52 Commits (master)

Author	SHA1	Message	Date
George Joseph	c873f2ae7e	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	3 days ago
George Joseph	3d5ae0b5e1	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/	2 months ago
Holger Hans Peter Freyther	9fc631d028	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.	4 months ago
George Joseph	dfc8652b55	docs: Add version information to ARI resources and methods. ... * Dump a git blame of each file in rest-api/api-docs. * Get the commit for each "resourcePath" and "httpMethod" entry. * Find the tags for each commit (same as other processes). * Insert a "since" array after each "resourcePath" and "httpMethod" entry.	4 months 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.	10 months 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	6 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	6 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	7 years ago
Kevin Harwell	b088cddc03	pjsip_options: wrongly applied "UNKNOWN" status ... A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89	8 years ago
Mark Michelson	7a8d6bc81b	Bump ARI version to 2.0.0 ... In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665	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
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
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	9 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
Matt Jordan	3e2a994c71	ARI: Update version to 1.7.0 ... This patch updates the version of ARI to 1.7.0 to reflect the backwards compatible changes that will be introduced in 13.4.0. Change-Id: I6c36e6144da426412f25828a868e4df916bff60a (cherry picked from commit 9d8a462356)	10 years ago
Matthew Jordan	894153b8b1	res/ari: Fix model validation for ChannelHold event ... When the ChannelHold event was added, the 'musicclass' parameter was erroneously removed. This caused the ChannelHold events to be rejected as they failed model validation. This patch updates the Swagger schema such that it now properly reflects the event that is being created. Hooray for tests that catch things like this. ........ Merged revisions 434597 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	c2f50ba6f4	ARI: Add the ability to intercept hold and raise an event ... For some applications - such as SLA - a phone pressing hold should not behave in the fashion that the Asterisk core would like it to. Instead, the hold action has some application specific behaviour associated with it - such as disconnecting the channel that initiated the hold; only playing MoH to channels in the bridge if the channels are of a particular type, etc. One way of accomplishing this is to use a framehook to intercept the hold/unhold frames, raise an event, and eat the frame. Tasty. This patch accomplishes that using a new dialplan function, HOLD_INTERCEPT. In addition, some general cleanup of raising hold/unhold Stasis messages was done, including removing some RAII_VAR usage. Review: https://reviewboard.asterisk.org/r/4549/ ASTERISK-24922 #close ........ Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	fe6cbf455a	AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features ... AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per semantic versioning, that warrants a bump in the minor version number, as it reflects a backwards compatible change. Hence, this commit. ........ Merged revisions 429091 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	fe7671fee6	Add new AMI and ARI events for connected line changes on a channel. ... The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 ........ Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9bf039346a	rest-api/api-docs/events.json: Remove non-compliant 'extends' attribute ... Prior to the release of Swagger 1.2, the attribute 'extends' was being promoted as a possible way to show that a particular object extends an existing object. Instead, the Swagger specification went with the 'subTypes' attribute in the base object. This patch removes the unsupported attribute; the object that the offending objects proposed to extend already lists them in its 'subTypes' attribute. ASTERISK-24300 #close Reported by: Bradley Watkins ........ Merged revisions 423620 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423621 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	de6e467db7	rest-api/api-docs: Correct basePath in resources to match top resources file ... The resources.json file that defines the resource JSON files used with ARI references a basePath of 'http://localhost:8088/ari'. This does not match what is defined in the resource files themselves, 'http://localhost:8088/stasis'. The correct base path is the one that includes 'ari' in the URL; this patch updates the various resource JSON files to have the correct basePath. ASTERISK-24339 #close Reported by: Bradley Watkins ........ Merged revisions 423617 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423618 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	36f4bff943	Stasis: Add information to blind transfer event ... When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ffccae8269	AMI/ARI: Update version to 2.5.0/1.5.0 respectively ... This is to support the backwards compatible changes made in the next version of Asterisk. ........ Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420808 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications ... This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	47bf7efc4d	Multiple revisions 420089-420090,420097 ... ........ r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5a3023a114	manager/ARI: Update version to 2.4.0/1.4.0; Update UPGRADE.txt ... ........ Merged revisions 418182 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sam Galarneau	aa370d6105	ARI: Improvements to body parameters documentation ... The variables body parameter under the originate and originate with id operations of the channel resource showed invalid JSON in its description. The variables body parameter under the userEvent operation of the event resource made no mention that the custom key/value pairs should be wrapped in a variables key in order to be added to the custom user event. ASTERISK-23975 #close Review: https://reviewboard.asterisk.org/r/3692/ ........ Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	53968c00b3	TALK_DETECT: A channel function that raises events when talking is detected ... This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6107712857	AMI/ARI: Update version numbers ... Update the semantic versioning of ARI to 1.3.0 and AMI to 2.3.0 to account for backwards compatible changes going from 12.2.0 to 12.3.0. ........ Merged revisions 414765 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	cf21644d6a	ARI: Add ability to raise arbitrary User Events ... User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	597f25db69	Update API versions and UPGRADE/CHANGES for 12.2.0 ... This patch does the following: * It updates the AMI version to 2.2.0 to indicate backwards compatible changes have been made since the last release * It updates the ARI version to 1.2.0 to indicate backwards compatible changes have been made since the last release * It updates the UPGRADE/CHANGES files with changes that were not mentioned ........ Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9b93917896	ARI/AMI: Update versions; update UPGRADE/CHANGES notes for 12.1.0 changes ... Due to backwards compatible changes made to AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0, respectively. ........ Merged revisions 407402 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e5899852cc	res_stasis: Enable transfers and provide events when they occur. ... This change enables transfers within ARI created bridges and adds events for when they occur. Unlike other events these will be received if *any* subscribed object is involved in the transfer. (closes issue ASTERISK-22984) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/3120/ ........ Merged revisions 407153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ec79aabdb9	ari: Bump the version of ARI to 1.0.0 ... (closes issue ASTERISK-23007) ........ Merged revisions 404184 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3a5cc054ed	res_stasis: Expose event for call forwarding and follow forwarded channel. ... This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ ........ 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	12 years ago
Joshua Colp	e93fbf41e6	res_ari: Add Recording events to the validator. ... ........ Merged revisions 403240 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	ed48377994	ARI: Implement device state API ... Created a data model and implemented functionality for an ARI device state resource. The following operations have been added that allow a user to manipulate an ARI controlled device: Create/Change the state of an ARI controlled device PUT /deviceStates/{deviceName}&{deviceState} Retrieve all ARI controlled devices GET /deviceStates Retrieve the current state of a device GET /deviceStates/{deviceName} Destroy a device-state controlled by ARI DELETE /deviceStates/{deviceName} The ARI controlled device must begin with 'Stasis:'. An example controlled device name would be Stasis:Example. A 'DeviceStateChanged' event has also been added so that an application can subscribe and receive device change events. Any device state, ARI controlled or not, can be subscribed to. While adding the event, the underlying subscription control mechanism was refactored so that all current and future resource subscriptions would be the same. Each event resource must now register itself in order to be able to properly handle [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3025/ ........ 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	12 years ago
Joshua Colp	14a7452934	ari: Add events for playback and recording. ... While there were events defined for playback and recording these were not actually sent. This change implements the to_json handlers which produces them. (closes issue ASTERISK-22710) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/3026/ ........ Merged revisions 403119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	5701c51bb6	Fixed a typ. ... ........ Merged revisions 402738 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d8a760307e	ARI recordings: Issue HTTP failures for recording requests with file conflicts ... If a file already exists in the recordings directory with the same name as what we would record, issue a 422 instead of relying on the internal failure and issuing success. (closes issue ASTERISK-22623) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2922/ ........ Merged revisions 401973 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	32b4e79434	The Swagger 1.2 specification for type extension ended up being ... slightly different than my proposal. Instead of putting an 'extends' field on the subtype, the base type has a 'subTypes' field, which is a list of the subTypes. Given that its a messaging model and not an object model, kinda makes sense. This patch changes the events.json api-doc, and the python translators to take the new format into account. Other changes that are in Swagger 1.2 were not adopted, since the spec is still in flux, and could change before it's finalized. A summary of changes to the Swagger-1.2 spec can be found at https://github.com/wordnik/swagger-core/wiki/1.2-transition. (closes issue ASTERISK-22440) Review: https://reviewboard.asterisk.org/r/2909/ ........ Merged revisions 401701 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8d7873b836	ARI: Add subscription support ... This patch adds an /applications API to ARI, allowing explicit management of Stasis applications. * GET /applications - list current applications * GET /applications/{applicationName} - get details of a specific application * POST /applications/{applicationName}/subscription - explicitly subscribe to a channel, bridge or endpoint * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe from a channel, bridge or endpoint Subscriptions work by a reference counting mechanism: if you subscript to an event source X number of times, you must unsubscribe X number of times to stop receiveing events for that event source. Review: https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) Reported by: Matt Jordan ........ Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	537ecebd2d	ARI - implement allowMultiple for parameters ... Swagger allows parameters to be specified as 'allowMultiple', meaning that the parameter may be specified as a comma separated list of values. I had written some of the API docs using that, but promptly forgot about implementing it. This patch finally fills in that gap. The codegen template was updated to represent 'allowMultiple' fields as array/size fields in the _args structs. It also parses the comma separated list using ast_app_separate_args(), so quoted strings in the argument will be handled properly. Review: https://reviewboard.asterisk.org/r/2698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	01c21c7aea	Document MissingParams error message for /ari/events ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c9a3d4562d	Update events to use Swagger 1.3 subtyping, and related aftermath ... This patch started with the simple idea of changing the /events data model to be more sane. The original model would send out events like: { "stasis_start": { "args": [], "channel": { ... } } } The event discriminator was the field name instead of being a value in the object, due to limitations in how Swagger 1.1 could model objects. While technically sufficient in communicating event information, it was really difficult to deal with in terms of client side JSON handling. This patch takes advantage of a proposed extension[1] to Swagger which allows type variance through the use of a discriminator field. This had a domino effect that made this a surprisingly large patch. [1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ In changing the models, I also had to change the swagger_model.py processor so it can handle the type discriminator and subtyping. I took that a big step forward, and using that information to generate an ari_model module, which can validate a JSON object against the Swagger model. The REST and WebSocket generators were changed to take advantage of the validators. If compiled with AST_DEVMODE enabled, JSON objects that don't match their corresponding models will not be sent out. For REST API calls, a 500 Internal Server response is sent. For WebSockets, the invalid JSON message is replaced with an error message. Since this took over about half of the job of the existing JSON generators, and the .to_json virtual function on messages took over the other half, I reluctantly removed the generators. The validators turned up all sorts of errors and inconsistencies in our data models, and the code. These were cleaned up, with checks in the code generator avoid some of the consistency problems in the future. * The model for a channel snapshot was trimmed down to match the information sent via AMI. Many of the field being sent were not useful in the general case. * The model for a bridge snapshot was updated to be more consistent with the other ARI models. Another impact of introducing subtyping was that the swagger-codegen documentation generator was insufficient (at least until it catches up with Swagger 1.2). I wanted it to be easier to generate docs for the API anyways, so I ported the wiki pages to use the Asterisk Swagger generator. In the process, I was able to clean up many of the model links, which would occasionally give inconsistent results on the wiki. I also added error responses to the wiki docs, making the wiki documentation more complete. Finally, since Stasis-HTTP will now be named Asterisk REST Interface (ARI), any new functions and files I created carry the ari_ prefix. I changed a few stasis_http references to ari where it was non-intrusive and made sense. (closes issue ASTERISK-21885) Review: https://reviewboard.asterisk.org/r/2639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	dcf03554a0	Shuffle RESTful URL's around. ... This patch moves the RESTful URL's around to more appropriate locations for release. The /stasis URL's are moved to /ari, since Asterisk REST Interface was a more appropriate name than Stasis-HTTP. (Most of the code still has stasis_http references, but they will be cleaned up after there are no more outstanding branches that would have merge conflicts with such a change). A larger change was moving the ARI events WebSocket off of the shared /ws URL to its permanent home on /ari/events. The Swagger code generator was extended to handle "upgrade: websocket" and "websocketProtocol:" attributes on an operation. The WebSocket module was modified to better handle WebSocket servers that have a single registered protocol handler. If a client connections does not specify the Sec-WebSocket-Protocol header, and the server has a single protocol handler registered, the WebSocket server will go ahead and accept the client for that subprotocol. (closes issue ASTERISK-21857) Review: https://reviewboard.asterisk.org/r/2621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a5bbc790e7	Stasis-HTTP: Flesh out bridge-related capabilities ... This adds support for Stasis applications to receive bridge-related messages when the application shows interest in a given bridge. To supplement this work and test it, this also adds support for the following bridge-related Stasis-HTTP functionality: * GET stasis/bridges * GET stasis/bridges/{bridgeId} * POST stasis/bridges * DELETE stasis/bridges/{bridgeId} * POST stasis/bridges/{bridgeId}/addChannel * POST stasis/bridges/{bridgeId}/removeChannel Review: https://reviewboard.asterisk.org/r/2572/ (closes issue ASTERISK-21711) (closes issue ASTERISK-21621) (closes issue ASTERISK-21622) (closes issue ASTERISK-21623) (closes issue ASTERISK-21624) (closes issue ASTERISK-21625) (closes issue ASTERISK-21626) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	10ba6bf8a8	This patch implements the REST API's for POST /channels/{channelId}/play ... and GET /playback/{playbackId}. This allows an external application to initiate playback of a sound on a channel while the channel is in the Stasis application. /play commands are issued asynchronously, and return immediately with the URL of the associated /playback resource. Playback commands queue up, playing in succession. The /playback resource shows the state of a playback operation as enqueued, playing or complete. (Although the operation will only be in the 'complete' state for a very short time, since it is almost immediately freed up). (closes issue ASTERISK-21283) (closes issue ASTERISK-21586) Review: https://reviewboard.asterisk.org/r/2531/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago