asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	ad0e70ba83	Say: If SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF Similar to how background works, if a say application is called with this variable set to 'true', 'yes', 'on', etc. then using DTMF while the say action is in progress will result in the channel jumping to that extension in the dialplan. Review: https://reviewboard.asterisk.org/r/3011/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	67b650543c	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. ........ Merged revisions 402804 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f6593b4156	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 ........ Merged revisions 402793 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4be01b4598	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 ........ Merged revisions 402787 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	94f19c8218	Switch to a scoped lock to avoid missing unlocks in failure returns. ........ Merged revisions 402769 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c0bc3f6b4c	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. ........ Merged revisions 402767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	12a0edac69	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 ........ Merged revisions 402757 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4f61528fba	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. ........ Merged revisions 402755 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	5701c51bb6	Fixed a typ. ........ Merged revisions 402738 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	95bc7626aa	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 ........ Merged revisions 402710 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d720bc7686	Confbridge: add test events for dynamic menus test Adds a couple of test events for conference menu actions so that it's easy to discern when those menu actions have been triggered. (issue ASTERISK-22760) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2999/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	92cf776119	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 ........ Merged revisions 402687 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	483d127d55	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 ........ Merged revisions 402647 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	094db82a73	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 ........ Merged revisions 402606 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7323809607	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/ ........ Merged revisions 402593 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bf5492abd2	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/ ........ Merged revisions 402584 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	518f091a1a	Clarify an ambiguous error message. ........ Merged revisions 402582 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4c128198c8	res_pjsip: Print a helpful error message if sorcery registration fails ........ Merged revisions 402570 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b83a3965b8	Changes from make ari-stubs after r402560 ........ Merged revisions 402561 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4f1bdeed1c	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 ........ Merged revisions 402560 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	97a8debd90	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/ ........ Merged revisions 402555 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2564ed26f7	app_dahdiras: Use waitpid instead of wait4. Several places in the code were using wait4 while other places were using waitpid. This change makes all places use waitpid in order to make things more consistent and since the 'rusage' object passed in/out of wait4 was never used. (closes issue ASTERISK-22557) Reported by: YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman (license 6537) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3c645e8520	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. ........ Merged revisions 402537 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7d0d1a1efb	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/ ........ Merged revisions 402528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	cdfbc02df1	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/ ........ Merged revisions 402517 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	029ce1e962	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. ........ Merged revisions 402507 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f6bd22b0fd	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 ........ Merged revisions 402505 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	aa05fde22d	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. ........ Merged revisions 402503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	aff0faf6ba	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/ ........ Merged revisions 402501 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	fe47684b43	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 ........ Merged revisions 402452 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7d2f2d6ef8	vector: Uppercase API to follow C convention. C does not support templates like C++. ........ Merged revisions 402438 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	629a5fc39b	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/ ........ Merged revisions 402429 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a84cff117d	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 ........ Merged revisions 402427 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0721b1de83	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 ........ Merged revisions 402416 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5401b2bfbf	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. ........ Merged revisions 402398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4b7ff87492	app_confbridge: Make the CONFBRIDGE function be able to create dynamic menus Also adds the ability to clear all profile items and makes behavior more consistent with documentation as when choosing whether to use CONFBRIDGE datastore profiles or the application arguments to the confbridge application. (closes issue ASTERISK-22760) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2971/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	3b36687a56	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/ ........ Merged revisions 402387 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2d0fb7f613	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) ........ Merged revisions 402367 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7678fd040e	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/ ........ Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	98dea21bc1	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 ........ Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4053f36a71	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/ ........ Merged revisions 402346 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d17a780333	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/ ........ Merged revisions 402336 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	dd221c74c5	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 ........ Merged revisions 402327 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e9fc321053	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 ........ Merged revisions 402289 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	981983bfde	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. ........ Merged revisions 402285 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	069da1e75a	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. ........ Merged revisions 402276 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4746c068dc	pjsip_messaging: Added debug for in dialog messaging (issue ASTERISK-22777) Reported by: Matt Jordan ........ Merged revisions 402265 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	74dc7983d1	Updates for 1.4.25 core sounds and 1.4.14 extra sounds, plus new en_GB language set The new sound packages relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, ASTERISK-20782 Modified sounds/Makefile for the new sound versions and to account for the new en_GB language set. (issue ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged revisions 402224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	076b29dd5b	Remove some spammy debug messages; improve clarity of others Debug messages aren't free. Even when the debug level is sufficiently low such that the messages are never evaluated, there is a cost to having to parse Asterisk logs that contain debug messages that (a) fail to convey sufficient information or (b) occur so frequently as to be next to meaningless. Based on having to stare at lots of DEBUG messages, this patch makes the following changes: * channel.c: When copying variables from a parent channel to a child channel, specify the channels involved. Do not log anything for a variable that is not inherited; the fact that it doesn't have an _ or __ already signifies that it won't be inherited. * pbx.c: Specify what function evaluation has occurred that created the result. * translate.c: Bump up the translator path messages to 10. I've never once had to use these debug messages, and for each format that is registered (on startup) and unregistered (on shutdown) the entire f^2 matrix is logged out. For short tests in the Asterisk Test Suite, this should make finding the actual test much easier. * xmldoc.c: The debug message that 'blah' is not found in the tree is expected. Often, description elements - which are not required - are not provided. This debug message adds no additional value, as it is not indicative of an error or helpful in debugging which element did not contain a 'blah' element as a child. If an element is supposed to contain a child element, then that XML tree should have failed validation in the first place. Review: https://reviewboard.asterisk.org/r/2966/ ........ Merged revisions 402150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402151 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402154 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	aa7f9e55f2	ARI: Remove channels/{channelId}/dial This removes the /ari/channels/{channelId}/dial URI since it is redundant, overly complex, is likely to become more externally complex over time, and is too high-level compared with other ARI operations. See the following for further information: http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html (closes issue ASTERISK-22784) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2968/ ........ Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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