165 Commits (808f2998085b78225addfd73289fdc6c473b7fa1)

Author	SHA1	Message	Date
George Joseph	b55d21ad91	make ari-stubs so doc periodic jobs can run ... The periodic doc job does a make ari-stubs and checks that there are no changes before generating the docs. Since I changed the mustache template (and the generated code directly) recently and forgot to regenerate the stubs, the doc job thinks they're out of date. Change-Id: I94b97035311eccf52b0101b8590223265a7881d4	9 years ago
George Joseph	6691606723	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f3)	9 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	9 years ago
Matt Jordan	a72ef38113	res/ari/resource_bridges: Add the ability to manipulate the video source ... In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	9 years ago
Sebastien Duthil	c6d755de11	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	9 years ago
Matt Jordan	c30d677333	res/stasis: Add CLI commands for displaying/debugging ARI apps ... This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Mark Michelson	3bd76dd679	ARI: Add duplicate channel ID checking for channel creation. ... This is similar to what is done for origination, but for the 14 and up channel creation method. When attempting to create a channel, if a channel ID is specified and a channel already exists with that ID, then a 409 is returned. Change-Id: I77f9253278c6947939c418073b6b31065489187c	9 years ago
Mark Michelson	e459b8dadf	ARI: Detect duplicate channel IDs ... ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06	9 years ago
Joshua Colp	f31772ec20	ari: Update model validator based on addition of asterisk_id. ... ASTERISK-26470 Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541	9 years ago
Matt Jordan	dd5129d84a	res/ari: Add the Asterisk EID field to outgoing events ... This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d	9 years ago
Jean Aunis	91993ebaa5	resource_channels.c: add hangup reason "answered_elsewhere". ... In ARI, the channels API allows to hangup a channel with a hangup reason. This commit adds a new reason "answered_elsewhere". When using a SIP channel, this will eventually allow Asterisk to add a proper "Reason" header to a CANCEL message. ASTERISK-26321 Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d	9 years ago
Kevin Harwell	e711e57106	resource_channels: Sync with ARI stubs ... This file was out of sync with the current ARI definitions. Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a	9 years ago
George Joseph	886f2cab23	rest_api/channels: Fix multiple issues with create and dial ... * We weren't properly subscribing to the channel and it's originator on create. * We weren't doing a publish_dial after calling ast_call on dial. * We weren't calling depart_bridge when a channel left the dial bridge. The first 2 issues were causing events to not be generated and the third was actually causing channels to not get properly destroyed when hung up. Together these 3 issues were causing the new rest_apichannels/create_dial_bridge tests to fail. As a result of the fixes, the cdr state machine had to be slightly tweaked to allow bridge leave events without asserting and the tests themselves had to be updated to account for the channels now cleaning themselves up. Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8	9 years ago
Mark Michelson	11caa10cf5	ARI: Ensure announcer channels are destroyed. ... Announcer channels were not being destroyed because the stasis_app_control structure that referenced them was not being destroyed. The control structure was not being destroyed because it was not being unlinked from its container. It was not being unlinked from its container because the after bridge callback for the announcer channel was not being run. The after bridge callback was not being run because the after bridge datastore was not being removed from the channel on destruction. The channel was not being destroyed because the hangup that used to destroy the channel was now only reducing the reference count to one. The reference count of the channel was only being reduced to one because the stasis_app_control structure was holding the final reference... The control structure used to not keep a reference to the channel, so that loop described above did not happen. The solution is to manually remove the control structure from its container when the playback on a bridge is complete. ASTERISK-26083 #close Reported by Joshua Colp Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4	9 years ago
zuul	39e6d80937	Merge "ARI: Ensure proper channel state on operations."	10 years ago
Mark Michelson	1fd3a7849e	ARI: Ensure proper channel state on operations. ... ARI was recently outfitted with operations to create and dial channels. This leads to the ability to try funny stuff. You could create a channel and then immediately try to play back media on it. You could create a channel, dial it, and while it is ringing attempt to make it continue in the dialplan. This commit attempts to fix this by adding a channel state check to operations that should not be able to operate on outbound channels that have not yet answered. If a channel is in an invalid state, we will send a 412 response. ASTERISK-26047 #close Reported by Mark Michelson Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8	10 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	10 years ago
George Joseph	a2f820e8dc	ari/resource_channels: Add 'formats' to channel create/originate ... If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b	10 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	10 years ago
Mark Michelson	88d997913f	ARI: Re-implement the ARI dial command, allowing for early bridging. ... ARI dial had been implemented using the Dial API. This made great sense when dialing was 100% separate from bridging. However, if a channel were to be added to a bridge during the dial attempt, there would be a conflict between the dialing thread and the bridging thread. Each would be attempting to read frames from the dialed channel and act on them. The initial attempt to make the two play nice was to have the Dial API suspend the channel in the bridge and stay in charge of the channel until the dial was complete. The problem with this was that it was riddled with potential race conditions. It also was not well-suited for the case where the channel changed which bridge it was in during the dial. This new approach removes the use of the Dial API altogether. Instead, the channel we are dialing is placed into an invisible ARI dialing bridge. The bridge channel thread handles incoming frames from the channel. If the channel is added to a real bridge, it is departed from the invisible bridge and then added to the real bridge. Similarly, if the channel is removed from the real bridge, it is automatically added back to the invisible bridge if the dial attempt is still active. This approach keeps the threading simple by always having the channel being handled by bridge channel threads. ASTERISK-25925 Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb	10 years ago
Matt Jordan	e773e3a9bb	ARI: Add the ability to download the media associated with a stored recording ... This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181	10 years ago
Matt Jordan	03d88b5656	ARI: Add the ability to play multiple media URIs in a single operation ... Many ARI applications will want to play multiple media files in a row to a resource. The most common use case is when building long-ish IVR prompts made up of multiple, smaller sound files. Today, that requires building a small state machine, listening for each PlaybackFinished event, and triggering the next sound file to play. While not especially challenging, it is tedious work. Since requiring developers to write tedious code to do normal activities stinks, this patch adds the ability to play back a list of media files to a resource. Each of the 'play' operations on supported resources (channels and bridges) now accepts a comma delineated list of media URIs to play. A single Playback resource is created as a handle to the entire list. The operation of playing a list is identical to playing a single media URI, save that a new event, PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final media URI. When the entire list is finished being played, a PlaybackFinished event is raised. In order to help inform applications where they are in the list playback, the Playback resource now includes a new, optional attribute, 'next_media_uri', that contains the next URI in the list to be played. It's important to note the following: - If an offset is provided to the 'play' operations, it only applies to the first media URI, as it would be weird to skip n seconds forward in every media resource. - Operations that control the position of the media only affect the current media being played. For example, once a media resource in the list completes, a 'reverse' operation on a subsequent media resource will not start a previously completed media resource at the appropiate offset. - This patch does not add any new operations to control the list. Hopefully, user feedback and/or future patches would add that if people want it. ASTERISK-26022 #close Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f	10 years ago
Sean Bright	14938184a3	res_ari: Correct Location headers returned by some ARI resources ... The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	10 years ago
Joshua Colp	72ef79dc2d	Merge "ARI: Add method to Dial a created channel."	10 years ago
Joshua Colp	3b71f09bb7	Merge "ARI: Add method to create a new channel."	10 years ago
Mark Michelson	abbb2edd4c	ARI: Add method to Dial a created channel. ... This adds a new ARI method that allows for you to dial a channel that you previously created in ARI. By combining this with the create method for channels, it allows for a workflow where a channel can be created, manipulated, and then dialed. The channel is under control of the ARI application during all stages of the Dial and can even be manipulated based on channel state changes observed within an ARI application. The overarching goal for this is to eventually be able to add a dialed channel to a Stasis bridge earlier than the "Up" state. However, at the moment more work is needed in the Dial and Bridge APIs in order to facilitate that. ASTERISK-25889 #close Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205	10 years ago
Mark Michelson	dd48d60c5b	ARI: Add method to create a new channel. ... This adds a new ARI method to the channels resource that allows for the creation of a new channel. The channel is created and then placed into the specified Stasis application. This is different from the existing originate method that creates a channel, dials it, and then places the answered channel into the dialplan or a Stasis application. This method does not attempt to call the channel at all. Dialing is left as a later step after channel creation. This allows for pre-dialing channel manipulation if desired. ASTERISK-25889 Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8	10 years ago
zuul	e2a8033d09	Merge "res_stasis: Add control ref to playback and recording structs."	10 years ago
Richard Mudgett	0ea742d33a	res_stasis: Add control ref to playback and recording structs. ... The stasis_app_playback and stasis_app_recording structs need to have a struct stasis_app_control ref. Other threads can get a reference to the playback and recording structs from their respective global container. These other threads can then use the control pointer they contain after the control struct has gone. * Add control ref to stasis_app_playback and stasis_app_recording structs. With the refs added, the control command queue can now have a circular control reference which will cause the control struct to never get released if the control's command queue is not flushed when the channel leaves the Stasis application. Also the command queue needs better protection from adding commands if the control->is_done flag is set. * Flush the control command queue on exit. ASTERISK-25882 #close Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d	10 years ago
Richard Mudgett	ece2edaa04	res_ari: Cannot get control also means channel is unavailable. ... The only caller of ari_bridges_play_found() has this note: If ari_bridges_play_found fails because the channel is unavailable for playback, The channel will be removed from the playback list soon. We can keep trying to get channels from the list until we either get one that will work or else there isn't a channel for this bridge anymore, in which case we'll revert to ari_bridges_play_new. Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6	10 years ago
Richard Mudgett	12feec0bf7	res/ari/config.c: Fix user sort compare function. ... Made use the ao2 sort compare template function and OBJ_SEARCH_xxx identifiers. Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c	10 years ago
Richard Mudgett	3f4fa245e5	res/ari/config.c: Optimize conf_alloc() object init. ... * Now conf_alloc() has more off nominal error checking. * Eliminated RAII_VAR() use in conf_alloc(). * Eliminated a dubius shortcut when destroying cfg->general in conf_destructor() that would cause a crash if cfg->general failed to get allocated. * Add some ACO registration section comments. Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a	10 years ago
Richard Mudgett	aa00df62ee	res/ari/config.c: Fix conf_alloc() object init. ... Need to finish initializing the string fields in the ao2 object before putting any default strings into them. ASTERISK-25383 #close Reported by: yaron nahum Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84	10 years ago
Matt Jordan	b99a705262	ARI: Add the ability to subscribe to all events ... This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6	10 years ago
Matt Jordan	5206aa9d30	ARI: Add events for Contact and Peer Status changes ... This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9	10 years ago
Mark Michelson	aacb46b56a	Merge "res_ari_events: Fix shutdown ref leak."	10 years ago
Richard Mudgett	21d419e4fc	ari/ari_websockets.c: Fix ast_debug parameter type mismatch. ... This is a type mismatch fix of the debugging commit c63316eec1 made to find out why a testsuite test was failing only on one of the continuous integration build agents. Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75	10 years ago
Richard Mudgett	03eb6cbc10	res_ari_events: Fix shutdown ref leak. ... ASTERISK-25308 #close Reported by: Joshua Colp Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082	10 years ago
Scott Emidy	12e6f5ac01	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	10 years ago
Scott Emidy	b91ca7ba49	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	10 years ago
Scott Emidy	f19c4930c2	ARI: Deleting log channels ... An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6	10 years ago
Mark Michelson	c63316eec1	res_http_websocket: Debug write lengths. ... Commit 39cc28f6ea attempted to fix a test failure observed on 32 bit test agents by ensuring that a cast from a 32 bit unsigned integer to a 64 bit unsigned integer was happening in a predictable place. As it turns out, this did not cause test runs to succeed. This commit adds several redundant debug messages that print the payload lengths of websocket frames. The idea here is that this commit will not cause tests to succeed for the faulty test agent, but we might deduce where the fault lies more easily this way by observing at what point the expected value (537) changes to some ungangly huge number. If you are wondering why something like this is being committed to the branch, keep in mind that in commit 39cc28f6ea I noted that the observed test failures only happen when automated tests are run. Attempts to run the tests by hand manually on the test agent result in the tests passing. Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d	10 years ago
Mark Michelson	35a98161df	res_http_websocket: Avoid passing strlen() to ast_websocket_write(). ... We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21 solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67	10 years ago
Mark Michelson	b002e09214	Merge "ARI: Channels added to Stasis application during WebSocket creation ..."	10 years ago
Benjamin Ford	1f02d20da4	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	10 years ago
Ashley Sanders	fe804b09b3	ARI: Channels added to Stasis application during WebSocket creation ... ... Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis applications were registered. This was done such that the WebSocket would be ready when an application is registered. However, by creating the WebSocket first, the client had the ability to make requests for the Stasis application it thought had been created with the initial handshake request. The inevitable conclusion of this scenario was the cart being put before the horse. ASTERISK-24988 resolved half of the problem by ensuring that the applications were created and registered with Stasis prior to completing the handshake with the client. While this meant that Stasis was ready when the client received the green-light from Asterisk, it also meant that the WebSocket was not yet ready for Stasis to dispatch messages. This patch introduces a message queuing mechanism for delaying messages from Stasis applications while the WebSocket is being constructed. When the ARI event processor receives the message from the WebSocket that it is being created, the event processor instantiates an event session which contains a message queue. It then tries to create and register the requested applications with Stasis. Messages that are dispatched from Stasis between this point and the point at which the event processor is notified the WebSocket is ready, are stashed in the queue. Once the WebSocket has been built, the queue's messages are dispatched in the order in which they were originally received and the queue is concurrently cleared. ASTERISK-25181 #close Reported By: Matt Jordan Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17	10 years ago
Matt Jordan	254d07b15b	ARI: Add support for push configuration of dynamic object ... This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c	10 years ago
Benjamin Ford	e01d93e092	ARI: Fixed unload mode for unload module. ... Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, which would unload a module even if it was in use. * Changed unload mode to proper mode ASTERISK-25173 Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533	10 years ago
Benjamin Ford	9d458b8311	ARI: Added new functionality to reload a single module. ... An http request can be sent to reload an Asterisk module. If the module can not be reloaded or is not already loaded, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, based on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be reloaded through http requests ASTERISK-25173 Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1	10 years ago