Setting svn:ignore

git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403748 f38db490-d61c-443f-a65b-d21fe96a405b
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/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403749 f38db490-d61c-443f-a65b-d21fe96a405b
ARI: Allow specifying channel variables during a POST /channels

Added the ability to specify channel variables when creating/originating a
channel in ARI.  The variables are sent in the body of the request and should
be formatted as a single level JSON object.  No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.

(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/



git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403752 f38db490-d61c-443f-a65b-d21fe96a405b
bridge_native_rtp: Deadlock during 4-way conference creation

The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore).  A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback.  Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.

(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
     lock_inversion.diff uploaded by kmoore (license 6273)


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403767 f38db490-d61c-443f-a65b-d21fe96a405b
app_voicemail: Voicemail callback registration/unregistration function improvements.

* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.

* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
........

Merged revisions 403643 from http://svn.asterisk.org/svn/asterisk/trunk


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403779 f38db490-d61c-443f-a65b-d21fe96a405b
test_voicemail_api: Add check for a registered voicemail provider before tests.

It is much nicer diagnosing a test failure if app_voicemail is actually
loaded.
........

Merged revisions 403726 from http://svn.asterisk.org/svn/asterisk/trunk


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403780 f38db490-d61c-443f-a65b-d21fe96a405b
Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably

There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.

(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403781 f38db490-d61c-443f-a65b-d21fe96a405b
test.c: Fix too sticky unit test failed status.

Rerunning a failed unit test after loading any required modules should
allow the test to report a pass status if it now passes.


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/12@403782 f38db490-d61c-443f-a65b-d21fe96a405b