24837 Commits (57967ec04a9cc7eb4e6bca2170a5f5a132d57e73)

 

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Author	SHA1	Message	Date
Matthew Jordan	57967ec04a	res_pjsip_sdp_rtp: Fix use of uninitialized value in PJSIP ... In PJMEDIA, pjmedia_sdp_rtpmap_to_attr will attempt to use the string rtpmap.param regardless of its length value. Simply setting the length to 0 does not prevent the garbage on the stack in rtpmap.param.ptr from being formatted in a sprintf call. This patch initializes the string to NULL so that at the very least, something is provided to the function that is predictable. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ef01e90ecf	res_pjsip_mwi: Fix memory leak of MWI subscriptions container ... This patch fixes a reference counting memory leak on the ao2_container created as part of create_mwi_subscriptions. When we create the container in this routine, the intent is to hand lifetime ownership over to the global container unsolicited_mwi. When ao2_global_obj_replace_unref is called, the reference count on mwi_subscriptions (the container) will be bumped by 1; however, the function does not decrement the reference count on mwi_subscriptions when this occurs. This will prevent the container from being fully disposed of when Asterisk exits (or on any subsequent call to this operation, such as during a reload). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4fd244e273	ari: Add silence generator controls ... This patch adds the ability to start a silence generator on a channel via ARI. This generator will play silence on the channel (avoiding audio timeouts on the peer) until it is stopped, or some other media operation is started (like playing media, starting music on hold, etc.). (closes issue ASTERISK-22514) Review: https://reviewboard.asterisk.org/r/3019/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	527a8b08de	res_pjsip_caller_id: Don't overwrite user portion of the From header when fromuser is set. ... The fromuser option is used to explicitly set the user within the From header. The res_pjsip_caller_id module did not take this setting into account when determining if the From header could be modified or not. (closes issue ASTERISK-22866) Reported by: Anthony Messina git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	18a2f38b76	res_pjsip: Add support for building against pjproject with SIP transaction group lock support. ... SIP transaction group lock support has been backported into our pjproject. Since the code now internally uses a group lock the code is now changed to unlock it if present. Note that the act of finding the transaction is what actually returns it locked. For further information about group locks check out the wiki page at: http://trac.pjsip.org/repos/wiki/Group_Lock (issue ASTERISK-22818) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b72cc1c96f	CEL: Fix crash when using CELGenUserEvent ... This fixes a crash when CELGenUserEvent is called from the dialplan while CEL is disabled. Currently, CEL does not create its topics and forwards if it is not enabled and external entities may depend on these topics blindly since they should always be available. This patch breaks up route creation and topic/forward creation such that the CEL topics and forwards will always exist while the router and its associated routes will be torn down and recreated as necessary. (closes issue ASTERISK-22799) Review: https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	13ce1cee50	stasis: Fixed scoping problem with bridge tracking. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0f7db6c413	res_ari_channels: Add the ability to stop locally generated ringing on a channel. ... Using the 'ring' operation it is possible to start locally generated ringback if the channel is answered. This change adds the ability to stop it by using DELETE. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	129cd55355	ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404 ... Was returning a 404 on a valid technology with an empty list of endpoints. Now checking against the channel tech to make sure the tech itself is valid and not just an empty list of endpoints. (issue ASTERISK-22803) Reported by: David M. Lee git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	59dc668622	ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404 ... Implementation listing endpoints by technology returned an empty array if no matching endpoints were found. Fixed so a "404 Not Found" will be returned instead. (closes issue ASTERISK-22803) Reported by: David M. Lee git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	68d09e0bf1	Switch to a scoped lock to avoid missing unlocks in failure returns. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	15feef01a0	Move a NULL check to a place that makes more sense. ... Two variables were being checked for NULLity immediately after being declared NULL. I moved the NULL check until after the variables are allocated. This allows for the "channelvars" option in manager.conf to work as intended again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1d6201ab9c	pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences ... Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to crash because they were trying to dereference a NULL pointer. In the case of res_pjsip_messaging it was attempting to "print" a contact header that did not exist. In fact contact headers should not be part of a SIP MESSAGE, so the offending code was simply removed. In the case of res_pjsip_header_funcs a null private channel tech was being passed to the function and then later dereferenced. Added null checks (and error logging) to the read/write function handlers to guard against crashing. (closes issue ASTERISK-22821) Reported by: Anthony Messina git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	22801822bc	CELGenUserEvent: Fix error message from ast_json_pack ... This prevents NULL from being passed into an ast_json_pack call when no extra information is passed to the application which prevents an error message about NULL arguments from being generated. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1432a9d59d	Fixed a typ. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	192732b83d	chan_dahdi: Fix crash during caller ID read ... Asterisk will sometimes core dump during caller id read on analog channels due to a negative return value from the read() in my_get_callerid that slips through as a negative length argument to callerid_feed() if the errno returned by DAHDI is ELAST. This change ensures that the negative return is treated properly even when it is ELAST. (closes issue ASTERISK-22746) Reported by: Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502) ........ Merged revisions 402708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402709 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8d1527fd7a	Get rid of some inaccurate comments. ... I'm doing some unrelated work in app_confbridge and finding these "invalid pin" comments to be annoying. Get out! ........ Merged revisions 402686 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	33a9e71b7e	app_queue: Honor penalty limits of 0 ... In the current app_queue code from 1.8 up to trunk the upper and lower penalties can be set to 0 but the value is interpreted to be disabled instead of actually setting limits. This is especially evident if min and max limits are set to 0 and members with penalties of 0 and 1 are in the queue since the member with penalty 1 will still receive calls. This patch adjusts the special disabled value to be INT_MAX instead of 0. (closes issue ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/ Reported by: Schmooze Com ........ Merged revisions 402645 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402646 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	96048afd61	chan_sip: keep same local (from) tag for outgoing register requests ... For outbound register requests the tag on the From line was updated every 20 seconds prior to a successful registration and also once for each registration renewal. That behavior can possibly cause the registration to be denied because of the different tag, and is not aligned with the intention of RFC 3261 8.1.3.5 "... request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request...". This updates chan_sip to have a field to keep the local tag in the registration structure and use that tag for registration requests where the callid is also unchanged. (closes issue ASTERISK-12117) Reported by: Pawel Pierscionek Review: https://reviewboard.asterisk.org/r/2988/ ........ Merged revisions 402604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402605 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	82c2db6592	res_stasis.c: Fix locking issues with the app_bridge_moh container. ... * Fix unlinking from the app_bridges_moh container in remove_bridge_moh() without a lock under normal circumstances. * Made check ast_bridge_set_after_callback() return value in bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK() locking over too much scope in stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop(). * Fixed unusual usage of ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge from off nominal path in stasis_app_bridge_create(). * Fixed strange construct in stasis_app_unsubscribe(). From a bad merge? * Made load_module() cleanup on failure. Review: https://reviewboard.asterisk.org/r/2962/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	92628d1654	security_events: Push out security events over AMI events ... Security Events will now be written to any listener of the new 'security' class Review: https://reviewboard.asterisk.org/r/2998/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	6ced38003d	Clarify an ambiguous error message. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7d1bfe62b6	res_pjsip: Print a helpful error message if sorcery registration fails ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	13c51ecc13	Changes from make ari-stubs after r402560 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	952cd73827	ARI playback: Rename ARI Playback to Playbacks ... Before playback was the only non plural resource. It has been renamed to playbacks for consistency. (closes issue ASTERISK-22737) Reported by: Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a274df3a21	ari: Add application/x-www-form-urlencoded parameter support ... ARI POST calls only accept parameters via the URL's query string. While this works, it's atypical for HTTP API's in general, and specifically frowned upon with RESTful API's. This patch adds parsing for application/x-www-form-urlencoded request bodies if they are sent in with the request. Any variables parsed this way are prepended to the variable list supplied by the query string. (closes issue ASTERISK-22743) Review: https://reviewboard.asterisk.org/r/2986/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	7194a6e59c	PJSIP: Improve error handling in digest authenticator ... Previously, regardless of whether failure to authenticate was due to lacking any authentication or actually failing authentication, the Digest Authenticator would simply return that a challenge was still needed. It will continue to do that when no authentication information is in the received SIP digest, but when authentication information is present and does not pass authentication, that will be treated as an authentication error. This is to ensure that PJSIP will issue security events indicated failed auths. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2f92695c22	ari: User better nicknames for ARI operations ... While working on building client libraries from the Swagger API, I noticed a problem with the nicknames. channel.deleteChannel() channel.answerChannel() channel.muteChannel() Etc. We put the object name in the nickname (since we were generating C code), but it makes OO generators redundant. This patch makes the nicknames more OO friendly. This resulted in a lot of name changing within the res_ari_*.so modules, but not much else. There were a couple of other fixed I made in the process. * When reversible operations (POST /hold, POST /unhold) were made more RESTful (POST /hold, DELETE /unhold), the path for the second operation was left in the API declaration. This worked, but really the two operations should have been on the same API. * The POST /unmute operation had still not been REST-ified. Review: https://reviewboard.asterisk.org/r/2940/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	e327b401da	app_queue: crash if first agent is "busy" ... If the first agent/member (via CLI "queue show") in a queue is "busy" (dnd, circuit busy, etc...) and no agents answered then app_queue would crash. This occurred because while the calling of agent(s) remained valid the channel on "busy" agent would be set to NULL and then later dereferenced upon a second "rna" function call. The original intention of the code is to have only valid "call attempt" objects (channels != NULL) checked while attempting to call agent(s). It does this by building a "call_next" list of valid "call attempt" objects. In the case of the "busy" agent subsequent builds of the valid "call attempt" list would sometimes include (the case mentioned above) an invalid "call attempt" object. The fix was to make sure the "call attempt" list was appropriately built on every iteration. A NULL sanity check was also added at the original offending spot of the crash just in case another one slipped by somehow. (closes issue ASTERISK-22644) Reported by: Marco Signorini Review: https://reviewboard.asterisk.org/r/2983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	0d66d14d4d	chan_sip: Use AST_AF* defined constant when calling ast_get_ip ... While the structure passed to ast_get_ip should be set memset to 0, thus initializing the ss_family member to 0, explicitly setting it to AST_AF_UNSPEC is more portable. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3a6b76c69e	chan_iax2: Fix incorrect usage of ast_get_ip involving uninitialized struct ... This started off as a fix for the failing IAX2 acl_call test in the Asterisk Test Suite. When inspecting why that test was failing, it became clear that all attempts to bind to any local loopback address was failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787] netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28] DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", "(null)", ...): ai_family not supported [Nov 2 15:56:28] WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's conceivably other ways for getaddrino to return EAI_FAMILY, the most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not provided as the desired family. The culprit was the call to ast_get_ip, defined in acl.h. This function uses the family from the passed in addr object (which it will also populate when it returns!) when it eventually calls getaddrinfo. This patch fixes the use of ast_get_ip that were not specifying the family in chan_iax2. This prevents uninitialized use of the structure, so that the addresses resolve correctly. Review: https://reviewboard.asterisk.org/r/2991 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	135ac61bc5	netsock2: Define AST_AF_* enum constants to their AF_* equivalents ... This patch explicitly defines AST_AF_* enum constants to their sys/socket.h defined equivalents. It is certainly unclear why these constants actually have to exist, given that netsock2.h includes sys/socket.h; however, since the code base is already liberally sprinkled with the usage of AST_AF_* (as well as with direct calls to AF_*), this will at least keep the semantics consistent between their usage across systems. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6fa219804d	stasis_channels: Don't give preference to ANI info in channel snapshots ... When publishing channel snapshots, we currently compute the caller ID name and number by giving preference first to ani.{name|number}, then to id.{name|number}. However, when a channel driver (such as chan_sip) updates the caller ID, it typically only updates the caller ID stored in id.{name|number}. This means that we are currently giving preference to stale information. When looking at the rest of the code base, the only other place where we appear to use this same logic is in app_amd. Everywhere else, we treat the party information in ani as being separate to the party information in id. This patch publishes only the caller ID name and number in the snapshot field for caller_name and caller_num. Note that the information in ANI is still available in caller_ani. Review: https://reviewboard.asterisk.org/r/2992/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	89e6640ad8	chan_sip: notify dialog info ignores presentation indicator in callerid ... The presentation indicator in a callerid (e.g. set by dialplan function Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies are generated during extension monitoring. Added a check to make sure the name and/or number presentations on the callee (remote identity) are set to allow. If they are restricted then "anonymous" is used instead. (closes issue AST-1175) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2976/ ........ Merged revisions 402450 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c981cad0b8	vector: Uppercase API to follow C convention. ... C does not support templates like C++. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0bc642bc38	vector: Update API to be more flexible. ... Made the vector macro API be more like linked lists. 1) Added a name parameter to ast_vector() to name the vector struct. 2) Made the API take a pointer to the vector struct instead of the struct itself. 3) Added an element cleanup macro/function parameter when removing an element from the vector for ast_vector_remove_cmp_unordered() and ast_vector_remove_elem_unordered(). 4) Added ast_vector_get_addr() in case the vector element is not a simple pointer. * Converted an inline vector usage in stasis_message_router to use the vector API. It needed the API improvements so it could be converted. * Fixed topic reference leak in router_dtor() when the stasis_message_router is destroyed. * Fixed deadlock potential in stasis_forward_all() and stasis_forward_cancel(). Locking two topics at the same time requires deadlock avoidance. * Made internal_stasis_subscribe() tolerant of a NULL topic. * Made stasis_message_router_add(), stasis_message_router_add_cache_update(), stasis_message_router_remove(), and stasis_message_router_remove_cache_update() tolerant of a NULL message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as intended in dispatch_message(). Review: https://reviewboard.asterisk.org/r/2903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	55da2b319d	confbridge: Separate user muting from system muting overrides. ... The system overrides the user muting requests when MOH is playing or a waitmarked user is waiting for a marked user to join. System muting overrides interfere with what the user may wish the muting to be when the system override ends. * User muting requests are now independent of the system muting overrides. The effective muting is now the logical or of the user request and system override. * Added a Muted flag to the CLI "confbridge list <conference>" command. * Added a Muted header to the AMI ConfbridgeList action ConfbridgeList event. (closes issue AST-1102) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........ Merged revisions 402425 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ff9809c0a8	config: Allow ConfBridge DTMF menus to have '#' as the first digit. ... ConfBridge allows custom DTMF menus to be created in the confbridge.conf file by assigning a DTMF key sequence to a sequence of actions as follows: DTMF-sequence = action,action... Unfortunately, the normal config file processing code interprets an initial '#' character as starting a directive such as #include. * Add the ability to escape the first non-blank character in a config line so the '#' character can be used without triggering the directive processing code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified) Review: https://reviewboard.asterisk.org/r/2969/ ........ Merged revisions 402407 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	05a838e7ab	voicemail: Simplify callback pointer declarations and add doxygen. ... * Typedefed and added doxegen for the voicemail callback functions. * Simplified the prototypes for ast_install_vm_functions() and ast_install_vm_test_functions() to use the new function typedefs. * Simplified the voicemail callback function pointer variable declarations to use the new function typedefs. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	e55d4418cc	Manager: Add equivalent AMI actions for the bridge CLI commands. ... Adds the following AMI events, closely following their CLI counterparts: BridgeDestroy BridgeKick BridgeTechnologyList BridgeTechnologySuspend BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one channel off the bridge. When kicking a channel, specifying the bridge also (optional) insures it is not removed from the wrong bridge. The BridgeTechnology events allow viewing and changing suspension status, which affects only subsequent not active bridging. (closes ASTERISK-22356) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/2973/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	11fa5ca2c0	ari wiki docs: add notes about allowMultiple parameters. ... This patch adds a note to any parameter that has 'allowMultiple' set in the Swagger documentation. (closes issue ASTERISK-22704) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	26458aa615	res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media. ... The ring operation sends ringing to the specified channel it is invoked on. The dtmf operation can be used to send DTMF digits to the specified channel of a specific length with a wait time in between. Finally hangup reasons allow you to specify why a channel is being hung up (busy, congestion). Early media behavior has also been tweaked slightly. When playing media to a channel it will no longer automatically answer. If it has not been answered a progress indication is sent instead. (closes issue ASTERISK-22701) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2916/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	de407196b8	chan_sip: Fix RTCP port for SRFLX ICE candidates ... This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ ........ Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6a6332b86d	res_ari_channels: Fix a deadlock when originating multiple channels close to eachother. ... If a Stasis application is specified an implicit subscription is done on the originated channel. This was previously done with the channel lock held which is dangerous as the underlying code locks the container and iterates items. This change releases the lock on the originated channel before subscribing occurs. (closes issue ASTERISK-22768) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	adf78c11ea	res_stasis: Ensure the channel is always departed from the bridge when it leaves. ... This change adds a command to the command queue to explicitly depart the channel from the bridge when it is told it has left. If the channel has already been departed or has entered a different bridge this command will become a no-op. (closes issue ASTERISK-22703) Reported by: John Bigelow (closes issue ASTERISK-22634) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2965/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1ad06e7772	Update the conversion script from sip.conf to pjsip.conf ... (closes issue ASTERISK-22374) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2846 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6a660c9eeb	core/loader: Don't call dlclose in a while loop ... For awhile now, we've noticed continuous integration builds hanging on CentOS 6 64-bit build agents. After resolving a number of problems with symbols, strange locks, and other shenanigans, the problem has persisted. In all cases, gdb shows the Asterisk process stuck in loader.c on one of the infinite while loops that calls dlclose repeatedly until success. The documentation of dlclose states that it returns 0 on success; any other value on error. It does not state that repeatedly calling it will eventually clear those errors. Most likely, the repeated calls to dlclose was to force a close by exhausting the references on the library; however, that will never succeed if: (a) There is some fundamental error at work in the loaded library that precludes unloading it (b) Some other loaded module is referencing a symbol in the currently loaded module This results in Asterisk sitting forever. Since we have matching pairs of dlopen/dlclose, this patch opts to only call dlclose once, and log out as an ERROR if dlclose fails to return success. If nothing else, this might help to determine why on the CentOS 6 64-bit build agent things are not closing successfully. Review: https://reviewboard.asterisk.org/r/2970 ........ Merged revisions 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402288 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f8cd9b61f6	medix_index: Display errors when library calls fail ... Based on feedback from ipengineer in #asterisk, when the media indexer cannot access a sound file on the system (or otherwise fails) Asterisk displays a "Cannot frob file" error but fails to tell you why. This is especially problematic as the media_indexer failing will rpevent Asterisk from starting, as it is in the core. We now display the errno error messages so folks can figure out what they've done wrong. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a181e534e0	stasis: add functions embarrassingly missing from r400522 ... I neglected to implement two of the endpoint subscription functions when I did the work. Normally, you'll only hit that when you unsubscribe from a specific endpoint. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	8a08f73fe0	pjsip_messaging: Added debug for in dialog messaging ... (issue ASTERISK-22777) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago