 18th, 2010.
+ Patch committed on January 15th, 2013. Ouch. (closes issue
+ ASTERISK-15456) Reported by: Thomas Omerzu patches:
+ chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
+ 5927) ........ Merged revisions 379145 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379146 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * CHANGES, addons/chan_mobile.c: Add busy detection to chan_mobile
+ From the patch author: "First this patch adds general support for
+ busy detection. It also adds support for the ECAM command at Sony
+ Ericsson phones and also signals busy when only early media was
+ received but the call got not answered." Review:
+ https://reviewboard.asterisk.org/r/323 (closes issue
+ ASTERISK-14527) Reported by: Artem Makhutov Tested by: Artem
+ Makhutov patches: busy-full5.patch uploaded by artem (license
+ 5757)
+
+2013-01-15 22:23 +0000 [r379128] Richard Mudgett <rmudgett@digium.com>
+
+ * main/bridging.c: Fix ast_bridge_features_register() not
+ registering builtin features. I broke. Ooops.
+
+2013-01-14 21:47 +0000 [r379021-379070] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/test.h: Fixed doc comment for ast_test_validate
+
+ * UPGRADE.txt, include/asterisk/manager.h, main/channel.c: Gently
+ reduce masquerade insanity Masquerades are an insane
+ implementation detail within Asterisk. It generates a number of
+ useless and confusing events, and manipulates channels in a way
+ that semantically doesn't make sense. I've given a fairly
+ thorough review of masquerade code and its usage on the wiki at
+ https://wiki.asterisk.org/wiki/x/IwBRAQ. While ultimately it
+ makes the most sense to abandon masquerades altogether, it will
+ take some time to completely irradicate. Even then, there may
+ always be code that's not worth rewriting to get rid of the
+ masquerade. This patch does two things to make masquerades
+ slightly less insane: * When swapping the names of the original
+ and clone channel, only emit a single rename event of original ->
+ original<ZOMBIE>. The original code issued three rename events to
+ accomplish the same end. * In addition to swapping the names of
+ the channels, also swap their uniqueid's. This allows the
+ 'Uniqueid' field to be used as a stable identifier for a channel
+ from and external interface, such as AMI. Review:
+ https://reviewboard.asterisk.org/r/2266/
+
+ * /, channels/chan_sip.c: Fix XML encoding of 'identity display' in
+ NOTIFY messages, continued. When r378933 was merged into 1.8, it
+ should have also escaped remote_display, since it will have the
+ same XML encoding problem when the caller/callee roles are
+ reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
+ ........ Merged revisions 379001 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379020 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-13 22:07 +0000 [r378985] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Reset RTP timestamp; sequence number
+ on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling
+ of RTP was modified to better account for out of order RTP
+ packets. This was accomplished by using the RTP timestamp and
+ sequence number to check for out of order packets. However, when
+ a SSRC change occurs, the timestamp and sequence number will no
+ longer have any relation to the previously received packets. The
+ variables tracking the timestamp and sequence number therefore
+ have to be reset. (closes issue ASTERISK-20906) Reported by:
+ Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
+ Brolman (license #6442) ........ Merged revisions 378967 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378984 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-12 06:43 +0000 [r378935] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/utils.h, /, channels/chan_sip.c,
+ tests/test_xml_escape.c (added), main/utils.c: Fix XML encoding
+ of 'identity display' in NOTIFY messages. XML encoding in
+ chan_sip is accomplished by naively building the XML directly
+ from strings. While this usually works, it fails to take into
+ account escaping the reserved characters in XML. This patch adds
+ an 'ast_xml_escape' function, which works similarly to
+ 'ast_uri_encode'. This is used to properly escape the
+ local_display attribute in XML formatted NOTIFY messages. Several
+ things to note: * The Right Thing(TM) to do would probably be to
+ replace the ast_build_string stuff with building an ast_xml_doc.
+ That's a much bigger change, and out of scope for the original
+ ticket, so I refrained myself. * It is with great sadness that I
+ wrote my own ast_xml_escape function. There's one in libxml2, but
+ it's knee-deep in libxml2-ness, and not easily used to one-off
+ escape a string. * I only escaped the string we know is causing
+ problems (local_display). At least some of the other strings are
+ URI-encoded, which should be XML safe. Rather than figuring out
+ what's safe and escaping what's not, it would be much cleaner to
+ simply build an ast_xml_doc for the messages and let the XML
+ library do the XML escaping. Like I said, that's out of scope.
+ (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter Review:
+ http://reviewboard.digium.internal/r/365/ ........ Merged
+ revision 378919 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 378933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378934 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-11 23:05 +0000 [r378918] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c, /: Retain XMPP filters across reconnections so
+ external modules continue to function as expected. Previously if
+ an XMPP client reconnected any filters added by an external
+ module were lost. This issue exhibited itself with chan_motif not
+ receiving and reacting to Jingle signaling. (closes issue
+ ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-11 22:31 +0000 [r378915] David M. Lee <dlee@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile,
+ res/res_json.exports.in (added), configure.ac,
+ include/asterisk/json.h (added), makeopts.in, tests/test_json.c
+ (added), contrib/scripts/install_prereq, res/res_json.c (added),
+ include/asterisk/test.h: Add JSON API for Asterisk. This provides
+ a JSON API by pulling in and wrapping the Jansson JSON
+ library[1]. The Asterisk API basically mirrors the Jansson
+ functionality, with a few minor tweaks. * Some names have been
+ asteriskified to protect the innocent. * Jansson provides both
+ reference-stealing and reference-borrowing versions of several
+ API's. The Asterisk API is exclusively reference-stealing for
+ operations that put elements into arrays and objects. * No
+ support for doubles, since we usually don't need that. * Coming
+ along for the ride is the ast_test_validate macro, which made the
+ unit tests much easier to write. [1]:
+ http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes
+ issue ASTERISK-20888) Review:
+ https://reviewboard.asterisk.org/r/2264/
+
+2013-01-10 02:40 +0000 [r378789-378889] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: * Simplify native bridge code in
+ ast_channel_bridge(). * Fix an unbalanced
+ manager_bridge_event(unlink) call if AST_SOFTHANGUP_UNBRIDGE is
+ set in ast_channel_bridge(). * Make ast_channel_bridge() use
+ common cleanup code when leaving the bridge.
+
+ * main/channel.c: * Removed some noop code and restructured an
+ else-if ladder in ast_generic_bridge(). * Trivial changes in
+ ast_channel_bridge().
+
+ * main/channel.c: * Simple optimization of bridge_playfile(). *
+ Squeezed some redundancy out of update_bridge_vars(). * Wrapped
+ long line in __ast_change_name_nolink().
+
+ * bridges/bridge_softmix.c, bridges/bridge_multiplexed.c: Trivial
+ misc bridge code changes. * softmix_bridge_thread() was
+ redundantly initializing an 8K buffer. * Promoted a debug message
+ to a warning in multiplexed_add_or_remove().
+
+ * main/logger.c: Fix logger.c function definition.
+
+ * include/asterisk/bridging_features.h, bridges/bridge_simple.c,
+ bridges/bridge_multiplexed.c, main/bridging.c: Trivial misc
+ bridge code changes.
+
+ * include/asterisk/test.h, main/test.c: Tweaked
+ __ast_test_suite_assert_notify() and
+ __ast_test_suite_event_notify() to be void functions.
+
+ * include/asterisk/test.h, main/test.c: * Whitespace changes. *
+ Made ast_test_init() match its prototype.
+
+ * main/udptl.c, main/rtp_engine.c: * Found some more places to use
+ ast_channel_lock_both(). * Minor optimization in
+ ast_rtp_instance_early_bridge().
+
+2013-01-09 20:30 +0000 [r378735-378783] David M. Lee <dlee@digium.com>
+
+ * main/rtp_engine.c, /: Fix end condition in
+ ast_rtp_lookup_mime_multiple2. The erroneous end condition would
+ never include the AST_RTP_CISCO_DTMF flag in the debug output.
+ (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........
+ Merged revisions 378776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378780 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, include/asterisk/strings.h: Move declaration of
+ ast_regex_string_to_regex_pattern futher down strings.h. The
+ prior location is before the declaration of struct ast_str, which
+ causes compiler warnings. (closes issue ASTERISK-20852) Reported
+ by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller
+ (license 6302) ........ Merged revisions 378747 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, include/asterisk/causes.h: Replace errant tabs with spaces in
+ causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
+ Patches: notabs.dif uploaded by snuffy (license 5024) ........
+ Merged revisions 378733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378734 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-09 00:05 +0000 [r378688-378691] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, /: app_queue: Fix incorrect assertion. (issue
+ ASTERISK-16115) ........ Merged revisions 378689 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378690 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+ apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+ that is in only one queue. When ringinuse=no queue members can
+ receive more than one call if these calls happen at nearly the
+ same time. * Fix so a queue member does not receive more than one
+ call from a queue. NOTE: This fix does not prevent multiple calls
+ to a member if the member is in more than one queue. * Did some
+ refactoring to eliminate some code redundancy. (issue
+ ASTERISK-16115) Reported by: nik600 Patches:
+ jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+ changes adding the queues.conf check_state_unknown option as it
+ was added in an attempt to fix this problem. The fix did not need
+ to be optional. The fix should not have tried to explicitly set
+ the device state. Setting the device state by something other
+ than the device introduces a race condition. I also could not see
+ how the change would be effective other than delaying the
+ app_queue code long enough for the device state to propagate to
+ app_queue. ........ Merged revisions 378663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378683 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378687 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-06 21:37 +0000 [r378623-378634] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Skinny blob cleanup Cleanup of red blobs
+ in chan_skinny and possible other small formatting issues.
+ Review: https://reviewboard.asterisk.org/r/2262/
+
+ * channels/chan_skinny.c: Add group and namedgroup pickup to skinny
+ Above says it all. Code by snuff, cleaned up by me. Review:
+ https://reviewboard.asterisk.org/r/2246/
+
+ * /, channels/chan_skinny.c: Rewrite skinny dialing to remove
+ threaded simpleswitch This rewrite changes skinny dialing from
+ the threaded simpleswitch to a scheduled timeout approach. There
+ were some underlying issues with the threaded simple switch with
+ occasional corruption and possible segfaults. Review:
+ https://reviewboard.asterisk.org/r/2240/ ........ Merged
+ revisions 378622 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-04 23:14 +0000 [r378593] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_srtp.c: res_srtp: Prevent a crash from occurring due
+ to srtp_create failures in srtp_create Under some circumstances,
+ libsrtp's srtp_create function deallocates memory that it wasn't
+ initially responsible for allocating. Because we weren't
+ initially aware of this behavior, this memory was still used in
+ spite of being unallocated during the course of the
+ srtp_unprotect function. A while back I made a patch which would
+ set this value to NULL, but that exposed a possible condition
+ where we would then try to check a member of the struct which
+ would cause a segfault. In order to address these problems,
+ ast_srtp_unprotect will now set an error value when it ends
+ without a valid SRTP session which will result in the caller of
+ srtp_unprotect observing this error and hanging up the relevant
+ channel instead of trying to keep using the invalid session
+ address. (closes issue ASTERISK-20499) Reported by: Tootai
+ Review:
+ https://reviewboard.asterisk.org/r/2228/diff/#index_header
+ ........ Merged revisions 378591 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378592 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-04 22:19 +0000 [r378585] Kinsey Moore <kmoore@digium.com>
+
+ * res/pjproject/aconfigure.ac, /, res/pjproject/build/common.mak,
+ res/pjproject/aconfigure: Fix pjproject compilation in certain
+ circumstances On a fresh checkout of Asterisk 11, running make
+ before ./configure could cause the pjproject subdirectory to get
+ in an odd state that would prevent compilation. This patch by
+ Tilghman prevents that from occurring. (closes issue
+ ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo
+ borges, Steve Lang patches: 20121208__ccar_solved.diff.txt
+ uploaded by Tilghman Lesher (license 5003) ........ Merged
+ revisions 378582 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-04 21:20 +0000 [r378565] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix SIP Notify Messages To Have The
+ Proper IP Address In The FROM Field On a multihomed server when
+ sending a NOTIFY message, we were not figuring out which network
+ should be used to contact the peer. This patch fixes the problem
+ by calling ast_sip_ouraddrfor() and then build_via() so that our
+ NOTIFY message contains the correct IP address. Also, a debug
+ message is being added to help follow the call-id changes that
+ occur. This was helpful for confirming that the IP address was
+ set properly since the call-id contains the IP address. It also
+ will be helpful for troubleshooting purposes when following a
+ call in the debug logs. (closes issue ASTERISK-20805) Reported
+ by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches:
+ asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2255/
+ ........ Merged revisions 378554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378559 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-04 21:18 +0000 [r378557] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Don't pass STUN packets through the
+ SRTP unprotect function. (closes issue AST-1036) Reported by:
+ jbigelow ........ Merged revisions 378553 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378555 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-04 16:44 +0000 [r378543] Andrew Latham <lathama@gmail.com>
+
+ * res/res_config_ldap.c: Doxygen Cleanups Baseline clean up of
+ formating to make room for extended documentation (issue
+ ASTERISK-20259)
+
+2013-01-03 22:14 +0000 [r378516] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_queue.c, /: Fix Queue Log Reporting Every Call
+ COMPLETECALLER With "h" Extension Present When the "h" extension
+ is present within the context of the queue, all calls are being
+ reported COMPLETECALLER even when the agent is hanging up the
+ call. This patch checks to see if the agent hung-up or not
+ instead of only relying on checking if the queue (caller) channel
+ hung-up or not. It would appear that having the h extension in
+ the mix, the pbx goes to the h extension, "hanging-up" the queue
+ channel and triggering the reporting of COMPLETECALLER. (closes
+ issue ASTERISK-20743) Reported by: call Tested by: call, Michael
+ L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2256/ ........ Merged
+ revisions 378514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378515 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-03 19:42 +0000 [r378488] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_agent.c: chan_agent: Fix wrapup time wait
+ response. * Made agent_cont_sleep() and agent_ack_sleep() stop
+ waiting if the wrapup time expires. agent_cont_sleep() had tried
+ but returned the wrong value to stop waiting. * Made
+ agent_ack_sleep() take a struct agent_pvt pointer instead of a
+ void pointer for better type safety. ........ Merged revisions
+ 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378487 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-03 18:51 +0000 [r378460] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, /: Add missing test event This test event was
+ missing from channel.c causing the dial_LS_options test to fail
+ intermittently because of a race condition where most code paths
+ emitted the test event but this one did not. The dial_LS_options
+ test should stop bouncing now. ........ Merged revisions 378455
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 378459 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-03 18:47 +0000 [r378429-378458] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_agent.c, /: chan_agent: Misc code cleanup. * Fix
+ off-nominal path resource cleanup in agent_request(). * Create
+ agent_pvt_destroy() to eliminate inlined versions in many places.
+ * Pull invariant code out of loop in add_agent(). * Remove
+ redundant module user references in login_exec(). * Remove unused
+ struct agent_pvt logincallerid[] member. ........ Merged
+ revisions 378456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_agent.c: chan_agent: Fix agent_indicate()
+ locking. Avoid deadlock potential with local channels and
+ simplify the locking. ........ Merged revisions 378427 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378428 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-03 16:04 +0000 [r378414] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configs/voicemail.conf.sample, apps/app_directory.c,
+ contrib/realtime/mysql/voicemail.sql: Add aliases to the
+ Directory. This is an interesting feature that allows additional
+ strings to be used to search the Directory, primarily intended to
+ be used with nicknames, but could be used with affiliations and
+ the like. Because the name field is used in more than one place
+ (such as email notifications), it is important that these
+ additional strings not be placed in the name field, but be
+ specified separately. Review:
+ https://reviewboard.asterisk.org/r/2244/
+
+2013-01-03 15:40 +0000 [r378412] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_xmpp.c: Prevent exhaustion of system resources through
+ exploitation of event cache This patch changes res_xmpp to no
+ longer cache events under certain circumstances. (issue
+ ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua
+ Colp Tested by: kmoore ........ Merged revisions 378411 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-03 15:37 +0000 [r378377-378410] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_xmpp.c: Prevent crashes in res_xmpp when receiving
+ large messages Similar to r378287, res_xmpp was marshaling data
+ read from an external source onto the stack. For a sufficiently
+ large message, this could cause a stack overflow. This patch
+ modifies res_xmpp in a similar fashion to res_jabber by removing
+ the stack allocation, as it was unnecessary. (issue
+ ASTERISK-20658) Reported by: wdoekes ........ Merged revisions
+ 378409 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * addons/app_mysql.c: Clean up app_mysql's application entry points
+ to properly parse arguments When parsing arguments, application
+ entry points should not attempt to directly modify the parameters
+ to the function. This patch properly duplicates the passed in
+ parameters before attempting to parse them. (issue
+ ASTERISK-20658) Reported by: wdoekes patches:
+ issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license
+ 5674)
+
+ * main/config.c, funcs/func_realtime.c, /: Prevent crashes from
+ occurring when reading from data sources with large values When
+ reading configuration data from an Asterisk .conf file or when
+ pulling data from an Asterisk RealTime backend, Asterisk was
+ copying the data on the stack for manipulation. Unfortunately, it
+ is possible to read configuration data or realtime data from some
+ data source that provides a large blob of characters. This could
+ potentially cause a crash via a stack overflow. This patch
+ prevents large sets of data from being read from an ARA backend
+ or from an Asterisk conf file. (issue ASTERISK-20658) Reported
+ by: wdoekes Tested by: wdoekes, mmichelson patches: *
+ issueA20658_dont_process_overlong_config_lines.patch uploaded by
+ wdoekes (license 5674) * issueA20658_func_realtime_limit.patch
+ uploaded by wdoekes (license 5674) ........ Merged revisions
+ 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378376 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-02 21:23 +0000 [r378374] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, main/features.c, include/asterisk/channel.h:
+ Fix AMI redirect action with two channels failing to redirect
+ both channels. The AMI redirect action can fail to redirect two
+ channels that are bridged together. There is a race between the
+ AMI thread redirecting the two channels and the bridge thread
+ noticing that a channel is hungup from the redirects. * Made the
+ bridge wait for both channels to be redirected before exiting. *
+ Made the AMI redirect check that all required headers are present
+ before proceeding with the redirection. * Made the AMI redirect
+ require that any supplied ExtraChannel exist before proceeding.
+ Previously the code fell back to a single channel redirect
+ operation. (closes issue ASTERISK-18975) Reported by: Ben Klang
+ (closes issue ASTERISK-19948) Reported by: Brent Dalgleish
+ Patches: jira_asterisk_19948_v11.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak
+ Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/
+ ........ Merged revisions 378356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378358 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-02 18:11 +0000 [r378288-378322] Matthew Jordan <mjordan@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, channels/chan_skinny.c,
+ include/asterisk/event_defs.h, main/features.c, main/event.c,
+ apps/app_confbridge.c, apps/confbridge/conf_state_empty.c,
+ funcs/func_devstate.c, res/res_calendar.c,
+ include/asterisk/devicestate.h, channels/chan_local.c, /,
+ main/ccss.c, channels/chan_sip.c, apps/app_meetme.c,
+ main/channel_internal_api.c, channels/chan_agent.c,
+ main/devicestate.c, include/asterisk/channel.h, res/res_jabber.c,
+ apps/app_queue.c, channels/chan_iax2.c: Prevent exhaustion of
+ system resources through exploitation of event cache Asterisk
+ maintains an internal cache for devices in the event subsystem.
+ The device state cache holds the state of each device known to
+ Asterisk, such that consumers of device state information can
+ query for the last known state for a particular device, even if
+ it is not part of an active call. The concept of a device in
+ Asterisk can include entities that do not have a physical
+ representation. One way that this occurred was when anonymous
+ calls are allowed in Asterisk. A device was automatically created
+ and stored in the cache for each anonymous call that occurred;
+ this was possible in the SIP and IAX2 channel drivers and through
+ channel drivers that utilized the res_jabber/res_xmpp resource
+ modules (Gtalk, Jingle, and Motif). These devices are never
+ removed from the system, allowing anonymous calls to potentially
+ exhaust a system's resources. This patch changes the event cache
+ subsystem and device state management to no longer cache devices
+ that are not associated with a physical entity. (issue
+ ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua
+ Colp Tested by: kmoore patches: event-cachability-3.diff uploaded
+ by jcolp (license 5000) ........ Merged revisions 378303 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378320 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378321 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_jabber.c, channels/sip/include/sip.h, /,
+ channels/chan_sip.c, main/http.c: Resolve crashes due to large
+ stack allocations when using TCP Asterisk had several places
+ where messages received over various network transports may be
+ copied in a single stack allocation. In the case of TCP, since
+ multiple packets in a stream may be concatenated together, this
+ can lead to large allocations that overflow the stack. This patch
+ modifies those portions of Asterisk using TCP to either favor
+ heap allocations or use an upper bound to ensure that the stack
+ will not overflow: * For SIP, the allocation now has an upper
+ limit * For HTTP, the allocation is now a heap allocation instead
+ of a stack allocation * For XMPP (in res_jabber), the allocation
+ has been eliminated since it was unnecesary. Note that the HTTP
+ portion of this issue was independently found by Brandon Edwards
+ of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
+ wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
+ ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
+ 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
+ wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
+ uploaded by wdoekes (license 5674) ........ Merged revisions
+ 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378286 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378287 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-01-01 19:02 +0000 [r378259] Andrew Latham <lathama@gmail.com>
+
+ * contrib/scripts/install_prereq: Add UUID packages now required to
+ configure In ASTERISK-20726 UUID was added to Asterisk. This
+ commit is to add the dependancies to the install script
+
+2013-01-01 17:10 +0000 [r378248-378249] Sean Bright <sean@malleable.com>
+
+ * main/translate.c: Revert 378248. I changed the logic of this
+ function unitentionally, pointed out by file.
+
+ * main/translate.c: Bail out early when building an ast_trans_pvt
+ and the translator doesn't supply a 'newpvt'
+
+2012-12-31 14:46 +0000 [r378220] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
+ without crypto info This ensures that Asterisk rejects encrypted
+ media streams (RTP/SAVP audio and video) that are missing
+ cryptographic keys and ensures that the incoming SDP is
+ consistent with RFC4568 as far as having a crypto attribute
+ present for any SAVP streams. Review:
+ https://reviewboard.asterisk.org/r/2204/ ........ Merged
+ revisions 378217 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378218 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378219 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-20 21:51 +0000 [r378166] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Give the causes[] a struct name. ........
+ Merged revisions 378164 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378165 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-18 17:48 +0000 [r378122] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, /: Add test events for time limit-related hangups
+ This patch adds hangup-related test events in order to support
+ testing of time-limited bridges. This aids in testing the S() and
+ L() bridge options. (issue SWP-4713) ........ Merged revisions
+ 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378120 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378121 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-17 23:10 +0000 [r378081-378095] Richard Mudgett <rmudgett@digium.com>
+
+ * main/loader.c, /: Fix potential double free when unloading a
+ module. ........ Merged revisions 378092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378093 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378094 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_local.c, /: Make chan_local module references tied
+ to local_pvt lifetime. The chan_local module references were
+ manually tied to the existence of the ;1 and ;2 channel links. *
+ Made chan_local module references tied to the existence of the
+ local_pvt structure as well as automatically take care of the
+ module references. * Tweaked the wording of the local_fixup()
+ failure warning message to make sense. Review:
+ https://reviewboard.asterisk.org/r/2181/ ........ Merged
+ revisions 378088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378089 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378090 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_local.c: chan_local: Parse dial string
+ consistently. * Fix local_alloc() unexpected limitation of exten
+ and context length from a combined length of 80 characters to a
+ normal 80 characters each. * Made local_alloc() and
+ local_devicestate() parse the same way.
+
+2012-12-17 20:59 +0000 [r378074] Jason Parker <jparker@digium.com>
+
+ * /, main/Makefile: Make libasteriskssl.so symlink use a relative
+ path. This was causing issues when using DESTDIR, since the path
+ to which the link pointed is not likely to exist (and not useful
+ to exist) on the target system. (issue ASTNOW-284) ........
+ Merged revisions 378073 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-17 20:34 +0000 [r378072] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: chan_local: Misc lock and ref tweaks. *
+ awesome_locking() does not need to thrash the pvt lock as much. *
+ local_setoption() does not need to check for NULL pvt on cleanup
+ since it will never be NULL. * Made ref the pvt before locking
+ for consistency.
+
+2012-12-14 22:45 +0000 [r378064] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_agent.c: chan_agent: Remove some duplicated code.
+ No need to check for an agent twice. Santa does that.
+
+2012-12-14 22:34 +0000 [r378063] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c, UPGRADE.txt, CHANGES: Features: BRIDGE_FEATURES
+ variable automixmonitor support and use proper party
+ BRIDGE_FEATURES did not previously support the automixmonitor
+ feature. Now it does. In addition, the BRIDGE_FEATURES variable
+ would not apply features to the proper party based on whether the
+ feature option letter was in caps or in lowercase (both ways
+ would apply it to the caller). Now uppercase applies to the
+ caller while lowercase applies to the callee (like with the dial
+ option)
+
+2012-12-14 21:35 +0000 [r378029-378039] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, /: app_queue: Revert bad ringinuse=no patch.
+ With the option ringinuse=no set, the patch committed for
+ ASTERISK-16115 causes non-SIP queue members to never be called
+ because the device state is checked after a channel is created to
+ determine if the member is busy. These queue members always get
+ the "Member %s is busy, cannot dial" message. Most channel
+ drivers other than chan_sip use the default device state
+ handling. The default device-state state is considered in use or
+ unknown if the channel exists or not respectively. (closes issue
+ ASTERISK-20801) Reported by: rmudgett Patches:
+ jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 378036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378037 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 378038 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_queue.c: app_queue: Make update_status() not return
+ anything.
+
+2012-12-14 01:55 +0000 [r378006-378011] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix skinny to recognise vmexten in
+ general section of conf Fixup the vmexten so if globally set in
+ general section will be honored by chan_skinny. Also get rid of
+ the 'global_' part of variable name to match regexten. (closes
+ issue ASTERISK-20790) Reported by: snuffy Tested by: snuffy,
+ myself Patches: skinny-vm.diff uploaded by snuffy (license 5024)
+ ........ Merged revisions 378010 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_skinny.c: Add g722 codec support to skinny (closes
+ issue ASTERISK-20788) Reported by: snuffy Tested by: snuffy,
+ myself Patches: skinny-g722.diff uploaded by snuffy (license
+ 5024)
+
+2012-12-13 21:28 +0000 [r378002] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c,
+ apps/confbridge/conf_state.c, /,
+ apps/confbridge/include/confbridge.h,
+ include/asterisk/bridging.h, apps/app_confbridge.c: confbridge:
+ Fix MOH on simultaneous user entry to a new conference. When two
+ users entered a new conference simultaneously, one of the callers
+ hears MOH. This happened if two unmarked users entered
+ simultaneously and also if a waitmarked and a marked user entered
+ simultaneously. * Created a confbridge internal MOH API to
+ eliminate the inlined MOH handling code. Note that the conference
+ mixing bridge needs to be locked when actually starting/stopping
+ MOH because there is a small window between the conference join
+ unsuspend MOH and actually joining the mixing bridge. * Created
+ the concept of suspended MOH so it can be interrupted while
+ conference join announcements to the user and DTMF features can
+ operate. * Suspend any MOH until the user is about to actually
+ join the mixing bridge of the conference. This way any pre-join
+ file playback does not need to worry about MOH. * Made post-join
+ actions only play deferred entry announcement files. Changing the
+ user/conference state during that time is not protected or
+ controlled by the state machine. (closes issue ASTERISK-20606)
+ Reported by: Eugenia Belova Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2232/ ........ Merged
+ revisions 377992 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377993 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-13 21:25 +0000 [r378001] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Minor fixes for chan_skinny
+ Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
+ correct len of 2 strcmp in skinny_setdebug(). (see opticron's
+ review on https://reviewboard.asterisk.org/r/2240/) ........
+ Merged revisions 377991 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-13 21:20 +0000 [r378000] Sean Bright <sean@malleable.com>
+
+ * res/res_calendar_exchange.c: Make generate_exchange_uuid() always
+ return the passed ast_str pointer. I changed this code earlier to
+ return NULL if it wasn't able to generate a UUID, whereas the
+ earlier code would always return the ast_str that was passed in.
+ Switch back to returning the ast_str, only set it to the empty
+ string instead if UUID generation fails. We still do a validity
+ check later which will catch this and blow up if necessary.
+
+2012-12-13 21:15 +0000 [r377994] David M. Lee <dlee@digium.com>
+
+ * /: Fixed svn merge property breakage from r377986
+
+2012-12-13 18:28 +0000 [r377986] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix skinny debug tab completion Review
+ the syntax of the 'skinny debug' command to show more than just
+ 'show' for options to 'skinny debug' command. (closes issue
+ ASTERISK-20789) Reported by: snuffy Tested by: snuffy, myself
+ Patches: skinny-debug.diff uploaded by snuffy (license 5024)
+ ........ Merged revisions 377985 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-13 16:43 +0000 [r377981] David M. Lee <dlee@digium.com>
+
+ * configure.ac, configure, include/asterisk/autoconfig.h.in: Bail
+ configure if it can't find libuuid.
+
+2012-12-13 16:18 +0000 [r377977] Russell Bryant <russell@russellbryant.com>
+
+ * main/utils.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Remove compile time check HAVE_DEV_URANDOM. The
+ code was doing a runtime check, anyway. The compile time check
+ isn't always valid (cross-compiling, packages). Review:
+ https://reviewboard.asterisk.org/r/2245/
+
+2012-12-13 15:40 +0000 [r377975] Mark Michelson <mmichelson@digium.com>
+
+ * main/taskprocessor.c: Re-add taskprocessor cleanup code that was
+ removed by the UUID merge.
+
+2012-12-13 15:37 +0000 [r377974] Sean Bright <sean@malleable.com>
+
+ * res/res_calendar_exchange.c: Use the UUID API to generate and
+ validate UUIDs for res_calendar_exchange. Currently the
+ res_calendar_exchange module uses its own method of generating
+ UUIDs using ast_random(). Now that we have a UUID API we should
+ use that instead.
+
+2012-12-13 15:37 +0000 [r377973] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_clialiases.c: The UUID commit removed changes made in
+ res_clialiases.c This puts back in the changes that are designed
+ to work around a memory leak fix in the CLI code.
+
+2012-12-13 15:24 +0000 [r377972] David M. Lee <dlee@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Fixed
+ configure.ac to look for proper uuid.h file Introduced in
+ r377846, the configure script was looking for uuid.h instead of
+ uuid/uuid.h.
+
+2012-12-13 15:22 +0000 [r377971] Brent Eagles <beagles@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: This change adds a SIP peer configuration
+ feature to allow the peer's configured codecs to take precedence
+ on an outgoing call. This change introduces a new peer
+ configuration property named 'ignore_requested_pref' that causes
+ the requested codec to be ignored when determining the preferred
+ codec for an outgoing call leg. The consequence is that
+ Asterisk's usual efforts to prefer avoiding transcoding can be
+ overridden on a peer-by-peer basis where appropriate.
+
+2012-12-13 14:28 +0000 [r377966] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure Min-SE is included in outbound
+ INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+ value is not 90 (the default) and session timers are not
+ disabled. This has the effect of Asterisk following RFC4028 more
+ closely with regard to 422 responses and preventing situations in
+ which Asterisk would be forced to temporarily accept a call to
+ tear it down based on a Session-Expires below the locally
+ configured Min-SE. (issue SWP-5051) Review:
+ https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+ Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 377947 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377948 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-12 22:43 +0000 [r377925] Rusty Newton <rnewton@digium.com>
+
+ * sounds/Makefile, /: Incremented EXTRA_SOUNDS_VERSION in
+ sounds/Makefile to 1.4.12 for new Extra Sounds releases See
+ CHANGES-* files in English extra 1.4.12 tarballs for new sound
+ prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
+ (closes AST-755) Reported by: John Bigelow ........ Merged
+ revisions 377922 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377923 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377924 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-12 04:43 +0000 [r377915] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/features.c: Convert Dynamic Features Buffer To Use ast_str
+ Currently, the buffer for the dynamic features list is set to a
+ fixed size of 128. If the list is bigger than that, it results in
+ the dynamic feature(s) not being recognized. This patch changes
+ the buffer from a fixed size to a dynamic one. (closes issue
+ ASTERISK-20680) Reported by: Clod Patry Tested by: Michael L.
+ Young Patches: asterisk-20680-dynamic-features-v2.diff uploaded
+ by Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2221/
+
+2012-12-12 00:02 +0000 [r377906-377911] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix a potential deadlock in chan_sip
+ during transfers. The issue comes from the fact that transfers
+ may perform a redirecting update on a channel. The issue is that
+ lock inversion between the channel and its tech_pvt occurs since
+ the channel lock is released during the transfer process. The fix
+ is to move when the redirecting update occurs to a place where
+ neither the tech_pvt or the channel is locked so that the two can
+ be locked in the proper order. (closes issue ASTERISK-20708)
+ reported by Mark Michelson patches: ASTERISK-20708-3.patch
+ uploaded by Mark Michelson (License #5049) Tested by: Tim
+ Ringenbach at Asteria Solutions Group ........ Merged revisions
+ 377910 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/features.c: Add test events necessary for bridging tests to
+ be able to properly run.
+
+2012-12-11 22:03 +0000 [r377884] Richard Mudgett <rmudgett@digium.com>
+
+ * main/aoc.c, main/image.c, main/cel.c, main/timing.c,
+ main/channel.c, main/data.c, main/stun.c, /, main/file.c,
+ main/http.c: Cleanup CLI commands on exit for several files.
+ (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ unregister-cli-multiple-all.patch (license #5909) patch uploaded
+ by Corey Farrell ........ Merged revisions 377881 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377882 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377883 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-11 21:53 +0000 [r377878-377880] Mark Michelson <mmichelson@digium.com>
+
+ * /: And remove svnmerge-integrated property.
+
+ * /: Remove automerge properties.
+
+2012-12-11 21:22 +0000 [r377867] Richard Mudgett <rmudgett@digium.com>
+
+ * main/udptl.c, /: Cleanup udptl on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+ Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377848 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377849 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-11 21:04 +0000 [r377844-377846] Mark Michelson <mmichelson@digium.com>
+
+ * configure.ac, include/asterisk/uuid.h (added),
+ main/taskprocessor.c, tests/test_uuid.c (added), main/asterisk.c,
+ main/uuid.c (added), res/res_clialiases.c, /, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile: Add UUID support
+ to Asterisk. This provides a common API for dealing with unique
+ identifiers. The API provides methods to create, parse, copy, and
+ stringify UUIDs. An accompanying unit test is provided that tests
+ all operations. (closes issue ASTERISK-20726) reported by Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2217
+
+ * /, res/res_clialiases.c: Fix crash that can occur if CLI
+ registration fails for an aliased command. A recent memory leak
+ fix in main/cli.c causes an ast_cli_entry's command field to be
+ freed and NULLed if ast_cli_register() fails. res_clialiases was
+ ignoring the return value of ast_cli_register() and was then
+ passing the NULL command off to a a hash function. This resulted
+ in a crash. The fix is not to ignore the erroneous return value.
+ If ast_cli_register() fails, then we do not continue trying to
+ process the current alias. ........ Merged revisions 377840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377842 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377843 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-11 20:46 +0000 [r377707-377841] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
+ CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell
+ taskprocessor-cleanup-10-only.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377838 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377839 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
+ exit. * Unreference hints and statecbs containers on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
+ Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
+ Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377807 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377808 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
+ destroy verbosers and logchannels lists on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ logger-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell Modified ........ Merged revisions 377771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377772 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377773 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/indications.c, /: Cleanup indications on exit. * Made
+ ast_unregister_indication_country() unlink the found tone zone
+ before selecting a new default_tone_zone to make it impossible to
+ select the tone zone being unregistered again. * Ringcadence is
+ no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
+ commands and destroy default_tone_zone on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ indications-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell Modified ........ Merged revisions 377740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377741 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377742 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/event.c: Cleanup event on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ event_shutdown-10-only.patch (license #5909) patch uploaded by
+ Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909)
+ patch uploaded by Corey Farrell ........ Merged revisions 377708
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 377709 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377710 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/dnsmgr.c, /: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
+ and CLI commands on exit. (issue ASTERISK-20649) Reported by:
+ Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
+ patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell Modified ........
+ Merged revisions 377704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377705 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377706 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-10 16:56 +0000 [r377626-377658] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
+ When using res_fax_digium, the T.38 CED tone was not being
+ provided properly which would cause some incoming faxes to fail.
+ This was not an issue with res_fax_spandsp since it does not
+ strictly honor the send_ced flag and sends the CED tone whenever
+ receiving a T.38 fax. (closes issue FAX-343) Reported-by:
+ Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions
+ 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377657 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Handle Session-Expires less than local
+ Min-SE in 200 OK Ensure that a call is immediately torn down if a
+ Session-Expires value received in a 200 OK is less than the local
+ Min-SE. This also prevents Asterisk from allowing calls with
+ Session-Expires below the RFC4028-mandated minimum (90s). (closes
+ issue ASTERISK-20653) Review:
+ https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
+ ........ Merged revisions 377623 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377624 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377625 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-10 07:03 +0000 [r377579-377595] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Add firmware information to CLI devices
+ listing
+
+ * channels/chan_unistim.c, /: Fix codec mismatch Fix code to send
+ in both rx and tx open stream messages correct codecs. Found that
+ on phase 0/1 phones wrong codecs cause to no audio in some
+ situations. (issue ASTERISK-20183) ........ Merged revisions
+ 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377592 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377593 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_unistim.c, /: Remove trailing whitespaces in number
+ from incoming redial list. Reported by: Igor Olhovskiy ........
+ Merged revisions 377577 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-10 01:41 +0000 [r377506-377512] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/xmldoc.c, /: Improve documentation by making all of the
+ colors used readable, no matter what the background color is.
+ Dark blue on a black background is unreadable, as is yellow on a
+ light background. This patch turns on the bright attribute for
+ colors when on a dark background and turns *off* the bright
+ attribute when the -W command line option is used (indicating a
+ _light_ background). This ensures that text is readable in both
+ cases. Patch by: tilghman Review:
+ https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
+ 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377511 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, addons/cdr_mysql.c: Remove some dead code and additionally
+ handle a case that wasn't handled. ........ Merged revisions
+ 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377504 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377505 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-09 01:23 +0000 [r377463] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Add missing support for "who hung up"
+ to chan_motif. (closes issue ASTERISK-20671) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2208/ ........
+ Merged revisions 377462 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-08 00:30 +0000 [r377402-377434] Richard Mudgett <rmudgett@digium.com>
+
+ * contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
+ allow/disallow in MySQL contrib script. Using the contrib
+ sippeers.sql script to create the sippeers MySQL table would
+ result in being unable to place calls if you set the disallow
+ value to all. (closes issue ASTERISK-20756) Reported by: Andre
+ Luis Patches: sippeers.patch patch uploaded by Andre Luis
+ ........ Merged revisions 377431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377432 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377433 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
+ allocation dumps. ........ Merged revisions 377398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377399 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377401 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-07 22:08 +0000 [r377384] Kinsey Moore <kmoore@digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
+ show" CLI command. In r306010 "Asterisk media architecture
+ conversion - no more format bitfields", the logic for
+ incrementing encoders and decoders when opening transcoder
+ channels was changed without making the corresponding change when
+ decrementing encoder / decoder channels. The result being that
+ when a channel was destroyed, codec_dahdi couldn't properly tell
+ if it was an encoder or decoder, and the default case is to
+ assume it was a decoder. This could result in negative numbers
+ for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
+ encoders/decoders of 92 channels are in use. (closes issue
+ ASTERISK-19921) Patch-by: Shaun Ruffell ........ Merged revisions
+ 377382 from http://svn.asterisk.org/svn/asterisk/branches/10
+ ........ Merged revisions 377383 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-07 00:00 +0000 [r377356] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c, apps/confbridge/conf_config_parser.c, /:
+ confbridge: Fix some resource leaks on conference teardown. *
+ Made destroy_conference_bridge() destroy a missed ast_mutex_t and
+ ast_cond_t. * Made join_conference_bridge() init the
+ ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
+ destroy them unconditionally. * Made join_conference_bridge()
+ abort if the new conference could not be added to the conferences
+ container. * Made leave_conference() discard any post-join
+ actions if join_conference_bridge() had to abort early. * Made
+ the join_conference_bridge() diagnostic messages better describe
+ what happened. * Renamed leave_conference_bridge() to
+ leave_conference() and made it only take a conference user
+ pointer. The conference pointer was redundant. * Made
+ conf_bridge_profile_copy() use struct copy instead of memcpy(). *
+ No need to lock the conference in start_conf_record_thread()
+ since all of the callers already have it locked. ........ Merged
+ revisions 377354 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377355 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-06 17:29 +0000 [r377329-377341] Russell Bryant <russell@russellbryant.com>
+
+ * /: Recorded merge of revisions 377340 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Add CLI
+ tab completion to 'acl show'. The 'acl show' CLI command allows
+ you to show the details about a specific named ACL in acl.conf.
+ This patch adds tab completion to the command. Review:
+ https://reviewboard.asterisk.org/r/2230/
+
+ * main/named_acl.c: Minor code cleanup in named_acl.c. This patch
+ makes a few little cleanups to named_acl.c. A couple non-public
+ functions were made static and an opening brace for a function
+ was moved to its own line, per the coding guidelines.
+
+ * main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl
+ show' CLI command allows you to show the details about a specific
+ named ACL in acl.conf. This patch adds tab completion to the
+ command. Review: https://reviewboard.asterisk.org/r/2230/
+
+2012-12-06 14:26 +0000 [r377324] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c, /: Fix memory leak in 'manager show event' when
+ command entered incorrectly When the CLI command 'manager show
+ event' was run incorrectly and its usage instructions returned, a
+ reference to the event container was leaked. This would prevent
+ the container from being reclaimed when Asterisk exits. We now
+ properly decrement the count on the ao2 object using the nifty
+ RAII_VAR macro. Thanks to Russell for helping me stumble on this,
+ and Terry for writing that ridiculously helpful macro. ........
+ Merged revisions 377319 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-05 17:17 +0000 [r377263] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_srtp.c: res_srtp: Fix a crash caused by srtp_dealloc
+ on an already dealloced session When srtp_create fails, the
+ session may be dealloced or just not alloced. At the same time
+ though, the session pointer might not be set to NULL in this
+ process and attempting to srtp_dealloc it again will cause a
+ segfault. This patch checks for failure of srtp_create and sets
+ the session pointer to NULL if it fails. (closes issue
+ ASTERISK-20499) Reported by: tootai Review:
+ https://reviewboard.asterisk.org/r/2228/ ........ Merged
+ revisions 377256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377261 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377262 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-05 16:51 +0000 [r377260] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
+ connections. During the TLS re-work in chan_sip some TLS specific
+ code was moved into a separate function. This function operates
+ on a copy of the incoming SIP request. This copy was never
+ deinitialized causing a memory leak for each request processed.
+ This function is now given a SIP request structure which it can
+ use to copy the incoming request into. This reduces the amount of
+ memory allocations done since the internal allocated components
+ are reused between packets and also ensures the SIP request
+ structure is deinitialized when the TLS connection is torn down.
+ (closes issue ASTERISK-20763) Reported by: deti ........ Merged
+ revisions 377257 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377258 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377259 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-05 02:23 +0000 [r377214-377246] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/_private.h, main/asterisk.c, main/format.c:
+ Remove init_framer(). It no longer does anything.
+
+ * main/format.c, /: Fix registering core show codecs/codec CLI
+ commands twice. ........ Merged revisions 377241 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377244 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_config_parser.c, /: confbridge: Fix several
+ small issues. * Made func_confbridge_helper() allow an empty
+ value when setting options. You previously could not
+ Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
+ dialplan. * Made func_confbridge_helper() handle its datastore
+ better if multiple threads attempt to set the first CONFBRIDGE
+ option value on the channel. * Made the func_confbridge_helper()
+ only output one diagnostic message concerning the option. * Made
+ the bridge video_mode able to repeatedly change in the config
+ file and CONFBRIDGE dialplan function. The video_mode option
+ values are an enum and not independent of each other. * Made
+ handle_cli_confbridge_show_bridge_profile() better handle the
+ video_mode option. * Simplified datastore handling code in
+ conf_find_user_profile() and conf_find_bridge_profile(). (closes
+ issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter
+ ........ Merged revisions 377227 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377228 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_confbridge.c: confbridge: Update online XML
+ documentation. ........ Merged revisions 377212 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377213 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-04 13:01 +0000 [r377196] Russell Bryant <russell@russellbryant.com>
+
+ * contrib/scripts/install_prereq, /: Add libuuid to install_prereq
+ for Fedora. I ran this script and my build failed. pjproject
+ requires this. ........ Merged revisions 377195 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-03 23:00 +0000 [r377040-377168] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
+ Convert atexits list to a mutex instead of a rd/wr lock. The lock
+ is only write locked. * Move CLI verbose Asterisk ending message
+ to where AMI message is output in really_quit() to avoid further
+ surprises about using stuff already shutdown. (issue
+ ASTERISK-20649) Reported by: Corey Farrell ........ Merged
+ revisions 377165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377166 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377167 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/asterisk.c, /, include/asterisk/_private.h,
+ main/stdtime/localtime.c: Cleanup core main on exit. * Cleanup
+ time zones on exit. * Make exit clean/unclean report consistent
+ for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported
+ by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license
+ #5909) patch uploaded by Corey Farrell
+ core-cleanup-11-trunk.patch (license #5909) patch uploaded by
+ Corey Farrell Modified ........ Merged revisions 377135 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377136 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377137 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/config.c, /: Cleanup config cache on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ config-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell ........ Merged revisions 377104 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377105 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377106 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/cli.c: Cleanup CLI resources on exit and CLI command
+ registration errors. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377074 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377075 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
+ do_reload() return handling since it never returned anything
+ other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
+ Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377070 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377071 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/ccss.c: Fix CCSS CLI commands and logger level not
+ unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell ........ Merged revisions 377037 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377038 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 377039 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-03 16:45 +0000 [r377035] Olle Johansson <oej@edvina.net>
+
+ * res/res_rtp_asterisk.c: Formatting fixes
+
+2012-12-03 14:56 +0000 [r377022] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Fix an RTP instance reference count
+ leak in chan_motif. When setting up an RTP instance the RTCP
+ portion of the instance keeps a reference to the instance itself.
+ In order to release this reference and stop RTCP the stop API
+ call must be called before destroying the instance. (closes issue
+ ASTERISK-20751) Reported by: joshoa ........ Merged revisions
+ 377021 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-12-03 14:46 +0000 [r376998-377018] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Move functions to AFTER the block of forward
+ declarations of functions. It was a mess. The first part of
+ chan_sip.c is constants, declarations, structures and stuff, then
+ forward declarations and then actual code. It's still a mess, but
+ a bit less messy ;-)
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c: Formatting changes
+ Found a large amount of missing {} in the code before patching in
+ another branch
+
+2012-12-01 00:47 +0000 [r376984] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c, configs/motif.conf.sample: Tweak
+ extension used for incoming calls received on Motif. Based on
+ feedback from numerous individuals this patch tweaks incoming
+ calls to first look for an extension with the name of the
+ endpoint. If no such extension exists the call will silently fall
+ back to the "s" extension as it previously did. ........ Merged
+ revisions 376983 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-30 21:38 +0000 [r376953] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, /: chan_misdn: Fix sending
+ RELEASE_COMPLETE in response to SETUP. Fix sending a
+ RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+ have a B channel available to assign to the call. (closes issue
+ ABE-2869) Reported by: Guenther Kelleter Patches:
+ setup-reject_2.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified ........ Merged revision 376949 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 376950 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376951 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376952 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-30 17:08 +0000 [r376922] Sean Bright <sean@malleable.com>
+
+ * /, funcs/func_volume.c: Minor spelling fix to the VOLUME
+ documentation. ........ Merged revisions 376919 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376920 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376921 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-30 16:56 +0000 [r376918] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix potential crashes during SIP attended
+ transfers. The principal behind this patch is simple. During a
+ transfer, we manipulate channels that are owned by a separate
+ thread than the one we currently are running in, so it makes
+ sense that we need to grab a reference to the channels so that
+ they cannot disappear out from under us. In the wild, crashes
+ were sometimes seen when the transferring party would hang up the
+ call before the transfer target answered the call. The most
+ common place to see the crash occur was when attempting to send a
+ connected line update to the transferer channel. (closes issue
+ ASTERISK-20226) Reported by Jared Smith Patches:
+ ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+ Tested by: Jared Smith ........ Merged revisions 376901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376917 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-29 23:01 +0000 [r376867-376871] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
+ local_devicestate(). Regression introduced by ASTERISK-20390 fix.
+ (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
+ rmudgett ........ Merged revisions 376868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376869 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376870 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+ ........ Merged revisions 376864 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376865 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376866 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-29 21:58 +0000 [r376837] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Improve Code Readability And Fix Setting
+ natdetected Flag For 1.8, 10, 11 and trunk we are are improving
+ the code readability. For 11 and trunk, auto nat detection was
+ added. The natdetected flag was being set to 1 when the host
+ address in the VIA header did not specifiy a port. This patch
+ fixes this by setting the port on the temporary sock address used
+ to SIP_STANDARD_PORT in order for the sock address comparison to
+ work properly. (closes issue ASTERISK-20724) Reported by: Michael
+ L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2206/ ........ Merged
+ revisions 376834 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376835 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376836 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-29 17:16 +0000 [r376821] David M. Lee <dlee@digium.com>
+
+ * main/utils.c: Fixed ast_random's comment about locking. The
+ original comment was separated from the code at some point, and
+ didn't reflect the use of libc's other than glibc for Linux.
+
+2012-11-29 16:44 +0000 [r376820] Pedro Kiefer <pedro@kiefer.com.br>
+
+ * channels/chan_sip.c: Fix chan_sip websocket payload handling
+ Websocket by default doesn't return an ast_str for the payload
+ received. When converting it to an ast_str on chan_sip the last
+ character was being omitted, because ast_str functions expects
+ that the given length includes the trailing 0x00. payload_len
+ only has the actual string length without counting the trailing
+ zero. For most cases this passed unnoticed as most of SIP
+ messages ends with \r\n. (closes issue ASTERISK-20745) Reported
+ by: Iñaki Baz Castillo Review:
+ https://reviewboard.asterisk.org/r/2219/
+
+2012-11-29 00:48 +0000 [r376761-376791] Richard Mudgett <rmudgett@digium.com>
+
+ * main/asterisk.c, /, main/astmm.c: Add MALLOC_DEBUG atexit
+ unreleased malloc memory summary. * Adds the following CLI
+ commands to control MALLOC_DEBUG reporting of unreleased malloc
+ memory when Asterisk is shut down. memory atexit list on memory
+ atexit list off memory atexit summary byline memory atexit
+ summary byfunc memory atexit summary byfile memory atexit summary
+ off * Made check all remaining allocated region blocks atexit for
+ fence violations. * Increased the allocated region hash table
+ size by about three times. It still isn't large enough
+ considering the number of malloced blocks Asterisk uses. * Made
+ CLI "memory show allocations anomalies" use
+ regions_check_all_fences(). Review:
+ https://reviewboard.asterisk.org/r/2196/ ........ Merged
+ revisions 376788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376789 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376790 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/astmm.c, /: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
+ "memory show allocations" misspelling of anomalies option. The
+ command will still accept the original misspelling. *
+ Miscellaneous tweaks to CLI "memory show allocations" command
+ output format. * Made CLI "memory show summary" summarize by line
+ number instead of by function if a filename is given. * Made CLI
+ "memory show summary" sort its output by filename or
+ function-name/line-number depending upon request. * Miscellaneous
+ tweaks to CLI "memory show summary" command output format.
+ ........ Merged revisions 376758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376759 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376760 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-28 16:47 +0000 [r376728] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c, /: manager: Make challenge work with
+ allowmultiplelogin=no Prior to this patch, challenge would yield
+ a multiple logins error if used without providing the username
+ (which isn't really supposed to be an argument to challenge) if
+ allowmultiplelogin was set to no because allowmultiplelogin finds
+ a user with a zero length login name. This check is simply
+ disabled for the challenge action when the username is empty by
+ this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+ Patches: challenge_action_nomultiplelogin.diff uploaded by
+ Jonathan Rose (license 6182) ........ Merged revisions 376725
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 376726 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376727 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-28 00:13 +0000 [r376630-376691] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
+ char. The '-' char is supposed to be ignored by the dialplan
+ extension matching. Unfortunately, it's treatment is not handled
+ consistently throughout the extension matching code. * Made the
+ old exten matching code consistently ignore '-' chars. * Made the
+ old exten matching code consistently handle case in the matching.
+ * Made ignore empty character sets. * Fixed ast_extension_cmp()
+ to return -1, 0, or 1 as documented. The only user of it in
+ pbx_lua.c was testing for -1. It was originally returning the
+ strcmp() value for less than which is not usually going to be -1.
+ * Fix character set sorting if the sets have the same number of
+ characters and start with the same character. Character set [0-9]
+ now sorts before [02-9a] as originally intended. * Updated some
+ extension label and priority already in use warnings to also
+ indicate if the extension is aliased. (closes issue
+ ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+ Harzenetter Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2201/ ........ Merged
+ revisions 376688 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376689 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376690 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
+ pbx/pbx_dundi.c: Remove unnecessary channel module references. *
+ Removed call to ast_module_user_hangup_all() in
+ res_config_mysql.c since it is effectively a noop. No channels
+ can attach a reference to that module. * Removed call to
+ ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+ of unload_module() has already called it. * Removed redundant
+ channel module references in pbx_dundi.c. The registered dialplan
+ function callback dispatchers for the read/read2/write callbacks
+ already reference the module before calling. * pbx_dundi: Moved
+ unregistering CLI commands, DUNDi switch, and dialplan functions
+ to the first thing the unload_module() does. This will reduce the
+ chance of new channels using DUNDi services while the module is
+ being torn down. ........ Merged revisions 376657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376658 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376659 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
+ and use better names. * Update doxygen of AST_LIST_REMOVE().
+ ........ Merged revisions 376627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376628 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376629 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-23 00:02 +0000 [r376589] Matthew Jordan <mjordan@digium.com>
+
+ * main/logger.c, include/asterisk/lock.h, main/lock.c, /:
+ Re-initialize logmsgs mutex upon logger initialization to prevent
+ lock errors Similar to the patch that moved the fork earlier in
+ the startup sequence to prevent mutex errors in the recursive
+ mutex surrounding the read/write thread registration lock, this
+ patch re-initializes the logmsgs mutex. Part of the start up
+ sequence before forking the process into the background includes
+ reading asterisk.conf; this has to occur prior to the call to
+ daemon in order to read startup parameters. When reading in a
+ conf file, log statements can be generated. Since this can't be
+ avoided, the mutex instead is re-initialized to ensure a reset of
+ any thread tracking information. This patch also includes some
+ additional debugging to catch errors when locking or unlocking
+ the recursive mutex that surrounds locks when the DEBUG_THREADS
+ build option is enabled. DO_CRASH or THREAD_CRASH will cause an
+ abort() if a mutex error is detected. (issue ASTERISK-19463)
+ Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
+ 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376587 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376588 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-21 18:33 +0000 [r376575] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, main/test.c, tests/test_astobj2.c,
+ channels/chan_iax2.c, main/astobj2.c, include/asterisk/test.h,
+ main/channel.c: Add red-black tree container type to astobj2. *
+ Add red-black tree container type. * Add CLI command "astobj2
+ container dump <name>" * Added ao2_container_dump() so the
+ container could be dumped by other modules for debugging
+ purposes. * Changed ao2_container_stats() so it can be used by
+ other modules like ao2_container_check() for debugging purposes.
+ * Updated the unit tests to check red-black tree containers.
+ (closes issue ASTERISK-19970) Reported by: rmudgett Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/2110/
+
+2012-11-20 22:06 +0000 [r376562] David M. Lee <dlee@digium.com>
+
+ * /, res/res_http_websocket.c: Added missing newlines to websocket
+ ast_logs. Without these newlines, log messages just continue
+ tacking onto the same line, and do not flush immediately.
+ ........ Merged revisions 376561 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-20 19:09 +0000 [r376551] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Add "Require:
+ timer" to 200 OK responses when appropriate. The method by which
+ the Require header is added to 200 responses is inspired by the
+ method that Olle Johansson uses in his darjeeling-prack branch.
+ (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
+ behest of Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
+ 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376550 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-20 17:39 +0000 [r376541] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: Reduce CLI spam of "Extension Changed"
+ device state messages. Asterisk 11 follows RFC3265 that states
+ that after every subscribe or resubscribe a notify should be
+ sent. Thus the console if filled continuously with the following
+ after every subscribe; == Extension Changed 8512[phones] new
+ state IDLE for Notify User cisco1 In Asterisk 1.8 only changes
+ would be sent. Thus only when a device state changed was anything
+ emitted to the console. fix: Only print to console when device
+ state isn't forced. (closes issue ASTERISK-20706) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) ........
+ Merged revisions 376540 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-19 20:03 +0000 [r376472] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c, main/security_events.c,
+ main/indications.c: Fix most leftover non-opaque ast_str uses.
+ Instead of calling str->str, one should use ast_str_buffer(str).
+ Same goes for str->used as ast_str_strlen(str) and str->len as
+ ast_str_size(str). Review:
+ https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+ 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376471 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-19 02:14 +0000 [r376416-376457] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_astobj2.c: Fix uninitialized in this function error
+ With some versions of gcc, n_buckets will be flagged as being
+ uninitialized before use. While its technically impossible (since
+ the switch statement, even without a default, accounts for all
+ possibilities), we'll initialize the variable to 0 anyway.
+
+ * main/utils.c, main/asterisk.c, /: Reorder startup sequence to
+ prevent lockups when process is sent to background Although it is
+ very rare and timing dependent, the potential exists for the call
+ to 'daemon' to cause what appears to be a deadlock in Asterisk
+ during startup. This can occur when a recursive mutex is obtained
+ prior to the daemon call executing. Since daemon uses fork to
+ send the process into the background, any threading primitives
+ are unsafe to re-use after the call. Implementations of pthread
+ recursive mutexes are highly likely to store the thread
+ identifier of the thread that previously obtained the mutex. If
+ the mutex was locked prior to the fork, a subsequent unlock
+ operation will potentially fail as the thread identifier is no
+ longer valid. Since the mutex is still locked, all subsequent
+ attempts to grab the mutex by other threads will block. This
+ behavior exhibited itself most often when DEBUG_THREADS was
+ enabled, as this compile time option surrounds the mutexes in
+ Asterisk with another recursive mutex that protects the storage
+ of thread related information. This made it much more likely that
+ a recursive mutex would be obtained prior to daemon and unlocked
+ after the call. This patch does the following: a) It backports a
+ patch from Asterisk 11 that prevents the spawning of the
+ localtime monitoring thread. This thread is now spawned after
+ Asterisk has fully booted. b) It re-orders the startup sequence
+ to call daemon earlier during Asterisk startup. This limits the
+ potential of threading primitives being accessed by
+ initialization calls before daemon is called. c) It removes calls
+ to ast_verbose/ast_log/etc. prior to daemon being called.
+ Developers should send error messages directly to stderr prior to
+ daemon, as calls to ast_log may access recursive mutexes that
+ store thread related information. d) It reorganizes when thread
+ local storage is created for storing lock information during the
+ creation of threads. Prior to this patch, the read/write lock
+ protecting the list of threads in ast_register_thread would
+ utilize the lock in the thread local storage prior to it being
+ initialized; this patch prevents that. On a very related note,
+ this patch will *greatly* improve the stability of the Asterisk
+ Test Suite. Review: https://reviewboard.asterisk.org/r/2197
+ (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
+ mjordan ........ Merged revisions 376428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376431 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376441 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/confbridge/conf_state.c: Add a test event that reports
+ changes in ConfBridge state This patch adds a test event to
+ ConfBridge that reports transitions between states in ConfBridge.
+ This is used by tests in the Asterisk Test Suite that verify
+ state changes based on the entering/leaving of conference
+ participants. ........ Merged revisions 376414 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376415 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-16 00:15 +0000 [r376341-376345] David M. Lee <dlee@digium.com>
+
+ * utils/extconf.c, /: Fixed extconf.c breakage introduced in
+ r376306. To quote wdoekes: > Note that I'm not confirming
+ legitimacy of having that file in tree at > all. Is anyone using
+ aelparse/conf2ael? ........ Merged revisions 376340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376342 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376343 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /: Somehow I put in svn-1.6 merge information. Oops.
+
+ * utils/hashtest.c (removed), tests/test_hashtab_thrash.c (added),
+ utils/hashtest2.c (removed), include/asterisk/hashtab.h,
+ utils/Makefile, tests/test_astobj2_thrash.c (added),
+ utils/utils.xml, /: Migrate hashtest/hashtest2 to be unit tests.
+ Both hashtest and hashtest2 are manual testing apps that thrash
+ hash tables (hashtab and ao2 containers, respectively), by
+ spinning up several threads that randomly insert, delete, lookup
+ and iterate over the hash table. If the app doesn't crash, the
+ hash table probably passes the test. Those utils are not a part
+ of the typical Asterisk build, so they do not usually get
+ compiled. This all makes them less that useful. This patch
+ removes those manual test programs and replaces them with
+ Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+ also attempts to make the tests more deterministic. * Rather than
+ spinning up some number of threads that operate on the hash table
+ randomly, spin up four threads that concurrenly add, remove,
+ lookup and iterate over the hash table. * Each thread checks the
+ state of the hash table both during and after execution, and
+ indicates a test failure if things are not as expected. * Each
+ thread times out after 60 seconds to prevent deadlocking the unit
+ test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+ revisions 376306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376315 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376339 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-15 23:10 +0000 [r376312] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Fix channels lingering when
+ hung up under certain conditions Channels would get stuck and
+ MeetMe would repeatedly display an Unable to write frame to
+ channel error in the conf_run function if hung up during certain
+ sound prompts such as during user count announcements. This patch
+ fixes that by reintroducing a hangup check in the meetme's main
+ loop (also in conf_run). (closes issue ASTERISK-20486) Reported
+ by: Michael Cargile Review:
+ https://reviewboard.asterisk.org/r/2187/ Patches:
+ meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 376307 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376308 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376310 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-15 14:35 +0000 [r376291] Brent Eagles <beagles@digium.com>
+
+ * main/channel.c, /: Patch to prevent stopping the active generator
+ when it is not the silence generator. This patch introduces an
+ internal helper function to safely check whether the current
+ generator is the one that is expected before deactivating it. The
+ current externally accessible ast_channel_stop_generator()
+ function has been modified to be implemented in terms of the new
+ function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
+ ........ Merged revisions 376217 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-15 02:29 +0000 [r376282] Rusty Newton <rnewton@digium.com>
+
+ * apps/app_voicemail.c, /: Patch to play correct sound file when a
+ voicemail's urgent status is removed We were attempting to play
+ "vm-urgent-removed", which didn't exist. Now we play
+ "vm-marked-nonurgent" which exists and is the correct sound file.
+ Previous behavior was silence and a warning on the CLI. (issue
+ ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
+ Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
+ uploaded by Rusty Newton (license 5829) ........ Merged revisions
+ 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376263 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376264 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-14 19:55 +0000 [r376235] Richard Mudgett <rmudgett@digium.com>
+
+ * pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
+ Future dated call files are ignored when astspooldir is relative
+ to the current directory. The queue_file() assumed that the qdir
+ needed to be prepended if the given filename did not start with a
+ '/'. If astspooldir is relative it is not going to start from the
+ root directory obviously so it will not start with a '/'. The
+ filename used in queue_file() ultimately results in qdir
+ prepended multiple times. * Made queue_file() not prepend qdir if
+ the filename contains a '/'. (closes issue ASTERISK-20593)
+ Reported by: James Le Cuirot Patches:
+ 0004-Fix-future-call-files-from-relative-directories.patch
+ (license #6439) patch uploaded by James Le Cuirot ........ Merged
+ revisions 376232 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376233 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376234 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-13 19:42 +0000 [r376219] Jonathan Rose <jrose@digium.com>
+
+ * CHANGES, channels/chan_sip.c: chan_sip: Add SubscribeContext
+ field to SIPshowpeer AMI response The new field is will show up
+ within the response if the requested peer has a subscribe context
+ set. (closes issue ASTERISK-20626) Reported by: Jaco Kroon
+ Patches: asterisk-sip-ami-SubscrContext.patch uploaded by jkroon
+ (license 5671) -with modifications by jrose to conform to style
+ guidelines Review: https://reviewboard.asterisk.org/r/2195/
+
+2012-11-12 20:46 +0000 [r376169] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Properly check if the "Context" and "Extension"
+ headers are empty in a ShowDialPlan action. The code which
+ handles the ShowDialPlan action wrongly assumed that a non-NULL
+ return value from the function which retrieves headers from an
+ action indicates that the header has a value. This is incorrect
+ and the contents must be checked to see if they are blank.
+ (closes issue ASTERISK-20628) Reported by: jkroon Patches:
+ asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
+ ........ Merged revisions 376166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376167 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376168 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-12 20:18 +0000 [r376148] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
+ Problem When adding a dynamic hint, if an extension contains an
+ underscore no variable subsitution is being performed. This patch
+ changes from checking if the extension contains an underscore to
+ checking if the extension begins with an underscore. (closes
+ issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
+ Steven T. Wheeler, Michael L. Young Patches:
+ asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2188/ ........ Merged
+ revisions 376142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376143 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376144 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-11 17:15 +0000 [r376131] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, /, channels/chan_sip.c,
+ configs/sip.conf.sample: Remove a fixed size limitation for
+ producing SDP and change how ICE support is disabled by default.
+ With ICE support enabled in chan_sip and a large number of
+ interfaces on the system it was possible for the produced SDP to
+ be truncated due to some fixed size buffers. These buffers have
+ now been changed so they will dynamically grow as needed. ICE
+ support is now also enabled by default in res_rtp_asterisk to
+ provide a smoother experience for chan_motif users where it is
+ required. To maintain the previous behavior in chan_sip it is no
+ longer enabled by default there. (closes issue ASTERISK-20643)
+ Reported by: coopvr ........ Merged revisions 376130 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-08 22:10 +0000 [r376092] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
+ Turns out the "helpful" setting of ms and res in this macro is
+ completely useless after the timeout antipattern fix. If you're a
+ new guy looking to write code, don't write a macro like this one.
+ ........ Merged revisions 376087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376088 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376089 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-08 21:12 +0000 [r376049-376061] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, /: chan_dahdi/SS7: Made reject incoming call
+ for an in-alarm or blocked channel. If a SS7 call comes in
+ requesting a CIC that is in-alarm, the call is accepted and
+ connects if the extension exists in the dialplan. The call does
+ not have any audio. * Made release the call immediately with
+ circuit congestion cause. (closes issue ASTERISK-20204) Reported
+ by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376059 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376060 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/asterisk.c, include/asterisk/utils.h,
+ include/asterisk/astmm.h, /, main/utils.c, main/astmm.c: Add
+ MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc().
+ It will return a memory block filled with 0x55. A nonzero value.
+ * Makes free() fill the released memory block and boundary
+ fence's with 0xdeaddead. Any pointer use after free is going to
+ have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is
+ usually an invalid memory address so a crash is expected. * Puts
+ the freed memory block into a circular array so it is not reused
+ immediately. * When the circular array rotates out a memory block
+ to the heap it checks that the memory has not been altered from
+ 0xdeaddead. * Made the astmm_log message wording better. * Made
+ crash if the DO_CRASH menuselect option is enabled and something
+ is found. * Fixed a potential alignment issue on 64 bit systems.
+ struct ast_region.data[] should now be aligned correctly for all
+ platforms. * Extracted region_check_fences() from
+ __ast_free_region() and handle_memory_show(). * Updated
+ handle_memory_show() CLI usage help. Review:
+ https://reviewboard.asterisk.org/r/2182/ ........ Merged
+ revisions 376029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376030 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376048 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-07 19:15 +0000 [r376015] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_jack.c, include/asterisk/time.h, apps/app_dial.c,
+ main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c,
+ res/res_fax.c, apps/app_record.c, channels/chan_agent.c,
+ main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
+ channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
+ channels/chan_dahdi.c, apps/app_waitforring.c,
+ channels/sig_analog.c: Multiple revisions 375993-375994 ........
+ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov
+ 2012) | 30 lines Fix misuses of timeouts throughout the code.
+ Prior to this change, a common method for determining if a
+ timeout was reached was to call a function such as
+ ast_waitfor_n() and inspect the out parameter that told how many
+ milliseconds were left, then use that as the input to
+ ast_waitfor_n() on the next go-around. The problem with this is
+ that in some cases, submillisecond timeouts can occur, resulting
+ in the out parameter not decreasing any. When this happens
+ thousands of times, the result is that the timeout takes much
+ longer than intended to be reached. As an example, I had a
+ situation where a 3 second timeout took multiple days to finally
+ end since most wakeups from ast_waitfor_n() were under a
+ millisecond. This patch seeks to fix this pattern throughout the
+ code. Now we log the time when an operation began and find the
+ difference in wall clock time between now and when the event
+ started. This means that sub-millisecond timeouts now cannot play
+ havoc when trying to determine if something has timed out. Part
+ of this fix also includes changing the function ast_waitfor() so
+ that it is possible for it to return less than zero when a
+ negative timeout is given to it. This makes it actually possible
+ to detect errors in ast_waitfor() when there is no timeout.
+ (closes issue ASTERISK-20414) reported by David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
+ mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
+ lines Remove some debugging that accidentally made it in the last
+ commit. ........ Merged revisions 375993-375994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375995 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 376014 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-06 19:05 +0000 [r375967] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c, include/asterisk/channel.h,
+ include/asterisk/features.h, main/channel.c, /,
+ main/channel_internal_api.c: Fix stuck DTMF when bridge is
+ broken. When a bridge is broken by an AMI Redirect action or the
+ ChannelRedirect application, an in progress DTMF digit could be
+ stuck sending forever. * Made simulate a DTMF end event when a
+ bridge is broken and a DTMF digit was in progress. (closes issue
+ ASTERISK-20492) Reported by: Jeremiah Gowdy Patches:
+ bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by
+ Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch
+ jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2169/ ........ Merged
+ revisions 375964 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375965 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375966 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-06 12:15 +0000 [r375926] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Fix a bug where our Motif ICE
+ candidates were not quite proper, and make us more forgiving. An
+ issue was reported on the mailing list where calling would result
+ in an "Incomplete ICE-UDP candidate received on session" error
+ message. This is the result of the ICE-UDP candidate code not
+ placing a "network" attribute within the candidates. This is now
+ done. To increase compatibility though I have removed the
+ requirement for the "network" attribute to exist within ICE-UDP
+ candidates that are received since we don't actually require the
+ value. Reported on the mailing list by Jean-Denis Girard.
+ ........ Merged revisions 375925 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-05 23:10 +0000 [r375896] Matthew Jordan <mjordan@digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+ include/asterisk/timing.h, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c: Refactor ast_timer_ack to return an
+ error and handle the error in timer users Currently, if an
+ acknowledgement of a timer fails Asterisk will not realize that a
+ serious error occurred and will continue attempting to use the
+ timer's file descriptor. This can lead to situations where errors
+ stream to the CLI/log file. This consumes significant resources,
+ masks the actual problem that occurred (whatever caused the timer
+ to fail in the first place), and can leave channels in odd
+ states. This patch propagates the errors in the timing resource
+ modules up through the timer core, and makes users of these
+ timers handle acknowledgement failures. It also adds some
+ defensive coding around the use of timers to prevent using bad
+ file descriptors in off nominal code paths. Note that the patch
+ created by the issue reporter was modified slightly for this
+ commit and backported to 1.8, as it was originally written for
+ Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
+ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
+ jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
+ 6358) ........ Merged revisions 375893 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375894 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375895 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-05 21:42 +0000 [r375865] Richard Mudgett <rmudgett@digium.com>
+
+ * main/loader.c, /: Add safety NULL pointer check in module user
+ references. Made __ast_module_user_remove() check for NULL
+ pointers. ........ Merged revision 375860 from C.3 ........
+ Merged revisions 375862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375864 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-05 18:00 +0000 [r375848] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE.txt: chan_sip: Document a change to user-field
+ encoding introduced with r303509 The change in question was added
+ to improve compliance with RFC3261, but at the time of commit, it
+ wasn't adequately documented in the UPGRADE notes. (closes issue
+ ASTERISK-20561) Reported by: Deniz Review:
+ https://reviewboard.asterisk.org/r/2177/ ........ Merged
+ revisions 375846 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375847 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-04 03:10 +0000 [r375730-375803] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c, /: Don't attempt to purge sessions when no
+ sessions exist Manager's tcp/tls objects have a periodic function
+ that purge old manager sessions periodically. During shutdown,
+ the underlying container holding those sessions can be disposed
+ of and set to NULL before the tcp/tls periodic function is
+ stopped. If the periodic function fires, it will attempt to
+ iterate over a NULL container. This patch checks for whether or
+ not the sessions container exists before attempting to purge
+ sessions out of it. If the sessions container is NULL, we simply
+ return. Note that this error was also caught by the Asterisk Test
+ Suite. ........ Merged revisions 375800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375801 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375802 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_fax.c: Only deref a reserved gateway session if we
+ actually reserved one Its perfectly acceptable to have a gateway
+ session unreserved when we go to first allocate one. Unreffing
+ the reserved gateway session - when its NULL - will result in an
+ assertion error. This problem was caught by the Asterisk Test
+ Suite (once we had enough of the debugging flags enabled)
+ ........ Merged revisions 375797 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375798 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /: Properly clean up manager resources on exit
+ This patch does two things: 1) It properly unregisters the
+ manager CLI commands 2) It cleans up AMI users on exit. Prior to
+ this patch, the AMI users were not being disposed of properly,
+ resulting in a memory leak. (closes issue ASTERISK-20646)
+ Reported by: Corey Farrell patches: manager_shutdown.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 375793 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375794 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375795 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/db.c: Properly finalize prepared SQLite3 statements to
+ prevent memory leak The AstDB uses prepared SQLite3 statements to
+ retrieve data from the SQLite3 database. These statements should
+ be finalized during Asterisk shutdown so that the SQLite3
+ database can be properly closed. Failure to finalize the
+ statements results in a memory leak and a failure when closing
+ the database. This patch fixes those issues by ensuring that all
+ prepared statements are properly finalized at shutdown. (closes
+ issue ASTERISK-20647) Reported by: Corey Farrell patches:
+ astdb-sqlite3_close.patch uploaded by Corey Farrell (license
+ 5909) ........ Merged revisions 375761 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375763 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/xmldoc.c: Fix memory leaks in XML documentation This
+ patch fixes two memory leaks: 1) When building XML documentation
+ items, the 'name' attribute was extracted from XML elements but
+ not properly freed after being copied into the item being built.
+ 2) When unloading XML documentation, the doctree container
+ objects were not properly freed. This patch corrects these memory
+ leaks. Note that this patch was modified slightly for this
+ commmit, as the case where the 'name' attribute doesn't exist
+ also wasn't handled in the item construction. This patch also
+ checks for that attribute not existing. (closes issue
+ ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan
+ patches: xmldoc-memory_leak.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 375756 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cdr.c, /: Prevent multiple CDR batches from conflicting when
+ scheduling the CDR write The Asterisk Test Suite caught an error
+ condition where a scheduled CDR batch write can be deleted twice
+ if two channels attempt to post their CDRs at the same time. The
+ batch CDR mutex is locked while the CDRs are appended to the
+ current batch list; however, it is unlocked prior to actually
+ scheduling the CDR write. As such, two threads can attempt to
+ remove the currently scheduled batch write at the same time,
+ resulting in an assertion error. This patch extends the time that
+ the mutex is locked to encompass actually scheduling the write.
+ This prevents two threads from unscheduling the currently
+ scheduled write at the same time. ........ Merged revisions
+ 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375728 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375729 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-02 21:03 +0000 [r375663] Damien Wedhorn <voip@facts.com.au>
+
+ * /, channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports
+ open Skinny wasn't closing RTP sockets. This patch includes
+ ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes
+ the problem. Also add destroy for VRTP (which I believe is
+ unused, but exists). Review:
+ https://reviewboard.asterisk.org/r/2176/ ........ Merged
+ revisions 375660 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-02 21:01 +0000 [r375628-375662] Richard Mudgett <rmudgett@digium.com>
+
+ * main/format_pref.c, main/channel.c, channels/chan_misdn.c, /,
+ main/ccss.c: Things don't need to be that const. ........ Merged
+ revisions 375658 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375659 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375661 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Multiple
+ revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
+ 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
+ primitives must be handled first. The frm->addr is a different
+ "address space" than the stack/instance address of other Lx
+ primitives. The test for B channel instance address could fail.
+ Patches: patch01_timers.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
+ 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
+ chan_misdn: Free memory in error paths and other memory leaks.
+ The one line commented with BUG is not easily fixable because
+ there is no de-init function one can call. Patches:
+ patch02_memory.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
+ 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
+ L2 de-establish/establish * An NT-PTMP cannot de/establish L2
+ since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
+ is finally active in handle_l1. * L2 deactivation logging
+ cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
+ as "UNKN". * Removed unused functions and code for L2 handling.
+ Patches: patch03_L2estab.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
+ rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
+ lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
+ prim via lower_id layer (3 or 1) simply does not work. For TE (3)
+ it returns an error (len=-6) which is not evaluated by
+ handle_l1(), so the L1 layer status ends up wrong. Instead PH
+ must be sent via L4, only then does it reach L1 without an error
+ message. And NT PH prims only reach L1 when they are sent to
+ layer 2 id. --> use upper_id to send PH primitives. * Check for
+ errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
+ improved. * The lower_id is now not used for anything, except:
+ Why is lower_id layer deleted when it wasn't created? I removed
+ this code since it looks very wrong. Patches:
+ patch04_l1activation.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
+ 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
+ chan_misdn: Fix loss of B channels if L1 is down. If you make 2
+ calls out an NT PTMP port which is not connected to any phone,
+ the B channel associated with that call becomes unusable until
+ Asterisk is restarted. The problem is the EVENT_SETUP is queued
+ when L1 is not up in misdn_lib_send_event(). If L1 cannot be
+ activated the event won't be dequeued. It gets even worse when
+ the call is hung up. The queued EVENT_SETUP will be overwritten
+ by an EVENT_DISCONNECT. The reserved B channel then will never be
+ freed. If later someone connects a phone to the port, L1 will
+ eventually activate and the queued EVENT_DISCONNECT is sent down
+ the stack. However, it is ignored because it is the wrong call
+ state. The real fix would be that activation and queueing for a
+ new SETUP is done by the NT stack. But since it doesn't, the
+ workaround must be removed because it doesn't always work. Fix:
+ The event is no longer queued but immediately sent to the stack.
+ If L1 cannot be activated, the L3 state machine that was started
+ by the EVENT_SETUP will do its work, i.e. a timeout will release
+ the B channel properly. The SETUP possibly cannot be sent the
+ first time but is resent by T303 in case L1 could be activated.
+ Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
+ by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
+ rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
+ lines chan_misdn: Remove some calls to exit(). Try proper cleanup
+ when something goes wrong in misdn_lib_init(). Especially do not
+ call exit()! * Fix memory leak because stack_destroy() does not
+ free the stack struct. Patches: patch06_cleanup-init.diff
+ (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
+ ABE-2888 ........ Merged revisions 375519-375524 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 375625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375626 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375627 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-02 17:27 +0000 [r375614] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
+ Origin Processing While looking at some debug logs, I noticed
+ that it was being reported that the SDP origin line was
+ unsupported or failed. Upon looking into this on my local
+ machine, I found that I too was getting this debug message yet
+ everything seemed to be getting processed properly. What was
+ discovered is, that, the variable to determine what is displayed
+ in the debug message for the SDP line that was processed, was not
+ being set for the origin line when the result was successful.
+ This patch fixes this and was tested on local machine. ........
+ Merged revisions 375594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375601 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375613 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-11-01 15:03 +0000 [r375576] Jonathan Rose <jrose@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: chan_sip: Fix a
+ bug causing SIP reloads to remove all entries from the registry A
+ regression was introduced in chan_sip by changes to sip reload
+ introduced by r349097. That patch moved peer purging from the
+ beginning of the reload to after the general configuration was
+ finished. This patch fixes that by undoing the repositioning of
+ the original peer purging code and using a similar function after
+ performing general configuration that purges only autocreated
+ peers that were created when persist mode isn't enabled. (closes
+ issue ASTERISK-20611) Reported by: Alisher Review:
+ https://reviewboard.asterisk.org/r/2171/ ........ Merged
+ revisions 375575 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-31 18:01 +0000 [r375560] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_http_websocket.exports.in: Fix an issue with
+ res_http_websocket where the chan_sip WebSocket handler could not
+ be registered. On some systems the optional API support uses the
+ GCC compiler attribute "weakref" to provide its functionality.
+ This code changes the function names and prefixes "__" to the
+ front. The res_http_websocket exports file did not take this into
+ account, thereby not allowing those functions to be global and
+ ultimately found. (closes issue ASTERISK-20631) Reported by:
+ danjenkins ........ Merged revisions 375559 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-31 14:58 +0000 [r375533] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_calendar_ews.c: Properly extract the Body information
+ of an EWS calendar item Unlike all other calendar modules,
+ res_calendar_ews fails to extract the Body information for a
+ calendar item. This is due, in part, to a quirk in the schema in
+ the XML - not only does a CalendarItem contain a Body element,
+ but the CalendarItem exists as a descendant of a different Body
+ element. The neon parser was erroneously skipping all Body
+ elements. This patch fixes that by bypassing Body elements that
+ are not a child of CalendarItem, and parsing the Body element out
+ if it is a child. Note that the original patch by Terry Wilson
+ only needed slight modifications to make it properly pull the
+ Body information out; as such, while I've linked to the patch
+ that I uploaded for Dmitry, I've attributed the patch to Terry.
+ (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
+ by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
+ uploaded by Terry Wilson (license 6283) ........ Merged revisions
+ 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375531 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375532 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-30 19:31 +0000 [r375511] Richard Mudgett <rmudgett@digium.com>
+
+ * /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing
+ module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
+ Patches: smfix.patch (license #6099) patch uploaded by feyfre
+ Modified for coding guidelines. ........ Merged revisions 375496
+ from http://svn.asterisk.org/svn/asterisk/branches/10 ........
+ Merged revisions 375506 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-30 19:20 +0000 [r375472-375498] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, /: mixmonitor: Add a test event This test
+ event is being used to fix the mixmonitor_audiohook_inherit test.
+ ........ Merged revisions 375484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375485 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375486 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_confbridge.c: confbridge: Fix a bug which made
+ conferences not record with AMI/CLI commands When confbridge was
+ changed to handle conference status with a state machine in
+ r374658. The function responsible for starting recording for a
+ conference was refactored with the function actually responsible
+ for launching the recording thread being split into a function
+ with another name. The old function name was still used for
+ manually started recordings through AMI or CLI. This patch fixes
+ that by switching which function is used to start recording the
+ conference. (closes issue ASTERISK-20601) Reported by: Vilius
+ Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 375470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375471 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-29 21:38 +0000 [r375442-375443] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Prevent resetting of NATted realtime peer
+ address on reload. If a "sip reload" is issued for a SIP peer,
+ then his IP address will be cleared, thus resulting in forgetting
+ the public IP address. Asterisk will then attempt to route SIP
+ traffic to the private IP address. The fix here is to make "sip
+ reload" ignore realtime peers when "host = dynamic" is spotted.
+ Realtime peers can now only have their IP address reset if they
+ have gone from being not dynamic to being dynamic. (closes issue
+ ASTERISK-18203) reported by daren ferreira (closes issue
+ ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
+ uploaded by JoshE (license #6075) ........ Merged revisions
+ 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375417 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375437 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * funcs/func_strings.c, UPGRADE.txt, channels/chan_mgcp.c,
+ main/pbx.c, apps/app_osplookup.c, channels/chan_sip.c,
+ channels/chan_skinny.c: Make evaluation of channel variables
+ consistently case-sensitive. Due to inconsistencies in how
+ variable names were evaluated, the decision was made to make all
+ evaluations case-sensitive. See the UPGRADE.txt file or
+ https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for
+ more details. (closes issue ASTERISK-20163) reported by Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2160
+
+2012-10-29 21:02 +0000 [r375416] Matthew Jordan <mjordan@digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c: Ensure that CDRs for a caller in a
+ Queue that is not answered is NO ANSWER. When a caller enters a
+ queue and no queue member answers the call, the current behaviour
+ can be a little odd depending on the paused status of the queue
+ members. If any queue member is paused, but not all, the CDR
+ disposition will be BUSY. If all queue members are paused, then
+ the CDR disposition is based instead on the disposition of the
+ call prior to entering the Queue. This patch modifies the
+ behaviour in the following ways: * If no queue members are
+ paused, the CDR disposition is whatever the disposition was prior
+ to going into Queue. If the call was answered this will be
+ ANSWERED; otherwise, it is NO ANSWER. * If some queue members are
+ pused, the CDR result is NO ANSWER. (This is a change in
+ behaviour, as the result would previously have been BUSY) * If
+ all queue members are paused, the CDR result is whatever the
+ result was prior to going into Queue. This is the same as the
+ behaviour prior to this patch. * If the caller hangs up, times
+ out, or presses '*' with the 'h' option, the CDR disposition is
+ again not set and is dependent on whether or not the caller was
+ Answered prior to entering Queue. This patch was based on one
+ provided by Thomas Arimont, but has been modified to accomodate
+ findings by the reviewers. Review:
+ https://reviewboard.asterisk.org/r/2064/ (closes issue AST-906)
+ Reported by: Thomas Arimont (closes issue ASTERISK-17776)
+ Reported by: Attila Megyeri
+
+2012-10-29 19:31 +0000 [r375364-375391] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix the Park 'r' option when a channel parks
+ itself. When a channel uses the Park appliation to park itself
+ with the 'r' option, the channel hears music-on-hold instead of
+ the requested ringing. * Added a missing check for the 'r' option
+ when a channel parks itself. (closes issue ASTERISK-19382)
+ Reported by: James Stocks Patches by: dsessions Review:
+ https://reviewboard.asterisk.org/r/2148/ ........ Merged
+ revisions 375388 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375389 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375390 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
+ a NULL tech_pvt. The tech support customer was using the AMI
+ Redirect action shortly after a call was placed. While the
+ channel tried to do an ast_read(), the masquerade resulting from
+ the channel redirect took place. The masquerade in the middle of
+ the ast_read() resulted in the segfault. (closes issue AST-1025)
+ Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
+ (license #5621) patch uploaded by rmudgett ........ Merged
+ revisions 375361 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375362 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375363 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-23 16:22 +0000 [r375291-375328] Jonathan Rose <jrose@digium.com>
+
+ * contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better
+ response for various exit conditions to openssl (closes issue
+ ASTERISK-20260) Reported by: Daniel O'Connor Patches:
+ ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
+ 6419) ........ Merged revisions 375325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375326 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375327 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/app.c: core: Fix a memory leak in app.c from an early
+ return ast_app_group_match_get_count allocates memory with the
+ regcomp function and we previously forgot to free it when bailing
+ out due to a regex compilation failure against category. (closes
+ issue AST-1018) Reported by: Guenther Kelleter Patches:
+ regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
+ ........ Merged revisions 375299 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375301 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * codecs/gsm/src/code.c, /: GSM: Fix encoding problems with GSM
+ (closes issue ASTERISK-20457) Reported by: Richard Miller
+ Patches: code.patch uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 375272 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375273 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375288 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-18 21:49 +0000 [r375240-375249] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE
+ notes describing behavioral changes to rrmemory strategy caused
+ by 375216 (issue AST-989) Reported by: Thomas Arimont
+
+ * /, apps/app_queue.c: app_queue: Make ordering of
+ rrmemory/rrordered persist over add/remove members Prior to this
+ patch, adding, removing or reloading members to rrmemory would
+ cause the order to become completely jumbled. Now it behaves more
+ or less like rrordered other than the fact that it stores the
+ members on a hash table rather than a linked list. This patch
+ also prevents removal of members and member reloads from jumbling
+ rrordered queues. (issue AST-989) Reported by: Thomas Arimont
+ Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
+ revisions 375216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375217 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375219 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-18 20:31 +0000 [r375215] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_alarmreceiver.c: Fix XML Document Validation Failure Fix
+ documentation error when validating the xml in trunk caused by
+ r375150. Moved the description end tag down to below the
+ variablelist element end tag. Found when compiling with
+ --dev-mode-enabled. (issue ASTERISK-20289)
+
+2012-10-18 20:13 +0000 [r375192] Richard Mudgett <rmudgett@digium.com>
+
+ * Makefile, /, build_tools/make_version, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ build_tools: Allow Asterisk to report git SHAs in version string.
+ Make git more attractive for managing work-in-progress.
+ Especially convenient when a potential patch set needs to be
+ tested on multiple platforms since one can use git to keep all
+ the test environments in sync independent of a subversion server.
+ Now the Asterisk version will show the exact git SHA5 that was
+ used when building (still appended by "M" if there are local
+ modifications) from a git clone of the Asterisk repository so the
+ developer can more easily know what is actually under test. You
+ will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of
+ this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported
+ This has zero impact for those not using git with the exception
+ of an extra test in the configure script to gather git's path.
+ This is necessary to prevent "sudo make install" from failing
+ since git may not be in the path in make's shell environment.
+ (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches:
+ 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 375189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375190 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375191 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-18 14:17 +0000 [r375182] Andrew Latham <lathama@gmail.com>
+
+ * main/manager.c, include/asterisk/compat.h, main/features.c,
+ include/asterisk/module.h, main/logger.c, main/http.c,
+ main/app.c, include/asterisk/doxygen/commits.h,
+ include/asterisk/udptl.h, include/asterisk.h, main/dnsmgr.c,
+ contrib/asterisk-ng-doxygen, codecs/log2comp.h, main/cel.c,
+ main/named_acl.c, Makefile, include/asterisk/paths.h,
+ include/asterisk/doxygen/asterisk-git-howto.h,
+ include/asterisk/doxygen/reviewboard.h,
+ include/asterisk/doxygen/licensing.h, pbx/pbx_dundi.c,
+ include/asterisk/smdi.h, main/asterisk.c, main/dsp.c,
+ main/udptl.c, include/asterisk/doxygen/architecture.h,
+ main/ccss.c, Makefile.moddir_rules, Makefile.rules, main/enum.c,
+ main/cli.c, main/cdr.c, include/asterisk/doxygen/releases.h,
+ include/asterisk/doxyref.h: Doxygen Updates - Title update Update
+ and extend the configuration_file group and enable linking.
+ Commit other cleanups from multi-version Doxygen testing. Update
+ title that was left behind many years ago. (issue ASTERISK-20259)
+
+2012-10-17 20:34 +0000 [r375175] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c: manager: remove curses dependent stuff from
+ r375103 Upon further examination, this code was causing
+ compliation problems on CentOS at the least (possibly on any
+ machine without curses) and also the local value of COLS is used
+ even with a remote console, so it is less than ideal. (issue
+ ASTERISK-20396) Reported by: Johan Wilfer
+
+2012-10-17 19:02 +0000 [r375150] Pedro Kiefer <pedro@kiefer.com.br>
+
+ * apps/app_alarmreceiver.c, configs/alarmreceiver.conf.sample: Adds
+ new formats to app_alarmreceiver, ALAW calls support and enhanced
+ protection. Commiting this on behalf of Kaloyan Kovachev (license
+ 5506). AlarmReceiver now supports the following DTMF signaling
+ types: - ContactId - 4x1 - 4x2 - High Speed - Super Fast We are
+ also auto-detecting which signaling is being received. So support
+ for those protocols should work out-the-box. Correctly identify
+ ALAW / ULAW calls. Some enhanced protection for broken panels and
+ malicious callers where added. (closes issue ASTERISK-20289)
+ Reported by: Kaloyan Kovachev Review:
+ https://reviewboard.asterisk.org/r/2088/
+
+2012-10-17 19:01 +0000 [r375149] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/tcptls.c: Ensure Asterisk fails TCP/TLS SIP calls when
+ certificate checking fails When placing a call to a TCP/TLS SIP
+ endpoint whose certificate is not signed by a configured CA
+ certificate, Asterisk would issue a warning and continue to
+ process the call as if there was not an issue with the
+ certificate. Asterisk now properly fails the call if the
+ certificate fails verification or if the certificate does not
+ exist when certificate checking is enabled (the default
+ behavior). (closes issue ASTERISK-20559) Reported by: kmoore
+ Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged
+ revisions 375146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375147 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375148 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-17 14:24 +0000 [r375110-375137] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * cdr/cdr_odbc.c, res/res_rtp_asterisk.c, main/pbx.c,
+ channels/chan_sip.c: Change a few warnings to debug and the
+ inverse. Remove the "RTP Read too short" warning for RTP
+ keepalives. Remove the the warning about the application
+ delimiter switch from pipe to comma. (You should've done this by
+ now.) Make cdr_odbc report more when an insert fails. Make
+ chan_sip warn less when the peer wants SRTP (and we don't) or
+ sends a zero port to disable a media type. Review:
+ https://reviewboard.asterisk.org/r/2167 (closes issue
+ ASTERISK-20538)
+
+ * /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
+ Don't crash on large user input. Allow SIP headers without space.
+ Optimize code a bit. Review:
+ https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
+ 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375112 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375113 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_sip.c: Don't do SIP contact/route DNS if we're not
+ using the result. In many cases (for peers behind NAT or for TCP
+ sockets) we do not need to look up any hostname in the Contact
+ (or Route) when sending an in-dialog request. This should reduce
+ netsock2.c: getaddrinfo errors in certain scenarios. Review:
+ https://reviewboard.asterisk.org/r/2156
+
+2012-10-16 20:45 +0000 [r375103] Jonathan Rose <jrose@digium.com>
+
+ * CHANGES, main/manager.c: manager: Change display of 'manager show
+ commands' and 'manager show command' manager show commands now
+ shows the full name of the command being displayed regardless of
+ size. The privilege column has also been removed from this
+ display. It will also now use the full length of the terminal if
+ curses is available. Manager show command will now always display
+ the privilege of the manager command within the CLI. (closes
+ ASTERISK-20396) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/2143/
+
+2012-10-16 19:26 +0000 [r375081] Pedro Kiefer <pedro@kiefer.com.br>
+
+ * apps/app_alarmreceiver.c: Fixes two small regressions from
+ ASTERISK-20157 - receive_dtmf_digits had the wrong buffer length
+ - app_alarmreceiver should wait 100ms before sending the second
+ part of handshake (closes issue ASTERISK-20484) Reported by:
+ Jean-Philippe Lord Tested by: Jean-Philippe Lord, Pedro Kiefer
+ Patches: ASTERISK-20484_v2.diff uploaded by Kaloyan Kovachev
+ (license 5506)
+
+2012-10-16 19:25 +0000 [r375080] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Update sip_request_call SIP dial string
+ documentation. This was missed when merging review r1859.
+ ........ Merged revisions 375074 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375078 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375079 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-16 14:09 +0000 [r375052] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_iax2.c: Remove a log message that was left in
+ accidentally from call-id logging development. ........ Merged
+ revisions 375051 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-15 21:25 +0000 [r375044] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/strings.h, channels/chan_iax2.c,
+ apps/app_dial.c, /, main/ccss.c: Fix some potential misuses of
+ ast_str in the code. Passing an ast_str pointer by value that
+ then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or
+ ast_str_append_va() can result in the pointer originally passed
+ by value being invalidated if the ast_str had to be reallocated.
+ This fixes places in the code that do this. Only the example in
+ ccss.c could result in pointer invalidation though since the
+ other cases use a stack-allocated ast_str and cannot be
+ reallocated. I've also updated the doxygen in strings.h to
+ include notes about potential misuse of the functions mentioned
+ previously. Review: https://reviewboard.asterisk.org/r/2161
+ ........ Merged revisions 375025 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375026 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 375027 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-15 08:26 +0000 [r375017] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix underscreen buttons warnings
+ apeared while transfer process ........ Merged revisions 375016
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-14 21:59 +0000 [r375003-375009] Andrew Latham <lathama@gmail.com>
+
+ * addons/app_mysql.c, addons/chan_mobile.c: Doxygen Updates Update
+ and extend the configuration_file group and enable linking.
+ (issue ASTERISK-20259)
+
+ * utils/muted.c, utils/extconf.c: Doxygen Updates Update and extend
+ the configuration_file group and enable linking. (issue
+ ASTERISK-20259)
+
+ * formats/Makefile, sounds/Makefile, funcs/Makefile,
+ bridges/Makefile, agi/Makefile, codecs/Makefile, utils/Makefile,
+ tests/Makefile, cel/Makefile, main/Makefile, addons/Makefile,
+ pbx/Makefile: Title update Update title that was left behind many
+ years ago. Used revision 6596 as my guide for what it should be.
+ (issue ASTERISK-20259)
+
+ * channels/chan_phone.c, channels/chan_dahdi.c,
+ channels/iax2-parser.h, channels/chan_misdn.c,
+ channels/chan_skinny.c, channels/chan_motif.c,
+ channels/chan_h323.c, channels/iax2.h, channels/chan_alsa.c,
+ channels/chan_mgcp.c, channels/chan_vpb.cc, channels/chan_sip.c,
+ channels/chan_gtalk.c, channels/chan_console.c,
+ channels/Makefile, channels/chan_iax2.c, channels/chan_oss.c,
+ channels/chan_jingle.c: Doxygen Updates - Title update Update and
+ extend the configuration_file group and enable linking. Update
+ title that was left behind many years ago. (issue ASTERISK-20259)
+
+ * cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c,
+ cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_syslog.c, cdr/Makefile,
+ cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Doxygen Updates - Title
+ update Update and extend the configuration_file group and enable
+ linking. Update title that was left behind many years ago. (issue
+ ASTERISK-20259)
+
+ * apps/app_festival.c, apps/app_fax.c, apps/app_skel.c,
+ apps/app_alarmreceiver.c, apps/app_amd.c, apps/app_confbridge.c,
+ apps/app_followme.c, apps/app_queue.c, apps/app_adsiprog.c,
+ apps/app_voicemail.c, apps/Makefile, apps/app_meetme.c: Doxygen
+ Updates - Title update Update and extend the configuration_file
+ group and enable linking to the application. Update title that
+ was left behind many years ago. (issue ASTERISK-20259)
+
+ * res/res_phoneprov.c, res/Makefile, res/res_xmpp.c,
+ res/res_musiconhold.c, res/res_jabber.c, res/res_config_sqlite.c,
+ res/res_smdi.c, res/res_curl.c, res/res_config_ldap.c,
+ res/res_odbc.c, res/res_clialiases.c, res/res_calendar.c,
+ res/res_config_sqlite3.c, res/res_config_pgsql.c, res/res_snmp.c,
+ res/res_limit.c, res/res_fax.c: Doxygen Updates - Title update
+ Update and extend the configuration_file group and enable linking
+ to the resource. Update title that was left behind many years
+ ago. (issue ASTERISK-20259)
+
+2012-10-14 12:23 +0000 [r374996] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * config.sub, /, config.guess: Update config.guess and config.sub:
+ 2012-10-10 Update config.guess and config.sub to revision
+ fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
+ savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
+ 64bit). config.guess:timestamp='2012-09-25'
+ config.sub:timestamp='2012-10-10' ........ Merged revisions
+ 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374991 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374995 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-13 19:58 +0000 [r374940-374970] Andrew Latham <lathama@gmail.com>
+
+ * CREDITS: Update CREDITS Update Jean-Denis and add myself (issue
+ ASTERISK-20259)
+
+ * Makefile: Multiplatform Makefile Update Paul Belanger pointed out
+ that using sed in the Makefile is an issue with multiple
+ platforms. We are cleaning up the Doxygen config as a following
+ step so I just switched the sed inplace changes to be an echo
+ append instead. (issue ASTERISK-20259)
+
+ * apps/app_dial.c, main/app.c: Doxygen Clean ups Add app_skel.c as
+ an example in app.c and fix some formating for the "Dial Privacy
+ scripts" so it actually shows up in the Doxygen output. (issue
+ ASTERISK-20259)
+
+ * Makefile: Test for Asterisk Version info Doxygen uses the
+ ASTERISKVERSION as a sub header. If a SVN export is done and no
+ .svn or .version file exists it defualts to
+ UNKNOWN__and_probably_unsupported which is honest but not great
+ for the online docs. During the "make progdocs" I added a test
+ for this and just warned and ommitted the version. (issue
+ ASTERISK-20259)
+
+ * contrib/asterisk-ng-doxygen: Correct output directory During
+ testing I used an alternate output directory and mistakenly
+ committed it. Matt Jordan noticed and I reverted. This is the
+ correct setting for local output to match with all branches.
+ (issue ASTERISK-20259)
+
+ * static-http/ajamdemo.html, static-http/astman.css: Add
+ licens/copyright header Begin update of static-http files and
+ general clean ups. This only adds the standard header to the
+ files. (issue ASTERISK-20503)
+
+ * makeopts.in, Makefile, configure, configure.ac: Add check for
+ Doxygen The autoconf configuration system had a test for DOT but
+ not for Doxygen. I added the test for Doxygen and did an overhaul
+ of the Makefile check to a much simpler process. (issue
+ ASTERISK-20259)
+
+2012-10-12 21:58 +0000 [r374933] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_voicemail.c, /: Avoid a segfault on invalid format names
+ If a format name was not found by ast_getformatbyname, a NULL
+ pointer would be passed into ast_format_rate and immediately
+ dereferenced. This ensures that a valid pointer is used since the
+ structure is already allocated on the stack. (closes issue
+ DPH-523) Reported-by: Steve Pitts ........ Merged revisions
+ 374932 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-12 16:31 +0000 [r374924] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ Do not use a FILE handle when doing SIP TCP reads. This is used
+ to solve an issue where a poll on a file descriptor does not
+ necessarily correspond to the readiness of a FILE handle to be
+ read. This change makes it so that for TCP connections, we do a
+ recv() on the file descriptor instead. Because TCP does not
+ guarantee that an entire message or even just one single message
+ will arrive during a read, a loop has been introduced to ensure
+ that we only attempt to handle a single message at a time. The
+ tcptls_session_instance structure has also had an overflow buffer
+ added to it so that if more than one TCP message arrives in one
+ go, there is a place to throw the excess. Huge thanks goes out to
+ Walter Doekes for doing extensive review on this change and
+ finding edge cases where code could fail. (closes issue
+ ASTERISK-20212) reported by Phil Ciccone Review:
+ https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
+ 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374906 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374914 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-11 23:40 +0000 [r374879-374897] Andrew Latham <lathama@gmail.com>
+
+ * contrib/scripts/install_prereq: Append Doxygen to Debian packages
+ list Add Doxygen to the Debian install list. I will check for
+ other platforms like Red Hat (issue ASTERISK-20259)
+
+ * static-http/mantest.html: Update JQuery URL to recent version The
+ JQuery URL to version 1.4 will be removed within the life span of
+ Asterisk 11. This is a compatible upgrade by using the URL for
+ 1.8. (issue ASTERISK-20503)
+
+ * main/manager.c, include/asterisk/module.h: Continue to group
+ config files (issue ASTERISK-20259)
+
+ * CREDITS: CREDITS clean up As discussed online
+ http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html
+ the credits file needs some cleaning. This is 95% whitespace with
+ a few additions found in file headers. Further additions should
+ be added here instead of in the file being updated. (issue
+ ASTERISK-20259)
+
+ * contrib/asterisk-ng-doxygen: Revert Local testing Config Revert a
+ local testing config that I made. This was not intended to be
+ committed. Thank you Matt Jordan for noticing this. (issue
+ ASTERISK-20259)
+
+2012-10-11 21:19 +0000 [r374852-374878] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Fix a bug where audio on Google Voice
+ would not work due to ignoring candidates. Instead of ignoring
+ parts of the message that are not known just ignore the ones we
+ know may be present and that would cause a problem. ........
+ Merged revisions 374877 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_motif.c: Fix an issue where outgoing calls would
+ fail to establish audio due to ICE negotiation failures. This
+ change removes the requirement for ufrag and pwd in the transport
+ stanza and also makes us the controlling agent. (closes issue
+ ASTERISK-20554) Reported by: mmichelson ........ Merged revisions
+ 374850 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-11 15:49 +0000 [r374849] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.exports.in (removed), main/sip_api.c (added),
+ /, channels/chan_sip.c, include/asterisk/sip_api.h: Don't make
+ chan_sip export global symbols. During testing, it was discovered
+ that having chan_sip export global symbols was problematic. The
+ biggest problem was that load order was affected. Trying to use
+ realtime could be problematic since in all likelihood the
+ necessary realtime driver(s) would not be loaded before chan_sip.
+ In addition, it was found that it was impossible to use the
+ Digium Phone Module for Asterisk since it must be loaded before
+ chan_sip since it must hook into chan_sip's configuration
+ parsing. The solution is to use a virtual table in the same
+ manner that other modules in Asterisk do, like app_voicemail.
+ (closes issue ASTERISK-20545) Reported by: kmoore ........ Merged
+ revisions 374842 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-11 15:44 +0000 [r374846] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c, /: Fix incorrect billing duration reported when batch
+ mode is enabled Similar to r369351, the billing duration can be
+ skewed when batch mode is enabled. This happened much more rarely
+ than the duration, as it only occured when the call was answered
+ (thereby indicating an actual answer time) and immediately hung
+ up on (indicating a billsec of 0). Since a billing time of '0'
+ can either mean that the call immediately ended or that the CDR
+ was improperly answered, we have to use additional information to
+ know whether or not we can trust the CDR billsec value. Prior to
+ this patch, we looked to see if we had a valid answer time. If we
+ did, and billsec was zero, we used the current time to calculate
+ what billsec value we could from the CDR being written. If batch
+ mode is enabled, this will incorrectly report a billsec value
+ being much greater than the actual duration of the call. Instead
+ of relying on the presence of an answer time to know whether or
+ not we can re-calculate the billsec for the CDR, we now also use
+ the presence of the CDR's end time to know if we need to
+ re-calculate or whether we can trust the billsec value that we
+ have. This prevents erroneous jumps in the billsec value, while
+ still making sure that in the worst case, some billing time will
+ be calculated. (closes issue AST-1016) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont ........ Merged revisions
+ 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374844 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374845 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-11 13:34 +0000 [r374834] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Consider the Google Talk content stanza
+ name (jin:content) valid. ........ Merged revisions 374833 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-10 21:05 +0000 [r374805] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, /: app_queue: Made pass connected line updates
+ from the caller to ringing queue members. Party A calls Party B
+ Party B puts Party A on hold. Party B calls a queue. Ringing
+ queue member D sees Party B identification. Party B transfers
+ Party A to the queue. Queue member D does not get a connected
+ line update for Party A. Queue member D answers the call and
+ still sees Party B information. However, if Party A later
+ transfers the call to Party C then queue member D gets a
+ connected line update for Party C. * Made pass connected line
+ updates from the caller to queue members while the queue members
+ are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
+ (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
+ rmudgett ........ Merged revisions 374801 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 374802 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374803 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374804 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-10 13:40 +0000 [r374793] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/manager.c: Fix segfault regression from r370681 Due to
+ usage of ast_hook_send_action, AMI action handling code should be
+ able to handle a NULL mansession->session. This would cause a
+ crash on NULL dereference if action_originate was called from
+ ast_hook_send_action. (closes issue ASTERISK-20544) ........
+ Merged revisions 374792 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-09 22:24 +0000 [r374778] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Fix execution of 'i' extension due to
+ uninitialized variable. The fix for ASTERISK-18243 added code
+ that could potentially use dst_exten[] uninitialized. As a result
+ the 'i' exten may not be executed when it should. (closes issue
+ ASTERISK-20455) Reported by: Richard Miller Patches:
+ pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
+ Miller Made some cosmetic modifications. ........ Merged
+ revisions 374758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374763 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374771 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-09 21:35 +0000 [r374757] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Improve logging for DTLS-SRTP failure
+ situations. (closes issue ASTERISK-20487) Reported by: mjordan
+ ........ Merged revisions 374756 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-08 22:31 +0000 [r374717-374730] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample, /: dahdi.conf.sample: Add
+ description for "buffers" setting. This contains an edited
+ version of the patch originally created by John Bigelow. (closes
+ issue ASTERISK-14435) Reported by: John Bigelow Patches:
+ buffers.patch (license #5091) patch uploaded by John Bigelow
+ 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 374727 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374728 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374729 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * pbx/pbx_spool.c, /: Fix deletion of unopenable spool files. If
+ scan_service() cannot open the spool file, it logs a message
+ saying that it will delete the file and calls remove_from_queue()
+ to do it. However, remove_from_queue() fails to delete the spool
+ file because struct outgoing has not yet been fully initialized.
+ * Merged allocating a new struct outgoing and init_outgoing()
+ into new_outgoing(). Allocation is initialization. * Made
+ apply_outgoing() not initialize the spool filename in struct
+ outgoing. * Made apply_outgoing() call ast_trim_blanks() and
+ ast_skip_blanks() rather than manually inlining them. * Reduced
+ indentation levels in apply_outgoing(). * Fixed a garbled comment
+ in remove_from_queue(). * Reworked scan_service() to simplify it.
+ (closes issue ASTERISK-17231) Reported by: David Chappell
+ Patches: spool_open_failure.diff (license #4997) patch uploaded
+ by David Chappell Started with this patch. ........ Merged
+ revisions 374686 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some
+ memory leaks on off nominal paths in init_outgoing() when merging
+ into the new_outgoing() function dealing with o->capabilities.
+ ........ Merged revisions 374695 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374708 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-08 20:39 +0000 [r374633-374677] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample: Disable ICE
+ support by default Since there are a number of legacy devices out
+ there that fail to handle ICE candidates properly (which is a
+ nice way of saying something much uglier), disable it by default.
+ Support for ICE candidates can be enabled in rtp.conf using the
+ icesupport setting. ........ Merged revisions 374676 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/confbridge/conf_state_single_marked.c (added), /,
+ apps/confbridge/include/confbridge.h,
+ apps/confbridge/include/conf_state.h (added),
+ apps/confbridge/conf_state_multi.c (added),
+ apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
+ (added), apps/confbridge/conf_state_empty.c (added),
+ apps/confbridge/conf_state.c (added),
+ apps/confbridge/conf_state_single.c (added),
+ apps/confbridge/conf_state_inactive.c (added): Resolve issues in
+ ConfBridge regarding marked, waitmarked, and unmarked users
+ Thank's to Neil Tallim (flan)'s tireless testing, issue
+ reporting, and patches it became clear that app_confbridge had
+ some complex logic in how it handled interactions between marked,
+ waitmarked, and unmarked users. In particular, there were some
+ areas in which the interactions between the users resulted in
+ inconsistent behavior, and app_confbridge was missing logic in
+ how to handle some corner cases. Some areas included: * Poor
+ handling of mixing unmarked and waitmarked users *
+ Inconsistencies in how MOH and muting was applied to various
+ users * Handling of various announcements for different user
+ profile options flan's patches seem to fix the various issues,
+ but highlighted how hard the code could be to maintain. In an
+ attempt to make things easier to maintain and to more fully
+ enumerate the various cases that exist, this patch breaks up the
+ logic into a state machine-like setup. Please note that the
+ various state transitioned are documented on the Asterisk wiki:
+ https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
+ Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
+ the following issues, mjordan uploaded the patch, although it was
+ written by twilson. Any contributor license discrepency is due to
+ that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
+ flan, mjordan, jrose patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
+ flan Tested by: flan patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
+ Jonathan White Tested by: Jonathan White patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) ........ Merged revisions 374652 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374657 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/pjproject/pjlib/src/pj/sock_bsd.c,
+ res/pjproject/pjlib/src/pj/sock_linux_kernel.c,
+ res/pjproject/pjlib/include/pj/sock.h,
+ res/pjproject/pjlib/src/pj/sock_symbian.cpp: pjproject: Fix for
+ Solaris builds. Do not undef s_addr. pjproject, in order to solve
+ build problems on Windows [1], undefines s_addr in one of it's
+ headers that is included in res_rtp_asterisk.c. On Solaris s_addr
+ is not a structure member, but defined to map to the real
+ strucuture member, therefore when building on Solaris it's
+ possible to get build errors like: [CC] res_rtp_asterisk.c ->
+ res_rtp_asterisk.o In file included from
+ /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
+ from res_rtp_asterisk.c:51:
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In
+ function `inaddrcmp':
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
+ error: structure has no member named `s_addr'
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
+ error: structure has no member named `s_addr' res_rtp_asterisk.c:
+ In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706:
+ warning: dereferencing type-punned pointer will break
+ strict-aliasing rules res_rtp_asterisk.c:710: warning:
+ dereferencing type-punned pointer will break strict-aliasing
+ rules res_rtp_asterisk.c: In function
+ `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error:
+ structure has no member named `s_addr' make[2]: ***
+ [res_rtp_asterisk.o] Error 1 make[1]: *** [res] Error 2 make[1]:
+ Leaving directory `/export/home/admin/asterisk-11-svn' gmake: ***
+ [_cleantest_all] Error 2 Unfortunately, in order to make this
+ work, I also had to make sure pjproject only used the typdef
+ pj_in_addr and not the struct pj_in_addr so that when building
+ Asterisk I could "typedef struct in_addr pj_in_addr". It's
+ possible then that the library and users of those interfaces in
+ Asterisk have a different idea about the type of the argument,
+ while on the surface it looks like they are all 32 bit big endian
+ values. [1] http://trac.pjsip.org/repos/changeset/484 (issues
+ ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang,
+ mjordan patches:
+ 0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch
+ uploaded by Shaun Ruffell (license 5417) ........ Merged
+ revisions 374642 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/acl.c: Trivial patch to make 'best_score' defined for all
+ architectures. Fixes trivial build error on Solaris: acl.c: In
+ function `get_local_address': acl.c:196: error: `best_score'
+ undeclared (first use in this function) acl.c:196: error: (Each
+ undeclared identifier is reported only once acl.c:196: error: for
+ each function it appears in.) make[2]: *** [acl.o] Error 1 (issue
+ ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang
+ patches:
+ 0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch
+ by Shaun Ruffell (license 5417) ........ Merged revisions 374632
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-06 03:22 +0000 [r374612-374623] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c, /: Handle capability stanzas that fail to provide
+ node or version information While XEP-0115 states that the node
+ and ver attributes are both required, some devices fail to
+ provide either field. Prior to this patch, failure to provide the
+ node or ver attribute would cause a crash in res_xmpp. While
+ failing to provide the node or ver attribute is technically
+ invalid, since this information is not utilized by Asterisk
+ except for reporting purposes, for interoperability reasons, we
+ continue to process the capability stanza anyways. (closes issue
+ ASTERISK-20495) Reported by: Martin W Tested by: Martin W
+ patches: 20495.patch uploaded by Martin W (license #6434)
+ ........ Merged revisions 374622 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, res/res_xmpp.c, main/message.c: Update documentation for
+ MessageSend application/command's From field for XMPP When using
+ the channel technology agnostic application/AMI command
+ MessageSend, the "From" field is technically optional for the SIP
+ channel driver. However, if being sent by the XMPP resource
+ module (either res_xmpp or res_jabber), the "From" field is
+ necessary, and must correspond to a defined account. This patch
+ updates the documentation for this application/AMI command to
+ reflect this. (closes issue ASTERISK-20405) Reported by: Leif
+ Madsen ........ Merged revisions 374611 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-05 20:33 +0000 [r374588] David M. Lee <dlee@digium.com>
+
+ * main/manager.c, /: Multiple revisions 374570,374581 ........
+ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
+ 22 lines Improve AMI long line error handling In AMI's parser,
+ when it receives a long line (> 1024 characters), it discards
+ that line, but continues to process the message normally.
+ Typically, this is not a problem because a) who has lines that
+ long and b) usually a discarded line results in an invalid
+ message. But if that line is specifying an optional field, then
+ the message will be processed, you get a 'Response: Success', but
+ things don't work the way you expected them to. This patch
+ changes the behavior when a line-too-long parse error occurs. *
+ Changes the log message to avoid way-too-long (and truncated
+ anyways) log messages * Adds a 'parsing' status flag to Response:
+ Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line
+ is too long * Responds with an appropriate error if parsing !=
+ MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow
+ Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581
+ | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
+ I've committed too much. Reverting part of r374570. ........
+ Merged revisions 374570,374581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374586 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374587 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-05 18:42 +0000 [r374539] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
+ channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Merged
+ revisions 374515-374535 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+ ........ Merged revisions 374536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374537 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374538 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-04 20:21 +0000 [r374478-374493] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * CHANGES, main/dsp.c, /, configs/dsp.conf.sample: dsp.c User
+ Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
+ a recompile, allow values to be adjusted in dsp.conf For binary
+ distributions allows easy adjustment for wobbly GSM calls, and
+ other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
+ DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2144/ ........ Merged
+ revisions 374479 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374481 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374485 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's
+ always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if
+ hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2145/ ........ Merged
+ revisions 374475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374476 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374477 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-04 15:48 +0000 [r374429] David M. Lee <dlee@digium.com>
+
+ * main/db.c, /, res/res_agi.c: Fix DBDelTree error codes for AMI,
+ CLI and AGI The AMI DBDelTree command will return Success/Key
+ tree deleted successfully even if the given key does not exist.
+ The CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/ ........ Merged
+ revisions 374426 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374427 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374428 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-04 13:49 +0000 [r374414] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c,
+ channels/chan_sip.c: Add support for applying direct media ACLs
+ between differing channel technologies. Review:
+ https://reviewboard.asterisk.org/r/2122/
+
+2012-10-04 04:50 +0000 [r374387] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * CHANGES, main/dsp.c, /, configs/dsp.conf.sample: dsp.c User
+ configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
+ Asterisk's DTMF Specifications are based on AT&T specs, which may
+ not be compatible in other countries. Various countries have
+ different specifications for the maximum power level differences
+ between the DTMF low group and high group of frequencies. Power
+ level difference between frequencies for different
+ Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
+ 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
+ = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
+ (2006-03) Now allow 4 variables to be individually configured in
+ dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
+ specifications Add's the following variables to dsp.conf
+ ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
+ ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
+ (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
+ tbsky,alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2141/ ........ Merged
+ revisions 374384 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374385 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374386 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-04 02:16 +0000 [r374302-374338] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_jabber.c: Check for presence of buddy in info/dinfo
+ handlers The res_jabber resource module uses the ASTOBJ library
+ for managing its ref counted objects. After calling
+ ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
+ the object has to be checked to see if the buddy existed. Prior
+ to this patch, the buddy object was not checked for NULL; with
+ this patch in both aji_client_info_handler and aji_dinfo_handler
+ the pointer is checked before used and, if no buddy object was
+ found, the handlers return an error code. This patch does not
+ take the approach that our JID can be used to log in from another
+ resource. If that approach is desired, an improvement could be
+ made to this patch to create the buddy on the fly. This patch
+ seeks only to prevent Asterisk from crashing. FYI: In Asterisk
+ 11+, you really should be using res_xmpp. It does not have this
+ problem, as it moved to the astobj2 library. Note that multiple
+ people have proposed patches for this issue; the patch being
+ committed here is based on those. (closes issue ASTERISK-19532)
+ Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
+ fix-jabber uploaded by Karsten Wemheuer (license #5930)
+ xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
+ (license #6157) (closes issue ASTERISK-19557) Reported by:
+ ulugutz ........ Merged revisions 374335 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374336 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374337 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/ccss.c: Destroy the generic_monitors container after the
+ core_instances in ccss For each item in core_instances disposed
+ of in the shutdown of ccss, any generic monitor instances
+ referenced by the objects will be removed from generic_monitors
+ during their destruction. Hilarity ensues if generic_monitors no
+ longer exists. Thanks to the Asterisk Test Suite's generic_ccss
+ test for complaining loudly when it ran into this. ........
+ Merged revisions 374300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374301 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-02 23:23 +0000 [r374269-374279] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: Missed an astobj2.c debug tag.
+
+ * main/astobj2.c: * Add ref debug tags to astobj2.c ref usage. *
+ Make container nodes not show up in the ref debug log.
+
+2012-10-02 21:26 +0000 [r374197-374259] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.c, /: Ensure Shutdown AMI event is still fired
+ during Asterisk shutdown Richard pointed out that having the
+ manager dispose of itself gracefully during shutdown meant that
+ the Shutdown event will no longer get fired. This patch moves the
+ AMI event just prior to running the atexit callbacks. ........
+ Merged revisions 374230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374231 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374248 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * utils/hashtest2.c: Modify hashtest2 to compile after r374213.
+ Someone, somewhere, may care. Because hashtest2 has to provide
+ symbols for things in asterisk that items it includes may use,
+ when astobj2 decided to use ast_register_atexit it needed to
+ provide a declaration for that as well. Otherwise - no linky. On
+ a related note, ASTERISK-20505 was filed to convert
+ hashtest/hashtest2 into actual unit tests, so we don't run into
+ this problem again.
+
+ * /, main/astobj2.c, main/message.c: Fix findings from check-in on
+ r374177 Richard pointed out two problems with the check-in from
+ r374177: * The ast_msg_shutdown function declaration doesn't
+ match the prototype in main/message.c. * The ref/alloc function
+ usage in astobj2 (in trunk) can use the ao2_t_* variants of the
+ functions to allow the REF_DEBUG flag to enable/disable their
+ debug counterparts. ........ Merged revisions 374210 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374211 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_xmpp.c, main/taskprocessor.c, res/res_musiconhold.c,
+ main/named_acl.c, main/cel.c, main/astobj2.c, main/format_pref.c,
+ main/indications.c, main/channel.c, main/data.c, main/manager.c,
+ main/features.c, main/config_options.c, main/event.c,
+ main/message.c, main/asterisk.c, main/db.c, main/xmldoc.c,
+ main/format.c, main/udptl.c, main/pbx.c, /, main/ccss.c,
+ include/asterisk/astobj2.h, channels/chan_agent.c: Fix a variety
+ of ref counting issues This patch resolves a number of ref leaks
+ that occur primarily on Asterisk shutdown. It adds a variety of
+ shutdown routines to core portions of Asterisk such that they can
+ reclaim resources allocate duringd initialization. Review:
+ https://reviewboard.asterisk.org/r/2137 ........ Merged revisions
+ 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374178 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374196 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-01 23:39 +0000 [r374164-374167] Andrew Latham <lathama@gmail.com>
+
+ * main/asterisk.c, addons/app_mysql.c, include/asterisk/doxyref.h,
+ contrib/asterisk-ng-doxygen, main/http.c: Doxygen Cleanup Start
+ adding configuration file linking and pages. Add module loading
+ doxygen block. Breaking up commits to keep it easy to track
+ (issue ASTERISK-20259)
+
+ * channels/chan_jingle.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, channels/chan_vpb.cc, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_motif.c,
+ channels/chan_alsa.c, channels/chan_console.c,
+ channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
+ channels/chan_mgcp.c: Doxygen Cleanup Start adding configuration
+ file linking and pages. Add module loading doxygen block.
+ Breaking up commits to keep it easy to track (issue
+ ASTERISK-20259)
+
+ * res/res_phoneprov.c, res/res_musiconhold.c, res/res_xmpp.c,
+ res/res_config_ldap.c, res/res_curl.c, res/res_config_sqlite.c,
+ res/res_timing_kqueue.c, res/res_odbc.c, res/res_calendar.c,
+ res/res_clialiases.c, res/res_config_sqlite3.c, res/res_smdi.c,
+ res/res_snmp.c, res/res_fax.c: Doxygen Cleanup Start adding
+ configuration file linking and pages. Add module loading doxygen
+ block. Breaking up commits to keep it easy to track (issue
+ ASTERISK-20259)
+
+ * apps/app_meetme.c, apps/app_festival.c, apps/app_skel.c,
+ apps/app_alarmreceiver.c, apps/app_amd.c, apps/app_confbridge.c,
+ apps/app_followme.c, apps/app_queue.c, apps/app_adsiprog.c,
+ apps/app_voicemail.c: Doxygen Cleanup Start adding configuration
+ file linking and pages. Add module loading doxygen block. (issue
+ ASTERISK-20259)
+
+2012-10-01 20:36 +0000 [r374134-374151] Sean Bright <sean@malleable.com>
+
+ * tests/test_db.c, apps/app_queue.c, main/db.c,
+ include/asterisk/astdb.h, /: app_queue: Support persisting and
+ loading of long member lists. Greenlight in #asterisk brought up
+ that he was receiving an error message "Could not create
+ persistent member string, out of space" when running app_queue in
+ Asterisk 10. dump_queue_members() made an assumption that 8K
+ would be enough to store the generated string, but with queues
+ that have large member lists this is not always the case. This
+ patch removes the limitation and uses ast_str instead of a fixed
+ sized buffer. The complicating factor comes from the fact that
+ ast_db_get requires a buffer and buffer size argument, which
+ doesn't let us pull back more than what we pass in, so I
+ introduced a new ast_db_get_allocated() which returns an
+ ast_strdup()'d copy of the value from astdb. As an aside, I did
+ some testing on the maximum size of data that we can store in the
+ BDB library we distribute and was able to store a 10MB string and
+ retrieve it with no problems, so I feel this is a safe patch.
+ Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged
+ revisions 374108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374135 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374150 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/db.c, /: Use ast_copy_string instead of strncpy to guarantee
+ a NUL terminated string. ........ Merged revisions 374132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374133 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-01 17:05 +0000 [r374109] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cli.c: Change core show help output format. The CLI "core
+ show help" output leaves something to be desired. 1) The command
+ is truncated to a maximum of 30 characters. 2) The output columns
+ are mirrored from the 31st column. Current output format: logger
+ mute Toggle logging output to a console logger reload Reopens the
+ log files logger rotate Rotates and reopens the log files logger
+ set level {DEBUG|NOTICE Enables/Disables a specific logging level
+ for this console logger show channels List configured log
+ channels New format: logger mute -- Toggle logging output to a
+ console logger reload -- Reopens the log files logger rotate --
+ Rotates and reopens the log files logger set level
+ {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} --
+ Enables/Disables a specific logging level for this console logger
+ show channels -- List configured log channels Review:
+ https://reviewboard.asterisk.org/r/2133/
+
+2012-10-01 16:26 +0000 [r374107] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/confbridge/conf_config_parser.c: Don't destroy confbridge
+ config when error is encountered during a reload. Not panicking
+ means that the old config is kept. (closes issue ASTERISK-20458)
+ Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded
+ by Mark Michelson(license #5049) Tested by Leif Madsen ........
+ Merged revisions 374106 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-10-01 12:29 +0000 [r374096] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/speech.h, res/res_speech.c,
+ apps/app_speech_utils.c: Add support for retrieving engine
+ specific settings using the speech API and from dialplan. (closes
+ issue ASTERISK-17136) Reported by: kenner
+
+2012-09-29 03:56 +0000 [r374086] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fix ref leak when adding ICE candidates
+ to an SDP There was a missing decrement to the reference count
+ for the current ICE candidate when local candidates are being
+ added to an outbound SDP. This patch corrects that. ........
+ Merged revisions 374085 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-28 22:11 +0000 [r374075] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_agi.c: Include channel uniqueid in "AsyncAGI" and
+ "AGIExec" events. * Added AMI event documentation for AsyncAGI
+ and AGIExec events. (closes issue ASTERISK-20318) Reported by:
+ Dan Cropp Patches: res_agi_patch.txt (license #6422) patch
+ uploaded by Dan Cropp modified for trunk.
+
+2012-09-28 19:37 +0000 [r374060] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
+ The opinion of development was that it is both improper to have
+ Matt's personal email address used in the source and that the
+ command wouldn't be useful without it. (closes issue AST-467)
+ Reported by: Malcolm Davenport ........ Merged revisions 374032
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 374045 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 374059 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-28 18:27 +0000 [r374030] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt, main/app.c, apps/app_senddtmf.c,
+ channels/chan_dahdi.c, channels/sig_analog.c: Add pause one
+ second W dial modifier. * The following dialplan applications now
+ recognize 'W' to pause sending DTMF for one second in addition to
+ the previously existing 'w' that paused sending DTMF for half a
+ second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog
+ port dialing and deferred DTMF dialing for PRI now distinguishes
+ between 'w' and 'W'. The 'w' pauses dialing for half a second.
+ The 'W' pauses dialing for one second. * Created dahdi_dial_str()
+ in chan_dahdi that eliminated a lot of duplicated dialing code
+ and diagnostic messages for the channel driver. (closes issue
+ ASTERISK-20039) Reported by: Jeremiah Gowdy Patches:
+ jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by
+ Jeremiah Gowdy Expanded patch to add support in chan_dahdi.
+ Tested by: rmudgett
+
+2012-09-28 13:04 +0000 [r374020] Brent Eagles <beagles@digium.com>
+
+ * res/res_xmpp.c, main/message.c, /: Reset hangup flags on channels
+ created through messages and cleanup globals in res_xmpp on
+ unload. This patch fixes an issue where hangup flags were not
+ being reset on a channel, affecting subsequent use of that
+ channel. The patch also adds some additional cleanup to res_xmpp
+ to fix an issue with reloading the module. (closes
+ ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles
+ Review: https://reviewboard.asterisk.org/r/2134/ ........ Merged
+ revisions 374019 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-28 12:17 +0000 [r373992] Joshua Colp <jcolp@digium.com>
+
+ * res/res_agi.c, /: Update documentation to make it explicit that
+ "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373990 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373991 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-28 03:06 +0000 [r373979] Matthew Jordan <mjordan@digium.com>
+
+ * CHANGES, apps/app_senddtmf.c: Add Duration header for PlayDTMF
+ AMI Action This patch adds an optional header to the PlayDTMF AMI
+ action, Duration. It allows the duration of the DTMF digit to be
+ played on the channel to be specified in milliseconds. (closes
+ issue ASTERISK-18172) Reported by: Renato dos Santos patches:
+ send-dtmf.patch uploaded by Renato dos Santos (license #6267)
+ Modified slightly for this commit for Asterisk 12.
+
+2012-09-27 22:43 +0000 [r373965-373967] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Tweak app_dial documentation.
+
+ * main/app.c: Cleanup ast_dtmf_stream() * Made ast_dtmf_stream()
+ wait after starting the silence generator rather than before. *
+ Made ast_dtmf_stream() put the peer in autoservice for the whole
+ time things are being done to the chan.
+
+ * /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
+ leak using channel name parameter. The SendDTMF channel name
+ parameter has two issues. 1) Crashes if the channel name does not
+ exist. 2) Leaks a channel reference if the channel is the current
+ channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
+ documentation. * Renamed app to senddtmf_name and tweaked the
+ type. ........ Merged revisions 373945 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373946 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373954 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-27 17:12 +0000 [r373915] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/http_websocket.h,
+ res/res_http_websocket.c: Make res_http_websocket an optional
+ dependency on supported platforms for chan_sip. (closes issue
+ ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded
+ by sruffell (license 5417) ........ Merged revisions 373914 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-27 17:02 +0000 [r373913] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_voicemail.c, CHANGES: Add VoicemailRefresh AMI Action
+ Currently, if there are modifications to mailboxes that Asterisk
+ is not aware of, the user needs to add "pollmailboxes" to their
+ mailbox configuration, which repeatedly polls the subscribed
+ mailboxes for changes. This results in a lot of extra work for
+ the CPU. This patch introduces the AMI command VoicemailRefresh
+ which permits external applications to trigger the refresh
+ themselves. The refresh can apply to a specified mailbox only, an
+ entire context, or all configured mailboxes. Even a refresh
+ performed on every mailbox would not consume as much CPU as the
+ pollmailboxes option, given that pollmailboxes runs continuously
+ and this only runs on demand. (closes issue ASTERISK-17206)
+ (closes issue ASTERISK-19908) Reported-by: Jeff Hutchins
+ Reported-by: Tilghman Lesher Patch-by: Tilghman Lesher
+
+2012-09-27 16:53 +0000 [r373881-373912] Joshua Colp <jcolp@digium.com>
+
+ * /, main/loader.c: loader: Ensure dependent modules are properly
+ initialized. If an Asterisk module specifies a dependency in
+ ast_module_info.nonoptreq, it is possible for Asterisk to skip
+ calling the modules's .load function. Asterisk was loading and
+ linking the module via load_dynamic_module() but was not adding
+ the module to the resource_heap. Therefore the module was not
+ initialized based on it's priority along with the other modules
+ in the heap. Now use load_resource() instead of
+ load_dynamic_module() for non-optional requirement. This will add
+ the module to the resource_heap so the module can be properly
+ initialized in the correct order. This is required if there are
+ any module global data structures initialized in the .load()
+ callback for the module on platforms which do not support weak
+ references. (issue ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-loader-Ensure-dependent-modules-are-properly-initial.patch
+ uploaded by sruffell (license 5417) ........ Merged revisions
+ 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373910 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373911 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_local.c, /: Fix an issue where Local channels
+ dialed by app_queue are considered in use immediately. The
+ chan_local channel driver returns a device state of in use even
+ if a created Local channel has not yet been dialed. This fix
+ changes the logic to return a state of not in use until the
+ channel itself has been dialed. (closes issue ASTERISK-20390)
+ Reported by: tim_ringenbach Review:
+ https://reviewboard.asterisk.org/r/2116/ ........ Merged
+ revisions 373878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373879 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373880 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-26 21:17 +0000 [r373852] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Move handling of 408 response so there is
+ no misleading warning message. (closes issue ASTERISK-20060)
+ Reported by: Walter Doekes ........ Merged revisions 373848 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373849 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373850 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-26 18:23 +0000 [r373835] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Fixed meetme tab completion and command
+ documentation. * Removed unnecessary case sensitivity in meetme
+ list, lock, unlock, mute, unmute, and kick commands. * Separated
+ meetme lock/unlock, mute/unmute, and kick commands into their own
+ registered commands to simplify tab completion and parameter
+ checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
+ Merged revisions 373815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373816 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373818 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-26 08:31 +0000 [r373805] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, apps/app_queue.c: app_queue: 'agent available' hint, cleanup
+ restart, and initial state Fix previously untested senarios; 1).
+ On queue initialisation set queue_avail devstate to INUSE.
+ Previously was unavailable, which indicated an agent was
+ available. 2). When removing members, if there are no other
+ members available, set queue_avail to INUSE. Previously, if a
+ member interface had become 'unavailable', they were never going
+ to be removed, particularly when persistant queues is enabled.
+ 3). When adding a member, check that they are available, if they
+ are set queue_avail to NOT_INUSE. Previously on reloaded, members
+ may have been 'unavailable'. 4). When pausing or unpausing a
+ member, set appropriate queue availability. alecdavis (license
+ 585) Reported by: Alec Davis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2129/ ........ Merged
+ revisions 373804 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 23:10 +0000 [r373740-373776] Mark Michelson <mmichelson@digium.com>
+
+ * main/say.c, /: Fix saying of date in Dutch. The Dutch say the
+ date before the month. (closes issue ASTERISK-20353) Reported by:
+ Teun Ouwehand ........ Merged revisions 373773 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373774 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373775 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * configs/agents.conf.sample, /, channels/chan_agent.c: Remove dead
+ code and documentation for nonexistent feature. multiplelogin was
+ removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
+ was removed. (closes issue AST-948) reported by Steve Pitts
+ ........ Merged revisions 373768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373769 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373770 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_voicemail.c: Fix error where improper IMAP greetings
+ would be deleted. (closes issue ASTERISK-20435) Reported by:
+ fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
+ uploaded by Michael L. Young (License #5026) (with suggested
+ modification made by me) ........ Merged revisions 373735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373738 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 20:14 +0000 [r373708] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Fix T.38 support when used with
+ chan_local in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/ ........ Merged
+ revisions 373705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373706 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373707 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 19:29 +0000 [r373701] Mark Michelson <mmichelson@digium.com>
+
+ * main/callerid.c, include/asterisk/channel.h, CHANGES,
+ channels/sig_pri.c, funcs/func_callerid.c,
+ include/asterisk/callerid.h, main/channel.c,
+ channels/chan_misdn.c, channels/chan_sip.c: Allow for redirecting
+ reasons to be set to arbitrary strings. This allows for the
+ REDIRECTING dialplan function to be used to set the reason to any
+ string. The SIP channel driver has been modified to set the
+ redirecting reason string to the value received in a Diversion
+ header. In addition, SIP 480 response reason text will set the
+ redirecting reason as well. (closes issue AST-942) reported by
+ Malcolm Davenport (closes issue AST-943) reported by Malcolm
+ Davenport Review: https://reviewboard.asterisk.org/r/2101
+
+2012-09-25 19:08 +0000 [r373691] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
+ session timers The SIP session timer mechanism contains a
+ mandatory 'refresher' parameter (included in the Session-Expires
+ header) which is used in the session timer offer/answer signaling
+ within a SIP Invite dialog. It looks like asterisk is
+ interpreting the uac resp. uas role only as the initial role of
+ client and server (caller is uac, callee is uas). The standard
+ rfc 4028 however assigns the client role to the ((RE)-Invite)
+ requester, the server role to the ((RE)-Invite) responder. This
+ patch has Asterisk track the actual refresher as "us" or "them"
+ as opposed to relying on just the configured "uas" or "uac"
+ properties. (closes issue AST-922) Reported by: Thomas Airmont
+ Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
+ revisions 373652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373665 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373690 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 18:33 +0000 [r373689] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c, /: "show" completion option for "queue"
+ shouldn't appear twice When tab-completing CLI commands starting
+ with "queue", "show" appeared twice in the list due to the way
+ that Asterisk's tab completion functions and the order in which
+ the commands were registered. The registration order has been
+ altered to resolve this issue. (closes issue AST-940)
+ Reported-by: Steve Pitts ........ Merged revisions 373666 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373675 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373688 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 17:22 +0000 [r373636-373656] Richard Mudgett <rmudgett@digium.com>
+
+ * /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix
+ valgrind found memcpy issues in codec_ilbc. Valgrind found
+ codec_ilbc using memcpy instead of memmove for overlapping memory
+ blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
+ Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
+ #5674) patch uploaded by Walter Doekes ........ Merged revisions
+ 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373645 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373650 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if
+ the respective sources change. ........ Merged revisions 373618
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373633 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373635 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 16:45 +0000 [r373608-373634] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Set Quality of Service for
+ video rtp instance (closes issue ASTERISK-20201) Reported by:
+ ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
+ 6008) ........ Merged revisions 373617 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373631 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373632 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_agi.c: res_agi: async_agi responsiveness improvement on
+ datastore problems This patch changes get_agi_cmd so that the
+ return can be checked to differentiate between an empty list
+ success and something that triggered an error. This in turn
+ allows launch_asyncagi to detect these errors and break free from
+ the command processing loop so that the async agi can be ended
+ more cleanly (closes issue ASTERISK-20109) Reported by: Jeremiah
+ Gowdy Patches: jgowdy-7-9-2012.diff uploaded by Jeremiah Gowdy
+ (license 6358) (Modified by me to fix some logical issues and
+ apply to trunk) Review: https://reviewboard.asterisk.org/r/2117/
+
+2012-09-25 14:13 +0000 [r373583] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_presencestate.c, /: "He who go through turnstile
+ sideways is going to Bangkok" ........ Merged revisions 373582
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 13:29 +0000 [r373581] Kinsey Moore <kmoore@digium.com>
+
+ * /, configs/res_odbc.conf.sample: Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
+ 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373579 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373580 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-25 12:12 +0000 [r373553] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_multicast.c, /: Fix an issue where a caller to
+ ast_write on a MulticastRTP channel would determine it failed
+ when in reality it did not. When sending RTP packets via
+ multicast the amount of data sent is stored in a variable and
+ returned from the write function. This is incorrect as any
+ non-zero value returned is considered a failure while a return
+ value of 0 is success. For callers (such as ast_streamfile) that
+ checked the return value they would have considered it a failure
+ when in reality nothing went wrong and it was actually a success.
+ The write function for the multicast RTP engine now returns -1 on
+ failure and 0 on success, as it should. (closes issue
+ ASTERISK-17254) Reported by: wybecom ........ Merged revisions
+ 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373551 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373552 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-24 22:14 +0000 [r373503] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Be consistent, send From: "Anonymous"
+ <sip:anonymous@anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola ........ Merged revisions 373500
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373501 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373502 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-24 21:19 +0000 [r373479] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, funcs/func_audiohookinherit.c, /:
+ func_audiohookinherit: Document some missed sources. This patch
+ also mentions that AUDIOHOOK_INHERIT can be used to transfer
+ MixMonitor audiohooks. There is also wiki that addresses
+ audiohooks and the use of AUDIOHOOK_INHERIT at the following
+ link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
+ (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........
+ Merged revisions 373467 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373468 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373470 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-24 21:15 +0000 [r373471] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix potential reentrancy problems in
+ chan_sip. Asterisk v1.8 and later was not as vulnerable to this
+ issue. * Made find_call() lock each private as it processes the
+ found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 373424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373466 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373469 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-24 19:23 +0000 [r373414-373456] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+ ........ Merged revisions 373438 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373440 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373454 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_format_attr_h264.c, /, channels/chan_sip.c: Fix an issue
+ with H.264 format attribute comparison and fix an issue with
+ improper SDP being produced. The H.264 format attribute module
+ compares two format attribute structures to determine if they are
+ compatible or not. In some instances it was possible for this
+ check to determine that both structures were incompatible when
+ they actually should be considered compatible. This check has now
+ been made even more permissive by assuming that if no attribute
+ information is available the two structures are compatible. If
+ both structures contain attribute information a base level
+ comparison of the H.264 IDC value is done to see if they are
+ compatible or not. The above issue uncovered a secondary issue in
+ chan_sip where the SDP being produced would be incorrect if the
+ formats were considered incompatible. This has now been fixed by
+ checking that all information required to produce the SDP is
+ available instead of assuming it is. (closes issue
+ ASTERISK-20464) Reported by: Leif Madsen ........ Merged
+ revisions 373413 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-24 12:42 +0000 [r373404] Brent Eagles <beagles@digium.com>
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample:
+ res_rtp_asterisk: Make TURN and STUN server configurations
+ consistent. This patch removes the turnport configuration
+ property and changes the turnaddr property to be a combined
+ host[:port] configuration string. The patch also modifies the
+ documentation in the example configuration to reflect the
+ property changes and adds some additional text indicating how the
+ STUN port is configured. (closes issue ASTERISK-20344) Reported
+ by: beagles Tested by: beagles Review:
+ https://reviewboard.asterisk.org/r/2111/ ........ Merged
+ revisions 373403 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-22 20:43 +0000 [r373384] Andrew Latham <lathama@gmail.com>
+
+ * tests/test_gosub.c, include/asterisk/doxygen/mantisworkflow.h
+ (removed), contrib/asterisk-ng-doxygen, channels/chan_agent.c,
+ main/astfd.c, apps/app_queue.c, codecs/speex/speex_resampler.h,
+ res/res_config_sqlite.c, Makefile, cel/cel_odbc.c,
+ include/asterisk/doxyref.h, main/manager.c, doc/README.txt,
+ include/asterisk/xmpp.h, apps/app_minivm.c,
+ cel/cel_sqlite3_custom.c, include/asterisk/format.h,
+ main/audiohook.c, include/asterisk/pbx.h,
+ res/res_timing_kqueue.c, addons/chan_mobile.c, main/asterisk.c,
+ main/xmldoc.c, channels/chan_mgcp.c, apps/app_voicemail.c,
+ utils/refcounter.c, res/res_config_pgsql.c, main/pbx.c,
+ main/ccss.c, channels/chan_sip.c: Doxygen Updates Janitor Work *
+ Whitespace, doc-blocks, spelling, case, missing and incorrect
+ tags. * Add cleanup to Makefile for the Doxygen configuration
+ update * Start updating Doxygen configuration for cleaner output
+ * Enable inclusion of configuration files into documentation *
+ remove mantisworkflow... * update documentation README * Add
+ markup to Tilghman's email and talk with him about updating his
+ email, he knows... * no code changes on this commit other than
+ the mentioned Makefile change (issue ASTERISK-20259)
+
+2012-09-21 19:35 +0000 [r373369] Jonathan Rose <jrose@digium.com>
+
+ * channels/iax2-provision.c, /: iax2-provision: Fix improper return
+ on failed cache retrieval (closes issue ASTERISK-20337) reported
+ by: John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 373342 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373343 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373368 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-21 18:22 +0000 [r373320-373341] Andrew Latham <lathama@gmail.com>
+
+ * contrib/asterisk-ng-doxygen: Update Doxygen Config Comments This
+ annoying update is almost totally whitespace and updated config
+ comments. I did add Python to the documented file types. (issue
+ ASTERISK-20259)
+
+ * include/asterisk/doxygen/commits.h, res/res_curl.c,
+ main/asterisk.c, main/xmldoc.c,
+ include/asterisk/doxygen/architecture.h, cel/cel_pgsql.c,
+ main/strings.c, res/res_config_pgsql.c, apps/app_meetme.c,
+ main/ccss.c, include/asterisk/doxygen/mantisworkflow.h,
+ main/sha1.c, codecs/codec_speex.c, res/res_crypto.c,
+ channels/sip/reqresp_parser.c, main/acl.c, apps/app_ices.c,
+ cdr/cdr_pgsql.c, channels/chan_jingle.c,
+ include/asterisk/doxygen/releases.h, include/asterisk/app.h,
+ main/ast_expr2f.c, apps/app_skel.c, channels/chan_motif.c,
+ main/http.c, channels/chan_h323.c,
+ include/asterisk/doxygen/reviewboard.h, apps/app_confbridge.c,
+ res/res_config_ldap.c, include/asterisk/acl.h,
+ funcs/func_speex.c, cel/cel_radius.c, res/res_snmp.c,
+ include/asterisk/localtime.h, res/res_xmpp.c,
+ channels/chan_console.c, res/res_jabber.c, cdr/cdr_radius.c,
+ res/res_config_curl.c, include/asterisk/doxyref.h,
+ include/asterisk/res_odbc.h, res/res_smdi.c, main/manager.c,
+ channels/chan_misdn.c,
+ include/asterisk/doxygen/asterisk-git-howto.h, main/tdd.c,
+ include/asterisk/bridging_features.h, cdr/cdr_sqlite.c,
+ include/asterisk/sip_api.h, include/asterisk/xmpp.h,
+ include/asterisk/jabber.h, channels/sip/include/sdp_crypto.h:
+ Doxygen Updates - janitor work Doxygen updates including
+ mistakes, misspellings, missing parameters, updates for Doxygen
+ style. Some missing txt file links are removed but their content
+ or essense will be included in some later updates. A majority of
+ the txt files were removed in the 1.6 era but never noted. The HR
+ and EXTREF are simple changes that make the documentation more
+ compatable with more versions of Doxygen. Further updates coming.
+ (issue ASTERISK-20259)
+
+ * README: Start work on documentation janitor project with a little
+ commit. This adds a link to the Asterisk wiki at
+ https://wiki.asterisk.org to the README file. (issue
+ ASTERISK-20259)
+
+2012-09-21 15:41 +0000 [r373319] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_queue.c, /: app_queue: Make queue reload members and
+ variants of that work Prior to this patch, 'queue reload members'
+ cli command did not work at all. This also affects the manager
+ function 'QueueReload' when supplied with the 'members: yes'
+ field. (closes issue AST-956) Reported by: John Bigelow ........
+ Merged revisions 373298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373318 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-21 09:11 +0000 [r373275-373284] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: dsp.c: remove more whitespace mentioned in review2107
+
+ * main/dsp.c: dsp.c ast_dsp_call_progress use local short variable
+ in loop, plus other cleanup janitor cleanup. No functional
+ change. 1). ast_dsp_call_progress: use 'short samp' instead of
+ s[x] inside loop. apply same casting as other _init, dsp->energy
+ = (int32_t) samp * (int32_t) samp 2). ast_dtmf_detect_init: move
+ repeated setting of s->energy to outside of loop. do
+ goertzel_init loop first before setting s->lasthit and
+ s->current_hit, consistant with ast_dsp_digitreset() 3).
+ ast_mf_detect_init: do goertzel_init loop first before setting
+ s->hits[] and s->current_hit, consistant with
+ ast_dsp_digitreset() 4). Don't chain init different variables, as
+ the type may change Review
+ https://reviewboard.asterisk.org/r/2107/
+
+2012-09-20 19:16 +0000 [r373247] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow ........ Merged revisions 373242 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373245 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373246 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-20 18:44 +0000 [r373239] Matthew Jordan <mjordan@digium.com>
+
+ * CHANGES, apps/app_queue.c, configs/extensions.conf.sample, /: Add
+ queue monitoring hints This patch adds support for hints on a
+ queue. Hints can be added using the nomenclature 'Queue:name',
+ where name is the name of the queue being monitored. This nifty
+ feature was done by Alec Davis. Review:
+ https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
+ Tested by: alecdavis patches: review1619.diff2 by alecdavis
+ (license 585) ........ Merged revisions 373235 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-20 18:27 +0000 [r373234] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/rtp_engine.h,
+ channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Add support for
+ DTLS-SRTP to res_rtp_asterisk and chan_sip. As mentioned on the
+ review for this, WebRTC has moved towards choosing DTLS-SRTP as
+ the mechanism for key exchange for SRTP. This commit adds support
+ for this but makes it available for normal SIP clients as well.
+ Testing has been done to ensure that this introduces no
+ regressions with existing behavior and also that it functions as
+ expected. Review: https://reviewboard.asterisk.org/r/2113/
+ ........ Merged revisions 373229 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-20 18:02 +0000 [r373222] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c: Support all ways a member can be available for
+ 'agent available' hints Alec's patch in r373188 added the ability
+ to subscribe to a hint for when Queue members are available. This
+ patch modifies the check that determines when a Queue member is
+ available by refactoring the availability checks in
+ num_available_members into a shared function is_member_available.
+ This should now handle the ringinuse option, as well as device
+ state values other than AST_DEVICE_NOT_INUSE.
+
+2012-09-20 17:22 +0000 [r373221] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, include/asterisk/features.h,
+ main/channel.c, /, apps/app_directed_pickup.c,
+ funcs/func_channel.c, main/features.c: Named call pickup groups.
+ Fixes, missing functionality, and improvements. * ASTERISK-20383
+ Missing named call pickup group features: CHANNEL(callgroup) -
+ Need CHANNEL(namedcallgroup) CHANNEL(pickupgroup) - Need
+ CHANNEL(namedpickupgroup) Pickup() - Needs to also select from
+ named pickup groups. * ASTERISK-20384 Using the pickupexten, the
+ pickup channel selection could fail even though there was a call
+ it could have picked up. In a call pickup race when there are
+ multiple calls to pickup and two extensions try to pickup a call,
+ it is conceivable that the loser will not pick up any call even
+ though it could have picked up the next oldest matching call.
+ Regression because of the named call pickup group feature. * See
+ ASTERISK-20386 for the implementation improvements. These are the
+ changes in channel.c and channel.h. * Fixed some locking issues
+ in CHANNEL(). (closes issue ASTERISK-20383) Reported by: rmudgett
+ (closes issue ASTERISK-20384) Reported by: rmudgett (closes issue
+ ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2112/ ........ Merged
+ revisions 373220 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-20 13:04 +0000 [r373212] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Correct handling of unknown SDP stream
+ types When the patch to handle arbitrary SDP stream arrangements
+ went into Asterisk, it also included an ability to transparently
+ decline unknown stream types. The scanf calls used were not
+ checked properly causing this part of the functionality to be
+ broken. (closes issue ASTERISK-20203) ........ Merged revisions
+ 373211 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-20 11:05 +0000 [r373203] Sean Bright <sean@malleable.com>
+
+ * res/res_curl.c: When trying to unload res_curl.so, warn about all
+ dependent modules. Before this, attempting to unload res_curl.so
+ would warn you about the first module it found that was
+ dependent. We now warn about all of the loaded modules instead.
+
+2012-09-20 10:41 +0000 [r373188-373202] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: dsp.c: remove whitespace mentioned in review2107
+ Related https://reviewboard.asterisk.org/r/2107/
+
+ * configs/extensions.conf.sample, CHANGES, apps/app_queue.c:
+ app_queue: Support an 'agent available' hint Sets INUSE when no
+ free agents, NOT_INUSE when an agent is free. modifes
+ handle_statechange() scan members loop to scan for a free agent
+ and updates the Queue:queuename_avial devstate. Previously exited
+ early if the member was found in the queue. Now Exits later when
+ both a member was found, and a free agent was found. alecdavis
+ (license 585) Reported by: Alec Davis Tested by: alecdavis
+ Review: https://reviewboard.asterisk.org/r/2121/
+
+2012-09-18 20:19 +0000 [r373134-373142] Sean Bright <sean@malleable.com>
+
+ * main/logger.c: Make the casing of CALL_ID in debug messages
+ consistent to satisfy my OCD.
+
+ * main/manager.c, /: Don't crash when passing a NULL message to
+ __astman_get_header. Before this commit, __astman_get_header
+ would blindly dereference the passed in 'struct message *' to
+ traverse the header list. There are cases, however, such as
+ '*CLI> sip qualify peer foo' where the message pointer is NULL,
+ so we need to check for that. ........ Merged revisions 373131
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373133 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-18 15:50 +0000 [r373120] David M. Lee <dlee@digium.com>
+
+ * makeopts.in, Makefile, include/asterisk/utils.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Add
+ -fnested-functions compile flag, if needed. In order to use
+ nested functions on some versions of GCC (e.g. GCC on OS X), the
+ -fnested-functions flag must be passed to the compiler. This
+ patch adds detection logic to ./configure to add the flag if
+ necessary. It also adds a comment to utils.h as to why the nested
+ function needs a prototype. (closes issue ASTERISK-20399)
+ Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2102/ ........ Merged
+ revisions 373119 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-15 00:32 +0000 [r373108] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, /: Made companding law for SS7 calls only
+ determined by SS7 signaling type. For SS7, the companding law for
+ a call was chosen inconsistently depending upon ss7type (ITU vs
+ ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+ calls, the companding law was determined by ss7type. For outgoing
+ calls, the companding law was determined by the DAHDI default.
+ With the wrong combination you would get A-law/u-law conflicts.
+ An A-law/u-law conflict sounds like bad static on the line. SS7
+ ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+ noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+ with T1 line: ok * Fix the companding law used to be determined
+ by the SS7 signaling type only. ........ Merged revisions 373090
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373101 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373107 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-14 19:53 +0000 [r373080] Matthew Jordan <mjordan@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.c:
+ Resolve memory leaks in TLS initialization and TLS client
+ connections This patch resolves two sources of memory leaks when
+ using TLS in Asterisk: 1) It removes improper initialization (and
+ multiple re-initializations) of portions of the SSL library.
+ Asterisk calls SSL_library_init and SSL_load_error_strings during
+ SSL initialization; collectively this obviates the need for
+ calling any of the following during initialization or client
+ connection handling: * ERR_load_crypto_strings (handled by
+ SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+ SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+ SSL_library_init) 2) Failure to completely clean up all memory
+ allocated by Asterisk and by the SSL library for TLS clients.
+ This included not freeing the SSL_CTX object in the SIP channel
+ driver, as well as not clearing the error stack when the TLS
+ client exited. Note that these memory leaks were found by Thomas
+ Arimont, and this patch was essentially written by him with some
+ minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+ Arimont (license 5525) Review:
+ https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
+ 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373062 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373079 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-13 20:05 +0000 [r373046-373048] David M. Lee <dlee@digium.com>
+
+ * /, main/Makefile: Fixed make clean when configured
+ --disable-asteriskssl ........ Merged revisions 373047 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, include/asterisk/channel.h, main/channel.c: Fix timeouts for
+ ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
+ its timeout to ast_waitfor_nandfds, expecting it to decrement the
+ timeout by however many milliseconds were waited. This is a
+ problem if it consistently waits less than 1ms. The timeout will
+ never be decremented, and we wait... FOREVER! This patch makes
+ ast_waitfordigit_full manage the timeout itself. It maintains the
+ previously undocumented behavior that negative timeouts wait
+ forever. (closes issue ASTERISK-20375) Reported by: Mark
+ Michelson Tested by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2109/ ........ Merged
+ revisions 373024 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373025 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 373029 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-12 21:02 +0000 [r372997] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_astobj2.c, main/astobj2.c, main/channel.c,
+ include/asterisk/astobj2.h: Enhance astobj2 to support other
+ types of containers. The new API allows for sorted containers,
+ insertion options, duplicate handling options, and traversal
+ order options. * Adds the ability for containers to be sorted
+ when they are created. * Adds container creation options to
+ handle duplicates when they are inserted. * Adds container
+ creation option to insert objects at the beginning or end of the
+ container traversal order. * Adds OBJ_PARTIAL_KEY to allow
+ searching with a partial key. The partial key works similarly to
+ the OBJ_KEY flag. (The real search speed improvement with this
+ flag will come when red-black trees are added.) * Adds container
+ traversal and iteration order options: Ascending and Descending.
+ * Adds an AST_DEVMODE compile feature to check the stats and
+ integrity of registered containers using the CLI "astobj2
+ container stats <name>" and "astobj2 container check <name>". The
+ channels container is normally registered since it is one of the
+ most important containers in the system. * Adds
+ ao2_iterator_restart() to allow iteration to be restarted from
+ the beginning. * Changes the generic container object to have a
+ v_method table pointer to support other types of containers. *
+ Changes the container nodes holding objects to be ref counted.
+ The ref counted nodes and v_method table pointer changes pave the
+ way to allow other types of containers. * Includes a large
+ astobj2 unit test enhancement that tests the new features.
+ (closes issue ASTERISK-19969) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2078/
+
+2012-09-12 20:54 +0000 [r372996] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c, /: Skip any non-content information when
+ looking for and handling content. This fixes a bug with Jitsi and
+ conference calling. Jitsi implements XEP-0298 which places some
+ conference-info information in the session-initiate request which
+ chan_motif did not expect to occur. ........ Merged revisions
+ 372995 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-12 18:33 +0000 [r372976-372985] Jonathan Rose <jrose@digium.com>
+
+ * res/res_xmpp.c, /: res_xmpp: Fix a segfault caused by bodyless
+ messages (closes issue ASTERISK-20361) Reported by: Noah
+ Engelberth Review: https://reviewboard.asterisk.org/r/2108/
+ ........ Merged revisions 372984 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * configs/logger.conf.sample, main/logger.c: logger: Add
+ rotatestrategy option of 'none' which does not perform rotations
+ With this option in use, it may be necessary to regulate your log
+ files externally. (closes issue ASTERISK-20189) Reported by: Jaco
+ Kroon Patches: asterisk-logger-norotate-trunk.patch uploaded by
+ Jaco Kroon (license 5671)
+
+2012-09-12 15:21 +0000 [r372943] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue. ........ Merged revisions 372932 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372933 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372937 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-12 14:22 +0000 [r372931] David M. Lee <dlee@digium.com>
+
+ * /, main/Makefile: Fixed r372696 when configured
+ --disable-asteriskssl; properly install libasteriskssl.dylib on
+ OS X. I didn't realize that libasteriskssl.c was still compiled,
+ even when you disable asteriskssl; it simple gets statically
+ linked into asterisk. ........ Merged revisions 372930 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-11 22:40 +0000 [r372918] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_local.c, /: chan_local: Switch from using a random
+ 4 digit hex identifier to unique id Changes chan_local channels
+ to use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/ ........ Merged
+ revisions 372902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372917 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-11 21:17 +0000 [r372887-372891] Mark Michelson <mmichelson@digium.com>
+
+ * main/message.c, main/asterisk.c, /, include/asterisk/_private.h:
+ Fix inability to shutdown gracefully due to an unending channel
+ reference. message.c makes use of a special message queue channel
+ that exists in thread storage. This channel never goes away due
+ to the fact that the taskprocessor used by message.c does not get
+ shut down, meaning that it never ends the thread that stores the
+ channel. This patch fixes the problem by shutting down the
+ taskprocessor when Asterisk is shut down. In addition, the thread
+ storage has a destructor that will release the channel reference
+ when the taskprocessor is destroyed. (closes issue AST-937)
+ Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
+ Michelson (License #5049) Tested by Jason Parker ........ Merged
+ revisions 372885 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372888 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/features.c: Fix bad channel application data reference.
+ When channels get bridged due to an AMI bridge action or a DTMF
+ attended transfer, the two channels that get bridged have their
+ application data pointing to the other channel's name. This means
+ that if one channel is hung up but the other moves on, it means
+ that the channel that moves on will have its application data
+ pointing at freed memory. (issue ASTERISK-20335) Reported by:
+ aragon ........ Merged revisions 372840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372841 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372886 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-11 18:09 +0000 [r372874] David M. Lee <dlee@digium.com>
+
+ * Makefile, /: Corrects the astsbindir setting when installing the
+ sample asterisk.conf. (closes issue ASTERISK-20406) ........
+ Merged revisions 372863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372864 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-11 14:43 +0000 [r372808-372832] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt, CHANGES: chan_sip: Fix CHANGES and UPGRADE.txt for
+ r372808 (issue AST-969) Reported by John Bigelow
+
+ * channels/chan_sip.c: chan_sip: Change SIPQualifyPeer to improve
+ initial response time Prior to this patch, The acknowledgement
+ wasn't produced until after executing the sip_poke_peer action
+ actually responsible for qualifying the peer. Now the response is
+ given immediately once it is known that a peer will be qualified
+ and a SIPqualifypeerdone event is issued when the process is
+ finished. Thanks to OEJ for identifying the problem and helping
+ to come up with a solution. (issue AST-969) Reported by John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2098/
+
+2012-09-10 21:00 +0000 [r372796-372807] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
+ when expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert ........ Merged revisions 372804 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372805 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372806 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_jingle.c, include/asterisk/doxygen/architecture.h,
+ /, main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c:
+ Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
+ chan_jingle, and res_jabber are now deprecated in favor of using
+ chan_motif and res_xmpp. They are a feature-equivalent
+ replacement and are written to be more easily maintainable.
+ (closes issue ASTERISK-20298) Review:
+ https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen
+ ........ Merged revisions 372795 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-10 19:22 +0000 [r372787] David M. Lee <dlee@digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Eliminate
+ "type-punned pointer" build warning. Removes
+ "res_rtp_asterisk.c:706: warning: dereferencing type-punned
+ pointer will break strict-aliasing rules" warning from the build
+ on 32-bit platforms. The problem is that 'size' was referenced
+ aliased to both (pj_size_t *) and (pj_ssize_t *). Now just make a
+ copy of size that is the right type so there isn't any pointer
+ aliasing happening. It also adds comments and asserts regarding
+ what looks like an inappropriate use of pj_sock_sendto, but is
+ actually totally fine. (closes issue ASTERISK-20368) Reported by:
+ Shaun Ruffell Tested by: Michael L. Young Patches:
+ 0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
+ uploaded by Shaun Ruffell (license 5417) slightly modified by
+ David M. Lee. ........ Merged revisions 372777 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-10 18:58 +0000 [r372755-372769] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow ........ Merged revisions 372765 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372767 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372768 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/channel.c: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 372736 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372754 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-09 01:28 +0000 [r372712] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/sip/sdp_crypto.c: Only re-create an SRTP session when
+ needed In r356604, SRTP handling was fixed to accomodate multiple
+ crypto keys in an SDP offer and the ability to re-create an SRTP
+ session when the crypto keys changed. In certain circumstances -
+ most notably when a phone is put on hold after having been
+ bridged for a significant amount of time - the act of re-creating
+ the SRTP session causes problems for certain models of phones.
+ The patch committed in r356604 always re-created the SRTP session
+ regardless of whether or not the cryptographic keys changed.
+ Since this is technically not necessary, this patch modifies the
+ behavior to only re-create the SRTP session if Asterisk detects
+ that the remote key has changed. This allows models of phones
+ that do not handle the SRTP session changing to continue to work,
+ while also providing the behavior needed for those phones that do
+ re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+ 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372710 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372711 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-08 06:18 +0000 [r372699] David M. Lee <dlee@digium.com>
+
+ * /, main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
+ tcptls.c. Without this flag, those files will compile with the
+ system installed OpenSSL headers (if they exist). This is a real
+ bummer if a different path was specified using --with-ssl=
+ (closes issue ASTERISK-20392) ........ Merged revisions 372682
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Recorded merge of revisions 372695 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........
+ Recorded merge of revisions 372696 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-07 23:10 +0000 [r372623-372658] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astmm.c, /: Fix MALLOC_DEBUG version of ast_strndup().
+ (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
+ Merged revisions 372655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372657 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter ........ Merged revisions 372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372629 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372630 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_queue.c: Fix exception path typo in app_queue.c
+ try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
+ Pepper Patches: fix-local-channel-locking.patch (license #6350)
+ patch uploaded by Jeremy Pepper ........ Merged revisions 372624
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 372625 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372626 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
+ ServerEmail and MailCommand reported values. The AMI action
+ VoicemailUsersList VoicemailUserEntry event headers ServerEmail
+ and MailCommand did not report the global values if they were not
+ overridden. The VoicemailUserEntry event header ServerEmail was
+ not populated with the global value if the voicemail user did not
+ override it. The VoicemailUserEntry event header MailCommand was
+ never populated with a value. * Removed unused struct ast_vm_user
+ member mailcmd[]. (closes issue AST-973) Reported by: John
+ Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372621 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372622 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-07 21:04 +0000 [r372610-372612] David M. Lee <dlee@digium.com>
+
+ * res/pjproject/third_party/gsm/lib,
+ res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
+ res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
+ res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
+ res/pjproject/third_party/lib, res/pjproject/third_party/bin,
+ res/pjproject/lib, res/pjproject/pjlib/lib, /,
+ res/pjproject/pjmedia/lib, codecs/ilbc,
+ res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin:
+ svn:ignore cleanup. * pjproject bin and lib directories should
+ pretty much ignore everything * Ignore *.o in codecs/ilbc
+ ........ Merged revisions 372611 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/Makefile, /: Fix parallel make for res_asterisk_rtp. Fixes a
+ build regression introduced in r369517 "Add support for
+ ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
+ http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
+ When compiling asterisk in parallel like: $ make -j 10 It's
+ possible to get errors like the following:
+ .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
+ separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
+ Error 1 make[2]: ***
+ [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
+ Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
+ `+' to parent make rule. This is because the build system is
+ trying to build each of the libraries in pjproject in parallel.
+ Now the build will build pjproject in a single job and link the
+ results into res_asterisk_rtp. Parallel builds, on one test
+ system, saves ~1.5 minutes from a default Asterisk build: Single
+ job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
+ 2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
+ 0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
+ ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
+ 1m2.353s user 2m39.120s sys 0m18.850s (closes issue
+ ASTERISK-20362) Reported by: Shaun Ruffel Patches:
+ 0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
+ uploaded by Shaun Ruffel (License #5417) ........ Merged
+ revisions 372609 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-07 02:27 +0000 [r372538-372584] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_minivm.c: Free ast_str objects when temp file fails
+ to be created in MiniVM The previous commit (r372554) was from a
+ patch that was written before r366880, which ensured that ast_str
+ objects allocated in the sendmail routine were free'd in off
+ nominal paths. This commit frees the string objects in the off
+ nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372582 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372583 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
+ issue in MiniVM when sending mail When MiniVM sends an e-mail and
+ it has the volgain option set, it will spawn sox in a separate
+ process to handle the manipulation of the sound file. In doing
+ so, it creates a temporary file. There are two problems here: 1)
+ The file descriptor returned from mkstemp is leaked 2) The
+ finalfilename character pointer points to a buffer that loses
+ scope once volgain processing is finished. Note that in r316265,
+ Russell fixed some gcc warnings by using the return value of the
+ mkstemp call. A warning was placed in minivm that the file
+ descriptor was going to be leaked. This patch reverts that
+ change, as it handles the leak and 'uses' the file descriptor
+ returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+ Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+ Cohen (license #5035) ........ Merged revisions 372554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372555 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372556 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, apps/app_queue.c: Update QueueMemberStatus event documentation
+ to include member status values The Status: header in a
+ QueueMemberStatus event (and other QueueMember* events) is the
+ numeric value of the device state corresponding to that Queue
+ Member. As those values are not exactly obvious, listing them in
+ the documentation is useful. Matt Riddell reported this
+ indirectly through the wiki page. (closes issue ASTERISK-20243)
+ Reported by: Matt Riddell ........ Merged revisions 372531 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-06 22:14 +0000 [r372524] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
+ parking a call for the second time. Using the AMI redirect action
+ to take an ISDN call out of a parking lot causes the MOH state to
+ get confused. The redirect action does not take the call off of
+ hold. When the call is subsequently parked again, the call no
+ longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+ repeated AST_CONTROL_HOLD frames if it is already in a state
+ where it is supposed to be sending MOH. The MOH may have been
+ stopped by other means. (Such as killing the generator.) This
+ simple fix is done rather than making the AMI redirect action
+ post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+ channel and thus potentially breaking something with an
+ unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+ jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 372521 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 372522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372523 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-06 21:43 +0000 [r372520] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_queue.c: Ensure listed queues are not offered for
+ completion When using tab-completion for the list of queues on
+ "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow ........ Merged revisions 372517 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372518 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372519 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-06 15:57 +0000 [r372474] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
+ directmediapermit/deny ACL works r366547 introduced a change to
+ the directmedia ACL for chan_sip which modified the behavior
+ significantly. Prior to the patch, this option would bridge peers
+ with directmedia if a peer's IP address matched its own
+ directmedia ACL. After that patch, the peer would check the
+ bridged peer's ACL instead. This change has been present since
+ 1.8.14.0. That patched failed to document the change in
+ Upgrade.txt, so this patch adds mention of that change to
+ UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+ ........ Merged revisions 372471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372472 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372473 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-06 14:31 +0000 [r372447] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c, /: Ensure "rules" is tab-completable for "queue
+ show" Previously, tabbing at the end of "queue show" produced a
+ list of available queues about which information could be shown,
+ but did not include an alternative command, "rules", to access
+ information about queue rules. The "rules" item should now be
+ shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow ........ Merged revisions
+ 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372445 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372446 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-06 02:52 +0000 [r372393-372420] Matthew Jordan <mjordan@digium.com>
+
+ * pbx/pbx_dundi.c, /: Fix DUNDi message routing bug when
+ neighboring peer is unreachable Consider a scenario where DUNDi
+ peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+ and where PBX2 and PBX3 are also neighbors. If the connection is
+ temporarily broken between PBX1 and PBX3, PBX1 should not include
+ PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+ message, as it cannot send messages to PBX3. If it does, PBX2
+ will assume that PBX3 already received the message and fail to
+ forward the message on to PBX3 itself. This patch fixes this by
+ only including peers in a DPDISCOVER message that are reachable
+ by the sending node. This includes all peers with an empty
+ address (00:00:00:00:00:00) and that are have been reached by a
+ qualify message. This patch also prevents attempting to qualify a
+ dynamic peer with an empty address until that peer registers. The
+ patch uploaded by Peter was modified slightly for this commit.
+ (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+ dundi_routing.patch uploaded by Peter Racz (license 6290)
+ ........ Merged revisions 372417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372418 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372419 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_followme.c, /: Allow configured numbers for FollowMe to
+ be greater than 90 characters When parsing a 'number' defined in
+ followme.conf, FollowMe previously parsed the number in the
+ configuration file into a buffer with a length of 90 characters.
+ This can artificially limit some parallel dial scenarios. This
+ patch allows for numbers of any length to be defined in the
+ configuration file. Note that Clod Patry originally wrote a patch
+ to fix this problem and received a Ship It! on the JIRA issue.
+ The patch originally expanded the buffer to 256 characters.
+ Instead, the patch being committed duplicates the string in the
+ config file on the stack before parsing it for consumption by the
+ application. (closes issue ASTERISK-16879) Reported by: Clod
+ Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
+ by Clod Patry (license #5138) Slightly modified for this commit.
+ ........ Merged revisions 372390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372391 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372392 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 19:44 +0000 [r372374] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 372373 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix
+ compile error. ........ Merged revisions 372372 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:26 +0000 [r372344-372371] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c, /: Correct documentation for ModuleLoad AMI
+ action The documentation incorrectly listed 'rtp' as a reloadable
+ subsystem and left out many other reloadable subsystems. It is
+ now also documented that subsystems may only be reloaded, not
+ loaded or unloaded. (closes issue AST-977) Reported-by: John
+ Bigelow ........ Merged revisions 372354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372365 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/pbx.c, /: Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow ........ Merged revisions 372337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372338 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372340 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 18:56 +0000 [r372343] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
+ goertzel samples to 160, should be MF_GSIZE Remove unused
+ goertzel_state_t member 'samples'. Related
+ https://reviewboard.asterisk.org/r/2097/
+
+2012-09-05 17:38 +0000 [r372329] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Multiple revisions 372327-372328
+ ........ r372327 | rmudgett | 2012-09-05 12:33:11 -0500 (Wed, 05
+ Sep 2012) | 15 lines Fix RTP/RTCP read error message confusion.
+ The RTP/RTCP read error message can report "fail: success" when
+ the read failure is because of an ICE failure. * Changed
+ __rtp_recvfrom() to generate a PJ ICE message when ICE fails. *
+ Changed RTP/RTCP read error message to indicate an unspecified
+ error when errno is zero. (closes issue ASTERISK-20288) Reported
+ by: Joern Krebs Patches: jira_asterisk_20288_err_msg.patch
+ (license #5621) patch uploaded by rmudgett (modified) ........
+ r372328 | rmudgett | 2012-09-05 12:35:20 -0500 (Wed, 05 Sep 2012)
+ | 1 line Fix coding guidelines issue with a recent commit.
+ ........ Merged revisions 372327-372328 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 16:24 +0000 [r372310-372319] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp_engine.c, /, include/asterisk/rtp_engine.h,
+ res/res_rtp_asterisk.c: Re-fix sending unnegotiated payloads
+ during a P2P RTP bridge. The previous fix still would look in the
+ static_RTP_PT table, which is inappropriate since we specifically
+ want to find a codec that has been negotiated. (closes issue
+ ASTERISK-20296) reported by NITESH BANSAL Patches:
+ codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
+ ........ Merged revisions 372311 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_alarmreceiver.c: Add fixes and cleanup to
+ app_alarmreceiver. This work comes courtesy of Pedro Kiefer
+ (License #6407) The work was posted to review board by Kaloyan
+ Kovachev (License #5506) (closes issue ASTERISK-16668) Reported
+ by Grant Crawshay (closes issue ASTERISK-16694) Reported by Fred
+ van Lieshout (closes issue ASTERISK-18417) Reported by Kostas
+ Liakakis (closes issue ASTERISK-19435) Reported by Deon George
+ (closes issue ASTERISK-20157) Reported by Pedro Kiefer (closes
+ issue ASTERISK-20158) Reported by Pedro Kiefer (closes issue
+ ASTERISK-20224) Reported by Pedro Kiefer Review:
+ https://reviewboard.asterisk.org/r/2075
+
+2012-09-05 14:44 +0000 [r372302] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
+ using IMAP storage or realtime config This patch fixes two memory
+ leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+ 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372288 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372289 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 14:12 +0000 [r372290] Darren Sessions <dmsessions@gmail.com>
+
+ * configs/res_ldap.conf.sample, channels/chan_sip.c: LDAP Realtime
+ Peers Cannot Register Prior to 1.8, it was not necessary for an
+ explicit "type" to be set for an asterisk LDAP realtime peer. Now
+ the routine find_peer actually checks the type field during
+ registration and fails to find the peer if it is not set. The
+ attached patch makes the realtime type equal whatever type is
+ being searched for if the type is 0 upon return from routine
+ build_peer. (closes issue ASTERISK-17222) Reported by: John
+ Covert Patch by: David Vossel Tested by: Darren Sessions Review:
+ https://reviewboard.asterisk.org/r/2095/
+
+2012-09-05 12:18 +0000 [r372267] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix breakage caused by last merge.
+ Missing a variable for 11 and trunk. ........ Merged revisions
+ 372266 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 07:43 +0000 [r372215-372242] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit
+ delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss
+ detector to original -r349249 method with some changes, remove
+ unnecessary; 1. reseting of hits=0, when no signal, only need to
+ set it once. 2. incrementing of hits, when the hit is the same as
+ the current hit. 3. setting of lasthit, when it's the same as
+ before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
+ spelling mistakes (closes issue ASTERISK-19610) alecdavis
+ (license 585) Reported by: Jean-Philippe Lord Tested by:
+ alecdavis Review: https://reviewboard.asterisk.org/r/2085/
+ ........ Merged revisions 372239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372240 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372241 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/dsp.c, /: dsp.c: optimize goerztzel sample loops, in
+ dtmf_detect, mf_detect and tone_detect use a temporary short int
+ when repeatedly used to call goertzel_sample. alecdavis (license
+ 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2093/ ........ Merged
+ revisions 372212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372213 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372214 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 04:55 +0000 [r372200] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For
+ Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
+ place to increment the sequence number for retransmitted DTMF end
+ packets. With the introduction of the RTP engine API in 1.8, the
+ sequence number was no longer being incremented. This patch fixes
+ this regression as well as cleans up a few lines that were not
+ doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
+ Bansal Tested by: Michael L. Young Patches:
+ 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
+ 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2083/ ........ Merged
+ revisions 372185 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372198 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372199 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-05 02:26 +0000 [r372176] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully
+ written to PostgreSQL database PQClear is not called when the
+ result object of a call to PQExec has a status of
+ PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
+ handled properly, so this memory leak only occurred when CEL
+ records were successfully written. This patch properly clears the
+ result in the nominal code path. (closes issue ASTERISK-19991)
+ Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
+ mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
+ #6394) ........ Merged revisions 372158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372165 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372175 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-09-04 19:30 +0000 [r372148-372149] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: app_queue: PAUSEALL/UNPAUSEALL logged only if
+ interface is a queue member Adding UPGRADE.txt entry for r372148
+ (issue AST-946) Reported by: John Bigelow
+
+ * CHANGES, apps/app_queue.c: app_queue: Only log
+ PAUSEALL/UNPAUSEALL when 1+ memebers changed. Prior to this
+ patch, if pause or unpause was issued on an interface without
+ specifying a specific queue, a PAUSEALL or UNPAUSEALL event would
+ be logged in the queue log even if that interface wasn't a member
+ of any queues. This patch changes it so that these events are
+ only logged when at least one member of any queue exists for that
+ interface. (closes issue AST-946) Reported by: John Bigelow
+ Review: https://reviewboard.asterisk.org/r/2079/
+
+2012-09-04 15:50 +0000 [r372136-372138] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix issue where SIP devices were not
+ notified when custom devices changed to "ringing". The problem
+ had to do with logic used when checking for what the oldest
+ ringing channel was. The problem was that if no channel was
+ found, then no notification would be sent. For custom device
+ states, there is no associated channel, so no notification would
+ get sent. This fixes the issue by still sending the notification
+ even if no associated channel can be found for a ringing device
+ state change. (closes issue ASTERISK-20297) Reported by Noah
+ Engelberth ........ Merged revisions 372137 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/config_options.c, apps/app_confbridge.c, /: Prevent crash
+ from using app_page with no confbridge.conf file provided. Also
+ prevents other potential crashes when using aco API with
+ uninitialized aco_info structs. (closes issue ASTERISK-20305)
+ reported by Noah Engelberth Tested by Noah Engelberth Review:
+ https://reviewboard.asterisk.org/r/2086 ........ Merged revisions
+ 372135 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-31 21:15 +0000 [r372119] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_rtp_asterisk.c: Prevent local RTP bridges from sending
+ inappropriate formats to participants. A change for Asterisk 11
+ caused a check for failure to incorrectly check the return value.
+ This resulted in the possibility of transmitting media that a
+ party had not negotiated. If this media happened to be G.729,
+ then this could potentially result in one-way audio if no G.729
+ translators are installed. (closes issue ASTERISK-20296) reported
+ by NITESH BANSAL ........ Merged revisions 372118 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-30 20:54 +0000 [r372051-372092] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Prevent crash on shutdown due to refcount
+ error on queues container. When app_queue is unloaded, the queues
+ container has its refcount decremented, potentially to 0. Then
+ the taskprocessor responsible for handling device state changes
+ is unreferenced. If the taskprocessor happens to be just about to
+ run its task, then it will create and destroy an iterator on the
+ queues container. This can cause the refcount on the queues
+ container to increase to 1 and then back to 0. Going back to 0 a
+ second time results in double frees. This failure was seen
+ periodically in the testsuite when Asterisk would shut down.
+ ........ Merged revisions 372089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372090 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372091 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_queue.c, /: Help prevent ringing queue members from
+ being rung when ringinuse set to no. Queue member status would
+ not always get updated properly when the member was called, thus
+ resulting in the member getting multiple calls. With this change,
+ we update the member's status at the time of calling, and we also
+ check to make sure the member is still available to take the call
+ before placing an outbound call. (closes issue ASTERISK-16115)
+ reported by nik600 Patches: app_queue.c-svn-r370418.patch
+ uploaded by Italo Rossi (license #6409) ........ Merged revisions
+ 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372049 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372050 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-30 16:25 +0000 [r371964-372029] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_iax2.c, /: AST-2012-013: Resolve ACL rules being
+ ignored during calls by some IAX2 peers When an IAX2 call is made
+ using the credentials of a peer defined in a dynamic Asterisk
+ Realtime Architecture (ARA) backend, the ACL rules for that peer
+ are not applied to the call attempt. This allows for a remote
+ attacker who is aware of a peer's credentials to bypass the ACL
+ rules set for that peer. This patch ensures that the ACLs are
+ applied for all peers, regardless of their storage mechanism.
+ (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
+ mjordan, Alan Frisch ........ Merged revisions 372028 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /, README-SERIOUSLY.bestpractices.txt:
+ AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+ ExternalIVR The AMI Originate action can allow a remote user to
+ specify information that can be used to execute shell commands on
+ the system hosting Asterisk. This can result in an unwanted
+ escalation of permissions, as the Originate action, which
+ requires the "originate" class authorization, can be used to
+ perform actions that would typically require the "system" class
+ authorization. Previous attempts to prevent this permission
+ escalation (AST-2011-006, AST-2012-004) have sought to do so by
+ inspecting the names of applications and functions passed in with
+ the Originate action and, if those applications/functions matched
+ a predefined set of values, rejecting the command if the user
+ lacked the "system" class authorization. As noted by IBM X-Force
+ Research, the "ExternalIVR" application is not listed in the
+ predefined set of values. The solution for this particular
+ vulnerability is to include the "ExternalIVR" application in the
+ set of defined applications/functions that require "system" class
+ authorization. Unfortunately, the approach of inspecting fields
+ in the Originate action against known applications/functions has
+ a significant flaw. The predefined set of values can be bypassed
+ by creative use of the Originate action or by certain dialplan
+ configurations, which is beyond the ability of Asterisk to
+ analyze at run-time. Attempting to work around these scenarios
+ would result in severely restricting the applications or
+ functions and prevent their usage for legitimate means. As such,
+ any additional security vulnerabilities, where an
+ application/function that would normally require the "system"
+ class authorization can be executed by users with the "originate"
+ class authorization, will not be addressed. Instead, the
+ README-SERIOUSLY.bestpractices.txt file has been updated to
+ reflect that the AMI Originate action can result in commands
+ requiring the "system" class authorization to be executed. Proper
+ system configuration can limit the impact of such scenarios.
+ (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
+ X-Force Research ........ Merged revisions 371998 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371999 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 372000 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/cdr.c, main/channel.c, include/asterisk/crypto.h,
+ include/asterisk/config_options.h,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/audiohook.h,
+ apps/confbridge/include/confbridge.h, include/asterisk/format.h,
+ include/asterisk/netsock2.h, include/asterisk/rtp_engine.h,
+ include/asterisk/ccss.h, main/pbx.c, include/asterisk/utils.h,
+ channels/sip/srtp.c, channels/chan_sip.c,
+ include/asterisk/format_pref.h, include/asterisk/astobj2.h,
+ include/asterisk/presencestate.h, channels/chan_agent.c,
+ include/asterisk/config.h, pbx/pbx_lua.c,
+ formats/format_ogg_vorbis.c, include/asterisk/channel.h,
+ main/named_acl.c, codecs/speex/speex_resampler.h,
+ include/asterisk/manager.h, include/asterisk/format_cap.h,
+ include/asterisk/framehook.h, include/asterisk/heap.h,
+ channels/sig_pri.h, Makefile, include/asterisk/message.h,
+ include/asterisk/bridging.h, include/asterisk/datastore.h,
+ main/file.c, include/asterisk/strings.h, include/asterisk/pbx.h,
+ channels/sip/include/srtp.h, main/audiohook.c,
+ include/asterisk/translate.h: Clean up doxygen warnings This
+ patch fixes numerous doxygen warnings across Asterisk. It also
+ updates the makefile to regenerate the doxygen configuration on
+ the local system before running doxygen to help prevent
+ warnings/errors on the local system. Much thanks to Andrew for
+ tackling one of the Asterisk janitor projects! (issue
+ ASTERISK-20259) Reported by: Andrew Latham Patches:
+ doxygen_partial.diff uploaded by Andrew Latham (license 5985)
+ make_progdocs.diff uploaded by Andrew Latham (license 5985)
+
+ * doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to
+ doc folder In r294740, the CODING-GUIDELINES was removed from the
+ doc folder in favor of the content on the Asterisk wiki. Some
+ folks still look in the doc folder initially for coding guideline
+ suggestions; as such, this patch adds a CODING-GUIDELINES file
+ back into the doc folder. The content of the file merely points
+ to the correct page on the Asterisk wiki where the coding
+ guidelines currently live. (closes issue ASTERISK-20279) Reported
+ by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
+ Andrew Latham (license 5985) ........ Merged revisions 371961
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371962 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371963 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-29 22:48 +0000 [r371951-371952] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/md5.h: Ensure alignment of in[] field in
+ MD5Context struct. The struct MD5Context character buffer is cast
+ to an int32_t* without making sure that said buffer is aligned.
+ Since the buffer follows two uint32_t's, the chance of 'in' being
+ (32 bits) unaligned is nil in practice. But adding code to ensure
+ that 'in' stays aligned costs nothing and removes all doubts
+ about the casts being safe. (closes issue ASTERISK-20241)
+ Reported by: Walter Doekes Patches: tmp.diff (license #5674)
+ patch uploaded by Walter Doekes
+
+ * /, apps/app_meetme.c: Fix compile errors. ........ Merged
+ revisions 371950 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-29 21:15 +0000 [r371922] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Adding test events for
+ following activity in MeetMe. ........ Merged revisions 371919
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371920 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371921 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-29 19:57 +0000 [r371892-371894] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix theoretical compile error with HAVE_EPOLL.
+ Really shows how much epoll is used since it had not been
+ reported yet. ........ Merged revisions 371893 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/channel.c, /: Initialize file descriptors for dummy channels
+ to -1. Dummy channels usually aren't read from, but functions
+ like SHELL and CURL use autoservice on the channel. (closes issue
+ ASTERISK-20283) Reported by: Gareth Palmer Patches:
+ svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
+ (modified) ........ Merged revisions 371888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371890 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371891 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-29 19:38 +0000 [r371889] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, UPGRADE.txt: chan_sip: Change manager event
+ to confirm SIPqualifypeer into an ack Matt Jordan informed me
+ that it was more appropriate to use an astman_send_ack here
+ instead of making an event response. I've also used this
+ opportunity to update UPGRADE.txt to mention this change in
+ behavior. (issue AST-969) Reported by: John Bigelow
+
+2012-08-29 18:40 +0000 [r371863] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Fix hangup cause passthrough regression. The
+ v1.8 -r369258 change to fix the F and F(x) action logic
+ introduced a regression in passing the hangup cause from the
+ called channel to the caller channel. (closes issue
+ ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged
+ revisions 371860 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371861 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371862 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-29 17:35 +0000 [r371823-371851] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
+ instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
+ Doekes ........ Merged revisions 371824 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371825 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371845 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/chan_sip.c: chan_sip: Send a manager event to confirm
+ SIPqualifypeer completes Prior to this patch, Issuing
+ SIPqualifypeer either resulted in an error or if it succeeded, a
+ few \r\ns. This patch adds a SIPqualifypeerComplete event issued
+ as a response when the command is successfully executed. (closes
+ issue AST-969) Reported by: John Bigelow
+
+2012-08-27 21:51 +0000 [r371785-371791] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/agents.conf.sample: Fix misleading documentation in
+ agents.conf.sample regarding ackcall usage. The documentation
+ made it sound as if the DTMF acknowledgment was needed at the
+ time the agent logs in, rather than when the agent is called.
+ This is likely a relic from the days when there were multiple
+ ways of logging in agents. (closes issue AST-962) reported by
+ Steve Pitts ........ Merged revisions 371787 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371789 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371790 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c, /: Fix incorrect documentation of the
+ MailboxStatus manager command. The "Waiting" field was
+ misdocumented as reporting the number of messages waiting. In
+ reality, it simply indicated the presence or absence of waiting
+ messages. ........ Merged revisions 371782 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371783 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371784 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-27 18:16 +0000 [r371754] David M. Lee <dlee@digium.com>
+
+ * res/pjproject/pjlib-util/build/output, res/pjproject/pjnath/bin,
+ res/pjproject/pjlib/build/output, res/pjproject/pjlib-util/bin,
+ res/pjproject/pjnath/build/output, /, res/pjproject/pjlib/bin:
+ svn:ignore pjproject bin & output for all platforms. ........
+ Merged revisions 371753 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-27 17:52 +0000 [r371751] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, /: Fix incorrectly documented option
+ in queues.conf sharedlastcall defaults to "no" not "yes" (closes
+ issue AST-979) reported by Steve Pitts ........ Merged revisions
+ 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 371748 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371750 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-27 16:56 +0000 [r371721] David M. Lee <dlee@digium.com>
+
+ * /, main/lock.c: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
+ variants. The original implementations simply wrap pthread
+ functions, which take absolute time as an argument. The spinlock
+ version for systems without those functions treated the argument
+ as a delta. This patch fixes the spinlock version to be
+ consistent with the pthread version. (closes issue
+ ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
+ uploaded by Egor Gorlin (license 6416) ........ Merged revisions
+ 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 371720 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-27 14:13 +0000 [r371693] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash
+ When compiling with BETTER_BACKTRACES enabled, Asterisk will
+ sometimes crash when "core show locks" is run. This happens
+ regularly in the testsuite since several tests run "core show
+ locks" to help with debugging. This seems to be a fault with
+ libraries on certain operating systems (notably CentOS 6.2/6.3)
+ running on virtual machines and utilizing gcc 4.4.6. (closes
+ issue ASTERISK-20090) ........ Merged revisions 371690 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371691 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371692 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-26 23:10 +0000 [r371665] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of
+ MF_GSIZE ........ Merged revisions 371662 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371663 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371664 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-23 04:12 +0000 [r371633] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_scoped_lock.c (added): I forgot to add the unit tests
+ for scoped locks earlier today.
+
+2012-08-22 15:55 +0000 [r371620] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_motif.c: Add support for call-id logging to
+ chan_motif. Review: https://reviewboard.asterisk.org/r/2077/
+ ........ Merged revisions 371619 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-21 21:01 +0000 [r371572-371593] Mark Michelson <mmichelson@digium.com>
+
+ * cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c,
+ channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
+ main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
+ res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
+ res/res_config_sqlite.c: Fix misuses of asprintf throughout the
+ code. This fixes three main issues * Change asprintf() uses to
+ ast_asprintf() so that it pairs properly with ast_free() and no
+ longer causes MALLOC_DEBUG to freak out. * When ast_asprintf()
+ fails, set the pointer NULL if it will be referenced later. * Fix
+ some memory leaks that were spotted while taking care of the
+ first two points. (Closes issue ASTERISK-20135) reported by
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071
+ ........ Merged revisions 371590 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371591 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371592 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * include/asterisk/lock.h, main/config.c: Add scoped locks to
+ Asterisk. With the SCOPED_LOCK macro, you can create a variable
+ that locks a specific lock and unlocks the lock when the variable
+ goes out of scope. This is useful for situations where many
+ breaks, continues, returns, or other interruptions would require
+ separate unlock statements. With a scoped lock, these aren't
+ necessary. There are specializations for mutexes, read locks,
+ write locks, ao2 locks, ao2 read locks, ao2 write locks, and
+ channel locks. Each of these is a SCOPED_LOCK at heart though.
+ Review: https://reviewboard.asterisk.org/r/2060
+
+ * res/res_rtp_asterisk.c, /: Use thread-local storage to store
+ pj_thread_descs. pj_thread_register() takes a parameter of type
+ pj_thread_desc. It was assumed that pj_thread_register either
+ used this item temporarily or made a copy of it. Unfortunately,
+ all it does is keep a pointer to the structure in thread-local
+ storage. This means that if our pj_thread_desc goes out of scope,
+ then pjlib will be referencing bogus data quite often, most
+ commonly on operations involving a pj_mutex_t. In our case, our
+ pj_thread_desc was on the stack and went out of scope very
+ shortly after registering our thread with pjlib. With this
+ change, the pj_thread_desc is stored in thread-local storage so
+ the pointer that pjlib keeps in thread-local storage will
+ reference legitimate memory. (closes issue ASTERISK-20237)
+ reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded
+ by Mark Michelson (license #5049) Tested by Jeremy Pepper
+ ........ Merged revisions 371571 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-20 15:39 +0000 [r371535-371547] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/udptl.c: Ignore recovered zero-length secondary UDPTL
+ packets In some cases, recovering lost packets using the
+ secondary packet recovery mechanism with UDPTL/T.38 can result in
+ the recovery of zero-length packets. These must be ignored or the
+ frame generated from them can cause segfaults and allocation
+ failures. (closes issue ASTERISK-19762) (closes issue
+ ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
+ Gagnon (rgagnon) ........ Merged revisions 371544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371545 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371546 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/utils.c: Fix for commit r371535
+
+ * main/utils.c: Apply work-around for BETTER_BACKTRACES crash When
+ compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
+ crash when "core show locks" is run. This happens regularly in
+ the testsuite since several tests run "core show locks" to help
+ with debugging. This seems to be a fault with libraries on
+ certain operating systems (notably CentOS 6.2/6.3) running on
+ virtual machines and utilizing gcc 4.4.6. (issue ASTERISK-20090)
+
+2012-08-18 02:09 +0000 [r371493-371521] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/http.c: Remove old debug code from http configuration
+ loading (closes issue ASTERISK-20254) Reported by: Andrew Latham
+ Patches: http.diff uploaded by Andrew Latham (license #5985)
+ ........ Merged revisions 371520 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_xmpp.c, /: Fix typo in JabberSend that looked for '2'
+ instead of '@' in recipient argument The summary says about all
+ there is to say. (closes issue ASTERISK-20239) Reported by:
+ Gregory Porras ........ Merged revisions 371518 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * funcs/func_hangupcause.c, /: Make the name of the
+ "HangupCauseClear" application consistent The name of the
+ "HangupCauseClear" application is "HangupCauseClear", not
+ "HangupcauseClear". The incorrect case of 'cause' caused the XML
+ documentation to not register properly. As an aside, this commit
+ message felt very awkward, but I'm not sure how else to note that
+ "X", which has to be "X", was referred to as "x". (closes issue
+ ASTERISK-20253) Reported by: Andrew Latham Patches:
+ hangupcause.diff uploaded by Andrew Latham (license #5985)
+ ........ Merged revisions 371516 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * build_tools/cflags.xml, utils/utils.xml, /, res/res_fax.c,
+ sounds/sounds.xml, res/res_curl.c: Update module support level on
+ a variety of modules and compiler options Some core support
+ modules and compiler options were no longer tagged with a module
+ support level. This patch adds 'core' back to those options. Note
+ that this patch modifies a few of the patches provided by Andrew
+ Latham slightly. res_curl and res_fax are both 'core' supported
+ modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham
+ Tested by: mjordan Patches: astcanary.diff (license #5985)
+ uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded
+ by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew
+ Latham soundsxml.diff (license #5985) uploaded by Andrew Latham
+ ........ Merged revisions 371507 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/xmldoc.c: Fix memory leak in XML documentation When
+ formatting documentation fields, the XML documentation parser
+ calls xmldoc_get_formatted. This function allocates a string
+ buffer at the beginning of its routine. Unfortunately, on certain
+ code paths, it also calls xmldoc_string_cleanup, which assumes
+ that it will create the string buffer. The previously allocated
+ string buffer is then leaked by the xmldoc_string_cleanup
+ routine. Now: we don't do that. (closes issue AST-932) Reported
+ by: Alexander Homig ........ Merged revisions 371469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371491 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371492 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-17 19:50 +0000 [r371483] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: When a peer registers using WebSocket do
+ not resolve the Contact provided. (closes issue ASTERISK-20238)
+ Reported by: james.mortensen ........ Merged revisions 371482
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-17 16:01 +0000 [r371439] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, /: Add instrumentation to subsystem reloads When
+ Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
+ generate TestEvent AMI events on subsystem reloads such as cdr,
+ dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions
+ 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 371437 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371438 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-17 12:42 +0000 [r371428] Russell Bryant <russell@russellbryant.com>
+
+ * /, res/res_rtp_asterisk.c: rtp: Ensure defaults are set without
+ rtp.conf. While building up a new install to test chan_motif, I
+ ran into a failure due to icesupport being disabled. This was due
+ to me not having an rtp.conf. It was intended in the code for it
+ to be enabled by default, but it was only applied if rtp.conf
+ existed. This patch updates res_rtp_asterisk to be consistent in
+ how it handles defaults. A few options didn't have their default
+ values set globally, including icesupport. They are now set and
+ icesupport is enabled by default, even if you do not have an
+ rtp.conf. ........ Merged revisions 371425 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-17 12:25 +0000 [r371427] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_format_attr_h264.c: Add some additional H.264
+ attributes, "max-smbps" and "max-fps", for passthrough. (closes
+ issue ASTERISK-20206) Reported by: ddkprog Patches:
+ res_format_attr_h264.c.diff uploaded by ddkprog (license 6008)
+ ........ Merged revisions 371426 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-16 23:08 +0000 [r371400] Terry Wilson <twilson@digium.com>
+
+ * /, main/config.c: Handle integer over/under-flow in
+ ast_parse_args The strtol family of functions will return
+ *_MIN/*_MAX on overflow. To detect when an overflow has happened,
+ errno must be set to 0 before calling the function, then checked
+ afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
+ revisions 371392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371398 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371399 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-16 22:45 +0000 [r371396] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, /: Add module reload instrumentation for
+ TEST_FRAMEWORK This adds AMI events for module reloads when
+ Asterisk is built with TEST_FRAMEWORK enabled and corrects
+ generation of the module load AMI event. (issue PQ-1126) ........
+ Merged revisions 371393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371394 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371395 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-16 19:52 +0000 [r371356-371383] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable
+ to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING
+ flag was used instead, which will frequently flip during
+ reinvites. (closes issue AST-897) Reported by: Thomas Arimont
+ ........ Merged revisions 371357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371382 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP
+ answer is included in the SIP ACK Under certain conditions, a SIP
+ transaction involving directmedia wouldn't trigger a re-invite
+ because the SDP answer was included in an ACK instead of in a
+ message that we would have triggered the invite with. This patch
+ just queues a source change control frame if the dialog is using
+ directmedia when we find sdp for an ACK. (closes issue AST-913)
+ Reported by: Thomas Arimont ........ Merged revisions 371337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371338 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371355 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-15 23:35 +0000 [r371325] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Fix bug where final queue member would not
+ be removed from memory. If a static queue had realtime members,
+ then there could be a potential for those realtime members not to
+ be properly deleted from memory. If the queue's members were
+ loaded from realtime and then all the members were deleted from
+ the backend, then the queue would still think these members
+ existed. The reason was that there was a short- circuit in code
+ such that if there were no members found in the backend, then the
+ queue would not be updated to reflect this. Note that this only
+ affected static queues with realtime members. Realtime queues
+ with realtime members were unaffected by this issue. (closes
+ issue ASTERISK-19793) reported by Marcus Haas ........ Merged
+ revisions 371306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371313 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371324 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-15 20:43 +0000 [r371296] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Segfault When Registering SIP Over
+ WebSockets The helper function, get_address_family_filter, in
+ chan_sip for dns resolution by address family was not recognizing
+ the websockets transport and resulting in a null pointer being
+ sent to functions in netsock2, in an attempt to determine if we
+ are bound to ANY address ([::]) or not. This patch fixes this
+ issue by handling the transport types SIP_TRANSPORT_WS and
+ SIP_TRANSPORT_WSS which results in a sock address being set
+ properly for use in determining the address family. (closes issue
+ ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven
+ Beisiegel, James Mortensen Patches:
+ asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young
+ (license 5026) ........ Merged revisions 371295 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-15 20:18 +0000 [r371259-371277] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
+ relatedpeer on SIP dialog destruction The other instance of this
+ bug was fixed by jcolp/file in r121496. If we are destroying a
+ dialog only set the MWI dialog pointer on the related peer to
+ NULL if it is the dialog currently being destroyed. (closes issue
+ ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged
+ revisions 371270 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371272 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_analog.c,
+ /, channels/chan_sip.c, channels/chan_iax2.c, channels/sig_pri.c:
+ Add HANGUPCAUSE information to callee channels This adds
+ HANGUPCAUSE information to called channels so that hangup
+ handlers can, in conjunction with predial dialplan execution,
+ access the hangupcause information when the dialed channel hangs
+ up on a one-to-one basis instead of a many-to-one basis as with
+ HANGUPCAUSE usage on the caller channel. Review:
+ https://reviewboard.asterisk.org/r/2069/ (closes issue
+ ASTERISK-20198) ........ Merged revisions 371258 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-13 20:36 +0000 [r371228] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, /, apps/app_meetme.c: Add test instrumentation
+ This adds test instrumentation for loading and unloading of
+ modules and for certain actions in MeetMe to be used in the
+ testsuite or any other consumer of AMI events. These will only be
+ generated when Asterisk is built with TEST_FRAMEWORK enabled.
+ (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371203 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371227 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-13 20:02 +0000 [r371202] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix problem where incorrect pointer was
+ checked for nullity. ........ Merged revisions 371198 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371199 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371200 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-11 19:13 +0000 [r371170] Matthew Jordan <mjordan@digium.com>
+
+ * UPGRADE-11.txt (added), UPGRADE.txt: Add UPGRADE-11.txt file;
+ update UPGRADE.txt to reflect Asterisk 12
+
+2012-08-10 22:04 +0000 [r371147] Richard Mudgett <rmudgett@digium.com>
+
+ * CHANGES, /: Update CHANGES for private party ID. ........ Merged
+ revisions 371146 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-10 21:35 +0000 [r371144] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Fix a couple of documentation problems in
+ app_queue.c * The RemoveQueueMember app made mention of options
+ that could be passed in, but no options are supported. I have
+ removed the listing of options from the documentation. * The
+ RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
+ that could be set. (closes issue AST-949) reported by Steve Pitts
+ (closes issue AST-954) reported by Steve Pitts ........ Merged
+ revisions 371141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371142 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
+ revisions 371143 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2012-08-10 21:09 +0000 [r371134] Matthew Jordan <mjordan@digium.com>
+
+ * /: Remove 10 properties, add 11 properties
+
+2012-08-10 19:54 +0000 [r371120] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel_internal_api.c, main/features.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ funcs/func_callerid.c, main/cli.c, main/channel.c,
+ channels/chan_misdn.c, channels/chan_sip.c: Add private
+ representation of caller, connected and redirecting party ids.
+ This patch adds the feature "Private representation of caller,
+ connected and redirecting party ids", as previously discussed
+ with us (DATUS) and Digium. 1. Feature motivation Until now it is
+ quite difficult to modify a party number or name which can only
+ be seen by exactly one particular instantiated technology channel
+ subscriber. One example where a modified party number or name on
+ one channel is spread over several channels are supplementary
+ services like call transfer or pickup. To implement these
+ features Asterisk internally copies caller and connected ids from
+ one channel to another. Another example are extension
+ subscriptions. The monitoring entities (watchers) are notified of
+ state changes and - if desired - of party numbers or names which
+ represent the involving call parties. One major feature where a
+ private representation of party names is essentially needed, i.e.
+ where a party name shall be exclusively signaled to only one
+ particular user, is a private user-specific name resolution for
+ party numbers. A lookup in a private destination-dependent
+ telephone book shall provide party names which cannot be seen by
+ any other user at any time. 2. Feature Description This feature
+ comes along with the implementation of additional private party
+ id elements for caller id, connected id and redirecting ids
+ inside Asterisk channels. The private party id elements can be
+ read or set by the user using Asterisk dialplan functions. When a
+ technology channel is initiating a call, receives an internal
+ connected-line update event, or receives an internal redirecting
+ update event, it merges the corresponding public id with the
+ private id to create an effective party id. The effective party
+ id is then used for protocol signaling. The channel technologies
+ which initially support the private id representation with this
+ patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
+ (chan_dahdi). Once a private name or number on a channel is set
+ and (implicitly) made valid, it is generally used for any further
+ protocol signaling until it is rewritten or invalidated. To
+ simplify the invalidation of private ids all internally generated
+ connected/redirecting update events and also all
+ connected/redirecting update events which are generated by
+ technology channels -- receiving regarding protocol information -
+ automatically trigger the invalidation of private ids. If not
+ using the private party id representation feature at all, i.e. if
+ using only the 'regular' caller-id, connected and redirecting
+ related functions, the current characteristic of Asterisk is not
+ affected by the new extended functionality. 3. User interface
+ Description To grant access to the private name and number
+ representation from the Asterisk dialplan, the CALLERID,
+ CONNECTEDLINE and REDIRECTING dialplan functions are extended by
+ the following data types. The formats of these data types are
+ equal to the corresponding regular 'non-private' already existing
+ data types: CALLERID: priv-all priv-name priv-name-valid
+ priv-name-charset priv-name-pres priv-num priv-num-valid
+ priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
+ priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
+ priv-name priv-name-valid priv-name-pres priv-name-charset
+ priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
+ priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
+ REDIRECTING: priv-orig-name priv-orig-name-valid
+ priv-orig-name-pres priv-orig-name-charset priv-orig-num
+ priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
+ priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
+ priv-orig-subaddr-odd priv-orig-tag priv-from-name
+ priv-from-name-valid priv-from-name-pres priv-from-name-charset
+ priv-from-num priv-from-num-valid priv-from-num-pres
+ priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
+ priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
+ priv-to-name priv-to-name-valid priv-to-name-pres
+ priv-to-name-charset priv-to-num priv-to-num-valid
+ priv-to-num-pres priv-to-num-plan priv-to-subaddr
+ priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
+ priv-to-tag Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2030/
+
projects / thirdparty / asterisk.git / commitdiff
