2918 Commits (d1b70b8e43c84238306f91b0c3f00f7e8e8198ed)

Author	SHA1	Message	Date
Kevin Harwell	d1b70b8e43	channel locking: Add locking for channel snapshot creation ... Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f4d662add6	channels: Return allocated channels locked. ... This change makes ast_channel_alloc return allocated channels locked. By doing so no other thread can acquire, lock, and manipulate the channel before it is completely set up. (closes issue AST-1256) Review: https://reviewboard.asterisk.org/r/3067/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	1703d1e1fd	Several components: fixing Typos in comments and code, "avaliable" instead of "available" ... (issue ASTERISK-23021) (closes issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty Newton Patches: available.patch uploaded by Jeremy Lainé (license 6561) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	93a096fe9f	bridging: Give bridges a name and a known creator ... Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	66ee458f07	security: Inhibit execution of privilege escalating functions ... This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	db930057fd	res_pjsip: Apply outbound proxy to all SIP requests. ... Objects which are involved in SIP request creation and sending now allow an outbound proxy to be specified. For cases where an endpoint is used the outbound proxy specified there will be applied. (closes issue ASTERISK-22673) Reported by: Antti Yrjola Review: https://reviewboard.asterisk.org/r/3022/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6e7d64c79d	res_stasis: Expose event for call forwarding and follow forwarded channel. ... This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	95795aa36b	Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably ... There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be set on channels involved with blind and attended transfers. This would happen with features that were initialized by channel driver specific mechanisms in multiparty calls. This patch resolves those cases while attempted to keep the behavior for setting those variables as consistent as possible. (closes issue AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f22ac347fc	test_voicemail_api: Add check for a registered voicemail provider before tests. ... It is much nicer diagnosing a test failure if app_voicemail is actually loaded. ........ Merged revisions 403726 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5f0d590b2e	app_voicemail: Voicemail callback registration/unregistration function improvements. ... * The voicemail registration/unregistration functions now take a struct of callbacks instead of a lengthy parameter list of callbacks. * The voicemail registration/unregistration functions now prevent a competing module from interfering with an already registered callback supplying module. ........ Merged revisions 403643 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	9aea346918	bridge_native_rtp: Deadlock during 4-way conference creation ... The change contains a slightly adjusted patch that was on the issue (submitted by kmoore). A fix was made by adding in a bridge lock while calling bridge_start/stop from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. (closes issue ASTERISK-22749) Reported by: Kinsey Moore Review: https://reviewboard.asterisk.org/r/3066/ Patches: lock_inversion.diff uploaded by kmoore (license 6273) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	010983be91	ARI: Adding a channel to a bridge while a live recording is active blocks ... Added the ability to have rules that are checked when adding and/or removing channels to/from a bridge. In this case, if a channel is currently recording and someone attempts to add it to a bridge an "is recording" rule is checked, fails, and a 409 conflict is returned. Also command functions now return an integer value that can be descriptive of what kind of problems, if any, occurred before or during execution. (closes issue ASTERISK-22624) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2947/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	5cfa71a6b4	res_pjsip_messaging: send message to a default outbound endpoint ... In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c125499963	func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip ... This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan to use the CHANNEL function on a chan_pjsip channel to obtain run-time information about the channel from the PJSIP channel driver and the PJSIP stack. This includes: * RTP information, including source/destination media addresses, whether or not the media is secure, held, and other properties. * RTCP information. This includes sets of parseable information, as well as individual statistic attriutes. * PJSIP information. This includes URIs, local/remote signalling addresses, whether or not the signalling is secure, and other properties. * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT function to obtain more detailed endpoint information. Review: https://reviewboard.asterisk.org/r/3038/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c0b4e18900	Reverting r403311. It's causing ARI tests to hang. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3dfd810c6c	res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. ... Newer versions of PJSIP have changed to using a flag for the PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a configure check to detect the presence of the flag and use it if found. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1123e09e2f	sorcery, bucket: Change observer remove calls to take const callbacks struct. ... * Make ast_sorcery_observer_remove() accept a const callbacks struct. * Make ast_sorcery_observer_remove() tolerant of the sorcery parameter being NULL. Now it can be called within a module unload routine if the sorcery initialization fails. * Fix ast_sorcery_observer_add() to fail if the container link fails. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	4b7ba33125	Add channel locking for channel snapshot creation. ... This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e0cfdf5655	res_pjsip_transport_websocket: Fix security events and simplify implementation. ... Transport type determination for security events has been simplified to use the type present on the message itself instead of searching through configured transports to find the transport used. The actual WebSocket transport has also been simplified. It now leverages the existing PJSIP transport manager for finding the active WebSocket transport for outgoing messages. This removes the need for res_pjsip_transport_websocket to store a mapping itself. (closes issue ASTERISK-22897) Reported by: Max E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4a9e6d06a7	res_pjsip_session: Add configurable behavior for redirects. ... The action taken when a redirect occurs is now configurable on a per-endpoint basis. The redirect can either be treated as a redirect to a local extension, to a URI that is dialed through the Asterisk core, or to a URI that is dialed within PJSIP itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2963/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	094c5e2cca	ari:Add application/json parameter support ... The patch allows ARI to parse request parameters from an incoming JSON request body, instead of requiring the request to come in as query parameters (which is just weird for POST and DELETE) or form parameters (which is okay, but a bit asymmetric given that all of our responses are JSON). For any operation that does _not_ have a parameter defined of type body (i.e. "paramType": "body" in the API declaration), if a request provides a request body with a Content type of "application/json", the provided JSON document is parsed and searched for parameters. The expected fields in the provided JSON document should match the query parameters defined for the operation. If the parameter has 'allowMultiple' set, then the field in the JSON document may optionally be an array of values. (closes issue ASTERISK-22685) Review: https://reviewboard.asterisk.org/r/2994/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9a2df36dd4	Fix a configure issue with PJSIP transaction group lock detection. ... The configure check did not use the provided paths for pjproject if provided when looking for transaction group lock support. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	99d5cb8555	ARI: Implement device state API ... Created a data model and implemented functionality for an ARI device state resource. The following operations have been added that allow a user to manipulate an ARI controlled device: Create/Change the state of an ARI controlled device PUT /deviceStates/{deviceName}&{deviceState} Retrieve all ARI controlled devices GET /deviceStates Retrieve the current state of a device GET /deviceStates/{deviceName} Destroy a device-state controlled by ARI DELETE /deviceStates/{deviceName} The ARI controlled device must begin with 'Stasis:'. An example controlled device name would be Stasis:Example. A 'DeviceStateChanged' event has also been added so that an application can subscribe and receive device change events. Any device state, ARI controlled or not, can be subscribed to. While adding the event, the underlying subscription control mechanism was refactored so that all current and future resource subscriptions would be the same. Each event resource must now register itself in order to be able to properly handle [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	24568ea6c9	res_pjsip: AMI commands and events. ... Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	60e700565a	ari: Add Snoop operation for spying/whispering on channels. ... The Snoop operation can be invoked on a channel to spy or whisper on it. It returns a channel that any channel operations can then be invoked on (such as record to do monitoring). (closes issue ASTERISK-22780) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3003/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0045449943	ARI: Don't leak implementation details ... This change prevents channels used as implementation details from leaking out to ARI. It does this by preventing creation of JSON blobs of channel snapshots created from those channels and sanitizing JSON blobs of bridge snapshots as they are created. This introduces a framework for excluding information from output targeted at Stasis applications on a consumer-by-consumer basis using channel sanitization callbacks which could be extended to bridges or endpoints if necessary. This prevents unhelpful error messages from being generated by ast_json_pack. This also corrects a bug where BridgeCreated events would not be created. (closes issue ASTERISK-22744) Review: https://reviewboard.asterisk.org/r/2987/ Reported by: David M. Lee git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	01c44f6bed	voicemail: Fixup some doxygen comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402956 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	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
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
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
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
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	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
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
Matthew Jordan	5740a2bda6	Prevent CDR backends from unregistering while billing data is in flight ... This patch makes it so that CDR backends cannot be unregistered while active CDR records exist. This helps to prevent billing data from being lost during restarts and shutdowns. Review: https://reviewboard.asterisk.org/r/2880/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	23be89dfff	chan_pjsip: Fix a crash when direct media is enabled and an ACK is received after the channel is hung up. ... (closes issue ASTERISK-22731) Reported by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	330a2b4177	rtp_engine: fix rtp payloads copy and improve argument names ... In function ast_rtp_instance_early _bridge_make_compatible the use of instance 0/1 as arguments doesn't clearly communicate a direction that the copying of payloads from the source channel to the destination channel will occur, making it more probable to have the arguments to ast_rtp_codecs_payloads_copy() put in the reverse order. This patch renames the arguments with _dst and _src suffixes and corrects the copy direction. (closes issue ASTERISK-21464) Reported by: Kevin Stewart Review: https://reviewboard.asterisk.org/r/2894/ ........ Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows rtpmap:119 being copied per this change, but is not in sip invite ........ Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	1de75f4fae	pbx.c: fix confused match caller id that deleted exten still in hash ... This fixes a bug where a zero length callerid match adjacent to a no match callerid extension entry would be deleted together, which then resulted in hashtable references to free'd memory. A third state of the matchcid value has been added to indicate match to any extension which allows enforcing comparison of matchcid on/off without errors. (closes issue AST-1235) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401960 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	aa94b9c148	Bridging: Fix orphaned bridge if neither of the joining channels can join. ... The original issue noted that the bridge is orphaned when res_parking.so is not loaded and a call uses the dial kK flags. A similar issue happens when only one of the park flags is used. In this case you have the bridge with one or the other channel left in it. The channel and bridge will stay around until the channel hangs up. * Fixed the initial bridge channel push failure to act as if the channel were kicked out of the bridge. The bridge then decides if it needs to be dissolved. (closes issue ASTERISK-22629) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2928/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	aa4e8c4ab3	Add channel lock protection around translation path setup. ... Most callers of ast_channel_make_compatible() happen before the channels enter a two party bridge. With the new bridging framework, two party bridging technologies may also call ast_channel_make_compatible() when there is more than one thread involved with the two channels. * Added channel lock protection in set_format() and ast_channel_make_compatible_helper() when dealing with the channel's native formats while setting up a translation path. * Fixed best_src_fmt and best_dst_fmt usage consistency in ast_channel_make_compatible_helper(). The call to ast_translator_best_choice() got them backwards. * Updated some callers of ast_channel_make_compatible() and the function documentation. There is actually a difference between the two channels passed in. * Fixed the deadlock potential in res_fax.c dealing with ast_channel_make_compatible(). The deadlock potential was already there anyway because res_fax called ast_channel_make_compatible() with chan locked. (closes issue ASTERISK-22542) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2915/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0b625fcc69	Tweak ast_bridge_depart() doxygen. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	61a66eb214	Remove the bit about requiring ast_bridge_depart() to be called before ast_bridge_destroy(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	edc734c694	Clarify in ast_bridge_destroy() about how departable channels must be handled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3c20ac4f61	channel.h: whitespace changes. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1d72d481a7	Miscellaneous stand alone comment cleanups. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago