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
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(closes issue ASTERISK-23007)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404185 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.

(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge.  In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.

Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.

(closes issue ASTERISK-22624)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2947/
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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/
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found.  Fixed so a "404 Not Found" will be returned
instead.

(closes issue ASTERISK-22803)
Reported by: David M. Lee
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Before playback was the only non plural resource.  It has been renamed to
playbacks for consistency.

(closes issue ASTERISK-22737)
Reported by: Paul Belanger
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402562 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402529 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22722)
Reported by: Richard Mudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402140 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk will now issue 422 if recording is requested against channels
or bridges with an unknown format

(closes issue ASTERISK-22626)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2939/
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Merged revisions 402001 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/
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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/
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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
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Review: https://reviewboard.asterisk.org/r/2911/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.

(closes issue ASTERISK-22627)
Reported by: Joshua Colp
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change updates how we handle toggle events, rather then create two
different function names, we'll just use POST / DELETE from HTTP to handle it.

Review: https://reviewboard.asterisk.org/r/2906/
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........
  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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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
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........
  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)
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his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.

The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.

There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.

The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)

(closes issue ASTERISK-21582)
Review: https://reviewboard.asterisk.org/r/2693/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds error checking to ARI bridge operations, when
adding/removing channels to/from bridges.

In general, the error codes fall out as follows:
 * Bridge not found - 404 Not Found
 * Bridge not in Stasis - 409 Conflict
 * Channel not found - 400 Bad Request
 * Channel not in Stasis - 422 Unprocessable Entity
 * Channel not in this bridge (on remove) - 422 Unprocessable Entity

(closes issue ASTERISK-22036)
Review: https://reviewboard.asterisk.org/r/2769/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds ARI functions to be able to turn on/off music on hold in a
bridge. It actually functions more as a background music without
further actions on the bridge since if the rest of the channels
in the bridge aren't explicitly muted, they will still be able
to communicate.

(closes issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2688/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Both /asterisk/variable and /channel/{channelId}/variable requires a
?variable parameter to be passed into the query. But we weren't checking
for the parameter being missing, which caused a segfault.

All calls now properly return 400 Bad Request errors when the parameter
is missing. The Swagger api-docs were updated accordingly.

(closes issue ASTERISK-22273)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397306 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.

(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3