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

This patch adds the following:
(1) A new module, res_hep, which implements a generic packet capture agent for
the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based
on a patch provided by Alexandr Dubovikov; I basically just wrapped it up,
added configuration via the configuration framework, and threw in a
taskprocessor.
(2) A new module, res_hep_pjsip, which forwards all SIP message traffic that
passes through the res_pjsip stack over to res_hep for encapsulation and
transmission to a HEPv3 capture server.

Much thanks to Alexandr for his Asterisk patch for this code and for a *lot*
of patience waiting for me to port it to 12/trunk. Due to some dithering on
my part, this has taken the better part of a year to port forward (I still
blame CDRs for the delay).

ASTERISK-23557 #close

Review: https://reviewboard.asterisk.org/r/3207/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created

(closes issue ASTERISK-23460)

Reported by: Dmitry Melekhov
Patches:
	ASTERISK-23460-2.patch

Tested by: Dmitry Melekhov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It updates the AMI version to 2.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the ARI version to 1.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the UPGRADE/CHANGES files with changes that were not
   mentioned
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.

Also, the check for existence of a mandatory column checked for the first
column in the list instead of the key field lookup column. This patch fixes
that issue as well.

Finally, the compatibility option allow_empty_string_in_nontext, which was
added to previous revisions to allow for some database backends with certain
schemas to function, has been removed.

Review: https://reviewboard.asterisk.org/r/3335

ASTERISK-23459 #close
ASTERISK-23351 #close

(closes issue ASTERISK-23459)
Reported by: zvision
patches:
  res_config_odbc.diff uploaded by zvision (License 5755)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values.

This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.

This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It should be disabled
for database backends (such as PostgreSQL) that require NULL instead of an
empty string for Integer columns.

Review: https://reviewboard.asterisk.org/r/3375

(issue ASTERISK-23459)
Reported by: zvision
patches:
  res_config_odbc.diff uploaded by zvision (License 5755)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411513 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:

a) Client request comes from node.js user agent
   "Shred" via use of swagger-client library.

b) Asterisk and Client are *not* on the same
   host or TCP/IP stack

In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function.  The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission.  See review for more details.


ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411460 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since the relatime scripts are now managed by Alembic, the previous realtime
scripts were previously removed. However, the removal process messed up, as
the files were still in the repository. The contents were just empty.

This removes the files from the tree.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411443 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.

ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)

Review: https://reviewboard.asterisk.org/r/3396/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Update asterisk.h to reflect availability of ast_register_cleanup in 11.9.
* Use ast_register_cleanup for format_attr_shutdown.

(closes issue ASTERISK-23103)
Reported by: JoshE
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On graceful shutdown, sorcery wizards are all killed off, but it is
possible for sorcery instances to still have dangling pointers after
this, possibly causing a crash. Giving the sorcery instances a reference
to their wizards ensures that the wizard reference will remain valid for
the lifetime of the sorcery instance.

Review: https://reviewboard.asterisk.org/r/3401
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411296 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change fixes a bug where calling SayNumber with a number divisible by
100 using the Polish language would cause the code to attempt to play a
sound file with an empty name.

(closes issue ASTERISK-23509)
Reported by: zvision

Review: https://reviewboard.asterisk.org/r/3378/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411246 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior too 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@411192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, if ARI is not enabled it will still complain that there are no
configured users.  This patch checks to see if ARI is enabled before logging and
error or iterating the container to validate the users.

Review: https://reviewboard.asterisk.org/r/3391/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed bad use of ao2_find() in on_endpoint().

* Replaced use of find_endpoints() with find_an_endpoint() since only the
first found endpoint is ever needed.

* Fixed qualify_contact_cb() to update the contact with the aor
authenticate_qualify setting.  Otherwise, permanent contacts in the aor
type sections would have a config line order dependancy.

* Fixed off nominal path contact ref leak in qualify_contact().  The
comment saying the unref is not needed was wrong.

* Fixed off nominal path use of the endpoint parameter if it is NULL in
send_out_of_dialog_request().

* Added missing off nominal path unref of pjsip tdata in
send_out_of_dialog_request().

* Fixed off nominal path failing to call the callback in send_request_cb()
when the request is challenged for authentication.

* Eliminated silly RAII_VAR() use in qualify_contact_cb().

* Updated ast_sip_send_request() doxygen to better reflect reality.

(closes issue ASTERISK-23254)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/3381/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411142 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.

(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
    provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a channel in a stasis application is joined to a bridge, a subscription
for that bridge is created implicitly for the stasis application serving the
channel. Prior to this patch, subsequent removals of the channel from the
bridge would leave the subscription open.

Review: https://reviewboard.asterisk.org/r/3380/
........

Merged revisions 411086 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411087 65c4cc65-6c06-0410-ace0-fbb531ad65f3