Commit graph

61 commits

Author SHA1 Message Date
Matthew Jordan
df5dbbd878 res_stasis: Don't play MoH to channels by default when added to holding bridges
When ARI manipulates a bridge, it generally doesn't care what the mixing
technology is. Operations on a bridge initiated through ARI should perform
their action in generally the same way, regardless of the bridge's mixing
technology. While the mixing technology may determine how media flows to
channels, the actual operations on a bridge themselves should be the same.

Currently, this isn't the case with holding bridges. When a channel joins
without a role, MoH is started on that channel automatically. Subsequent bridge
operations that would stop MoH would fail (as there is no Announcer channel
playing MoH to the bridge). Starting MoH on the bridge will also create two
MoH streams: one from the MoH being played on the participant channel, and one
from the announcer channel. From the perspective of ARI users, this is
counter-intuitive - I would not expect MoH to be started for me. The mixing
technology determines how media is shared between participants, not the
application experience.

This patch does the following:
 * The Stasis bridge class now inspects channels as they are going into a
   bridge. If the bridge has a holding capability, and the channel has no
   roles, we give it a participant role and mark the default behaviour to have
   no entertainment. This allows addChannel operations to continue to set a
   participant role with an entertainment option if it felt like it (or could
   do it).
 * The music on hold channel is now Stasis approved (tm)

Review: https://reviewboard.asterisk.org/r/3929/

ASTERISK-24264 #close
Reported by: Samuel Galarneau
Tested by: Samuel Galarneau 
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2014-09-01 14:15:32 +00:00
Jonathan Rose
33835e17a0 ARI: Fix a crash caused by hanging during playback to a channel in a bridge
ASTERISK-24147 #close
Reported by: Edvin Vidmar
Review: https://reviewboard.asterisk.org/r/3908/
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2014-08-22 16:52:51 +00:00
Mark Michelson
56a1d4930a Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.
Because of the departable state of channels that enter Stasis bridges, Stasis has to
take responsibility for directing the channel to its intended after-bridge destination
if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures
that when such a move occurs, when the channel leaves the bridging system, any after
bridge gotos are honored.

Review: https://reviewboard.asterisk.org/r/3920
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2014-08-21 21:37:03 +00:00
Matthew Jordan
085d5a2629 Clean up files that do not end with newlines
Trivial patch to add new lines to several files missing them. This fixes
warnings when compiling with gcc 4.1.2 on CentOS 5.

ASTERISK-24245 #close
Reported by: Shaun Ruffell
patches:
  0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
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2014-08-21 14:52:28 +00:00
Kinsey Moore
36f4bff943 Stasis: Add information to blind transfer event
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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2014-08-20 13:06:33 +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
Richard Mudgett
aba07a0f6e res/stasis/command.c: Fix recent commit using spaces instead of tabs.
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2014-08-11 21:04:21 +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
ac5c75b45d stasis: Fix compilation issue with ao2 tagged objects
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2014-08-05 21:48:05 +00:00
Matthew Jordan
47bf7efc4d Multiple revisions 420089-420090,420097
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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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2014-08-05 21:44:09 +00:00
Richard Mudgett
f962448eee ARI: Make mixing bridges propagate linkedids and accountcodes.
* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.

* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.

* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.

* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.

AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/3720/
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2014-07-09 16:34:51 +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
Joshua Colp
10f4d0f65d res_stasis: Add progress indications to operations which perform media.
This change fixes operations which did not account for the fact that they may
be executed on channels which have not been answered. These operations will
now indicate progress when invoked.

ASTERISK-23560 #close
ASTERISk-23560 #comment Reported by: Jan Svoboda

Review: https://reviewboard.asterisk.org/r/3495/
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2014-04-30 12:42:42 +00:00
Joshua Colp
9b71a87108 res_stasis: Fix crash when handling a failed blind transfer message.
This changes fixes a crash that occurs when stasis determines if it
should send a message out to an application or not. The code
incorrectly assumed that a bridge snapshot would always be present
when in reality for failure cases it may not be.

ASTERISK-23573 #close
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2014-04-22 10:09:36 +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
Richard Mudgett
45ade68cb4 Remove unused RAII_VAR() declarations.
* Remove unused RAII_VAR() declarations.  The compiler cannot catch these
because the cleanup function "references" the unused variable.  Some
actually allocated and released resources that were never used.

* Fixed some whitespace issues in stasis_bridges.c.
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2014-04-15 18:01:47 +00:00
Matthew Jordan
2c5484c869 stasis/app.c: Add some extra debugging for subscription counts
Events are sent to a connected ARI application based on the things that ARI
application cares about. These subscriptions can be set up implicitly - such
as when that ARI application creates a new object - or explicitly, via the
application resource's subscription operations. Debugging *why* something was
being sent to an application - or why something was not being sent to an
application - was a bit tricky, as there was no debug information for the
subscriptions.

This patch adds some debug level 3 statements that show the subscription counts
for applications. (Level 3 was chosen as it matches the verbose level 3
statements elsewhere)
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2014-03-16 20:27:28 +00:00
Kinsey Moore
5247a0d990 ARI: Ensure managing application receives ChannelEnteredBridge messages
This fixes an issue where a Stasis application running over ARI and
subscribed to ari/events could miss the ChannelEnteredBridge event
because it did not subscribe to the new bridge fast enough.

To accomplish this, it subscribes the application controlling the
channel to the new bridge before adding it to that bridge which
required the stasis_app_control structure to maintain a reference to
the stasis_app.

(closes issue ASTERISK-23295)
Review: https://reviewboard.asterisk.org/r/3336/
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2014-03-13 19:33:22 +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
Joshua Colp
e5899852cc res_stasis: Enable transfers and provide events when they occur.
This change enables transfers within ARI created bridges and adds events
for when they occur. Unlike other events these will be received if *any*
subscribed object is involved in the transfer.

(closes issue ASTERISK-22984)
Reported by: David M. Lee

Review: https://reviewboard.asterisk.org/r/3120/
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2014-02-01 16:26:57 +00:00
Joshua Colp
3a5cc054ed res_stasis: Expose event for call forwarding and follow forwarded channel.
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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2013-12-14 17:19:41 +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
David M. Lee
fc70db3a81 ari: Fix deadlock problem with functions that use autoservice.
The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
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2013-12-05 22:09:20 +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/
........

Merged revisions 403134 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........

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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
d17a780333 res_stasis: Ensure the channel is always departed from the bridge when it leaves.
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.

(closes issue ASTERISK-22703)
Reported by: John Bigelow

(closes issue ASTERISK-22634)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2965/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 12:13:09 +00:00
David M. Lee
069da1e75a stasis: add functions embarrassingly missing from r400522
I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.
........

Merged revisions 402276 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 14:45:03 +00:00
Richard Mudgett
7eea4ab872 You'd think that new files would be free of whitespace issues. But you would be wrong.
........

Merged revisions 402003 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-10-25 22:03:04 +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
2de42c2a25 Multiple revisions 399887,400138,400178,400180-400181
........
  r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
  
  Minor performance bump by not allocate manager variable struct if we don't need it
........
  r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
  
  Stasis performance improvements
  
  This patch addresses several performance problems that were found in
  the initial performance testing of Asterisk 12.
  
  The Stasis dispatch object was allocated as an AO2 object, even though
  it has a very confined lifecycle. This was replaced with a straight
  ast_malloc().
  
  The Stasis message router was spending an inordinate amount of time
  searching hash tables. In this case, most of our routers had 6 or
  fewer routes in them to begin with. This was replaced with an array
  that's searched linearly for the route.
  
  We more heavily rely on AO2 objects in Asterisk 12, and the memset()
  in ao2_ref() actually became noticeable on the profile. This was
  #ifdef'ed to only run when AO2_DEBUG was enabled.
  
  After being misled by an erroneous comment in taskprocessor.c during
  profiling, the wrong comment was removed.
  
  Review: https://reviewboard.asterisk.org/r/2873/
........
  r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
  
  Taskprocessor optimization; switch Stasis to use taskprocessors
  
  This patch optimizes taskprocessor to use a semaphore for signaling,
  which the OS can do a better job at managing contention and waiting
  that we can with a mutex and condition.
  
  The taskprocessor execution was also slightly optimized to reduce the
  number of locks taken.
  
  The only observable difference in the taskprocessor implementation is
  that when the final reference to the taskprocessor goes away, it will
  execute all tasks to completion instead of discarding the unexecuted
  tasks.
  
  For systems where unnamed semaphores are not supported, a really
  simple semaphore implementation is provided. (Which gives identical
  performance as the original taskprocessor implementation).
  
  The way we ended up implementing Stasis caused the threadpool to be a
  burden instead of a boost to performance. This was switched to just
  use taskprocessors directly for subscriptions.
  
  Review: https://reviewboard.asterisk.org/r/2881/
........
  r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Optimize how Stasis forwards are dispatched
  
  This patch optimizes how forwards are dispatched in Stasis.
  
  Originally, forwards were dispatched as subscriptions that are invoked
  on the publishing thread. This did not account for the vast number of
  forwards we would end up having in the system, and the amount of work it
  would take to walk though the forward subscriptions.
  
  This patch modifies Stasis so that rather than walking the tree of
  forwards on every dispatch, when forwards and subscriptions are changed,
  the subscriber list for every topic in the tree is changed.
  
  This has a couple of benefits. First, this reduces the workload of
  dispatching messages. It also reduces contention when dispatching to
  different topics that happen to forward to the same aggregation topic
  (as happens with all of the channel, bridge and endpoint topics).
  
  Since forwards are no longer subscriptions, the bulk of this patch is
  simply changing stasis_subscription objects to stasis_forward objects
  (which, admittedly, I should have done in the first place.)
  
  Since this required me to yet again put in a growing array, I finally
  abstracted that out into a set of ast_vector macros in
  asterisk/vector.h.
  
  Review: https://reviewboard.asterisk.org/r/2883/
........
  r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Remove dispatch object allocation from Stasis publishing
  
  While looking for areas for performance improvement, I realized that an
  unused feature in Stasis was negatively impacting performance.
  
  When a message is sent to a subscriber, a dispatch object is allocated
  for the dispatch, containing the topic the message was published to, the
  subscriber the message is being sent to, and the message itself.
  
  The topic is actually unused by any subscriber in Asterisk today. And
  the subscriber is associated with the taskprocessor the message is being
  dispatched to.
  
  First, this patch removes the unused topic parameter from Stasis
  subscription callbacks.
  
  Second, this patch introduces the concept of taskprocessor local data,
  data that may be set on a taskprocessor and provided along with the data
  pointer when a task is pushed using the ast_taskprocessor_push_local()
  call. This allows the task to have both data specific to that
  taskprocessor, in addition to data specific to that invocation.
  
  With those two changes, the dispatch object can be removed completely,
  and the message is simply refcounted and sent directly to the
  taskprocessor.
  
  Review: https://reviewboard.asterisk.org/r/2884/
........

Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30 18:55:27 +00:00
Richard Mudgett
2a371cd80b Restore Dial, Queue, and FollowMe 'I' option support.
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.

* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.

* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.

* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.

* Made all callers of ast_bridge_impart() check the return value.  It is
important.  As a precaution, I also made the compiler complain now if it
is not checked.

* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.

An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.

(closes issue ASTERISK-22072)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/2845/
........

Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 22:19:23 +00:00
Kinsey Moore
53dbe10f5c Fix build warnings
When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.

(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
........

Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 18:53:32 +00:00
David M. Lee
451993f4f5 ARI: WebSocket event cleanup
Stasis events (which get distributed over the ARI WebSocket) are created
by subscribing to the channel_all_cached and bridge_all_cached topics,
filtering out events for channels/bridges currently subscribed to.

There are two issues with that. First was a race condition, where
messages in-flight to the master subscribe-to-all-things topic would get
sent out, even though the events happened before the channel was put
into Stasis. Secondly, as the number of channels and bridges grow in the
system, the work spent filtering messages becomes excessive.

Since r395954, individual channels and bridges have caching topics, and
can be subscribed to individually. This patch takes advantage, so that
channels and bridges are subscribed to on demand, instead of filtering
the global topics.

The one case where filtering is still required is handling BridgeMerge
messages, which are published directly to the bridge_all topic.

Other than the change to how subscriptions work, this patch mostly just
moves code around. Most of the work generating JSON objects from
messages was moved to .to_json handlers on the message types. The
callback functions handling app subscriptions were moved from res_stasis
(b/c they were global to the model) to stasis/app.c (b/c they are local
to the app now).

(closes issue ASTERISK-21969)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2754/
........

Merged revisions 397816 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-27 19:19:36 +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
David M. Lee
f5cca5e41e res_stasis: remove call to missing function control_continue.
In the shuffling around of res_stasis, control_continue was renamed to
stasis_app_control_continue, but the call in res_stasis wasn't updated.
In looking into it, it turns out it wasn't really the right thing to do
in res_stasis anyways.

This patch changes the handling of received a AST_CONTROL_HANGUP frame
to be the same as receiving a NULL frame, and removed the declaration of
control_continue(), since it doesn't exist any more.

(closes issue ASTERISK-22292)
Reported by: Denis Smirnov


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21 16:00:10 +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
Matthew Jordan
38236e54a8 Remove dead code from features.c; refactor pickup code into pickup.c
This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 02:32:44 +00:00
David M. Lee
e1b959ccbb Split caching out from the stasis_caching_topic.
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.

To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.

In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:

     single_topic  ---------------->  all_topic
           ^
           |
     single_topic_cached  ----+---->  all_topic_cached
                              |
                              +---->  cache

This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.

Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.

(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 13:49:34 +00:00
David M. Lee
b6f9b39830 Fix /stasis/res/app_replaced unit test.
A typo in recent changes caused the JSON ApplicationReplaced message to
fail to build, so the message wasn't being sent out the WebSocket.

Related, the replaced application would also unregister itself when it
disconnected, which would actually unregister the new application. This
was also fixed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-26 17:42:08 +00:00
Matthew Jordan
cafc115896 A great big renaming patch
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.

A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.

(closes issue ASTERISK-22130)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-25 04:06:32 +00:00
David M. Lee
f6a227a451 Continue events when ARI WebSocket reconnects
This patch addresses a bug in the /ari/events WebSocket in handling
reconnects.

When a Stasis application's associated WebSocket was disconnected and
reconnected, it would not receive events for any channels or bridges
it was subscribed to.

The fix was to lazily clean up Stasis application registrations,
instead of removing them as soon as the WebSocket goes away.

When an application is unregistered at the WebSocket level, the
underlying application is simply deactivated. If the application
WebSocket is reconnected, the application is reactivated for the new
connection.

To avoid memory leaks from lingering, unused application, the
application list is cleaned up whenever new applications are
registered/unregistered.

(closes issue ASTERISK-21970)
Review: https://reviewboard.asterisk.org/r/2678/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23 13:42:46 +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