When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.

(closes issue ASTERISK-19876)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368529 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was essentially duplicated functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE.  This removes the flag and converts that usage
into AST_CAUSE_ANSWERED_ELSEWHER usage.

Review: https://reviewboard.asterisk.org/r/1944
(closes issue ASTERISK-19865)
Patch-by: Birger Harzenetter


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Revision 351130 broke corect HANGUPCAUSE setting
for the 404 case in chan_sip. Other cases were also
potentially broken. This patch fixes the relaying
of causes to be what they used to be.

(closes issue ASTERISK-19914)
Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed later)
Patches:
	chan_sip.diff uploaded by Pavel Troller (license #6302)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368500 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19322)

(closes issue ASTERISK-19875)
Reported by: call
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19800)
Reported by: Billy Chia
Patches:
	asterisk.vim.patch uploaded by Billy Chia (license #6381)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368455 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368441 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.

Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.

Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.

chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.

Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.

Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().

* Simplify many calls to ast_do_masquerade() since it will never return a
failure now.  If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.

* Fixed the call to ast_bridged_channel() in ast_do_masquerade().  This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.

(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1915/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1952/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Attempting to remove a channel from autoservice with the channel lock held
will result in deadlock.

* Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held.

(closes issue ASTERISK-19764)
Reported by: rmudgett
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368311 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk should not accept SDP offers that contain unknown RTP profiles (for
audio/video streams) or unknown top-level media types. When it does, it answers
with an SDP that does not match the offer properly, and this will nearly
always result in a broken call. This patch causes such offers to be rejected.

Review: https://reviewboard.asterisk.org/r/1811/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 'Unsupported media type' is only reported when that is in fact the case,
   not when a supported media type is included in an 'm' line that has an
   invalid format.

* All warning messages related to parsing 'm' lines now include the 'm' line contents.

* (minor bugfix) newline added to port-number-zero warning messages.

* Warning messages improved to use RFC-specified terminology for various items.

* Warnings for offers that include more than one port for a single media type now
  include the media type.

Review: https://reviewboard.asterisk.org/r/1811/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This framework adds a way to register the various options in a config
file with Asterisk and to handle loading and reloading of that config
in a consistent and atomic manner.

Review: https://reviewboard.asterisk.org/r/1873/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368181 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.

This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.

Review: https://reviewboard.asterisk.org/r/1954



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ability to set "echocan_mode" and "buffers" through the dialplan was added
to chan_dahdi some time ago.  This patch adds some documentation to
func_channel.

(Closes issue ASTERISK-19911)
Reported by: Dale Noll
Tested by: Michael L. Young
Patches: 
  asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1949/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368094 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixes findings: 0-2,5,7-15,24-26,28-31

(issue ASTERISK-19648)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19854)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367982 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix sig_pri_lock_owner() to avoid deadlock properly.

* Code pri_grab() better.

* Fix sig_ss7_lock_owner() to avoid deadlock properly.

* Code ss7_grab() better.

(closes issue ASTERISK-19854)
Reported by: Jaxon
Patches:
      jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7)
Tested by: Jaxon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367979 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for
app_meetme.c in find_user().

* Change use of %i to %d in sscanf() in find_user().  The use of %i gives
unexpected parsing because it can accept hex, octal, and decimal integer
formats.

* Changed other uses of %i in app_meetme() to use %d for consistency.

(issue ASTERISK-19648)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367908 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a skinny session is unregistered, the corresponding device pointer is set
to NULL in the channel private data.  If the client was not in the on-hook state
at the time the connection was closed, the device pointer can later be
dereferened if a message or channel event attempts to use a line's pointer to
said device.

The patches prevent this from occurring by checking the line's pointer in
message handlers and channel callbacks that can fire after an unregistration
attempt.

(closes issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Damien Wedhorn
Patches:
  AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
  AST-2012-008-10.diff uploaded by mjordan (licesen 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made schedule_delivery() set the received frame f->data.ptr to NULL if 
the datalen is zero.  

* Fix queue_signalling() memcpy() size error.

* Made queue_signalling() not use C++ keyword variable names.

(closes issue ASTERISK-19597)
Reported by: mgrobecker
Patches:
      jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Michael L. Young
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The pvt_sip allowtransfer was not being set to that of the peer's setting.
Therefore, the global allowtransfer setting was being used instead which would
lead to calls not being transfered if the global setting was set to 'no' despite
the setting on the peer being 'yes' and vice versa, calls would be allowed to
transfer even if the peer's setting was 'no' but the global setting was 'yes'.

(Closes issue ASTERISK-19856)
Reported by: Jacek
Tested by: Michael L. Young, Jacek 
Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by 
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1923/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367732 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Dial and Queue I option is intended to block connected line updates
and redirecting updates.  However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity to setup
new connected line information.  Unfortunately, the Dial and Queue I
option is disabled for *all* forked calls if one of those calls is
redirected.

* Make the Dial and Queue I option apply to each outgoing call leg
independently.  Now if one outgoing call leg is locally redirected, the
other outgoing calls are not affected.

* Made Dial not pass any redirecting updates when forking calls.
Redirecting updates do not make sense for this scenario.

* Made Queue not pass any redirecting updates when using the ringall
strategy.  Redirecting updates do not make sense for this scenario.

* Fixed deadlock potential with chan_local when Dial and Queue send
redirecting updates for a local redirect.

* Converted the Queue stillgoing flag to a boolean bitfield.

(closes issue ASTERISK-19511)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1920/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When remotely bridging calls with directmedia, Asterisk would check
the address of the peers/users holding directmedia ACLs (set via
directmediapermit/directmediadeny) instead of the bridged peer. This
is similar to r366547, but trunk specific and involves changes to
the rtpengine instead of just chan_sip.

(closes issue AST-876)
review: https://reviewboard.asterisk.org/r/1924/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
chan_sip: Check the right channel's host address for directmediapermit/deny

Prior to this patch, when checking the addresses for directmediapermit and
directmediadeny, Asterisk would check the host address of the channel
permit/deny was specified, which differs from the expectations of both
our users and the development team. Instead, directmediapermit/deny now
checks against the address of the channel that the peer with the ACL is
connected to.

(issue AST-876)
Review: https://reviewboard.asterisk.org/r/1899/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A patch introduced in r354938 made it so that ConfBridge would not attempt to
play sound files if those files did not exist.  Unfortunately, ConfBridge uses
the same underlying function, play_sound_helper, to playback both sound files
and numbers to callers.  When a number is being played back, the name of the
sound file is expected to be NULL.  This NULL value was passed into a function
that tested for the existance of a sound file and is not tolerant to NULL
file names, causing a crash.

This patch fixes the behavior, such that if a sound file does not exist we
do not attempt to play it, but we only attempt that check if the a sound file
was specified in the first place.  If a sound file was not specified, we use
the 'play number' logic in the helper function.

(closes issue ASTERISK-19899)
Reported by: Florian Gilcher
Tested by: Florian Gilcher
patches:
  asterisk-19899.diff uploaded by mjordan (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367520 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The AST_CONTROL_HOLD MOH class from the WaitExten application can now be
queued onto a channel, passed over local channels with the /m option, and
passed over IAX channels.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367477 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Uncovered a nasty reference leak while I was writing some changes to
chan_dahdi/sig_analog. Slapped myself around a bit after seeing that I
performed the unchecked return causing this problem.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks to Paul Belanger for pointing out this error.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether
or not MWI NOTIFY requests had been sent to a specific peer.  When MWI
notifications were changed to use the internal event framework, this value was
no longer needed for its original purpose.  Hence, it was no longer updated
with the new/old message counts for a peer.  The value was previously removed
for Asterisk 10; however, since it was still present in Asterisk 1.8 and still
useful for reporting purposes, it was decided to re-add the value.

This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip
show peer [peer]' command, and makes it so that the value of lastmsgssent is
updated appropriately. The value should now display the new/old message counts
for a particular peer.

(closes issue ASTERISK-17866)
Reported by: Steve Davies
patches by:
  ast-17866-rb1272.patch (License #5041 by irroot)
  Modified slightly for this commit

Review: https://reviewboard.asterisk.org/r/1939
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367376 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes to situations that could cause the CEL LINKEDID_END event to
be missed.

1) During a core stop gracefully, modules are unloaded when ast_active_channels
== 0. The LINKDEDID_END event fires during the channel destructor. This means
that occasionally, the cel_* module will be unloaded before the channel is
destroyed. It seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and used it in the
shutdown code.

2) During a masquerade, ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids
container in cel.c. It didn't ref the new linkedid. Now it does. 

Review: https://reviewboard.asterisk.org/r/1900/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367309 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable
should definitely not be used after that. To solve this in the two cases
that affect subscribing for MWI notifications, we instead save the ref
locally, and unref them in the error conditions.

(closes issue ASTERISK-19827)
Reported by: B. R
Review: https://reviewboard.asterisk.org/r/1940/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It made no sense to trylock the channel and then unconditionally lock the
channel right after.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If chan_iax2 does not reject the PVT_CAUSE_CODE frames, the cause will not be
stored properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367189 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This should have been committed to my team trunk-digiumphones branch
instead of trunk.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change accommodates two methods by which calls can be directed to
a user's voicemail.

* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.

Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm". 

This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.

(closes issue AST-871)
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/1925



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367163 65c4cc65-6c06-0410-ace0-fbb531ad65f3