This is a change to align behavior with that of Asterisk 11 and previous versions.
In those versions, if a parked call were retrieved, and the call ended, the parked
call retriever would be hung up after the ParkedCall application ran. Prior to this
patch, in Asterisk 13, the same situation would result in the parked call retriever
falling through to additional priorities in the extension where the ParkedCall
application was called. With this patch, the behavior between Asterisk 11 and 13
aligns.

ASTERISK-24899 #close
Reported by Malcolm Davenport
Patches:
	ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then
ignoring the return.  Added OBJ_NODATA flag to prevent a reference leak.

ASTERISK-24935 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4578/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Outbound SIP MESSAGEs had the potential to be sent out
of order from how they were specified in a set of
dialplan steps.

This change creates a serializer for sending outbound
MESSAGE requests on. This ensures that the MESSAGEs are
sent by Asterisk in the same order that they were sent
from the dialplan.

ASTERISK-24937 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/4579
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A change in r430179 inserted a variable near the top of a
structure caused a problem when running DPMA in a version
of Asterisk compiled across the change.  This patch moves
the new variable to the end of the structure, eliminating
the problem.

Review: https://reviewboard.asterisk.org/r/4574/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but
for this to work correctly MALLOC_DEBUG must also be marked
as conflicting with DEBUG_CHAOS.

Review: https://reviewboard.asterisk.org/r/4557/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Resolve compile errors caused by r433863 by fixing the
documentation xml to comply with the schema.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds support for parsing SRV records and consuming their values
in an easy fashion. It also adds automatic sorting of SRV records according
to RFC 2782.

Tests have also been included which cover parsing, sorting, and off-nominal
cases where the record is corrupted.

ASTERISK-24931 #close
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/4528/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Resolve compile errors caused by r433839 by included the missing
header file, pbx.h.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When an error occurs while writing to a web socket, the web socket is
disconnected and the event is logged. A side-effect of this, however, is that
any application on the other side waiting for a response from Stasis is left
hanging indefinitely (as there is no mechanism presently available for
notifying interested parties about web socket error states in Stasis).

To remedy this scenario, this patch introduces a new channel variable:
STASISSTATUS.

The possible values for STASISSTATUS are:
SUCCESS         - The channel has exited Stasis without any failures
FAILED          - Something caused Stasis to croak. Some (not all) possible
                  reasons for this:
                    - The app registry is not instantiated;
                    - The app requested is not registered;
                    - The app requested is not active;
                    - Stasis couldn't send a start message

ASTERISK-24802
Reported By: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4519/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix misplaced parentheses in original fabs() expression.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
Re-add _ast_mem_backtrace_buffer variable for ABI compatibility.

Modules built prior to commit of r4502 expect to link at runtime
to the variable _ast_mem_backtrace_buffer.  This change re-adds
the variable to the C file only.

Review: https://reviewboard.asterisk.org/r/4558/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433796 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Binary modules are sometimes built against the latest release of
Asterisk in each branch, and need to be compatible with all
releases of that branch.  This change ensures that utils.h only
uses ast_log_safe from the core.  For modules and utilities ast_log
is used instead.

Review: https://reviewboard.asterisk.org/r/4548/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433774 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes several warnings caught by clang - in this case, usage of the
abs function on non-integer values. This patch uses labs and fabs, as
appropriate, in the various affected files.

Review: https://reviewboard.asterisk.org/r/4525

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4525.patch submitted by dkdegroot (License 6600)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes some invalid enum conversion warnings caught by clang. In
particular:
* chan_sip: Several functions mixed usage of the st_refresher_param
  enum and st_refresher enum. This patch corrects the functions to use the
  right enum.
* chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
* strings: Fixed incorrect usage of AO2 flags with strings container.
* res_stasis: Change a return enumeration to stasis_app_user_event_res.

Review: https://reviewboard.asterisk.org/r/4535

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4535.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The patch in r433720 caused a warning to be kicked back by gcc. It occurred
due to this check in unistd.h:

    if (__nbytes > __bos0 (__buf))
        return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf));

That is, if __nbytes is greater than the result of GCC's built-in object size
for the struct, we'll kick back a warning.

As it turns out, this is because there is an error in the code in the patch.
We are passing the address of the pointer to the struct, not iev, which is a
pointer to the struct. Hence, the number of bytes is probably going to be lot
larger than the number of bytes that make up a pointer! This patch changes
the code just read from the pointer to the struct - which fixes the warning.

ASTERISK-24917
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433745 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk's build system has a tendency to pass include directives for libraries
to everything compiled within a particular group of source files. This means
we pass the header for libxml2 to things that don't necessarily need it. As a
result, we ignore this particular warning.

Review: https://reviewboard.asterisk.org/r/4545/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4545.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a warning caught by clang, wherein a variable sized struct is
not located at the end of a struct. While the code in question actually
expected this, this is a good warning to watch for. Hence, this patch refactors
the code in question to not have two variable length elements in the same
struct.

Review: https://reviewboard.asterisk.org/r/4530/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4530.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433719 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:

* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
                    qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* main/security_events.c: removed unused TIMESTAMP_STR_LEN
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states

Review: https://reviewboard.asterisk.org/r/4526

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4526.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Assigning a variable to itself isn't super useful. However, the WAV format
modules make use of this in order to perform byte endian checks. This patch
works around the warning by only performing the self assignment if we are
going to do more than just assign it to ourselves. Which is odd, but true.

Review: https://reviewboard.asterisk.org/r/4544/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4544.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433692 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clang will treat ((a == b)) as a warning, as it reasonably expects that the
developer may have intended to write (a == b) or ((a = b)). This patch cleans
up all instances where equality, not assignment, was intended between two
parantheses.

Review: https://reviewboard.asterisk.org/r/4531/

ASTERISK-24917
Repoted by: dkdegroot
patches:
  rb4531.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by
clang, as it will truncate the -1 to a 1 implicitly.

Instead, we just assign the value a '1'.

Review: https://reviewboard.asterisk.org/r/4537/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4537.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes clange compiler warnings for initializer overrides.
Specifically:

res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
those enum values, we therefore initialize the value twice to two different
values, "tlsv1" and "default". This patch changes it to just initialize
the index in the array to "tlsv1".

Review: https://reviewboard.asterisk.org/r/4539/

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4539.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes clang compilers warnings for unused functions. Specifically:
 * channels/chan_iax2: removed user_ref function
 * main/dsp.c: removed goertzel_update function
 * main/config.c: made variable_list_switch static

Review: https://reviewboard.asterisk.org/r/4527

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4527.patch submitted by dkdegroot (License 6600)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433681 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/4504/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company

Patch 4488 includes all functionality needed for SAC's outside connectivity
and some externally accessed features, as well as outbound dialing.

Reported by: Malcolm Davenport
Tested by: Rusty Newton

Review: https://reviewboard.asterisk.org/r/4488/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage.

* Converted the contact_autoexpire container to use the ao2 template hash
and cmp functions.  Also made use the OBJ_SEARCH_xxx names instead of the
deprecated names.

* Eliminates several unnecessary uses of RAII_VAR().

Review: https://reviewboard.asterisk.org/r/4524/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.

This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.

ASTERISK-24920 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/4532/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Contact expiration object refs were leaked when the module was unloaded.

* Made empty the scheduler of entries before destroying it to release the
object ref held by the scheduler entry.

Review: https://reviewboard.asterisk.org/r/4523/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the kqueue timing module to conform to current timer API.

This fixes issues with using the kqueue timing source on Asterisk 13 on
FreeBSD 10. These issues include:

- Remove support for kevent64().  The values used to support Asterisk timers
  fit within 32bits and so can be handled on all platforms via kevent().

- Provide debug logging for, but do not track, unacked events.  This matches
  the behavior of all other timer implementations.

- Implement continuous mode by triggering and leaving active, a user event.
  This ensures that the file descriptor for the timer returns immediately from
  poll(), without placing the load of a high speed timer on the kernel.

- In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
  On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
  type kqueue supports for timers.

- In kqueue_timer_get_event(), assume the caller woke up from poll() and just
  return the mode the timer is currently in. This matches all other timer
  implementations.

- Adjust the test code now that unacked events are not tracked.

Review: https://reviewboard.asterisk.org/r/4465/

ASTERISK-24857 #close
Reported by: scsiguy
Tested by: Ed Hynan
patches:
  rb4465.patch submitted by scsiguy (License 6692)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use the standard ast_log instead of ast_log_safe for STANDALONE programs.

Review: https://reviewboard.asterisk.org/r/4538/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This introduces a new logger routine ast_log_safe.  This routine should be
used for all error messages in code that can be run as a result of ast_log.
ast_log_safe does nothing if run recursively.  All error logging in
astobj2.c, strings.c and utils.h have been switched to ast_log_safe.

This required adding support for raw threadstorage.  This provides direct
access to the void* pointer in threadstorage.  In ast_log_safe, NULL is used
to signify that this thread is not already running ast_log_safe, (void*)1 when
it is already running.  This was done since it's critical that ast_log_safe
do nothing that could log during recursion checking.

ASTERISK-24155 #close
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/4502/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add ast_register_cleanup to utils/clicompat.c to deal with
  any utils that copy sources from main.
* Asterisk 13+: remove unused variables from core_local.c.

Review: https://reviewboard.asterisk.org/r/4534/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433501 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups.  Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe.  ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.

Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.

ASTERISK-24142 #close
Reported by: David Brillert

ASTERISK-24683 #close
Reported by: Peter Katzmann

ASTERISK-24805 #close
Reported by: Badalian Vyacheslav

ASTERISK-24881 #close
Reported by: Corey Farrell

Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
  caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
  crashes in some cases.  In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
  shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.

ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
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Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.

ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
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Merged revisions 433446 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* In res/res_sorcery_realtime.c: Broke long line.

* In main/bucket.c: Eliminated unnecessary NULL check as
ast_sorcery_unref() is NULL tolerant and set the global object to NULL
after unref in the system shutdown bucket_cleanup().
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Merged revisions 433420 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
setting was never actually processed. This was due to not properly copying
over the global settings to the client settings when applying the
configuration to the run-time object.

Review: https://reviewboard.asterisk.org/r/4496/

ASTERISK-14233
ASTERISK-24780 #close
Reported by: Simon Arlott
patches:
  asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
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Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 433396 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.

This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.

Unit tests have also been written for all of the above to confirm the API and
functionality.

ASTERISK-24834 #close
Reported by: Matt Jordan

ASTERISK-24836 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens.  If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.

Consequences of these unnecessary messages:

* The caller can start hearing ringback before the far end even gets the
call.

* Many phones tend to grab the first connected line information and refuse
to update the display if it changes.  The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.

When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled.  When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.

* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages.  The default is "no" to disable sending the
unnecessary messages.

ASTERISK-24781 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4473/
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Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3