The main IVR was playing demo-congrats. I've switched it over to the
basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
has Allison prompting the user with the actual IVR menu.

ASTERISK-24892 #close

Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d

astmm.h includes defines that are meant to cause error's when standard
allocators (malloc, calloc, free, etc) are used.  It actually only
causes a warning, which is not always caught on certain sources.  In
modules this unknown symbol is not detected until runtime, where the
module fails to load.  This modifies the define's so that using one
of the blocked functions will cause a compile error regardless of
CFLAGS.

Moved spandsp header includes to before asterisk.h so the static inline
functions can continue using malloc and free.  Although these functions
are never called and optimized away, the updated replacement macro's
would still cause a failure.

Change-Id: I532640aca0913ba9da3b18c04a0f010ca1715af5

While trying to get WebRTC working with chan_pjsip, I was running
into the following error:

    Attempted to set an invalid DTLS-SRTP configuration on RTP
    instance...

Josh helpfully pointed out that res_srtp.so might not be loaded, and
sure enough, it wasn't. This patch adds a ERROR indiciating as much
to hopefully help others having a similar problem.

Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f

Added the necessary lines to the Makefile and sounds.xml so we'll have the
Swedish sounds in all available formats in menuselect.

See also: Swedish sounds were added into the core sounds release 1.4.27.

ASTERISK-24744 #close

Reported by: Tove Hjelm
Tested by: Rusty Newton

Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908

This switches codecs/lpc10/lpcini.c back to including "asterisk.h"
instead of <stdlib.h>.  lpcini.c allocates memory that is freed by
codec_lpc10.c, so it is important to use MALLOC_DEBUG allocator.
Added #define WRAP_LIBC_MALLOC to the start of the source to prevent
runtime symbol link error's.

Change-Id: I74f63fd09fdeb673ee7753122c3bb4722ab6e1ac

Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in

Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly.  Updated clean
targets as well.

Updated /.gitignore and doc/.gitignore.

Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622

 * Added a lookbehind to one-line comment matcher to skip escaped
   semicolons.
 * Added support for block comments.

Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7

The res_pjsip_exten_state module currently has a race condition between
processing the extension state callback from the PBX core and processing
the subscription shutdown callback from res_pjsip_pubsub. There is currently
no synchronization between the two. This can present a problem as while
the SIP subscription will remain valid the tree it points to may not.
This is in particular a problem as a task to send a NOTIFY may get queued
which will try to use the tree that may no longer be valid.

This change does the following to fix this problem:

1. All access to the subscription tree is done within the task that
sends the NOTIFY to ensure that no other thread is modifying or
destroying the tree. This task executes on the serializer for the
subscriptions.

2. A reference to the subscription serializer is kept to ensure it
remains valid for the lifetime of the extension state subscription.

3. The NOTIFY task has been changed so it will no longer attempt
to send a NOTIFY if the subscription has already been terminated.

ASTERISK-25057 #close
Reported by: Matt Jordan

Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643

* changes:
  CLI: Enable automatic references to modules.
  Modules: Make ast_module_info->self available to auxiliary sources.

After using the new vector stuff for real I found...

A bug in AST_VECTOR_INSERT_AT that could cause a seg fault.

The callbacks needed to be closer to ao2_callback in behavior
WRT to CMP_MATCH and CMP_STOP behavior and the ability to return
a vector of matched entries.

A pre-existing issue with APPEND and REPLACE was also fixed.

I also added a new macro to test.h that acts like ast_test_validate
but also accepts a return code variable and a cleanup label.  As well
as printing the error, it sets the rc variable to AST_TEST_FAIL and
does a goto to the specified label on error.  I had a local version
of this in test_vector so I just moved it.

ASTERISK-25045

Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc

Creating a snoop channel in ARI and spying only on a single direction (in or
out) results in CPU utilization continually increasing until the CPU is fully
consumed. This occurs because frames are being put in the opposing direction's
slin factory queue, but not being removed.

Fixed the problem by always reading and disposing of frames from the opposite
queue of the direction selected.

ASTERISK-24938 #closes

Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60

ASTERISK-25034
Reported by: Richard Mudgett

Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30

Change-Id: I26d4d2cb9cee924632ff59ef0b30a7e6a1e2b00d

A crash happens with this sequence of steps:
1) Party A is connected to party B.
2) Party B starts a DTMF attended transfer.
3) Party A hangs up while party B is dialing party C.

When party A hangs up the bridge that party A and party B are in is
dissolved and party B is kicked out of the bridge.  When party B finishes
dialing party C he attempts to move to the new bridge with party C.  Since
party B is no longer in a bridge the attempted move dereferences a NULL
bridge_channel pointer and crashes.

* Made the hold(), unhold(), ringing(), and the bridge_move() functions
tolerant of the channel not being in a bridge.  The assertion that party B
is always in a bridge is not true if the bridged peer of party B hangs up
and dissolves the bridge.  Being tolerant of not being in a bridge allows
the peer hangup stimulus to be processed by the FSM.

* Made the bridge_move() function return void since where the return value
for a failed move was checked generated a FSM coding ERROR message for a
normal off-nominal condition.

* Eliminated most uses of RAII_VAR in bridge_basic.c.

ASTERISK-25003 #close
Reported by: Artem Volodin

Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada

This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
parameters: position, original position and waiting time.

ASTERISK-25038 #close
Reported by: Etienne Lessard

Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618

Change-Id: I57081045c72b9fcf12d5c84493278f9272c31b32

ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by 
ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were 
called by SSL_load_error_strings already and got removed. These changes allow 
OpenSSL forks like BoringSSL to be used with Asterisk.

ASTERISK-25043 #close
Reported by: Alexander Traud
patches:
  asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520)

Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629

Missed this module in the previous commit.  res_ari_bridges uses symbols
from res_stasis_playback and res_stasis_recording.

ASTERISK-25027 #close
Reported by: Corey Farrell

Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f

Switch manager actions in pbx_config to use the registration macro that
passes the module pointer, allowing pbx_config reference to be bumped
while the manager actions run.

ASTERISK-25061 #close
Reported by: Corey Farrell

Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e

A recent change went into Asterisk which added reference counts to the
channels stored in a dial masquerade datastore. Unfortunately this
included a reference to the caller in a dialing operation. While all
of the dialed targets have the datastore removed from them upon dialing
completion this did not occur for the caller, causing it to have a
reference to itself that could go never go away (as it depended on
the destruction of the datastore which only happened when the channel
was destroyed). This resulted in the caller channel remaining on the
system despite it having hung up.

This change does the following to fix this issue:

1. The dial masquerade datastore is now removed from the caller upon
dialing completion, just like the dialed targets.
2. Upon destruction of the caller all the dialed targets are also
removed from the dial masquerade datastore (just in case).
3. The reference to the caller has been removed as it should not be
possible for the datastore to now be valid/useful after the lifetime
of the caller has ended.

ASTERISK-25025 #close

Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f