Removed d->registered which was mirroring d->session. Changed relevant
references to use d->session instead.

Moved setting and unsetting of l->device from session register to device 
configuration. As such, l->device will always be valid unless it is has not
been configured to a device. Revised various test where checking if a device
is registered to use l->device->session.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368675 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Regression from -r367080 ringinuse commit.

(issue ASTERISK-19536)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368674 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding multiple file support broke reloading an unchanged file. This
adds an enum for return values for the aco_process_* functions and
ensures that the config is not applied if res is not ACO_PROCESS_OK.

Review: https://reviewboard.asterisk.org/r/1979/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1970/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368668 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added documentation describing what flags and arguments to pass to
aco_option_register for default option types. Also changed the ACL
handler to use the flags parameter to differentiate between "permit"
and "deny" instead of adding an additional vararg parameter.

Review: https://reviewboard.asterisk.org/r/1969/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer.  If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.

* Release the channel and private locks when creating a new channel as a
result of a flash hook.

(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a dialog-starting INVITE contains a to-tag, then Asterisk
will respond with a 481. In this case, the resulting incoming
ACK would not be matched, so Asterisk would continue retransmitting
the 481 until the transaction times out.

There were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481, since there
was a to-tag in the INVITE, Asterisk would place this original to-tag
in the 481 response. When the ACK came in, Asterisk would attempt to
match the to-tag in the ACK to the generated local tag. Unfortunately,
Asterisk never actually transmitted a response with the generated local
tag, so the to-tag in the ACK would not match.

The other problem was that when the 481 was sent, nothing was set
on the sip_pvt to indicate what CSeq is expected in the ACK.

To fix the first problem, we zero out the to-tag seen in the incoming
INVITE. This way, Asterisk, when time to send a response, will send
its generated local tag instead.

To fix the second problem, we set the sip_pvt's pendinginvite to the
CSeq of the INVITE when we send a 481.

(closes issue ASTERISK-19892)
Reported by Mark Michelson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Certain branches, such as Certified Asterisk, may have a modifier added to
them that specifies the features available in that branch.  For branches, this
modifier is expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of /certified/branches/1.8.11
would have a feature modifier of 'certified'.  This is slightly different then
how features are determined for tags, where the feature is part of the actual
tag name, e.g., "10.5.0-digiumphones".

In keeping with the nomenclature used for tags, the feature specifier for
branches is translated and placed after the revision numbers.  For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When changing between different modes of hold, the flags were not being
cleared out properly causing a failure to change hold states.

(closes issue ASTERISK-19919)
Patch-by: Morten Tryfoss
Reported-by: Morten Tryfoss
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368588 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a parked call was retrieved from the parking lot, it could not do a
blind transfer because it caused the involved calls to be hung up
unconditionally.

* Made the ParkedCall application return the ast_bridge_call() return
value.

(closes issue ABE-2862)
Reported by: Vlad Povorozniuc
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368550 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
Resolve some build warnings

My newly upgraded compiler caught these usages of uninitialized values.
They weren't actually used.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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