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20 authors
  1. Apr 17, 2018
    • George Joseph's avatar
      bridge_softmix: Forward TEXT frames · 4fb7967c
      George Joseph authored 
      Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge.  res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.

res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame.  On a normal
point-to-point call, the frames are forwarded between the two
correctly.  bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants.  Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.

* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload.  A channel
driver can queue a frame of that type when it receives a message
from outside.  A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties.  If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this.  Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.

* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel.  This allows the chat client user to set a friendly name
for the chat.

* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).

* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.

* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.

* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.

Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
      4fb7967c
  3. Dec 22, 2017
  5. Sep 21, 2017
    • Joshua Colp's avatar
      bridge: Change participant SFU streams when source streams change. · f2985e31
      Joshua Colp authored 
      Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.

This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.

ASTERISK-27277

Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
      f2985e31
  7. Mar 07, 2017
    • Joshua Colp's avatar
      core: Add stream topology changing primitives with tests. · 3ed05bad
      Joshua Colp authored 
      This change adds a few things to facilitate stream topology changing:

1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.

2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.

Tests have also been written which confirm the multistream and
non-multistream behavior.

ASTERISK-26839

Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9
      3ed05bad
  9. Jan 23, 2017
    • Lorenzo Miniero's avatar
      media: Add experimental support for RTCP feedback. · 1061539b
      Lorenzo Miniero authored 
      This change adds experimental support for providing RTCP
feedback information to codec modules so they can dynamically
change themselves based on conditions.

ASTERISK-26584

Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857
      1061539b
  11. Oct 27, 2016
    • Corey Farrell's avatar
      Remove ASTERISK_REGISTER_FILE. · a6e5bae3
      Corey Farrell authored 
      ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
      a6e5bae3
  13. Apr 13, 2015
    • Matt Jordan's avatar
      git migration: Refactor the ASTERISK_FILE_VERSION macro · 4a582616
      Matt Jordan authored 
      Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
      4a582616
  15. Oct 03, 2014
  17. Jul 28, 2014
  19. Jul 25, 2014
  21. Jul 20, 2014
  23. May 09, 2014
  25. Mar 27, 2014
  27. Mar 17, 2014
    • Mark Michelson's avatar
      Fix stuck channel in ARI through the introduction of synchronous bridge actions. · d44aefee
      Mark Michelson authored 
      Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.

The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
  would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
  playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
  not to occur (the bridge dies, the channel is removed from the bridge), then we would
  never be notified.

The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.

Review: https://reviewboard.asterisk.org/r/3338
........

Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d44aefee
  29. Aug 06, 2013
    • David M. Lee's avatar
      ARI: Add recording controls · c7908487
      David M. Lee authored 
      This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c7908487
  31. May 21, 2013
  33. Jan 31, 2013
  35. Jan 22, 2013
    • Matthew Jordan's avatar
      Add ControlPlayback manager action · 7d9871b3
      Matthew Jordan authored 
      This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.

Review: https://reviewboard.asterisk.org/r/2265/

(closes issue ASTERISK-20882)
Reported by: mjordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7d9871b3
  37. May 14, 2012
    • Kinsey Moore's avatar
      Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) · b5a6de76
      Kinsey Moore authored 
      This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b5a6de76
  39. Jan 09, 2012
    • Terry Wilson's avatar
      Replace direct access to channel name with accessor functions · 04da92c3
      Terry Wilson authored 
      There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      04da92c3
  41. Sep 16, 2011
  43. Sep 09, 2011
    • Matthew Jordan's avatar
      Merged revisions 335078 via svnmerge from · 8b5ba33f
      Matthew Jordan authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
  
  Merged revisions 335064 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
    
    Updated SIP 484 handling; added Incomplete control frame
    
    When a SIP phone uses the dial application and receives a 484 Address 
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.
    
    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to 
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.
    
    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan
    
    Review: https://reviewboard.asterisk.org/r/1416/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8b5ba33f
  45. Jul 14, 2011
  47. Feb 07, 2011
  49. Feb 03, 2011
  51. Sep 20, 2010
    • David Vossel's avatar
      Merged revisions 287647 via svnmerge from · 2f3dee23
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
  
  Addition of the FrameHook API (AKA AwesomeHooks)
  
  So far all our tools for viewing and manipulating media streams
  within Asterisk have been entirely focused on audio.  That made
  sense then, but is not scalable now.  The FrameHook API lets us
  tap into and manipulate _ANY_ type of media or signaling passed
  on a channel present today or in the future.  This tool is a step
  in the direction of expanding Asterisk's boundaries and will help
  generate some rather interesting applications in the future.
  
  In addition to the FrameHook API, a simple dialplan function
  exercising the api has been included as well.  This function
  is called FRAME_TRACE().  FRAME_TRACE() allows for the internal
  ast_frames read and written to a channel to be output.  Filters
  can be placed on this function to debug only certain types of frames.
  This function could be thought of as an internal way of doing
  ast_frame packet captures.
  
  Review: https://reviewboard.asterisk.org/r/925/
........



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2f3dee23