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

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

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

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

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

New functionality has been added so the version has been
bumped to one over the 13 version.

Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69

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

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

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

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

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

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

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

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

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

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

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

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

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

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

An http request can be sent to unload an Asterisk module. If the
module can not be unloaded or is already unloaded, an error response
will be returned.

The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, depending
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 unloaded through http requests

ASTERISK-25173

Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57

An http request can be sent to load an Asterisk module. If the
module can not be loaded or is loaded already, an error response
will be returned.

The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending 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 loaded through http requests

ASTERISK-25173

Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33

An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.

The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending 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
* Information on a single module can now be retrieved

ASTERISK-25173

Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463

An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.

The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending 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
* Information on modules can now be retrieved

Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0

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 9d8a4623)

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.
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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
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Sending the following ARI commands caused Asterisk to crash if the JSON
body 'variables' object passes values of types other than strings.

POST /ari/channels
POST /ari/channels/{channelid}
PUT /ari/endpoints/sendMessage
PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),
ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and
ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close
Reported by:  jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/
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This change makes it so that if a channel variable is requested and it does not exist
a 404 response will be returned instead of an allocation failed response. This makes
it easier to debug and figure out what is going on for a user.

ASTERISK-24677 #close
Reported by: Joshua Colp
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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
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One of the canonical reasons for hanging up a channel is because the far end
failed to answer - or because someone else answered, and we want to get rid of
this channel. This patch adds the missing value to the 'reason' query parameter
for the DELETE /channels operation.

Review: https://reviewboard.asterisk.org/r/4400

ASTERISK-24745 #close
Reported by: Ben Merrills
patches:
  add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678)
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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
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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
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The channel "language" was already part of a channel snapshot, however is was
not sent out over AMI or ARI. This patch makes it so the channel "language" is
included in the appropriate AMI or ARI events.

ASTERISK-24553 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4245/
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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/
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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.
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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
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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
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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
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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/
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