This changes implement the "get_pvt_uniqueid" which is used to return the
technology specific unique identifier. In the case of SIP this is the Call-ID
of the dialog.

Review: https://reviewboard.asterisk.org/r/3480/
........

Merged revisions 413088 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When bridge locking was added for bridge snapshot creation, some
locations where bridge locking was added were not guaranteed to
actually have a bridge and locking NULL AO2 objects tends to cause
segfaults. This ensures that NULL bridges aren't locked.
........

Merged revisions 413073 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413074 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These events are controlled by two new modules, res_manager_devicestate
and res_manager_presencestate.

Review: https://reviewboard.asterisk.org/r/3417



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Introducing changes proposed to chan_unistim driver:
1) Added the unistim.conf variable dtmf_duration which can select the DTMF playback duration from 0ms to 150ms (0 is off and is the new default)
2) Enabled the transmission of month names, which are sent with the date and changed the dateformat variable to accept the values 0-3 as per the UNISTIM standard (2 & 3 match the previous 1 & 2 formats).
3) Enabled the "Mute" packet so muting microphone works as expected and microphone muted for all calls while LED light on
4) Changed Duree to Timer on i2004 display

(closes issue ASTERISK-23592)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On congested networks, it is possible for the DTLS handshake messages to get
lost. This patch adds a timer to res_rtp_asterisk that will periodically
check to see if the handshake has succeeded. If not, it will retransmit the
DTLS handshake.

Review: https://reviewboard.asterisk.org/r/3337

ASTERISK-23649 #close
Reported by: Nitesh Bansal
patches:
  dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418)
........

Merged revisions 413008 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413009 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413012 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The string lengths on certain columns created through alembic for PJSIP were
too short. For instance, columns containing URIs are currently set to 40
characters, but this can be too small and result in truncated values.  Added
an alembic migration script that increases the size of these columns and a
few others to 255.

ASTERISK-23639 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3475/
........

Merged revisions 412992 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are cases in Asterisk where it might be desirable to lock
a short critical code section but not incur the context switch
and yield penalty of a mutex or rwlock.  The primary spinlock
implementations execute exclusively in userspace and therefore
don't incur those penalties.  Spinlocks are NOT meant to be a
general replacement for mutexes.  They should be used only for
protecting short blocks of critical code such as simple compares
and assignments.  Operations that may block, hold a lock, or
cause the thread to give up it's timeslice should NEVER be
attempted in a spinlock.

The first use case for spinlocks is in astobj2 - internal_ao2_ref.
Currently the manipulation of the reference counter is done with
an ast_atomic_fetchadd_int which works fine.  When weak reference
containers are introduced however, there's an additional comparison
and assignment that'll need to be done while the lock is held.
A mutex would be way too expensive here, hence the spinlock.
Given that lock contention in this situation would be infrequent,
the overhead of the spinlock is only a few more machine instructions
than the current ast_atomic_fetchadd_int call.

ASTERISK-23553 #close
Review: https://reviewboard.asterisk.org/r/3405/
........

Merged revisions 412976 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Backport -r411687 and fix the fix because content_length is the length of
out plus the length of the file controlled by fd.

When a response has an out content length of 0, fwrite would be called to
write a buffer with no data in it.  This resulted in the following classic
error message:

  [Apr  3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success

This patch makes it so that we only attempt to write the content of out if
the out string is non-zero.
........

Merged revisions 412922 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412923 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412924 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412925 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For some odd reason, loading app_mixmonitor was fine, but res_monitor was not.
This patch fixes a set of issues related to func_periodic_hook exporting the
beep functions that gets res_monitor working again.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This changes fixes a crash that occurs when stasis determines if it
should send a message out to an application or not. The code
incorrectly assumed that a bridge snapshot would always be present
when in reality for failure cases it may not be.

ASTERISK-23573 #close
........

Merged revisions 412882 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-1301)

(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
........

Merged revisions 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412822 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412823 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412824 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.

(closes issue AST-1301)

(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski

Review: https://reviewboard.asterisk.org/r/3447/
........

Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the TCP_NODELAY option to accepted connections on the HTTP
server built into Asterisk. This option disables the Nagle algorithm
which controls queueing of outbound data and in some cases can cause
delays on receipt of response by the client due to how the Nagle
algorithm interacts with TCP delayed ACK. This option is already set on
all non-HTTP AMI connections and this change would cover standard HTTP
requests, manager HTTP connections, and ARI HTTP requests and
websockets in Asterisk 12+ along with any future use of the HTTP
server.

Review: https://reviewboard.asterisk.org/r/3466/
........

Merged revisions 412745 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412748 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412749 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412750 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds documentation for the "all" channel option for the
ConfbridgeKick AMI action and adjusts AMI responses accordingly.

(issue ASTERISK-23282)
Reported by: Dorian Logan
........

Merged revisions 412730 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After the ability to kick all attendees from a conference was added, a
rework removed the comment about that feature from the CLI
documentation. This adds that documentation and adds "all" to the
participant tab completion list for the confbridge kick command.

(closes issue ASTERISK-23282)
Reported by: Dorian Logan
........

Merged revisions 412728 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix wrong dialtone. The "modulation" should not be referenced for tone+tone as it refers to the on-off characteristic - this often resulted in a single tone rather than the multitone as in the UK.
........

Merged revisions 412712 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412713 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When ASTERISK-23265/ASTERISK-23320 was fixed, it inadvertently led to realtime
features breaking. This was due to features loading prior to realtime. This
patch fixes this by loading features after loading dynamic modules.

ASTERISK-23487 #close
Reported by: Denis
Tested by: Denis
........

Merged revisions 412698 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two issues in app_sms:
(1) Firstly, the 'flags' field on the stack in sms_exec() is uninitialised,
    causing it to use the wrong protocol in some cases. This patch correctly
    initializes the flags fields.

(2) Secondly, when disconnect supervision is not working or
    inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was failing to
    terminate the call after it sent the REL(ease) message and the peer stopped
    talking to it. This patch fixes the code to handle the 'bad stop bit'
    message more gracefully in that case, and hang up the call.

Review: https://reviewboard.asterisk.org/r/1392/

ASTERISK-18331 #close
Reported by: David Woodhouse
patches:
  asterisk-fix-sms.patch uploaded by David Woodhouse (License 5754)
........

Merged revisions 412655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412656 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412657 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously multiple play actions against a bridge at one time would cause
the sounds to play simultaneously on the bridge. Now if a sound is already
playing, the play action will queue playback to occur after the completion
of other sounds currently on the queue.

(closes issue ASTERISK-22677)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3379/
........

Merged revisions 412639 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412641 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In sounds/Makefile

 1 Adds and moves some lines necessary for the en_GB core set. I'm just following how the other sets are defined here.
 2 removes the ES extra sounds related lines as we don't have ES extra sound sets. 

In sounds/sounds.xml

 3 Adds member definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in extra sound sets

ASTERISK-23550 #close
Review: https://reviewboard.asterisk.org/r/3464/
........

Merged revisions 412586 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412587 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is useful for configuring multiple permanent contacts for an AOR when using
realtime AORs.

Review: https://reviewboard.asterisk.org/r/3462
........

Merged revisions 412582 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Restore the reason value set by pbx_outgoing_attempt() to use
AST_CONTROL_xxx values as all the consumers were expecting rather than
cause codes.

* Fixed the dial routines to set cause codes for more than just
ast_request() so pbx_outgoing_attempt() reason codes will function.

* Fix inconsistent locked_channel return status in pbx_outgoing_attempt().
The chanel may not have been locked or the channel may have been a stale
pointer.

* Fixed the OutgoingSpoolFailed channel to run dialplan whenever the
dialing fails for an originate exten and 1 < synchronous.

* Fix incorrect ast_cond_wait() usage in pbx_outgoing_attempt().
Indroduced by issue ASTERISK-22212 patch.

* Made struct pbx_outgoing use the ao2 lock instead of its own lock for
the cond wait mutex.  No sense in having two locks associated with the
same struct when only one is needed.

Review: https://reviewboard.asterisk.org/r/3421/
........

Merged revisions 412581 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed ast_channel_publish_dial_forward() not locking the forwarded
channel when taking the channel snapshot.

* Fixed app_dial.c:do_forward() using the wrong channel to get the
original call forwarding string.

* Removed unnecessary locking when calling ast_channel_publish_dial() and
ast_channel_publish_dial_forward() in app_dial and app_queue.  Holding
channel locks when calling ast_channel_publish_dial_forward() with a
forwarded channel could result in pausing the system while the stasis bus
completes processsing a forwarded channel subscription.

Review: https://reviewboard.asterisk.org/r/3451/
........

Merged revisions 412579 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds DEBUG level logging for ARI websocket events and HTTP
responses similar to what is available for AMI. Logging for ARI HTTP
requests is already adequate for debugging purposes.
........

Merged revisions 412565 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change fixes a problem where permanent contacts being qualified were not
being updated. This was caused by the permanent contacts getting a uuid and not a
known identifier, causing an inability to look them up when updating in the
qualify code. A bug also existed where the new configuration may not be available
immediately when updating qualifies.

(closes issue ASTERISK-23514)
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/3448/
........

Merged revisions 412551 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 412549 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412550 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds a tones URI type to the playback resource. The tone can be specified by
name (from indications.conf) or by a tone pattern. In addition, tonezone can
be specified in the URI (by appending ;tonezone=<zone>). Tones must be
stopped manually in order for a stasis control to move on from playback of
the tone. Tones may be paused, resumed, restarted, and stopped. They may
not be rewound or fast forwarded (tones can't be controlled in a way that
lets you skip around from note to note and pausing and resuming will also
restart the tone from the beginning). Tests are currently in development
for this feature (https://reviewboard.asterisk.org/r/3428/).

(closes issue ASTERISK-23433)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3427/
........

Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412536 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two issues when building on SmartOS:

- channels/chan_oss.c: it makes sure soundcard.h is found
- main/Makefile: only use "-Wl,--version-script" when GNU LD is used as the Sun
  Linker doesn't support that. Similar checks are already used elswhere in the
  Makefile

Review: https://reviewboard.asterisk.org/r/3426

ASTERISK-23576 #close
Reported by: Sebastian Wiedenroth
patches:
  fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
........

Merged revisions 412468 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412483 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is a continuation of https://reviewboard.asterisk.org/r/3349/,
committed in r412303.

It resolves a finding oej had that the phone-context be available in a
channel variable separate from SIPDOMAIN. This patch adds that variable as
SIPURIPHONECONTEXT. It also allows a local number (or global number specified
in the TEL URI) to be used to look up as a peer.

(issue ASTERISK-17179)

Review: https://reviewboard.asterisk.org/r/3349/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The SIPREFERTOHDR channel variable is not being set on any channel when
performing a blind transfer using PJSIP. The 'refer->refer_to' was not
being set during a blind transfer.  Updated so the 'refer_to' is set to
the target uri on a blind transfer.

(closes issue ASTERISK-23502)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3445/
........

Merged revisions 412453 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412454 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This function returns an ast_bridge without a refcount bump and the
caller must increment the count if it intends to hold the pointer.

(closes issue ASTERISK-23588)
Review: https://reviewboard.asterisk.org/r/3450/
Reported by: Matt Jordan
........

Merged revisions 412439 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412440 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an option to enable a periodic beep to be played into a call if it
is being recorded.  If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval.  This option is provided for both Monitor() and
MixMonitor().

Review: https://reviewboard.asterisk.org/r/3424/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

RAII_VAR() is not a hammer appropriate to pound all nails.
........

Merged revisions 412413 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Remove unused RAII_VAR() declarations.  The compiler cannot catch these
because the cleanup function "references" the unused variable.  Some
actually allocated and released resources that were never used.

* Fixed some whitespace issues in stasis_bridges.c.
........

Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized.  The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.

* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.

* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.

* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential.  The callers must hold references to the passed in channel and
rtp objects.

* Eliminated sip_hangup() trying to get the bridge peer.  It is futile at
this point because the channel could never be in a bridge.

Review: https://reviewboard.asterisk.org/r/3431/
........

Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end
of the function.  The unref needs to happen after the last use of the
pointer.
........

Merged revisions 412348 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412383 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

```yaml:frontmatter
  r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines

  chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

  Prior to this patch, the P-Asserted-Identity header would include anonymous
  caller id information which seems to go against the point of the
  P-Asserted-Identity header. Now the real caller ID information will be
  included in this header. Also, no privacy header would be included.
  This patch adds 'Privacy: id' to outgoing SIP messages that include the
  P-Asserted-Identity header.

  (closes issue AST-1301)
```
........

Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

autoservice acquires a local reference to the logger callid of each channel
in a loop.  This local reference was not released, causing the callid of
every channel in autoservice to leak.  This change moves the callid unref
inside the loop.

ASTERISK-23616 #close
Reported by: ibercom
........

Merged revisions 412305 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412306 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for handling TEL URIs in inbound INVITE requests.
This includes the Request URI and the From URI. The number specified in
the Request URI will be the destination of the inbound channel in the dialplan.
The phone-context specified in the Request URI will be stored in the
TELPHONECONTEXT channel variable.

Review: https://reviewboard.asterisk.org/r/3349

ASTERISK-17179 #close
Reported by: Geert Van Pamel
Tested by: Geert Van Pamel
patches:
  asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
  asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412292 65c4cc65-6c06-0410-ace0-fbb531ad65f3