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  1. Jul 20, 2014
  3. Jul 18, 2014
  5. Jul 15, 2014
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  15. May 21, 2014
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  25. Apr 15, 2014
  27. Apr 12, 2014
  29. Apr 08, 2014
  31. Apr 05, 2014
    • Russell Bryant's avatar
      func_periodic_hook: New function for periodic hooks. · ea290b2c
      Russell Bryant authored 
      This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically.  The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded.  The implementation is much
more generic though and could be used for many other things.

The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.

The other important bit of the implementation is how it figures out
when to trigger the beep playback.  This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way.  It's a convenient way to get a callback and check if it's time
to kick off another beep.  It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.

Review: https://reviewboard.asterisk.org/r/3362/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ea290b2c
  33. Mar 27, 2014
  35. 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
  37. Mar 06, 2014
    • George Joseph's avatar
      sorcery: Create AST_SORCERY dialplan function. · a4906e9f
      George Joseph authored 
      This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file.  It's similar to 
AST_CONFIG.

The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects.  The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify.  You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html

So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...

* Creates ast_variable_list_append which is a helper to append one ast_variable
  list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
  already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
  type preference...a single ast_variable with all values concatenated or an
  ast_variable list with multiple entries.  Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
  definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
  sorcery_fields_handler handlers so they return multiple occurrences as an
  ast_variable_list.
* Added a whole bunch of tests to test_sorcery.

(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a4906e9f
  39. Mar 05, 2014
  41. Mar 04, 2014
  43. Feb 07, 2014
  45. Feb 04, 2014
  47. Jan 14, 2014
    • Richard Mudgett's avatar
      verbosity: Fix performance of console verbose messages. · 828f339a
      Richard Mudgett authored 
      The per console verbose level feature as previously implemented caused a
large performance penalty.  The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version.  If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console.  If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.

* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.

* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.

* Added a silent option to the "core set verbose" command.

* Fixed "core set debug off" tab completion.

* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.

* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section.  The default is now to once again follow
the current root console level.  As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.

(closes issue AST-1252)
Reported by: Guenther Kelleter

Review: https://reviewboard.asterisk.org/r/3114/
........

Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      828f339a
  49. Jan 12, 2014
  51. Jan 02, 2014
  53. Dec 19, 2013
    • Richard Mudgett's avatar
      Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. · e4803bbd
      Richard Mudgett authored 
      This change is in preparation for external MWI support.

Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context.  The
only exception is the legacy hasvoicemail users.conf option.  The legacy
option will only work for app_voicemail mailboxes.  The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.

chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier.  chan_sip just stored and
compared the two components.  chan_dahdi actually used the box
information.

The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number.  As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.

Review: https://reviewboard.asterisk.org/r/3072/
........

Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e4803bbd
    • Matthew Jordan's avatar
      app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state · 7e9febbf
      Matthew Jordan authored 
      When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".

This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.

While going through this, the following changes were also made:
 * DISA, which can reset the CDR when a user successfully authenticates, now
   just uses the ResetCDR app to do this. This prevents having to duplicate
   the same Stasis synchronization logic in that application.
 * Answer no longer disables CDRs. It actually didn't work anyway - calling
   DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
   time - it just kills all CDRs on that channel, which isn't what the caller
   would intend.

(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)

Review: https://reviewboard.asterisk.org/r/3057/
........

Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7e9febbf
  55. Dec 18, 2013
  57. Dec 16, 2013
  59. Dec 11, 2013
  61. Dec 05, 2013
  63. Dec 03, 2013
  65. Oct 23, 2013
  67. Oct 08, 2013