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

Resolve compile errors caused by r433863 by fixing the
documentation xml to comply with the schema.
........

Merged revisions 433888 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Resolve compile errors caused by r433839 by included the missing
header file, pbx.h.
........

Merged revisions 433863 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When an error occurs while writing to a web socket, the web socket is
disconnected and the event is logged. A side-effect of this, however, is that
any application on the other side waiting for a response from Stasis is left
hanging indefinitely (as there is no mechanism presently available for
notifying interested parties about web socket error states in Stasis).

To remedy this scenario, this patch introduces a new channel variable:
STASISSTATUS.

The possible values for STASISSTATUS are:
SUCCESS         - The channel has exited Stasis without any failures
FAILED          - Something caused Stasis to croak. Some (not all) possible
                  reasons for this:
                    - The app registry is not instantiated;
                    - The app requested is not registered;
                    - The app requested is not active;
                    - Stasis couldn't send a start message

ASTERISK-24802
Reported By: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4519/
........

Merged revisions 433839 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23919 #close
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/3802



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The menuselect parser is very simple.  It looks for AST_MODULE_INFO and
uses any quoted string on that line as the module summary display.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.

This patch breaks the major components of res_stasis.c into individual
files.

 * res/stasis/app.c - Stasis application tracking
 * res/stasis/control.c - Channel control objects
 * res/stasis/command.c - Channel command object

This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.

The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.

Review: https://reviewboard.asterisk.org/r/2530/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I've noticed when doing a graceful shutdown that the res_stasis_http.so
module gets unloaded before the modules that use it, which causes some
asserts during their unload.

While r386928 was a quick hack to get it to not assert and die, this
patch increases the use counts on res_stasis.so and res_stasis_http.so
properly. It's a bigger change than I expected, hence the review instead
of just committing it.

Review: https://reviewboard.asterisk.org/r/2489/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After some discussion on asterisk-dev, it was decided that the bulk of
the logic in app_stasis actually belongs in a resource module instead
of the application module.

This patch does that, leaves the app specific stuff in app_stasis, and
fixes up everything else to be consistent with that change.

 * Renamed test_app_stasis to test_res_stasis
 * Renamed app_stasis.h to stasis_app.h
   * This is still stasis application support, even though it's no
     longer in an app_ module. The name should never have been tied to
     the type of module, anyways.
 * Now that json isn't a resource module anymore, moved the
   ast_channel_snapshot_to_json function to main/stasis_channels.c,
   where it makes more sense.

Review: https://reviewboard.asterisk.org/r/2430/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3

refactored to use these events rather than producing the events directly
in channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket.

The AMI events are completely backward compatible, including sending
events on transmitted DTMF, and sending DTMF start events.

The Stasis-HTTP events are somewhat simplified. Since DTMF start and
DTMF send events are generally less useful, Stasis-HTTP will only send
events on received DTMF end.

(closes issue ASTERISK-21282)
(closes issue ASTERISK-21359)
Review: https://reviewboard.asterisk.org/r/2439


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The hash and compare functions for the control container was reusing
the wrong ones, causing some problems. I fixed it, but in the wrong
branch. Oh well, it happens.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * A new Stasis payload has been defined for multi-channel messages. This
   payload can store multiple ast_channel_snapshot objects along with a single
   JSON blob. The payload object itself is opaque; the snapshots are stored
   in a container keyed by roles. APIs have been provided to query for and
   retrieve the snapshots from the payload object.
 * The Dial AMI events have been refactored onto Stasis. This includes dial
   messages in app_dial, as well as the core dialing framework. The AMI events
   have been modified to send out a DialBegin/DialEnd events, as opposed to
   the subevent type that was previously used.
 * Stasis messages, types, and other objects related to channels have been
   placed in their own file, stasis_channels. Unit tests for some of these
   objects/messages have also been written.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.

This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.

Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.

Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.

Other changes along for the ride are:
 * An ast_frame_dtor function that's RAII_VAR safe
 * Some common JSON encoders for name/number, timeval, and
   context/extension/priority

Review: https://reviewboard.asterisk.org/r/2361/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3