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 9d8a462356)
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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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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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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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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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
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r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
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r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
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r419565 | mjordan | 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines
ARI: report duration values in LiveRecording objects
This patch adds three new fields to the LiveRecording model:
- total_duration: the total length of the live recording
- talking_duration: optional. The duration of talking energy that was
detected while the recording was made.
- silence_duration: optional. The duration of silence that was detected while
the recording was made.
These values are reported in the RecordingFinished ARI event.
When a DSP is enabled on the channel during the recording - which occurs when
the recording is created with max_silence_seconds (indicating that the user
actually cares about how much silence is in the file), we will report the
talking_duration and silence_duration in addition to the total_duration.
Review: https://reviewboard.asterisk.org/r/3770/
ASTERISK-24037 #close
Reported by: Samuel Galarneau
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r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) | 1 line
Update CHANGES for r419565
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This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
for sending/receiving arbitrary out of call text messages through ARI in a
technology agnostic fashion.
The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
arbitrary technology defined URI. This is less straight forward, as
endpoints are formed from a tech + resource pair. We don't have a
mechanism to note that a technology that *may* have endpoints exists.
This patch provides such a mechanism, and fixes a few bugs along the way.
The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
most of the interesting bits (such as channel creation, destruction, state
changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
This resulted in endpoints missing the channel creation message, which
limited the usefulness of the subscription in the first place (a major use
case being 'tell me when this endpoint has a channel'). Unfortunately,
this meant another parameter to ast_channel_alloc. Since not all channel
technologies support an ast_endpoint, this patch makes such a call
optional and opts for a new function, ast_channel_alloc_with_endpoint.
When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.
Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:
channel PJSIP/foo-1 --
\
--> endpoint PJSIP/foo --
/ \
channel PJSIP/foo-2 -- \
---- > endpoint PJSIP
/
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --
ARI, through the applications resource, can:
- subscribe to endpoint:PJSIP/foo and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
- subscribe to endpoint:PJSIP/bar and get notifications for channels
PJSIP/bar-1 and endpoint PJSIP/bar
- subscribe to endpoint:PJSIP and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar
Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).
This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).
Review: https://reviewboard.asterisk.org/r/3760/
ASTERISK-23692
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This patch adds a new operation for stored recordings, copy. It takes an
existing stored recording and makes a copy of it in the same directory
or a relative directory under the stored recording directory.
/ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
This is particularly useful for voicemail-esque applications, which may need to
copy or move recordings around a directory structure.
Review: https://reviewboard.asterisk.org/r/3768/
ASTERISK-24036 #close
Reported by: Sam Galarneau
Tested by: Sam Galarneau
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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/
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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
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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/
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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/
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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
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This change turns the bridge type field into a comma separated list of attributes.
These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting
the various attributes a user can control the type of bridge created with the
behavior they need for their application.
(closes issue ASTERISK-23437)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3359/
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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/
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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
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Adds the following AMI commands:
PUT mailboxes/mailboxName
modifies mailbox state and implicitly creates new mailboxes
GET mailboxes/mailboxName
retrieves a JSON representation of a single mailbox if it exists
GET mailboxes
retrieves a JSON array of all mailboxes
DELETE mailbox/mailboxName
deletes a mailbox
Note that res_mwi_external must be loaded for these functions to
actually do anything.
Review: https://reviewboard.asterisk.org/r/3117/
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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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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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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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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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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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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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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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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 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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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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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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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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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/
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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/
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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)
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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/
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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/
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This patch adds basic system information access to ARI.
The results are roughly what you get from 'core show settings', with a
few minor differences.
* Data is structured, with 'build', 'system', 'config' and 'status'
sub-objects.
* Each sub-object is selectable, using the ?only= parameter. A comma
separated list can be provided to select multiple sections.
* A few config options are numeric, for which 0 means 'unlimited'.
Instead of having a special interpretation of those fields, they
are simply omitted if they're 0.
* The information is limited to what might be useful to building
external applications.
(closes issue ASTERISK-21575)
Review: https://reviewboard.asterisk.org/r/2702/
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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/
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Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
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Matt Jordan
Mark Michelson
Scott Emidy
Benjamin Ford
Richard Mudgett
Joshua Colp
Kevin Harwell
Kinsey Moore
Sam Galarneau
Scott Griepentrog
Jonathan Rose
David M. Lee
Paul Belanger
Jason Parker