After the review for buckets was completed (r2715), the handling of names in
the bucket core was deferred to the wizards. As such, the bucket unit tests
cannot expect that passing a URI with a scheme specified but no actual resource
name will automatically fail. The tests have been updated to not make this
check.
........

Merged revisions 397630 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The config options test requires the entire configuration item to be transparent from
the documentation system. So we let it do that too.

As an aside, please do not use this power for evil. Documentation is your friend, and
you really should document your configurations. Hiding your module's configuration
information from the system attempting to enforce some sanity in the universe is something
only a Bond villain would contemplate.
........

Merged revisions 397628 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The rtpengine configuration parameter was documented in the XML documentation,
but it was not actually registered with the sorcery object. This adds the
parameter with a default of "asterisk", such that res_rtp_asterisk is chosen as
the default RTP implementation.

(closes issue ASTERISK-22380)
Reported by: Rusty Newton
Tested by: Rusty Newton
........

Merged revisions 397621 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Note: This is why I commit on the weekend.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397613 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When originating channels, ast_pbx_outgoing_* caused the dialed channel
reference to be bumped twice. Ostensibly, this routine is bumping the channel
lifetime such that the channel doesn't get nuked in between locks/unlocks;
however, since the routine should return the dialed channel with its
reference bumped, it only needs to do this one time.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
Make libuuid an optional dependency for res_rtp_asterisk instead of a requirement.

Review: https://reviewboard.asterisk.org/r/2777/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397602 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Bucket is a URI based API for the creation, retrieval, updating, and deletion
of "buckets" and files contained within them.

Review: https://reviewboard.asterisk.org/r/2715/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This also adds a _nodoc equivalent.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397599 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-1200)
reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/2790/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Starting Asterisk would kick back an ERROR message stating that the Stasis
message type ast_channel_snapshot_type was used prior to initialization.
This occurred due to the caching topic being created prior to the message
type that it depended on.

This patch re-orders the start up such that the message type is initialized
prior to the caching topic. It also checks the return value of the
initialization of the agent login/logoff types.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Issuing hold/unhold would lead to odd behavior. Between two chan_sip devices,
a hold could cause an endless chain of updates while with pjsip a similar chain
would begin but then end somewhat randomly. This patch fixes that by no longer
tweaking the RTP glue on both sides of the call for every
HOLD/UNHOLD/UPDATE_RTP_PEER frame.

(issue ASTERISK-22217)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2794/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.

The following cases need to be handled when a channel is moved around in
the system.

* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.

* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)

* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.

* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.

The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.

(closes issue ASTERISK-22043)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2791/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When strict XML documentation checking was re-enabled, the test objects used in
sorcery would fail to register as the types were not marked internal and the
nodoc option wasn't used for the options. This fixes that problem, such that,
as one would hope, they once again pass.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.

* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().

* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.

* Fixed some formatting in ast_bt_get_symbols().

* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.

* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.

* Moved __dump_backtrace() because of compile issues with the utils
directory.

(closes issue ASTERISK-22221)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2778/
........

Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If an option is registered to a type and it is the last known type in the list
of registered types, and the option fails to register, an overrun of the types
array can occur due to the index variable having been already incremented.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most, if not all, of the backing features of a conf file should now be
implemented (e.g. multi-line comments, includes, templates, etc...).  A
few of the options still need to be mapped.  Those are currently listed
in the 'sip_to_res_sip.py' file.

Things to do:

(1) There is more work to do here, at least for the sip.conf items that
aren't currently parsed. An issue will be created for that.
(2) All of the scripts should probably be passed through pylint and have
as many PEP8 issues fixed as possible.
(3) A public review is probably warranted at that point of the entire script.

Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds error checking to ARI bridge operations, when
adding/removing channels to/from bridges.

In general, the error codes fall out as follows:
 * Bridge not found - 404 Not Found
 * Bridge not in Stasis - 409 Conflict
 * Channel not found - 400 Bad Request
 * Channel not in Stasis - 422 Unprocessable Entity
 * Channel not in this bridge (on remove) - 422 Unprocessable Entity

(closes issue ASTERISK-22036)
Review: https://reviewboard.asterisk.org/r/2769/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds pass through support for Opus and VP8. That includes:

* Format attribute negotiation for Opus. Note that unlike some other codecs,
  the draft RFC specifies having spaces delimiting the attributes in addition
  to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
  chan_sip, so a small tweak was also included in this patch for that.

* A format attribute negotiation module for Opus, res_format_attr_opus

* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
  than FIR, this really is specific to VP8 at this time.

Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.

Review: https://reviewboard.asterisk.org/r/2723/

(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
  asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are times when a configuration option should not have documentation.

1. Some options are registered with a particular object merely as a warning to
   users. These options aren't even really 'deprecated' - which has its own
   separate API call - they are actually provided by a different configuration
   file. The options are merely registered so that the user gets a warning that
   a different configuration file provides the item.

2. Some object types - most notably some used by modules that use sorcery - are
   completely internal and should never be shown to the user.

3. Sorcery itself has several 'hidden' fields that should never be shown to a
   user.

This patch updates the configuration framework and sorcery with additional API
calls that allow a module to register types as internal and options as not
requiring documentation. This bypasses the XML documentation checking.

This patch also re-enables the strict XML documentation checking in trunk, as
well as updates some documentation that was missing.

Review: https://reviewboard.asterisk.org/r/2785/

(closes issue ASTERISK-22359)
Reported by: Matt Jordan

(closes issue ASTERISK-22112)
Reported by: Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a response to an initial incoming INVITE results in a transport error
the INVITE transaction is removed from the INVITE session. Any attempts
to answer the INVITE session after this results in a crash as it requires
the INVITE transaction to exist. This change explicitly locks the dialog
and checks to ensure that the INVITE transaction exists before answering.

(closes issue AST-1203)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397514 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds ARI functions to be able to turn on/off music on hold in a
bridge. It actually functions more as a background music without
further actions on the bridge since if the rest of the channels
in the bridge aren't explicitly muted, they will still be able
to communicate.

(closes issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2688/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.

Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added the ability to handle 'raw' DTMF within the body of an INFO message.
Also made it so values 10-16 are mapped to valid DTMF values.

(closes issue ASTERISK-22144)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2776/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397482 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-22344)
(closes issue ASTERISK-22344)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.

* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.

(closes issue ASTERISK-22042)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2772/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397461 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This essentially makes app_queue usable again. From reviewboard:

* Reporting of transfers and call completion is done by creating stasis 
  subscriptions and listening for specific events in order to determine
  when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
  Mixmonitor API now instead of using ast_pbx_run()

In addition to the changes in app_queue, there are several supplementary changes as well:

* Queue logging now differentiates between attended and blind transfers. A
  note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
  includes which of the two local channels involved is the destination of
  the optimization, the channel that is replacing the destination local channel,
  and an identifier so that begin and end events can be matched to each other.
  The end events are now sent whether the optimization was successful or not and
  includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
  be set on a bridge. This is necessary because the queue requires that its
  bridge only allows move-swap local channel optimizations into the bridge.

(closes issue ASTERISK-21517)
Reported by Matt Jordan

(closes issue ASTERISK-21943)
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2694



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this change, we would reject SUBSCRIBE requests that had no Accept
headers. Now event package handlers that handle the default type for the
event package indicate that they do so. Therefore, if we have a handler that
can handle the default type, we can allow SUBSCRIBEs for the handler's event
package that have no Accept headers.

(closes issue ASTERISK-22067)
reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/2774


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Resync the abstract jitter buffer on the following additional control
frames:
AST_CONTROL_HOLD
AST_CONTROL_UNHOLD
AST_CONTROL_T38_PARAMETERS


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397440 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This modifies the behavior of the CEL engine to conform to documented
behavior for Asterisk 12 as defined on the wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification

The primary changes deal with removal of the peer field from function
calls since it is no longer directly relevant to the bridging system
and removal of the layer of CDR-like business logic that was providing
a partial emulation of Asterisk 11 CEL functionality. With this change,
there is no longer a distinction between "bridges" and "conferences"
and all participation changes are denoted with bridge enter and bridge
exit messages.

This updates the CEL unit tests to handle these changes and simplifies
some of the macros used in the process.

This also fixes a segfault when attempting to ref a configuration that
failed to load.

Review: https://reviewboard.asterisk.org/r/2788/
(issue ASTERISK-21567)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397431 65c4cc65-6c06-0410-ace0-fbb531ad65f3