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2013-12-16security: Inhibit execution of privilege escalating functionsDavid M. Lee
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 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-14res_pjsip: Apply outbound proxy to all SIP requests.Joshua Colp
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/ ........ Merged revisions 403811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-14res_stasis: Expose event for call forwarding and follow forwarded channel.Joshua Colp
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/ ........ Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliablyJonathan Rose
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/ ........ Merged revisions 403781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13bridge_native_rtp: Deadlock during 4-way conference creationKevin Harwell
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) ........ Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13ARI: Adding a channel to a bridge while a live recording is active blocksKevin Harwell
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/ ........ Merged revisions 403749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13test_voicemail_api: Add check for a registered voicemail provider before tests.Richard Mudgett
It is much nicer diagnosing a test failure if app_voicemail is actually loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11res_pjsip_messaging: send message to a default outbound endpointKevin Harwell
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/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11app_voicemail: Voicemail callback registration/unregistration function ↵Richard Mudgett
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsipMatthew Jordan
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/ ........ Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09app_page: Add predial handlers for app_page.Jonathan Rose
(closes issue AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09Switch PJSIP auth to use a vector.Mark Michelson
Since Asterisk has a vector API now, places where arrays are manually resized don't really make sense any more. Since the auth work in PJSIP was freshly-written, it was easy to reform it to use a vector. Review: https://reviewboard.asterisk.org/r/3044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05Reverting r403311. It's causing ARI tests to hang.David M. Lee
........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.Joshua Colp
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. ........ Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03sorcery, bucket: Change observer remove calls to take const callbacks struct.Richard Mudgett
* 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. ........ Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03Add channel locking for channel snapshot creation.Mark Michelson
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. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-01res_pjsip_transport_websocket: Fix security events and simplify implementation.Joshua Colp
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/ ........ Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_pjsip_session: Add configurable behavior for redirects.Joshua Colp
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/ ........ Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27astdb: Tweak some doxygen comments.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27ari:Add application/json parameter supportDavid M. Lee
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/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-26Fix a configure issue with PJSIP transaction group lock detection.Joshua Colp
The configure check did not use the provided paths for pjproject if provided when looking for transaction group lock support. ........ Merged revisions 403160 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23ARI: Implement device state APIKevin Harwell
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/ ........ Merged revisions 403134 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23res_pjsip: AMI commands and events.Kevin Harwell
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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23ari: Add Snoop operation for spying/whispering on channels.Joshua Colp
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/ ........ Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-22ARI: Don't leak implementation detailsKinsey Moore
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 ........ Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-21voicemail: Fixup some doxygen comments.Richard Mudgett
........ Merged revisions 402956 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-21ari: Add silence generator controlsDavid M. Lee
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/ ........ Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-16res_pjsip: Add support for building against pjproject with SIP transaction ↵Joshua Colp
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 ........ Merged revisions 402864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-13res_ari_channels: Add the ability to stop locally generated ringing on a ↵Joshua Colp
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
2013-11-08security_events: Push out security events over AMI eventsJonathan Rose
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
2013-11-05netsock2: Define AST_AF_* enum constants to their AF_* equivalentsMatthew Jordan
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
2013-11-02vector: Uppercase API to follow C convention.Richard Mudgett
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
2013-11-02vector: Update API to be more flexible.Richard Mudgett
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
2013-11-01voicemail: Simplify callback pointer declarations and add doxygen.Richard Mudgett
* 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
2013-11-01Manager: Add equivalent AMI actions for the bridge CLI commands.Scott Griepentrog
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
2013-11-01res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and ↵Joshua Colp
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
2013-11-01chan_sip: Fix RTCP port for SRFLX ICE candidatesKinsey Moore
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
2013-11-01res_ari_channels: Fix a deadlock when originating multiple channels close to ↵Joshua Colp
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
2013-10-27Prevent CDR backends from unregistering while billing data is in flightMatthew Jordan
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/ ........ Merged revisions 402081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26chan_pjsip: Fix a crash when direct media is enabled and an ACK is received ↵Joshua Colp
after the channel is hung up. (closes issue ASTERISK-22731) Reported by: Kinsey Moore ........ Merged revisions 402064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26rtp_engine: fix rtp payloads copy and improve argument namesScott Griepentrog
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 ........ Merged revisions 402043 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25pbx.c: fix confused match caller id that deleted exten still in hashScott Griepentrog
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 ........ Merged revisions 401961 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22Bridging: Fix orphaned bridge if neither of the joining channels can join.Richard Mudgett
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/ ........ Merged revisions 401424 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Add channel lock protection around translation path setup.Richard Mudgett
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/ ........ Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Tweak ast_bridge_depart() doxygen.Richard Mudgett
........ Merged revisions 401232 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Remove the bit about requiring ast_bridge_depart() to be called before ↵Mark Michelson
ast_bridge_destroy(). ........ Merged revisions 401223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Clarify in ast_bridge_destroy() about how departable channels must be handled.Mark Michelson
........ Merged revisions 401212 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11channel.h: whitespace changes.Richard Mudgett
........ Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-07Miscellaneous stand alone comment cleanups.Richard Mudgett
........ Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04Add IPv6 Support To chan_iax2Michael L. Young
This patch adds IPv6 support to chan_iax2. Yay! (closes issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged revisions 400567 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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