Reset peer outboundproxy on sip.conf reload

If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload.  This patch fixes that
by resetting it in set_peer_defaults().

Closes ASTERISK-19454
Review: https://reviewboard.asterisk.org/r/3065/
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Merged revisions 403634 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 403635 from http://svn.asterisk.org/svn/asterisk/branches/11
res_pjsip_messaging: send message to a default outbound endpoint

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/
chan_pjsip: Revert r403587

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
realtime: Create extensions in alembic ast-db-manage contribution

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/
channels/Makefile: clean pjsip directory
Setting svn:ignore
ARI: Adding a channel to a bridge while a live recording is active blocks

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/
ARI: Allow specifying channel variables during a POST /channels

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/
bridge_native_rtp: Deadlock during 4-way conference creation

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)
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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Merged revisions 403643 from http://svn.asterisk.org/svn/asterisk/trunk
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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Merged revisions 403726 from http://svn.asterisk.org/svn/asterisk/trunk
Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably

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/
test.c: Fix too sticky unit test failed status.

Rerunning a failed unit test after loading any required modules should
allow the test to report a pass status if it now passes.
documentation: Add PJSIP technology to messaging documentation
res_stasis: Expose event for call forwarding and follow forwarded channel.

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/
res_pjsip: Apply outbound proxy to all SIP requests.

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/
pjsip/dialplan_functions: Use the right buffer length when printing URIs

While entertaining, sizeof(buflen) is not the same as buflen. Doh.
app_sms: BufferOverflow when receiving odd length 16 bit message

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
pbx.c: put copy of ast_exten.data on stack to prevent memory corruption

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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Merged revisions 403862 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 403863 from http://svn.asterisk.org/svn/asterisk/branches/11
transfers: Fix bug setting both BLINDTRANSFER and ATTENDEDTRANSFER

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)