Commit Graph

53 Commits

Author SHA1 Message Date
Alexander Traud
6988386234 stasis: Fix for Doxygen.
ASTERISK-29750

Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9
2021-11-18 13:25:54 -06:00
Jean Aunis
034ac357ad ARI: Ability to inhibit COLP frames when adding channels to a bridge
This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.

ASTERISK-28629

Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc
2020-01-02 15:06:15 +00:00
Ben Ford
6626df586e res_stasis: Add ability to switch applications.
Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:

client.channels.move(channelId, app, appArgs)

The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.

ASTERISK-28267 #close

Change-Id: I43d12b10045a98a8d42541889b85695be26f288a
2019-03-07 07:53:01 -06:00
Kevin Harwell
8681fc9db7 ARI event type filtering
Event type filtering is now enabled, and configurable per application. An app is
now able to specify which events are sent to the application by configuring an
allowed and/or disallowed list(s). This can be done by issuing the following:

PUT /applications/{applicationName}/eventFilter

And then enumerating the allowed/disallowed event types as a body parameter.

ASTERISK-28106

Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b
2019-02-20 09:56:22 -06:00
Corey Farrell
527cf5a570 Remove redundant module checks and references.
This removes references that are no longer needed due to automatic
references created by module dependencies.

In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.

Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
2018-01-24 13:37:29 -05:00
Seán C McCord
e8bde6916a ari/bridge: Add mute, dtmf suppression controls
Add bridge_features structure to bridge creation.  Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.

ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI

Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
2017-10-11 08:23:05 -04:00
Corey Farrell
f9695dc057 Forward declare 'struct ast_json' in asterisk.h
The ast_json structure is used in many Asterisk headers and is often the
only part of json.h used.  This adds a forward declaration to asterisk.h
and removes the include of json.h from many headers.  The declaration
has been left in endpoints.h and stasis.h to avoid problems with source
files that use ast_json functions without directly including json.h.

ari.h continues to include json.h as it uses enum
ast_json_encoding_format.

Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769
2017-03-30 22:59:59 -05:00
George Joseph
6691606723 ari: Implement 'debug all' and request/response logging
The 'ari set debug' command has been enhanced to accept 'all' as an
application name.  This allows dumping of all apps even if an app
hasn't registered yet.  To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.

'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.

* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
  to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
  failed, the consumption of the body was moved from the ari stubs
  to ast_ari_callback in res_ari.c and the moustache templates were
  then regenerated.  The body is now passed to ast_ari_invoke and then
  on to the handlers.  This results in code savings since that template
  was inserted multiple times into all the stubs.

An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function.  The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.

Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874f3)
2017-01-23 10:25:58 -07:00
Matt Jordan
c30d677333 res/stasis: Add CLI commands for displaying/debugging ARI apps
This patch adds three new CLI commands:
 - ari show apps: list the registered ARI applications
 - ari show app: show detailed information about an ARI application
 - ari set debug: dump events being sent to an ARI application

Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.

ASTERISK-26488 #close

Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
2016-11-01 09:43:46 -05:00
Mark Michelson
11caa10cf5 ARI: Ensure announcer channels are destroyed.
Announcer channels were not being destroyed because the
stasis_app_control structure that referenced them was not being
destroyed. The control structure was not being destroyed because it was
not being unlinked from its container. It was not being unlinked from
its container because the after bridge callback for the announcer
channel was not being run. The after bridge callback was not being run
because the after bridge datastore was not being removed from the
channel on destruction. The channel was not being destroyed because the
hangup that used to destroy the channel was now only reducing the
reference count to one. The reference count of the channel was only
being reduced to one because the stasis_app_control structure was
holding the final reference...

The control structure used to not keep a reference to the channel, so
that loop described above did not happen.

The solution is to manually remove the control structure from its
container when the playback on a bridge is complete.

ASTERISK-26083 #close
Reported by Joshua Colp

Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4
2016-06-20 09:41:26 -05:00
Mark Michelson
88d997913f ARI: Re-implement the ARI dial command, allowing for early bridging.
ARI dial had been implemented using the Dial API. This made great sense
when dialing was 100% separate from bridging. However, if a channel were
to be added to a bridge during the dial attempt, there would be a
conflict between the dialing thread and the bridging thread. Each would
be attempting to read frames from the dialed channel and act on them.

The initial attempt to make the two play nice was to have the Dial API
suspend the channel in the bridge and stay in charge of the channel
until the dial was complete. The problem with this was that it was
riddled with potential race conditions. It also was not well-suited for
the case where the channel changed which bridge it was in during the
dial.

This new approach removes the use of the Dial API altogether. Instead,
the channel we are dialing is placed into an invisible ARI dialing
bridge. The bridge channel thread handles incoming frames from the
channel. If the channel is added to a real bridge, it is departed from
the invisible bridge and then added to the real bridge. Similarly, if
the channel is removed from the real bridge, it is automatically added
back to the invisible bridge if the dial attempt is still active.

This approach keeps the threading simple by always having the channel
being handled by bridge channel threads.

ASTERISK-25925

Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb
2016-05-27 09:08:49 -05:00
Joshua Colp
72ef79dc2d Merge "ARI: Add method to Dial a created channel." 2016-04-06 05:43:47 -05:00
Mark Michelson
abbb2edd4c ARI: Add method to Dial a created channel.
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
2016-04-05 18:14:17 -05:00
Richard Mudgett
0ea742d33a res_stasis: Add control ref to playback and recording structs.
The stasis_app_playback and stasis_app_recording structs need to have a
struct stasis_app_control ref.  Other threads can get a reference to the
playback and recording structs from their respective global container.
These other threads can then use the control pointer they contain after
the control struct has gone.

* Add control ref to stasis_app_playback and stasis_app_recording structs.

With the refs added, the control command queue can now have a circular
control reference which will cause the control struct to never get
released if the control's command queue is not flushed when the channel
leaves the Stasis application.  Also the command queue needs better
protection from adding commands if the control->is_done flag is set.

* Flush the control command queue on exit.

ASTERISK-25882 #close

Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d
2016-03-30 16:36:20 -05:00
Matt Jordan
b99a705262 ARI: Add the ability to subscribe to all events
This patch adds the ability to subscribe to all events. There are two possible
ways to accomplish this:
(1) On initial WebSocket connection. This patch adds a new query parameter,
    'subscribeAll'. If present and True, Asterisk will subscribe the
    applications to all ARI events.
(2) Via the applications resource. When subscribing in this manner, an ARI
    client should merely specify a blank resource name, i.e., 'channels:'
    instead of 'channels:12354'. This will subscribe the application to all
    resources of the 'channels' type.

ASTERISK-24870 #close

Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6
2015-09-22 09:59:47 -05:00
Matthew Jordan
29f66b0429 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
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2015-02-12 20:34:37 +00:00
Kevin Harwell
ca02121ef7 Investigate and fix memory leaks in Asterisk
Fixed memory leaks that were found in Asterisk.

ASTERISK-24693 #close
Reported by:  Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4347/
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2015-01-23 15:21:56 +00:00
Kinsey Moore
74e706878b Stasis: Fix StasisEnd message ordering
This change corrects message ordering in cases where a channel-related
message can be received after a Stasis/ARI application has received the
StasisEnd message. The StasisEnd message was being passed to
applications directly without waiting for the channel topic to empty.

As a result of this fix, other bugs were also identified and fixed:
* StasisStart messages were also being sent directly to apps and are
  now routed through the stasis message bus properly
* Masquerade monitor datastores were being removed at the incorrect
  time in some cases and were causing StasisEnd messages to not be sent
* General refactoring where necessary for the above
* Unsubscription on StasisEnd timing changes to prevent additional
  messages from following the StasisEnd when they shouldn't

A channel sanitization function pointer was added to reduce processing
and AO2 lookups.

Review: https://reviewboard.asterisk.org/r/4163/
ASTERISK-24501 #close
Reported by: Matt Jordan
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2014-11-13 15:46:48 +00:00
Matthew Jordan
ba5d5da60b Improve call forwarding reporting, especially with regards to ARI.
This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
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2014-08-18 00:57:01 +00:00
Kinsey Moore
6f735288b0 Stasis: Allow internal channels directly into bridges
The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.

Review: https://reviewboard.asterisk.org/r/3903/
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2014-08-11 18:38:15 +00:00
Kinsey Moore
0ac7f96057 Stasis: Convey transfer information to applications
This fixes a class of issues where Stasis applications were not made
aware that their channels were being manipulated or replaced by
external entitiessuch as transfers, AMI commands, or dialplan
applications such as Bridge(). Inconsistent information such as
StasisEnd events with unknown channels as a result of masquerades has
also been corrected. To accomplish these fixes, several new fields
were added to blind and attended transfer messages as well as
StasisStart and BridgeAttendedTransfer Stasis events.

ASTERISK-23941 #close
Review: https://reviewboard.asterisk.org/r/3865/
Review: https://reviewboard.asterisk.org/r/3857/
Review: https://reviewboard.asterisk.org/r/3852/
Review: https://reviewboard.asterisk.org/r/3816/
Review: https://reviewboard.asterisk.org/r/3731/
Review: https://reviewboard.asterisk.org/r/3729/
Review: https://reviewboard.asterisk.org/r/3728/
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2014-08-07 15:30:19 +00:00
Matthew Jordan
d4b436d0ea ARI/res_stasis: Subscribe to both Local channel halves when originating to app
This patch fixes two bugs:

1. When originating a channel into a Stasis application, we already create a
   subscription for the channel that is going into our Stasis app.
   Unfortunately, when you create a Local channel and pass it off to a Stasis
   app, you really aren't creating just one channel: you're creating two. This
   patch snags the second half of the Local channel pair (assuming it is a
   Local channel pair, but luckily core_local is kind about such assumptions)
   and subscribes to it as well.

2. Subscriptions are a bit sticky right now. If a subscription is made, the
   'interest' count gets bumped on the Stasis subscription - but unless
   something explicitly unsubscribes the channel, said subscription sticks
   around. This is not much of a problem is a user is creating the subscription
   - if they made it, they must want it. However, when we are creating
   implicit subscriptions, we need to make sure something clears them out.
   This patch takes a pessimistic approach: it watches the cache updates
   coming from Stasis and, if we notice that the cache just cleared out an
   object, we delete our subscription object. This keeps our ao2 container of
   Stasis forwards in an application from growing out of hand; it also is a
   bit more forgiving for end users who may not realize they were supposed to
   unsubscribe from that channel that just hung up.

Review: https://reviewboard.asterisk.org/r/3710/
#ASTERISK-23939 #close
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2014-07-07 02:15:00 +00:00
Scott Griepentrog
cf21644d6a ARI: Add ability to raise arbitrary User Events
User events can now be generated from ARI.  Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots.  An application must be specified which will receive
the event message (other applications can subscribe to it).  The message
will also be delivered via AMI provided a channel is attached.  Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.

This change also introduces the multi object blob mechanism used to send
multiple snapshot types in a single message.  The dialplan app UserEvent
was also changed to use multi object blob, and a new stasis message type
created to handle them.

ASTERISK-22697 #close
Review: https://reviewboard.asterisk.org/r/3494/
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2014-05-22 16:09:51 +00:00
Jonathan Rose
b9d7dfcc62 ARI: Make bridges/{bridgeID}/play queue sound files
Previously multiple play actions against a bridge at one time would cause
the sounds to play simultaneously on the bridge. Now if a sound is already
playing, the play action will queue playback to occur after the completion
of other sounds currently on the queue.

(closes issue ASTERISK-22677)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3379/
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2014-04-18 20:09:24 +00:00
Kinsey Moore
b4210a0081 Stasis: Add a usage note on stasis_app_get_bridge
This function returns an ast_bridge without a refcount bump and the
caller must increment the count if it intends to hold the pointer.

(closes issue ASTERISK-23588)
Review: https://reviewboard.asterisk.org/r/3450/
Reported by: Matt Jordan
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2014-04-16 19:14:37 +00:00
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/
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2014-03-07 15:47:55 +00:00
Jonathan Rose
b0bb03e916 bridging: Give bridges a name and a known creator
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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2013-12-17 23:25:49 +00:00
Kevin Harwell
ce18946de4 ARI: Adding a channel to a bridge while a live recording is active blocks
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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2013-12-13 16:38:57 +00:00
Kevin Harwell
ed48377994 ARI: Implement device state API
Created a data model and implemented functionality for an ARI device state
resource.  The following operations have been added that allow a user to
manipulate an ARI controlled device:

Create/Change the state of an ARI controlled device
PUT    /deviceStates/{deviceName}&{deviceState}

Retrieve all ARI controlled devices
GET    /deviceStates

Retrieve the current state of a device
GET    /deviceStates/{deviceName}

Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}

The ARI controlled device must begin with 'Stasis:'.  An example controlled
device name would be Stasis:Example.  A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events.  Any device state, ARI controlled or not, can be subscribed to.

While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same.  Each event resource must now register itself in order to be able
to properly handle [un]subscribes.

(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
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2013-11-23 17:48:28 +00:00
Kinsey Moore
d9015a5356 ARI: Don't leak implementation details
This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.

This prevents unhelpful error messages from being generated by
ast_json_pack.

This also corrects a bug where BridgeCreated events would not be
created.

(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
........

Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-22 20:10:46 +00:00
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
2013-11-21 15:56:34 +00:00
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
2013-11-13 23:11:32 +00:00
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
2013-11-01 14:38:21 +00:00
Joshua Colp
4053f36a71 res_ari_channels: Fix a deadlock when originating multiple channels close to eachother.
If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.

(closes issue ASTERISK-22768)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2979/
........

Merged revisions 402346 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 12:33:09 +00:00
Matthew Jordan
8d7873b836 ARI: Add subscription support
This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.

 * GET /applications - list current applications
 * GET /applications/{applicationName} - get details of a specific application
 * POST /applications/{applicationName}/subscription - explicitly subscribe to
   a channel, bridge or endpoint
 * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
   from a channel, bridge or endpoint

Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.

Review: https://reviewboard.asterisk.org/r/2862

(issue ASTERISK-22451)
Reported by: Matt Jordan
........

Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04 16:01:48 +00:00
Joshua Colp
83d55f85ec Allow specifying a channel to dial an extension and context in an ARI dial operation.
(issue ASTERISK-22625)
Reported by: Scott Griepentrog
........

Merged revisions 400254 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02 15:24:16 +00:00
David M. Lee
7a581537e8 ARI: Correct error codes for bridge operations
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
2013-08-23 17:19:02 +00:00
Jonathan Rose
21e22310c7 ARI: Music on Hold/Background Music for bridges
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
2013-08-23 00:26:19 +00:00
David M. Lee
987fdfb444 ARI: allow other operations to happen while bridged
This patch changes ARI bridging to allow other channel operations to
happen while the channel is bridged.

ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was bridged,
any other channel operations would queue up and execute only after the
channel left the bridge.

This patch changes ARI bridging so that channel commands can execute
while the channel is bridged. For most operations, things simply work
as expected. The one thing that ended up being a bit odd is recording.

The current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself may
be recording; it's recording a specific channel in the bridge that
fails. While this is an annoying limitation, channel recording is
still very useful for use cases such as voice mail, and bridge
recording makes up much of the difference for other use cases.

(closes issue ASTERISK-22084)
Review: https://reviewboard.asterisk.org/r/2726/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13 15:27:32 +00:00
Jonathan Rose
e47794ead1 ARI: bridges/{bridgeID}/addChannel: add roles parameter
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
2013-08-05 16:59:13 +00:00
Jonathan Rose
a6329a3acf ARI: MOH start and stop for a channel
(issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2680/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-19 19:40:27 +00:00
Jonathan Rose
17c546173f ARI: Bridge Playback, Bridge Record
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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-19 19:35:21 +00:00
Jason Parker
c1a7567d24 ARI: Add support for suppressing media streams.
Also convert res_mutestream to use the core feature behind this.

(closes issue ASTERISK-21618)

Review: https://reviewboard.asterisk.org/r/2652/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-18 16:03:12 +00:00
Jason Parker
87973eecff ARI: Add support for getting/setting channel and global variables.
This allows for reading and writing of functions on channels.

(closes issue ASTERISK-21868)

Review: https://reviewboard.asterisk.org/r/2641/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08 14:46:20 +00:00
Jason Parker
f820d24db1 ARI: Implement channel hold/unhold.
This puts the channel on hold (rather than queueing a frame from the channel).

(closes issue ASTERISK-21619)

Review: https://reviewboard.asterisk.org/r/2647/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-01 18:56:21 +00:00
Jason Parker
f41faf0b7d ARI: Implement channel dial.
This creates a new outbound channel, and bridges it to a channel already in
the Stasis application.

(closes issue ASTERISK-21620)

Review: https://reviewboard.asterisk.org/r/2634/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-01 18:19:15 +00:00
Jason Parker
609c42c854 ARI: Add support for continuing to a different location in dialplan.
This allows going elsewhere in the dialplan, so that the location can be
specified after exiting the Stasis application.

(closes issue ASTERISK-21870)

Review: https://reviewboard.asterisk.org/r/2644/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-26 19:29:57 +00:00
Kinsey Moore
a5bbc790e7 Stasis-HTTP: Flesh out bridge-related capabilities
This adds support for Stasis applications to receive bridge-related
messages when the application shows interest in a given bridge.

To supplement this work and test it, this also adds support for the
following bridge-related Stasis-HTTP functionality:
* GET stasis/bridges
* GET stasis/bridges/{bridgeId}
* POST stasis/bridges
* DELETE stasis/bridges/{bridgeId}
* POST stasis/bridges/{bridgeId}/addChannel
* POST stasis/bridges/{bridgeId}/removeChannel

Review: https://reviewboard.asterisk.org/r/2572/
(closes issue ASTERISK-21711)
(closes issue ASTERISK-21621)
(closes issue ASTERISK-21622)
(closes issue ASTERISK-21623)
(closes issue ASTERISK-21624)
(closes issue ASTERISK-21625)
(closes issue ASTERISK-21626)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10 13:07:11 +00:00
David M. Lee
557125664d This patch adds support for controlling a playback operation from the
Asterisk REST interface.

This adds the /playback/{playbackId}/control resource, which may be
POSTed to to pause, unpause, reverse, forward or restart the media
playback.

Attempts to control a playback that is not currently playing will
either return a 404 Not Found (because the playback object no longer
exists) or a 409 Conflict (because the playback object is still in the
queue to be played).

This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource.

(closes issue ASTERISK-21587)
Review: https://reviewboard.asterisk.org/r/2559


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 20:21:16 +00:00
David M. Lee
e8f4ac6c61 Break res_stasis into smaller files.
When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.

This patch breaks the major components of res_stasis.c into individual
files.

 * res/stasis/app.c - Stasis application tracking
 * res/stasis/control.c - Channel control objects
 * res/stasis/command.c - Channel command object

This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.

The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.

Review: https://reviewboard.asterisk.org/r/2530/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-14 21:45:08 +00:00