asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	efc4034d72	rest-api: Code out of sync with the model Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006	9 years ago
George Joseph	c27c232057	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
Matthew Jordan	1995baad71	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. New Feature A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. Bug fixes In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	7f9b28b0c6	ARI: Improve wiki documentation This patch improves the documentation of ARI on the wiki. Specifically, it addresses the following: * Allowed values and allowed ranges weren't documented. This was particularly frustrating, as Asterisk would reject query parameters with disallowed values - but we didn't tell anyone what the allowed values were. * The /play/id operation on /channels and /bridges failed to document all of the added media resource types. * Documentation for creating a channel into a Stasis application failed to note when it occurred, and that creating a channel into Stasis conflicts with creating a channel into the dialplan. * Some other minor tweaks in the mustache templates, including italicizing the parameter type, putting the default value on its own sub-bullet, and some other nicities. Review: https://reviewboard.asterisk.org/r/4351 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	42b342c6e2	Add the ability to continue and originate using priority labels. With this patch, the following two ARI commands POST /channels POST /channels/{id}/continue Accept a new parameter, label, that can be used to continue to or originate to a priority label in the dialplan. Because this is adding a new parameter to ARI commands, the API version of ARI has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 #close Reported by Nir Simionovich Review: https://reviewboard.asterisk.org/r/4285 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	74b032bb03	ari: Add support for specifying an originator channel when originating. If an originator channel is specified when originating a channel the linked ID of it will be applied to the newly originated outgoing channel. This allows an association to be made between the two so it is known that the originator has dialed the originated channel. ASTERISK-24552 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429153 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
Jonathan Rose	a69e0cd32a	ARI: Remove unnecessary \briefs from automatically generated documentation Review: https://reviewboard.asterisk.org/r/3440/ ........ Merged revisions 412653 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a8742e327f	ARI: Add tones playback resource Adds a tones URI type to the playback resource. The tone can be specified by name (from indications.conf) or by a tone pattern. In addition, tonezone can be specified in the URI (by appending ;tonezone=<zone>). Tones must be stopped manually in order for a stasis control to move on from playback of the tone. Tones may be paused, resumed, restarted, and stopped. They may not be rewound or fast forwarded (tones can't be controlled in a way that lets you skip around from note to note and pausing and resuming will also restart the tone from the beginning). Tests are currently in development for this feature (https://reviewboard.asterisk.org/r/3428/). (closes issue ASTERISK-23433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3427/ ........ Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1590d32ab0	ARI: Support channel variables in originate This adds back in support for specifying channel variables during an originate without compromising the ability to specify query parameters in the JSON body. This was accomplished by generating the body-parsing code in a separate function instead of being integrated with the URI query parameter parsing code such that it could be called by paths with body parameters. This is transparent to the user of the API and prevents manual duplication of code or data structures. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan ........ Merged revisions 406003 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	40a7f68e4b	ari: Remove support for specifying channel vars during origination. When we added support for specifying channel variables for an origination, we didn't consider how that would interact with another feature, namely specifying request parameters in a JSON request body. The method of specifying channel variables (as a flat JSON object passed in the JSON body) interferes with parsing parameters out of the request body. Unfortunately, fixing this would be a backward incompatible change. In the interest of keeping the API sane and keeping our release schedule, we're dropping the feature for specifying channel variables in the origination request. We will bring the feature back soon, as a backward compatible addition to the API. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3088 ........ Merged revisions 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f425c4a086	ARI: Allow specifying channel variables during a POST /channels Added the ability to specify channel variables when creating/originating a channel in ARI. The variables are sent in the body of the request and should be formatted as a single level JSON object. No nested objects allowed. For example: {"variable1": "foo", "variable2": "bar"}. (closes issue ASTERISK-22872) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3052/ ........ Merged revisions 403752 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	eda7126862	ari: Add Snoop operation for spying/whispering on channels. The Snoop operation can be invoked on a channel to spy or whisper on it. It returns a channel that any channel operations can then be invoked on (such as record to do monitoring). (closes issue ASTERISK-22780) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3003/ ........ Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	d1ad4a95f8	ari: Add silence generator controls This patch adds the ability to start a silence generator on a channel via ARI. This generator will play silence on the channel (avoiding audio timeouts on the peer) until it is stopped, or some other media operation is started (like playing media, starting music on hold, etc.). (closes issue ASTERISK-22514) Review: https://reviewboard.asterisk.org/r/3019/ ........ Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	67b650543c	res_ari_channels: Add the ability to stop locally generated ringing on a channel. Using the 'ring' operation it is possible to start locally generated ringback if the channel is answered. This change adds the ability to stop it by using DELETE. ........ Merged revisions 402804 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7d0d1a1efb	ari: User better nicknames for ARI operations While working on building client libraries from the Swagger API, I noticed a problem with the nicknames. channel.deleteChannel() channel.answerChannel() channel.muteChannel() Etc. We put the object name in the nickname (since we were generating C code), but it makes OO generators redundant. This patch makes the nicknames more OO friendly. This resulted in a lot of name changing within the res_ari_.so modules, but not much else. There were a couple of other fixed I made in the process. When reversible operations (POST /hold, POST /unhold) were made more RESTful (POST /hold, DELETE /unhold), the path for the second operation was left in the API declaration. This worked, but really the two operations should have been on the same API. * The POST /unmute operation had still not been REST-ified. Review: https://reviewboard.asterisk.org/r/2940/ ........ Merged revisions 402528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7678fd040e	res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media. The ring operation sends ringing to the specified channel it is invoked on. The dtmf operation can be used to send DTMF digits to the specified channel of a specific length with a wait time in between. Finally hangup reasons allow you to specify why a channel is being hung up (busy, congestion). Early media behavior has also been tweaked slightly. When playing media to a channel it will no longer automatically answer. If it has not been answered a progress indication is sent instead. (closes issue ASTERISK-22701) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2916/ ........ Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	aa7f9e55f2	ARI: Remove channels/{channelId}/dial This removes the /ari/channels/{channelId}/dial URI since it is redundant, overly complex, is likely to become more externally complex over time, and is too high-level compared with other ARI operations. See the following for further information: http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html (closes issue ASTERISK-22784) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2968/ ........ Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d183c6e134	Return a channel snapshot when originating using ARI, and subscribe the Stasis application to it. This change allows a user of ARI to know what channel it has originated and also follow any progress. If a Stasis application is provided it will be automatically subscribed to the originated channel immediately. (closes issue ASTERISK-22485) Reported by: David Lee Review: https://reviewboard.asterisk.org/r/2910/ ........ Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2fc0a8873c	Clarify documentation for channel and bridge list This makes it clear that the ARI API calls for listing channels and bridges will list all channels or bridges in the system and not just those that are in or are controlled by a Stasis application. (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9234804a3b	Multiple revisions 400508,400842-400843,400848 ........ r400508 \| dlee \| 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) \| 1 line Corrected response class for stopPlayback ........ r400842 \| dlee \| 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) \| 1 line Correct some ARI wiki rendering errors ........ r400843 \| dlee \| 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) \| 1 line Updated /play resource docs. The playback of http: resources isn't implemented... yet ........ r400848 \| dlee \| 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) \| 5 lines Fix a stupid copy/paste error in ARI docs. Patches: ari-doc-patch.txt uploaded by jbigelow (license 5091) ........ Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	0bcc676d09	Multiple revisions 398638-398639 ........ r398638 \| dlee \| 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) \| 1 line Added note about expected behavior of originate ........ r398639 \| dlee \| 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) \| 1 line Added note about expected behavior of originate (the rest of the commit) ........ Merged revisions 398638-398639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d8956f690e	Rename everything Stasis-HTTP to ARI This renames all files and API calls from several variants of Stasis-HTTP to ARI including: * Stasis-HTTP -> ARI * STASIS_HTTP -> ARI * stasis_http -> ari (ast_ari for global symbols, file names as well) * stasis http -> ARI Review: https://reviewboard.asterisk.org/r/2706/ (closes issue ASTERISK-22136) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

24 Commits (12aa25b2e150068abe3cac98d2d85498c3194ef0)