* 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.
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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/
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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.
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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)
```
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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
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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

The previous code left one error path where the module would be unref'd twice
instead of once.  It was done once in the error handling block, and again inside
of datastore destruction.

Now the module ref is only released in the datastore destructor and only acquired
when the datastore has been successfully allocated.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412280 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This module lacked necessary module ref count incrementing and decrementing
when used.  This patch adds it.  There's already a datastore used, so doing the
ref counting along with the lifetime of the datastore provides a convenient
place to do it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23620 #close
Reported by: Bradley Watkins
Patches:
      ASTERISK-23620_unlock_oldlist.patch (license #5021) patch uploaded by Bradley Watkins
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412228 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Removed no longer needed files from the svn:ignore property to make them visible.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While the vast majority of bridge snapshot creation is locked properly,
there are currently some instances that are not. This adds the missing
locking to ensure bridge state is not malleable during snapshot
creation.

(closes issue ASTERISK-22904)
Review: https://reviewboard.asterisk.org/r/3415/
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412180 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412168 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
    REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
    Every run will now blow away the previous run (as large ref files
    sometimes caused issues). We now also no longer open/close the file
    on each write, instead relying on fflush to make sure data gets written
    to the file (in case the ao2 call being performed is about to cause a
    crash)
(3) It goes with a comma delineated format for the ref debug file. This
    makes parsing much easier. This also now includes the thread ID of the
    thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
    contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.

Review: https://reviewboard.asterisk.org/r/3377/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This app is pretty ancient, so it was never converted to use the
option parsing helper code.  I'd like to add an option to this app
that takes an argument, and that's a pain to do when not using this
helper, so start by doing this conversion.

Review: https://reviewboard.asterisk.org/r/3429/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During discussions with Alexandr Dubovikov at Kamailio World, it became
apparent that while the SIP call ID is a useful identifier prior to an Asterisk
channel being created, it is far more preferable to use the channel name (or
some channel based identifier) when the channel is available. Homer is smart
enough to tie the various messages together. This patch opts to use the channel
name when it is available, falling back to the call ID otherwise.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The result of the "ast_sip_pubsub_generate_body_content" was not
set/initialized.  Consequently, the nominal path potentially returned
an invalid value, thus not sending mwi notifications.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a parsing error that occurred during the processing of
the AMI action. The error did not result in MixMonitor itself
misbehaving, but it could result in the AMI response not giving
correct information back.

The new header allows for one to specify a post-process command
to run when recording finishes. Previously, in order to do this,
the post-process command would have to be placed at the end of
the Options: header. 

Patches: mixmonitor_command_2.patch by jhardin (License #6512)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412050 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412035 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add notice messages during execution that the -I command line option and
the astersik.conf internal_timing option are no longer needed.  The
internal timing functionality is now always enabled if there is a timing
module loaded.

NOTE: Since the command line options and the asterisk.conf config file are
processed before the logging system is initialized, the messages are
output to stderr.

Change requested as a result of asterisk-dev list comments about the
commit for ASTERISK-22846 that removed the -I and internal_timing options.

Review: https://reviewboard.asterisk.org/r/3423/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411990 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a long standing bug in CB_ADD_LEN() behaving like CB_ADD().

ASTERISK-23546 #close
Reported by: Walter Doekes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed copy pasta error.

ASTERISK-23545 #close
Reported by: John Knott
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change makes it so if a transport is configured on an endpoint that is a WebSocket
type the option will be ignored. In practice this is fine because the WebSocket
transport can not create outgoing connections, it can only reuse existing ones. By
ignoring the option the existing PJSIP logic for using the existing connection will
be invoked and stuff will proceed.

(closes issue ASTERISK-23584)
Reported by: Rusty Newton
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This module makes use of some existing Asterisk components.  app_chanspy was
already listed as a dependency.  There are a few function modules used, as
well, so list them.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411897 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The change that fixed the pubsub test event's use of a dangling pointer
also changed when it was processed relative to the pjsip subscription
state change processing. This change corrects the order of events while
holding a reference to the pointer that was previously dangling.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411884 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AGI applications would trigger NewExten events every time the state of the AGI
application changed. This has historically not been the behavior and this
behavior was introduced with a CDR patch. This patch corrects that.

(closes issue ASTERISK-23390)
Reported by: Benjamin Keith Ford
Review: https://reviewboard.asterisk.org/r/3406/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by: Ibrahim22 (on IRC)
Tested by: Ibrahim22
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
configs: Clean up long line and typo in res_odbc.conf.sample.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411810 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Stasis() dialplan application monitors what bridge a channel is in
and so necessarily holds on to a bridge pointer. This change ensures
that it also holds on to a reference for that bridge to prevent the
bridge pointer from becoming a dangling pointer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411806 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The test event introduced in revision 411671 uses a dangling pointer to
access information about pubsub state changes. This moves the event to
within the lifetime of the pointer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically.  The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded.  The implementation is much
more generic though and could be used for many other things.

The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.

The other important bit of the implementation is how it figures out
when to trigger the beep playback.  This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way.  It's a convenient way to get a callback and check if it's time
to kick off another beep.  It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.

Review: https://reviewboard.asterisk.org/r/3362/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross.  Local channel optimization requires frames
flowing to trigger when optimization can happen.  When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing.  If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received.  With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.

* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed.  Asterisk now always uses internal
timing when needed if any timing module is loaded.  The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used.  The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.

* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().

* Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
ast_opt_internal_timing.

ASTERISK-22846 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3414/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Assert if a channel is destroyed but has the snapshot staging flag set.
In this case the final channel destruction snapshot would never get taken.

* Assert if what we just got out of the stasis cache is not what we were
looking for.  This assert would have saved several days searching for a
bug and a lot of my hair.

* Assert if the music on hold message posts could not find the associated
channel.  A crash will happen later when manager tries to send the MOH AMI
message.  This assert catches the problem when the stasis message is
posted instead of by the thread processing the defective message.

* Always generate a backtrace when an ast_assert() fails.

Review: https://reviewboard.asterisk.org/r/3411/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411702 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a response has a 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 out the content if the
calculated content_length is non-zero.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411688 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds a test event when subscription state changes so that
integration tests may trigger new actions at the appropriate times.

Review: https://reviewboard.asterisk.org/r/3383/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Parts of res_hep properly checked for a valid configuration object before
attempting to access the configuration. A check, however, was missed when
a packet is sent. This patch fixes the crash caused by not checking if the
configuration object is valid.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit contains several changes to sorcery:

1) Application of sorcery configuration based on module name is automatically performed
when sorcery is opened for a module.
2) Sorcery will not attempt to apply the same wizard to an object type more than once.
3) Sorcery gives more exact results when attempting to apply a wizard, whether as the
default or based on configuration.

Sorcery unit tests still pass for me after making these changes.

Review: https://reviewboard.asterisk.org/r/3326
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Use ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of
ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables.

* Use ast_copy_string() instead of inlining it.

* Remove an already done TODO comment.

* Some whitespace tweaks.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.

This patch causes realtime queue members to remain in queues during the reload process. Previously these
members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY".

ASTERISK-23547 #close
ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409)

Review: https://reviewboard.asterisk.org/r/3404/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411587 65c4cc65-6c06-0410-ace0-fbb531ad65f3