242 Commits (22.4.1)

Author	SHA1	Message	Date
George Joseph	ea344ecc14	ari_websockets: Fix frack if ARI config fails to load. ... ari_ws_session_registry_dtor() wasn't checking that the container was valid before running ao2_callback on it to shutdown registered sessions. (cherry picked from commit abf8986068)	6 months ago
George Joseph	2a55f064f3	ARI: REST over Websocket ... This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ (cherry picked from commit 1442c17141)	6 months ago
George Joseph	de3d6c03e3	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. ... Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211 (cherry picked from commit 169b9b9a4f)	7 months ago
George Joseph	619cb75688	resource_channels.c: Fix memory leak in ast_ari_channels_external_media. ... Between ast_ari_channels_external_media(), external_media_rtp_udp(), and external_media_audiosocket_tcp(), the `variables` structure being passed around wasn't being cleaned up properly when there was a failure. * In ast_ari_channels_external_media(), the `variables` structure is now defined with RAII_VAR to ensure it always gets cleaned up. * The ast_variables_destroy() call was removed from external_media_rtp_udp(). * The ast_variables_destroy() call was removed from external_media_audiosocket_tcp(), its `endpoint` allocation was changed to to use ast_asprintf() as external_media_rtp_udp() does, and it now returns an error on failure. * ast_ari_channels_external_media() now checks the new return code from external_media_audiosocket_tcp() and sets the appropriate error response. Resolves: #1109 (cherry picked from commit 1a1eb88211)	7 months ago
Holger Hans Peter Freyther	fe5ff3021e	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function. (cherry picked from commit a0d0c47d06)	7 months ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event ... A stasis event is now produced when using the TONE_DETECT dialplan function. This event is published over ARI using the ChannelToneDetected event. This change does not make it available over AMI. Fixes: #811 UserNote: Setting the TONE_DETECT dialplan function on a channel in ARI will now cause a ChannelToneDetected ARI event to be raised when the specified tone is detected.	1 year ago
Naveen Albert	d1fb397cfc	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430	2 years ago
Sean Bright	c19470497f	resource_channels.c: Explicit codec request when creating UnicastRTP. ... Fixes #394	2 years ago
Holger Hans Peter Freyther	69590ba33e	ari: Provide the caller ID RDNIS for the channels ... Provide the caller ID RDNIS when available. This will allow an application to follow the redirect.	2 years ago
sungtae kim	ddb5c377fd	res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters ... This commit introduces an extension to the endpoint and relevant resource sizes for PJSIP, transitioning from its current 40-character constraint to a more versatile 255-character capacity. This enhancement significantly overcomes limitations related to domain qualification and practical usage, ultimately delivering improved functionality. In addition, it includes adjustments to accommodate the expanded realm size within the ARI, specifically enhancing the maximum realm length. Resolves: #345 UserNote: With this update, the PJSIP realm lengths have been extended to support up to 255 characters. UpgradeNote: As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas.	2 years ago
Holger Hans Peter Freyther	b99606955e	ari/stasis: Indicate progress before playback on a bridge ... Make it possible to start a playback and the calling party to receive audio on a bridge before the call is connected. Model the implementation after play_on_channel and deliver a AST_CONTROL_PROGRESS before starting the playback. For a PJSIP channel this will result in sending a SIP 183 Session Progress.	2 years ago
George Joseph	c3c82441a2	Prepare master for Asterisk 22	2 years ago
Maximilian Fridrich	51a7b18038	core/ari/pjsip: Add refer mechanism ... This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.	2 years ago
George Joseph	b974a8f9eb	rest-api: Ran make ari stubs to fix resource_endpoints inconsistency	2 years ago
Joe Searle	8462154a03	res_stasis.c: Add new type 'sdp_label' for bridge creation. ... Add new type 'sdp_label' when creating a bridge using the ARI. This will add labels to the SDP for each stream, the label is set to the corresponding channel id. Resolves: #91 UserNote: When creating a bridge using the ARI the 'type' argument now accepts a new value 'sdp_label' which will configure the bridge to add labels for each stream in the SDP with the corresponding channel id.	2 years ago
Naveen Albert	0119f3ad48	res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters. ... The current STIR/SHAKEN signing process is inconsistent with the RFCs in a couple ways that can cause interoperability issues. RFC8225 specifies that the keys must be ordered lexicographically, but currently the fields are simply ordered according to the order in which they were added to the JSON object, which is not compliant with the RFC and can cause issues with some carriers. To fix this, we now leverage libjansson's ability to dump a JSON object sorted by key value, yielding the correct field ordering. Additionally, telephone numbers must have any leading + prefix removed and must not contain characters outside of 0-9, *, and # in order to comply with the RFCs. Numbers are now properly formatted as such. ASTERISK-30407 #close Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2	3 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ... ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	3 years ago
Joshua C. Colp	52ed64e38a	ari: Destroy body variables in channel create. ... When passing a JSON body to the 'create' channel route it would be converted into Asterisk variables, but never freed resulting in a memory leak. This change makes it so that the variables are freed in all cases. ASTERISK-30344 Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d	3 years ago
Moritz Fain	4bf2473ac4	ari: expose channel driver's unique id to ARI channel resource ... This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Alexander Traud	826233b550	progdocs: Fix Doxygen left-overs. ... Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016	4 years ago
Alexander Traud	634e3ebdb8	res_ari: Fix for Doxygen. ... ASTERISK-29756 Change-Id: I2f1c1eea1c902492b77b74de9950f20ebbb7e758	4 years ago
Josh Soref	9ae9893c63	res: Spelling fixes ... Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Joshua C. Colp	0aac38c0ac	ari: Ignore invisible bridges when listing bridges. ... When listing bridges we go through the ones present in ARI, get their snapshot, turn it into JSON, and add it to the payload we ultimately return. An invisible "dial bridge" exists within ARI that would also try to be added to this payload if the channel "create" and "dial" routes were used. This would ultimately fail due to invisible bridges having no snapshot resulting in the listing of bridges failing. This change makes it so that the listing of bridges ignores invisible ones. ASTERISK-29668 Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a	4 years ago
Sungtae Kim	a1fa8df0ae	resource_channels.c: Fix external media data option ... Fixed the external media creation handle to handle the 'data' option correctly. ASTERISK-29629 Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129	4 years ago
sungtae kim	79d6d222d6	resource_channels.c: Fix wrong external media parameter parse ... Fixed ARI external media handler to accept body parameters. ASTERISK-29622 Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d	4 years ago
Igor Goncharovsky	99d44f0c5a	res_ari: Fix audiosocket segfault ... Add check that data parameter specified when audiosocket used for externalMedia. ASTERISK-29514 #close Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617	4 years ago
Jean Aunis	61116d5dbc	resource_endpoints.c: memory leak when providing a 404 response ... When handling a send_message request to a non-existing endpoint, the response's body is overriden and not properly freed. ASTERISK-29108 Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8	5 years ago
Sungtae Kim	aae0904c7d	res_stasis.c: Added video_single option for bridge creation ... Currently, it was not possible to create bridge with video_mode single. This made hard to put the bridge in a vidoe_single mode. So, added video_single option for Bridge creation using the ARI. This allows create a bridge with video_mode single. ASTERISK-29055 Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae	5 years ago
sungtae kim	bbe0f2230d	res_ari: Fix create channel request channelId parameter parsing ... If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. Fixed it to parse the channelId, other_channel_id parameter correclty. ASTERISK-28948 Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157	5 years ago
sungtae kim	fa7c69f40f	res_ari: Fix create request body parameter parsing. ... If parameters were passed in the body as JSON to the create route they were not being parsed before checking to ensure that required fields were set. This change moves the parsing so it occurs before checking. ASTERISK-28940 Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf	5 years ago
Joshua C. Colp	15cbff9d54	ari: Allow variables to be set on channel create. ... This change adds the same variable functionality that is available for originating a channel to the create call. Now when creating a channel you can specify dialplan variables to set instead of having to do another API call. ASTERISK-28896 Change-Id: If13997ba818136d7c070585504fc4164378aa992	6 years ago
sungtae kim	9ad3d2829c	res_ari_channels: Fixed endpoint 80 characters limit ... Fixed it to copy the entire string from the requested endpoint body except tech-prefix. ASTERISK-28847 Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25	6 years ago
Kevin Harwell	a715cf5aaa	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. ... This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	6 years ago
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge ... This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	6 years ago
Kevin Harwell	bdd785d31c	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
George Joseph	990a91b44a	stasis: Don't hold app_registry and session locks unnecessarily ... resource_events:stasis_app_message_handler() was locking the session, then attempting to determine if the app had debug enabled which locked the app_registry container. res_stasis:__stasis_app_register was locking the app_registry container then calling app_update which caused app_handler (which locks the session) to run. The result was a deadlock. * Updated resource_events:stasis_app_message_handler() to determine if debug was set (which locks the app_registry) before obtaining the session lock. * Updated res_stasis:__stasis_app_register to release the app_registry container lock before calling app_update (which locks the sesison). ASTERISK-28423 Reported by Ross Beer Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4	6 years ago
George Joseph	d71d0f9489	ExternalMedia: Change return object from ExternalMedia to Channel ... When we created the External Media addition to ARI we created an ExternalMedia object to be returned from the channels/externalMedia REST endpoint. This object contained the channel object that was created plus local_address and local_port attributes (which are also in the Channel variables). At the time, we thought that creating an ExternalMedia object would give us more flexibility in the future but as we created the sample speech to text application, we discovered that it doesn't work so well with ARI client libraries that a) don't have the ExternalMedia object defined and/or b) can't promote the embedded channel structure to a first-class Channel object. This change causes the channels/externalMedia REST endpoint to return a Channel object (like channels/create and channels/originate) instead of the ExternalMedia object. Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9	6 years ago
George Joseph	2ae1a22e0e	ARI: External Media ... The Channel resource has a new sub-resource "externalMedia". This allows an application to create a channel for the sole purpose of exchanging media with an external server. Once created, this channel could be placed into a bridge with existing channels to allow the external server to inject audio into the bridge or receive audio from the bridge. See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI for more information. Change-Id: I9618899198880b4c650354581b50c0401b58bc46	6 years ago
sungtae kim	613a335de5	res/ari/resource_channels.c: Added hangup reason code for channels ... Currently, DELETE /ari/channels/<channelID> supports only few hangup reasons. It's good enough for simple use, but when it needs to set the detail reason, it comes challenges. Added reason_code query parameter for that. ASTERISK-28385 Change-Id: I1cf1d991ffd759d0591b347445a55f416ddc3ff2	6 years ago
George Joseph	26cdf042f4	ARI: Run 'make ari-stubs' ... An earlier contributor apparently forgot to run 'make ari-stubs' before committing after making ARI model changes. Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6	7 years ago
George Joseph	2f13cdd315	Merge "res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics"	7 years ago
Friendly Automation	6a83c99c36	Merge "main/json.c: Added app_name, app_data to channel type"	7 years ago
sungtae kim	76768ad6ce	main/json.c: Added app_name, app_data to channel type ... It was difficult to check the channel's current application and parameters using ARI for current channels. Added app_name, app_data items to show the current application information. ASTERISK-28343 Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c	7 years ago
sungtae kim	71c0c7f631	res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics ... Added ARI resource for channel statistics. GET /ari/channels/{channelId}/rtp_statistics : It returns given channel's rtp statistics detail. ASTERISK-28320 Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376	7 years ago
sungtae kim	e2eb19b363	res/res_ari: Added timestamp as a requirement for all ARI events ... Changed to requirement to having timestamp for all of ARI events. The below ARI events were changed to having timestamp. PlaybackStarted, PlaybackContinuing, PlaybackFinished, RecordingStarted, RecordingFinished, RecordingFailed, ApplicationReplaced, ApplicationMoveFailed ASTERISK-28326 Change-Id: I382c2fef58f5fe107e1074869a6d05310accb41f	7 years ago
George Joseph	5f6890a8f9	Merge "res_stasis: Add ability to switch applications."	7 years ago
Ben Ford	6626df586e	res_stasis: Add ability to switch applications. ... Added the ability to move between Stasis applications within Stasis. This can be done by calling 'move' in an application, providing (at minimum) the channel's id and the application to switch to. If the application is not registered or active, nothing will happen and the channel will remain in the current application, and an event will be triggered to let the application know that the move failed. The event name is "ApplicationMoveFailed", and provides the "destination" that the channel was attempting to move to, as well as the usual channel information. Optionally, a list of arguments can be passed to the function call for the receiving application. A full example of a 'move' call would look like this: client.channels.move(channelId, app, appArgs) The control object used to control the channel in Stasis can now switch which application it belongs to, rather than belonging to one Stasis application for its lifetime. This allows us to use the same control object instead of having to tear down the current one and create another. ASTERISK-28267 #close Change-Id: I43d12b10045a98a8d42541889b85695be26f288a	7 years ago