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/
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On session registration, if device is already reporting that it is
connected to a device, an innocuous packet (update time) is sent to
the already connected device. If the tcp connection is down, the
device will be unregistered and the new connection allowed.

Without this patch, network issues can see a situation where a device
can not reregister until after 3*timeout.
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Review: https://reviewboard.asterisk.org/r/3039/
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Removed channel lock as it is now being down in ast_channel_alloc
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Original commit message by mmichelson (asterisk 12 r403311):

"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Introduce new config option 'aniasdni'. If yes then asterisk set dialed number as own id back to the caller
on incoming h.323 calls. Option can be set globally or per user section.

(closes issue ASTERISK-22020)
Reported by: Ross Beer



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.

(closes issue AST-1256)

Review: https://reviewboard.asterisk.org/r/3067/
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(close issue ASTERISK-22817)

Patches:
	ooh323_module_reload.patch



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404198 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-23007)
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This change adds a missing channel lock when adding a datastore
to a channel.
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Example output was inaccurate.

(issue ASTERISK-22970)
(closes issue ASTERISK-22970)
Reported by: Gareth Palmer
Patches:
   func_strings.patch uploaded by Gareth Palmer (license 5169)
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r404042 gave ast_bridge_base_new two new arguments for setting a bridge creator
and name. Unfortunately since a couple test modules aren't compiled by default,
I missed the fact that this change impacted those tests and caused compilation
failures against them.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-23021)
(closes issue ASTERISK-23021)
Reported by: Jeremy Lainé
Tested by: Rusty Newton
Patches:
   available.patch uploaded by Jeremy Lainé (license 6561)
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Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.

(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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If object creation fails an error message will now be output with the id, type,
and configuration file.
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Framehooks can be used in a reactive manner to execute specific logic
when a frame is received with a certain type and payload. Since it is
possible for framehooks to provide frames it was possible for this
reactive framehook to be unaware of frames it is looking for.

This change makes it so that when framehooks return a modified frame
the code will now re-iterate (from the beginning) and call any
previous framehooks that have not provided a modified frame themselves.

Review: https://reviewboard.asterisk.org/r/3046/
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Multiple revisions 403779-403780

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  r403779 | rmudgett | 2013-12-13 13:48:05 -0600 (Fri, 13 Dec 2013) | 12 lines
  
  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.
  ........
  
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  r403780 | rmudgett | 2013-12-13 13:55:31 -0600 (Fri, 13 Dec 2013) | 8 lines
  
  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.
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When creating channels via ARI, the current code fails to provide any default
format capabilities. For non-virtual channels this isn't really a problem -
the channels typically receive their capabilities as a result of the
underlying channel driver configuration. For virtual channels (such as Local
channels), the lack of any format capabilities causes the Asterisk core to
make some 'odd' choices with respect to the translation paths. The issue
reporter had some paths that had 3 hops on each channel leg, causing multiple
transcodings and some really crappy audio/performance.

By specifying a baseline of SLIN, we prevent that from occurring. Note that
this is what AMI does when it performs an Originate, as does res_clioriginate.

Review: https://reviewboard.asterisk.org/r/3068/

(issue ASTERISK-22962)
Reported by: Matt DiMeo
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This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.

A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.

Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.

Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.

(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
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The ast_bridge_set_transfer_variables function is supposed to wipe whichever
variable isn't being set. Instead it was setting both to the new value.  Oops.

(issue AFS-24)
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During dialplan execution in pbx_extension_helper(), the contexts global
read lock prevents link list corruption, but was released with a pointer
to the ast_exten and data later used in variable substitution.  Instead,
this patch removes pbx_substitute_variables() and locates a copy of the
ast_exten data on the stack before releasing the lock, where ast_exten
could get free'd by another thread performing a module reload.

(issue AST-1179)
Reported by: Thomas Arimont
(issue AST-1246)
Reported by: Alexander Hömig
Review: https://reviewboard.asterisk.org/r/3055/
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This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.

(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
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While entertaining, sizeof(buflen) is not the same as buflen. Doh.
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Objects which are involved in SIP request creation and sending
now allow an outbound proxy to be specified. For cases where
an endpoint is used the outbound proxy specified there will
be applied.

(closes issue ASTERISK-22673)
Reported by: Antti Yrjola

Review: https://reviewboard.asterisk.org/r/3022/
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This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.

(closes issue ASTERISK-22719)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/3054/
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Rerunning a failed unit test after loading any required modules should
allow the test to report a pass status if it now passes.
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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/
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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)
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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/
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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/
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It is much nicer diagnosing a test failure if app_voicemail is actually
loaded.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the alembic scripts were written for creating Asterisk
realtime databases the extensions table for dialplan wasn't
included.  This update creates the extensions table.

(closes issue ASTERISK-22815)
Reported by: Zone Conkle
Review: https://reviewboard.asterisk.org/r/3064/
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This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.

(issue ASTERISK-22936)
Reported by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint.  When sending
messages, if no endpoint can be found then the default one is used.

To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.

Review: https://reviewboard.asterisk.org/r/2944/
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