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

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@411460 65c4cc65-6c06-0410-ace0-fbb531ad65f3