summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2014-06-06PJSIP: Remove premature write of raw formatsKinsey Moore
Currently, there are situations that can occur when using chan_pjsip and certain dialplan applications (notably ChanSpy()) that can cause the channel to get no audio with scrolling warnings about format mismatches. This is caused by a failure to update translation paths on a mid-call native format update since the raw formats have already been updated by res_pjsip_sdp_rtp.c in set_caps(). Removing the premature raw format updates allows the translation paths to be setup correctly and the raw read and write formats with them. AFS-63 #close ........ Merged revisions 415342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-06Split astobj2.c into more maintainable components.George Joseph
Split astobj2.c into the following files to improve maintainability. astobj2.c - object primitives, object primitive misc and initialization code. astobj2_private.h - internal object declarations needed by the containers. astobj2_container.c - generic conainer and container misc code. astobj2_container_hash.c - hash container specific code. astobj2_container_rbtree.c - rbtree container specific code. astobj2_container_private.h - generic container definitions and rtti prototypes. https://reviewboard.asterisk.org/r/3576/ ........ Merged revisions 415317 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-06configs/cli_aliases.conf: Two new aliases, plus enhancements for context names.Rusty Newton
Changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. Added alias for "features reload" to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do "module reload features" Added alias for "pjsip reload" to the friendly template. It is shorter than "module reload res_pjsip.so" and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering "pjsip reload" is just easier. ASTERISK-23654 #close ASTERISK-23654 #comment Fixed by adding two new aliases and enhancements for context names. Review: https://reviewboard.asterisk.org/r/3572/ ........ Merged revisions 415301 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05config: Fix indentation and missing curlies in config_text_file_load().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05config: Fix config files not reloading when only an included file changes.Richard Mudgett
The twisted logic determining if a config file should be reloaded was mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed actually reenabled the broken logic. The incorrect test was causing the timestamp to always be cleared which caused config files with includes to always be reloaded. * Made wildcard includes always cause a reload. Determining if a file was deleted cannot be determined without restructuring the cache to determine if any files are missing from the last files actually loaded. Also without refactoring config_text_file_load(), the glob loop couldn't check more than one file for changes anyway. * Made remove the cache entry if the file no longer exists when trying to get its timestamp or it is no longer a regular file. This fixes the corner case where the file was loaded, then deleted, then the config reloaded, then the file restored with the same timestamp, and then the config reloaded again. * Made remove the cache entry include list when actually loading the file. This gets rid of any stale includes the file had from the last time the file was loaded. ASTERISK-23683 #close Reported by: tootai Review: https://reviewboard.asterisk.org/r/3575/ ........ Merged revisions 415225 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415229 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415230 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05res_http_websocket: Create a websocket clientKevin Harwell
Added a websocket server client in Asterisk. Asterisk has a websocket server, but not a client. The ability to have Asterisk be able to connect to a websocket server can potentially be useful for future work (for instance this could allow ARI to connect back to some external system, although more work would be needed in order to incorporate that). Also a couple of things to note - proxy connection support has not been implemented and there is limited http response code handling (basically, it is connect or not). Also added an initial new URI handling mechanism to core. Internet type URI's are parsed into a data structure that contains pointers to the various parts of the URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/3541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05app_confbridge: Allow muting of users waiting to enter a ConfBridgeMatthew Jordan
Prior to this patch, users waiting to enter a ConfBridge were not considered when muted via the CLI or via AMI. Instead, a confusing message would be emitted stating that the channel did not exist. This patch allows a user to be muted when waiting to enter a ConfBridge conference. This is equivalent to start when muted, only toggled via the CLI or AMI. Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824 #close patches: rb3582.patch uploaded by tm1000 (License 6524) ........ Merged revisions 415206 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05PJSIP: Send initial connected line informationKinsey Moore
This makes chan_pjsip send connected line information when it is called so that connected line information is available on the connected channel. (closes issue DPMA-442) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3584/ ........ Merged revisions 415191 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04safe_asterisk: Cleanup and debian compatibility.Walter Doekes
Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND option that allows the debian asterisk init script to capture the right pid. * Drop the vim #modeline which wasn't used. Use test consistently without the odd configure xno syntax. Double quote all paths. General cleanup. * Don't output message()s to the console but only to TTY if set. * Allow TTY to be "no" as well as empty (debian compatibility with debian/patches/safe_asterisk-config). * Add option to export ASTSAFE_FOREGROUND=1 from the init script that calls this to disable backgrounding. Debian uses a similar method in debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: https://reviewboard.asterisk.org/r/3574/ ........ Merged revisions 415132 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415171 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04chan_pjsip: Add debug in RTP Engine glue callbackMatthew Jordan
This patch adds some debug statements that aid with determining why a direct media request may or may not be initiated. ........ Merged revisions 415117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04res_pjsip_session: Add debug statement for session refreshesMatthew Jordan
This small patch adds a debug level 3 statement indicating how a session refresh is being sent - either as a re-INVITE or as an UPDATE - and where the session refresh is going. ........ Merged revisions 415115 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04app_confbridge: Correct verification of conference name lengthCorey Farrell
Conference names were not checked for maximum length, allowing unexpected behaviour. This change adds checking to ensure the maximum length is not exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909) confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415066 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415078 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-03func_odbc: Fix fixed size buffers fix (r414968).Walter Doekes
The change that removed the fixed size buffers in odbc-related code -- removing arbitrary column width limits -- was incomplete. This change adds: no segfault on writesql without insertsql and return value checks after strdup. While I was in the vicinity I cleaned up the linefeeds in the odbc function descriptions, moved some code for clarity, removed some blobs and noted (but didn't fix) that the 'odbc write ... exec' CLI command doesn't behave as the dialplan equivalent when insertsql= is used. ASTERISK-23582 #close Review: https://reviewboard.asterisk.org/r/3579/ ........ Merged revisions 414997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414998 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-01bridge_native_rtp: Take the bridge type choice of both channels into account.Joshua Colp
The bridge_native_rtp module currently uses the bridge result of the first channel that joins a bridge as the ultimate result. This means that if the first channel has direct media enabled but the second does not a direct media bridge will still occur. This change makes it so that both sides are taken into account. If either side forbids the bridge or responds with a local bridge result then either a generic or local bridge occurs. ASTERISK-23541 #close Reported by: Justin E Review: https://reviewboard.asterisk.org/r/3577/ ........ Merged revisions 414975 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-30PJSIP: Prevent crash on blind transferKinsey Moore
Blind transfers don't go too well with NULL channels which can occur if the channel has already been transferred away. (closes issue ASTERISK-23718) Reported by: Jonathan Rose ........ Merged revisions 414948 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-30TALK_DETECT: A channel function that raises events when talking is detectedMatthew Jordan
This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-30main/config.c: AMI action UpdateConfig EmptyCat clears all categoriesMatthew Jordan
When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk will make all categories empty in the config but the one requested with a Cat variable. This is due to a bug in ast_category_empty (main/config.c) that makes an incorrect comparison for a category name. This patch corrects the comparison such that only the requested category is cleared. Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803 #close Reported by: zvision patches: manager.c.diff uploaded by zvision (License 5755) ........ Merged revisions 414880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414881 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-29PBX: Prevent incorrect hint parsingKinsey Moore
Dynamic and pattern matching hints should not be checked for their last known state until they are instantiated by subscribers. (closes issue AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283) ........ Merged revisions 414813 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414859 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414860 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messagesMatthew Jordan
This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28pjsip.conf: privkey_file should be priv_key_file, mediaencryption=yes should ↵Rusty Newton
be mediaencryption=sdes privkey_file was missed in the snake case update. An example included an invalid value for the mediaencryption option. ........ Merged revisions 414780 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28AMI/ARI: Update version numbersMatthew Jordan
Update the semantic versioning of ARI to 1.3.0 and AMI to 2.3.0 to account for backwards compatible changes going from 12.2.0 to 12.3.0. ........ Merged revisions 414765 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28ast-db-manage/cdr/env.py: Don't fail if a config file can't be loadedMatthew Jordan
When generating SQL files via the repotools alembic_creator.py script, a configuration object is used programatically with SQLAlechemy, as opposed to a configuration file. This patch ignores failures to interpret a config file, as ... there isn't one in this case. ........ Merged revisions 414763 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28res_pjsip_session: Fix leaked video RTP ports.Richard Mudgett
Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak video RTP ports if the codec were not negotiated by an incoming call. * Made add_sdp_streams() associate the handler with the media stream if the handler handled the media stream. Otherwise, when the ast_sip_session_media object was destroyed it didn't know how to clean up the RTP resources. * Fixed sdp_requires_deferral() associating the handler with the media stream when deciding if the SDP processing needs to be deferred for T.38. Like the leaked video RTP ports, the T.38 handler needs to clean up allocated resources from deciding if SDP processing needs to be deffered. * Cleaned up some dead code in handle_incoming_sdp() and sdp_requires_deferral(). ASTERISK-23721 #close Reported by: cervajs Review: https://reviewboard.asterisk.org/r/3571/ ........ Merged revisions 414749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28app_agent_pool: Return to dialplan if the agent fails to ack the call.Richard Mudgett
Improvements to the agent pool functionality. * AgentRequest no longer hangs up the caller if the agent fails to connect with the caller. It now continues in the dialplan. * AgentRequest returns AGENT_STATUS set to NOT_CONNECTED if the agent failed to connect with the call. Most likely because the agent did not acknowledge the call in time or got disconnected. * The agent alerting play file configured by the agent.conf custom_beep option can now be disabled by setting the option to an empty string. The agent is effectively alerted to a call presence when MOH stops. * Fixed bridge reference leak when the agent connects with a caller. ASTERISK-23499 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3551/ ........ Merged revisions 414747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28res_config_odbc: Use dynamically sized buffers to store row data so values ↵Joshua Colp
do not get truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/3557/ ........ Merged revisions 414693 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414694 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414695 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28chan_unistim: Unlock mutex in rare OOM condition.Walter Doekes
#ASTERISK-23792 #close Reported by: Peter Whisker Review: https://reviewboard.asterisk.org/r/3567/ ........ Merged revisions 414677 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-27chan_sip: Start session timer at 200, not at INVITE.Walter Doekes
Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ ........ Merged revisions 414620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414628 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-27res_config_odbc: Fix old and new ast_string_field memory leaks.Walter Doekes
The ODBC realtime driver uses ^NN parameter encoding to cope with the special meaning of the semi-colon. A semi-colon in a field is interpreted as if the key was supplied twice, something which isn't otherwise possible with fixed database columns. E.g. allow=alaw;ulaw is parsed as allow=alaw and allow=ulaw. A literal semi-colon is rewritten to ^3B when stored in the database. The module uses a stringfield to efficiently store the encoded parameters. However, this stringfield wasn't always freed in some off-nominal cases. Commit r413241 fixed initialization so the encoding for INSERT and DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked apparently.) But that commit forgot the frees. This change cleans that up. Review: https://reviewboard.asterisk.org/r/3555/ ........ Merged revisions 414564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414565 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414566 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-25core_unreal: Prevent double free of core_unreal pvtMatthew Jordan
When a channel is destroyed (such as via ast_channel_release in off nominal paths in core_unreal), it will attempt to free (via ast_free) the channel tech pvt. This is problematic for a few reasons: 1. The channel tech pvt is an ao2 object in core_unreal. Free'ing the pvt directly is no good. 2. The channel tech pvt's reference count is dropped just prior to calling ast_channel_release, resulting in the pvt's destruction. Hence, the channel destructor is free'ing an invalid pointer. This patch keeps the dropping of the reference count, but sets the pvt to NULL on the channel prior to releasing it. This models what would occur if the channel was hung up directly. ........ Merged revisions 414542 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-23test_cel: Fix unit tests broken due to event def changes from res_corosyncMatthew Jordan
This patch instructs test_cel to skip any IE types it doesn't care about. The addition of the raw and bitfield types caused the tests to fail. ........ Merged revisions 414528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-23Fix signed/unsigned build warningsKinsey Moore
........ Merged revisions 414474 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22app_meetme: Don't interrupt MOH for waitmarked users.Richard Mudgett
Occasionally, when the last marked user leaves the conference, waitmarked users don't get MOH if MOH is supposed to be played while a waitmarked user is waiting for another marked user. * Made not interrupt MOH when the user is a waitmarked user. The waitmarked user doesn't need to hear any leave announcements from the conference as the user would have already heard different leave announcements if they were enabled. Apparently DAHDI occasionally sends unending non-silent streams to these users or a normal user still in the conference has continuous high background noise. These non-silent streams cause MOH to be suspended while the never ending "announcement" is played. Issue caused by ASTERISK-13680. AST-1349 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3543/ ........ Merged revisions 414401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414402 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414404 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22ARI: Add ability to raise arbitrary User EventsScott Griepentrog
User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfersJonathan Rose
PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22UPGRADE: Add note for REF_DEBUG flagMatthew Jordan
........ Merged revisions 414345 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414346 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414347 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22res_corosync: Update module to work with Stasis (and compile)Matthew Jordan
This patch fixes res_corosync such that it works with Asterisk 12. This restores the functionality that was present in previous versions of Asterisk, and ensures compatibility with those versions by restoring the binary message format needed to pass information from/to them. The following changes were made in the core to support this: * The event system has been partially restored. All event definition and event types in this patch were pulled from Asterisk 11. Previously, we had hoped that this information would live in res_corosync; however, the approach in this patch seems to be better for a few reasons: (1) Theoretically, ast_events can be used by any module as a binary representation of a Stasis message. Given the structure of an ast_event object, that information has to live in the core to be used universally. For example, defining the payload of a device state ast_event in res_corosync could result in an incompatible device state representation in another module. (2) Much of this representation already lived in the core, and was not easily extensible. (3) The code already existed. :-) * Stasis message types now have a message formatter that converts their payload to an ast_event object. * Stasis message forwarders now handle forwarding to themselves. Previously this would result in an infinite recursive call. Now, this simply creates a new forwarding object with no forwards set up (as it is the thing it is forwarding to). This is advantageous for res_corosync, as returning NULL would also imply an unrecoverable error. Returning a subscription in this case allows for easier handling of message types that are published directly to an aggregate topic that has forwarders. Review: https://reviewboard.asterisk.org/r/3486/ ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged revisions 414330 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-21core_unreal: Only block media frames when a generator is on both ends of an ↵Richard Mudgett
unreal channel. The fix for ASTERISK-12292 was a bit too aggressive. You could have generators pointed at each other on local channels but need to get other kinds of frames such as DTMF or CONNECTED_LINE frames accross. ........ Merged revisions 414269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414270 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-21pbx.c: prevent potential crash from recursive replace()Scott Griepentrog
Recurisve usage of replace() resulted in corruption of the temporary string storage and potential crash. By changing the string to be allocated separtely per instance, this is eliminated. ASTERISK-23650 #comment Reported by: Roel van Meer ASTERISK-23650 #close Review: https://reviewboard.asterisk.org/r/3539/ ........ Merged revisions 414214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414215 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-19Replace __ast_answer with ast_raw_answer in app_control_answerPaul Belanger
While load testing an ARI application, I noticed asterisk was returning HTTP 500 internal server errors on channels/:id/answer. After talking to #asterisk-dev, the issue appeared to be a lack of media flowing after __ast_answer() was called. So now, we call ast_raw_answer instead and no longer wait for media. ASTERISK-23758 #close Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged revisions 414195 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-19Undo r414123Matthew Jordan
The Test Suite caught a few problems, undoing until those are resolved git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-18bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hookMatthew Jordan
This patch fixes issues with direct media bridges that occur after a blind transfer. These issues were caught by the (currently failing) pjsip/transfers/blind_transfer/caller_direct_media test. The test currently fails primarily for two reasons: (1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself. (2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch adds a function to channel.h that allows the bridging framework to query for exactly why a channel is leaving a bridge via the channel's soft hangup flags. This allows it to only send the re-INVITE if there's a chance the channel will survive the native bridging experience. Review: https://reviewboard.asterisk.org/r/3535/ ........ Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-16chan_dahdi: Fix analog dialtone detection.Richard Mudgett
* Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to allow the DSP to operate early enough to detect dialtone. * Made use the correct variable in my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve Davies Patches: dialtone_detect_fix (license #5012) patch uploaded by Steve Davies Review: https://reviewboard.asterisk.org/r/3534/ ........ Merged revisions 414067 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414068 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-16sig_pri.c: Pull the pri_dchannel() PRI_EVENT_RING case into its own function.Richard Mudgett
* Populate the CALLERID(ani2) value (and the special CALLINGANI2 channel variable) with the ANI2 value in addition to the PRI specific ANI2 channel variable. * Made complete snapshot staging with the channel lock held. All channel snapshots need to be done while the channel lock is held. ........ Merged revisions 414050 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414051 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-15app_meetme: Fix overwrite of DAHDI conference data structure.Richard Mudgett
Starting a conference recording using the admin menu overwrites the DAHDI conference data structure used to modify the admin user's conference mute mode. * Made no longer pass the user's DAHDI conference data structure into the menu functions. The menu now uses its own DAHDI conference data structure to start the recording channel. * Moved the unlock conf->playlock to before playing the conf-full message. No sense keeping the lock while that prompt is playing. The user is never going to get into the conference at that point. ........ Merged revisions 413991 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413992 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413993 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-14res_musiconhold: Minor cleanup.Walter Doekes
Fix a few free()'s that should be ast_free()'s. Reverted an old workaround that isn't necessary. Reorder a tiny bit of code. Remove a bit of commented-out code. Review: https://reviewboard.asterisk.org/r/3536/ ........ Merged revisions 413894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413895 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'Jonathan Rose
ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ ........ Merged revisions 413876 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13h264: Fix H264 SDP payload format.Walter Doekes
https://tools.ietf.org/html/rfc3984#section-8.1 says profile-level-id takes 3 bytes in base16 (6 hex digits). This fixes video setup in certain cases. ASTERISK-23664 #close ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ ........ Merged revisions 413791 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413792 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13rtp: Fix case typo in H263+ mime.Walter Doekes
http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical mime subtype is "H263-1998", not "h263-1998". Original code was added in r183101 on 2009-03-19 02:26:50 +0100. This fixes issues with Polycom phones. ASTERISK-23665 #close ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me. Review: https://reviewboard.asterisk.org/r/3529/ ........ Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413788 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413789 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing ↵Richard Mudgett
is enabled. When overlap dialing is enabled, the lack of inband audio available information in the SETUP_ACKNOWLEDGE events causes an interoperability problem with SIP. sig_pri doesn't know if there is dialtone present when a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183 Session Progress and blocks the desired 180 Ringing message when the ALERTING message comes in. * Made the configure script detect if the installed version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * Using the new API, made generate an AST_CONTROL_PROGRESS frame on an incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio is present instead of assuming that dialtone is present. * Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio available indication only if dialtone is expected. The change also makes the fallback behaviour of sending the PROGRESS message better by sending it only if dialtone is expected. * Changed receiving a PROCEEDING message to not generate an AST_CONTROL_PROGRESS frame if the progress indication ie indicates non-end-to-end-ISDN. This helps interoperability with SIP. * Changed sending a PROCEEDING message in response to an AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It was silly to do so anyway because the channel driver doesn't know if inband audio is even available. This helps interoperability with SIP. This patch and a corresponding change in libpri work together to allow Asterisk to control the inband audio available progress indication ie on the SETUP_ACKNOWLEDGE message when dialtone is present. AST-1338 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ ........ Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13Fix compiler warning from GCC 4.10 fixup.Richard Mudgett
........ Merged revisions 413766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-10 21:46:30 +0000