2013-10-08 Asterisk Development Team * Asterisk 12.0.0-beta1 Released. 2013-10-08 22:58 +0000 [r400771] Matthew Jordan * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * channels/iax2/include/parser.h: chan_iax2: Fix compile error. 2013-10-04 21:40 +0000 [r400567] Michael L. Young * 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 * rest-api/api-docs/applications.json (added): Added missing file from r400522 2013-10-04 18:42 +0000 [r400532-400542] Jonathan Rose * 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 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 : 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * main/cel.c: cel: Some whitespace cleanups 2013-10-03 18:28 +0000 [r400384-400395] Kinsey Moore * 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 * 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 * channels/chan_vpb.cc: chan_vpb: Make compile again. 2013-10-03 14:56 +0000 [r400362] Mark Michelson * tests/test_cel.c: Get rid of uses of stasis_topic_wait() 2013-10-03 14:51 +0000 [r400360] Joshua Colp * 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 * 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 * 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 * 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 * 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 * rest-api/api-docs/asterisk.json: Correct allowable values for ARI general information filter 2013-10-02 18:57 +0000 [r400286] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_pjsip.c: Broke the build - Fixing XML DTD violation added in r399782, missing tags inside a 2013-09-25 19:28 +0000 [r399796] Michael L. Young * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * main/bridge.c: CLI bridge: Fix "bridge destroy " and "bridge kick " 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * /: Unblock r398558 2013-09-06 20:20 +0000 [r398533-398572] Richard Mudgett * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * Makefile: Fixed 'make clean' for wiki docs 2013-09-03 14:27 +0000 [r398196] Walter Doekes * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * / (added): __________ | \ |_______ | | | ______| | / | _ _ _ _ _ | _______| / \ ___| |_ ___ _ __(_)___| | __ / || | / _ \ / __| __/ _ \ '__| / __| |/ / | || |_______ / ___ \__ \| | __/ | | \__ \ < | || | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| \__________| 2013-08-23 22:20 +0000 [r397613] Joshua Colp * 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 * 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 * res/res_pjsip.c: Add some clarifying documentation to the rewrite_contact endpoint option. 2013-08-23 21:51 +0000 [r397602] Richard Mudgett * main/bridge_channel.c: Blank line tweaks. 2013-08-23 21:49 +0000 [r397599-397600] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configs/cel.conf.sample: Update CEL sample config 2013-08-23 00:26 +0000 [r397505] Jonathan Rose * 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 * 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 * 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 * 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 * res/res_pjsip.c: Add missing configOption close tags 2013-08-22 21:29 +0000 [r397482] Richard Mudgett * include/asterisk/musiconhold.h: Update MOH start/stop routine doxygen. 2013-08-22 21:21 +0000 [r397481] Rusty Newton * 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 * 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 * 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 * apps/app_queue.c: Remove set but unused variable 'meid'. 2013-08-22 19:52 +0000 [r397461] Kinsey Moore * main/cel.c: Fix crash when getting CEL config 2013-08-22 18:52 +0000 [r397441-397451] Mark Michelson * 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 * 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 * 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 * main/bridge.c: Update BUGBUG comment. 2013-08-22 12:28 +0000 [r397379-397415] Walter Doekes * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /: 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 * /, 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 * 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 * 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 * main/bridge.c: Update BUGBUG comment. 2013-08-19 14:54 +0000 [r396923] Jonathan Rose * 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 * 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 * 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 * 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 * 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 * include/asterisk/bridge.h, main/bridge.c, include/asterisk/bridge_channel.h, main/bridge_channel.c: Fix CLI "bridge kick " 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 " 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 * 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 * 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 ". 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * bridges/bridge_native_rtp.c: bridge_native_rtp: Remove some unnecessary NULL checks on c1. 2013-08-09 20:29 +0000 [r396505] Walter Doekes * 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 * 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 * 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 * tests/test_stasis.c: Fix stasis/core unit test. Should have had the CR/LF. 2013-08-08 22:09 +0000 [r396474] Tzafrir Cohen * channels/chan_dahdi.c: chan_dahdi: create channels at run-time This code adds chan_dahdi the command 'dahdi create channels ' (where is a single - 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/sig_pri.c: And get rid of another ast_bridged_channel() 2013-08-02 17:29 +0000 [r396136-396143] David M. Lee * 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 * res/snmp/agent.c: Get the SNMP code to compile. 2013-08-02 14:46 +0000 [r396119-396125] David M. Lee * 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 * 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 * 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 * 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 * 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 * channels/sig_analog.c: Fixed chan_dahdi compilation failure 2013-08-02 03:12 +0000 [r396060-396062] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_pjsip/pjsip_options.c: Found another missed "sip" -> "pjsip" CLI command. 2013-07-31 15:27 +0000 [r395881] Kinsey Moore * tests/test_cel.c: Disable CEL tests that need rearchitecting to operate properly 2013-07-31 14:45 +0000 [r395868] Mark Michelson * 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 * 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 * 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 * 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 * res/res_pjsip: Setting svn:ignore for res/res_pjsip 2013-07-30 19:10 +0000 [r395748-395779] Mark Michelson * 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 * 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 * res/res_pktccops.c: Fix compilation on gcc 4.8.1 2013-07-29 17:51 +0000 [r395686] David M. Lee * 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 * 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 * 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 * 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 * res/ari: Set svn:ignore in res/ari directory 2013-07-29 12:10 +0000 [r395619] Kinsey Moore * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/bridging.c: Let the compiler do more type checking with bridge hook callbacks. 2013-07-23 22:32 +0000 [r395227] Joshua Colp * 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 * 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 * 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 * 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 * channels/chan_gulp.c: Drop the reference count on the correct object. 2013-07-23 18:41 +0000 [r395154-395182] Richard Mudgett * 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 * 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 * 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 * 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'. 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 * 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 * 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 * res/res_sip/sip_options.c: Add missing newline 2013-07-23 12:27 +0000 [r395102] Joshua Colp * 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 * 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 * main/bridging.c: Remove some BUGBUG notes that have been handled. 2013-07-22 20:42 +0000 [r395074] Kinsey Moore * tests/test_cel.c: Make the CEL blind transfer test pass consistently 2013-07-22 13:52 +0000 [r394881-395034] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_http_websocket.c: Fixed null dereference when WebSocket subprotocol isn't specified 2013-07-18 16:49 +0000 [r394731] Jonathan Rose * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 ' 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 * 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 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 * res/stasis_http/ari_websockets.c: Debug logging to help with WebSocket connection problems 2013-07-16 20:00 +0000 [r394489] Richard Mudgett * 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 * 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 * 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 * 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 * res/res_http_websocket.c: Fixed null dereference when WebSocket protocol is omitted 2013-07-15 23:20 +0000 [r394417] Richard Mudgett * 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 * include/asterisk/stasis_channels.h: Remove misleading documentation for channel snapshot creation. 2013-07-15 21:22 +0000 [r394397] David M. Lee * res/res_stasis_http.c: Document the ari.conf allowed_origins setting 2013-07-15 13:43 +0000 [r394370] Joshua Colp * 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 * 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 * 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 * 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 * 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 * channels/chan_skinny.c: Fix a compiler warning. 2013-07-12 18:23 +0000 [r394203] David M. Lee * 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 * 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 * include/asterisk/bridging_technology.h: Fix bridge tech write callback parameter name. 2013-07-11 20:59 +0000 [r394156] David M. Lee * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * main/xmldoc.c: Fix printf NULL string (null) substituion for NULL config framework default. 2013-07-09 20:07 +0000 [r393897] Mark Michelson * 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 * include/asterisk/stasis_channels.h, include/asterisk/manager.h: Fix some stasis doxygen comments. 2013-07-09 11:05 +0000 [r393857-393870] Joshua Colp * 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 * 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 * 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 * 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 * 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 * res/stasis/control.c: Fix building. 2013-07-08 14:46 +0000 [r393804-393806] Jason Parker * 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 * 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 * 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 * 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 * 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 * 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 * main/bridging.c: OneTouchRecord: Add function defined earlier: ast_bridge_features_do() 2013-07-05 03:08 +0000 [r393716] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_sip_notify.c: Add pjproject dependency to res_sip_notify 2013-07-02 18:28 +0000 [r393463] Mark Michelson * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/stasis_http/resource_channels.c: Change some 500 errors to 400. 2013-06-28 02:14 +0000 [r393083-393100] David M. Lee * 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 * 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 * 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 * 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 * 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 * 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 * Makefile: Fix typo with XML docs. 2013-06-25 19:22 +0000 [r392864-392879] Joshua Colp * 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * rest-api/api-docs/channels.json: Fix typo. 2013-06-21 18:10 +0000 [r392437] Richard Mudgett * main/bridging.c: Add channel optimization interaction with frame hooks BUGBUG comments. 2013-06-21 18:05 +0000 [r392436] Mark Michelson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * include/asterisk/app.h: Fix a build warning with stasis messages. 2013-06-17 14:40 +0000 [r392004-392005] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * main/cel.c: Ensure that Asterisk still starts up when cel.conf is missing 2013-06-13 18:17 +0000 [r391676] Mark Michelson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_agi.c: Fix documentation generation Regression from r390701 2013-06-06 20:32 +0000 [r390729] Jason Parker * /: Remove props that people will yell at me for. I'm sorry I broke automerge. :( 2013-06-06 20:30 +0000 [r390728] Kinsey Moore * 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 * 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 * main/bridging.c: Parking: Enable code responsible for intercepting park exten transfers 2013-06-06 01:52 +0000 [r390612-390639] Richard Mudgett * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/channel.c: Remove unused RAII vars. 2013-05-29 03:22 +0000 [r389990] Matthew Jordan * 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 * 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 * /, 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/features.c: Fix inverted test preventing DTMF disconnect from working. 2013-05-23 18:39 +0000 [r389551-389568] Joshua Colp * 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 * 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 * /: 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 * main/stasis_bridging.c: Fix destruction order assert for stasis_bridging 2013-05-21 21:08 +0000 [r389426] Richard Mudgett * 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 * 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 2013-05-10 11:47 +0000 [r388254] Sean Bright * /, 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 * 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 * 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 * 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 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 * 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 * 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 * utils: Add version.c to list of ignored files in the utils directory. 2013-05-08 13:39 +0000 [r387932] David M. Lee * 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 * /, 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 * 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 * include/asterisk.h: Fix build breakage, from LOW_MEMORY fix. 2013-05-06 17:15 +0000 [r387740-387741] Richard Mudgett * 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 * main/asterisk.c: Fix building with LOW_MEMORY defined. 2013-05-06 15:58 +0000 [r387690] Russell Bryant * /, 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 * 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 * 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 * 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 * 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 * 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 * main/channel.c: Remove the ABI compatability ast_channel_alloc(). It is no longer needed. 2013-05-02 17:15 +0000 [r387423] Matthew Jordan * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * channels/chan_dahdi.c: Move some annoying chan_dahdi debug messages to level 5. 2013-04-30 22:50 +0000 [r387039] Matthew Jordan * 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 * 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 * channels/chan_gulp.c: Fix a log message. 2013-04-30 13:48 +0000 [r386931] Sean Bright * 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 * 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 * 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 * 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 * rest-api-templates/swagger_model.py: Fix spelling error in python doc 2013-04-27 19:03 +0000 [r386731-386760] Joshua Colp * 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 * 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 * 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 * /, 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 * main/loader.c: Removing stray printf from r386540 2013-04-26 20:32 +0000 [r386638] Mark Michelson * 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 * 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 * 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 * /: 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 2013-04-16 14:09 +0000 [r385835] David M. Lee * include/asterisk/stasis_channels.h: Fixed a typo 2013-04-15 17:26 +0000 [r385782] Jason Parker * 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 * 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 * 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 * /, 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 * /, 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 * 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 * include/asterisk/sorcery.h: Fix documentation. 2013-04-12 21:11 +0000 [r385522] Kinsey Moore * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * main/sorcery.c: Pass the object type name to the configuration framework. 2013-04-02 11:40 +0000 [r384514] Matthew Jordan * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * main/manager_channels.c: Fix typo * main/stasis.c: Fix missing ' ' around '=' 2013-03-25 19:28 +0000 [r383726-383747] David M. Lee * 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 * 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 * main/json.c, main/Makefile: Fixed another issue from r383579. Core modules don't honor flags in MODULEINFO, which broke jansson if specified --with-jansson to configure. 2013-03-22 20:43 +0000 [r383632] Michael L. Young * /, 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 * 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 * 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 * 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 * include/asterisk/astobj2.h: Fix astobj2 doxygen comment. 2013-03-20 20:27 +0000 [r383458-383462] Walter Doekes * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * main/threadpool.c: Changing log level of "Not changing threadpool size" from notice to debug. 2013-03-07 21:14 +0000 [r382636] Jason Parker * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * apps/app_confbridge.c: confbridge: Add flags column to CLI "confbridge list " * Added the following flags to the CLI "confbridge list " 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 * apps/app_confbridge.c: confbridge: Rename i iterator variables to iter. 2013-02-16 16:28 +0000 [r381615] Matthew Jordan * /, 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * /: 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 2013-01-23 00:30 +0000 [r379966] Richard Mudgett * 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 * /, 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 * 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 * /, 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 * /: Remove stray property. 2013-01-22 15:16 +0000 [r379828-379830] Matthew Jordan * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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. 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /: Fixed svn merge property breakage from r377986 2012-12-13 18:28 +0000 [r377986] Damien Wedhorn * /, 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 * 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 * 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 * main/taskprocessor.c: Re-add taskprocessor cleanup code that was removed by the UUID merge. 2012-12-13 15:37 +0000 [r377974] Sean Bright * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /: And remove svnmerge-integrated property. * /: Remove automerge properties. 2012-12-11 21:22 +0000 [r377867] Richard Mudgett * 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /: 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * res/res_rtp_asterisk.c: Formatting fixes 2012-12-03 14:56 +0000 [r377022] Joshua Colp * /, 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 " * 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 * /, 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * /, channels/chan_sip.c: Be consistent, send From: "Anonymous" When setting CALLERID(pres)=unavailable in the dialplan, the From header in the SIP message contains "Anonymous" . 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 " and "astobj2 container check ". 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * /, 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 * /, 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * 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 * /, 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 * /: Remove 10 properties, add 11 properties 2012-08-10 19:54 +0000 [r371120] Richard Mudgett * 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/