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20 authors
  1. Apr 10, 2015
  3. Apr 07, 2015
    • Matthew Jordan's avatar
      ARI: Add the ability to intercept hold and raise an event · c2f50ba6
      Matthew Jordan authored 
      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
........

Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c2f50ba6
  5. Feb 27, 2015
  7. Feb 21, 2015
  9. Feb 12, 2015
    • Matthew Jordan's avatar
      ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app · 29f66b04
      Matthew Jordan authored 
      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
........

Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      29f66b04
  11. Feb 09, 2015
  13. Jan 27, 2015
    • Matthew Jordan's avatar
      ARI: Improve wiki documentation · fb8a2e03
      Matthew Jordan authored 
      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
........

Merged revisions 431145 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      fb8a2e03
  15. Jan 07, 2015
  17. Dec 09, 2014
  19. Dec 08, 2014
  21. Sep 20, 2014
  23. Aug 20, 2014
  25. Aug 11, 2014
  27. Aug 07, 2014
  29. Aug 05, 2014
    • Matthew Jordan's avatar
      Multiple revisions 420089-420090,420097 · 47bf7efc
      Matthew Jordan authored 
      ........
  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
........
  r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
  
  Remove automerge properties :-(
........
  r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
  
  test_message: Fix strict-aliasing compilation issue
........

Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      47bf7efc
  31. Jul 25, 2014
    • Matthew Jordan's avatar
      Multiple revisions 419565-419566 · 355dc3d2
      Matthew Jordan authored 
      ........
  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
........
  r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) | 1 line
  
  Update CHANGES for r419565
........

Merged revisions 419565-419566 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      355dc3d2
  33. Jul 22, 2014
    • Matthew Jordan's avatar
      ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech · bb87796f
      Matthew Jordan authored 
      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
........

Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bb87796f
  35. Jul 18, 2014
  37. Jul 08, 2014
  39. Jul 03, 2014
  41. May 30, 2014
  43. May 28, 2014
  45. May 22, 2014
    • Scott Griepentrog's avatar
      ARI: Add ability to raise arbitrary User Events · cf21644d
      Scott Griepentrog authored 
      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/
........

Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cf21644d
  47. Apr 18, 2014
  49. Apr 17, 2014
  51. Mar 28, 2014
  53. Mar 19, 2014
  55. Mar 07, 2014
    • Scott Griepentrog's avatar
      uniqueid: channel linkedid, ami, ari object creation with id's · 80ef9a21
      Scott Griepentrog authored 
      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/
........

Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      80ef9a21
  57. Mar 06, 2014
  59. Feb 05, 2014
  61. Feb 01, 2014
  63. Jan 21, 2014
  65. Jan 14, 2014
  67. Dec 20, 2013
  69. Dec 18, 2013
  71. Dec 17, 2013
  73. Dec 14, 2013