The CDR was overwriting the start time when the call continued the
dialplan from the ARI stasis or a Local channel was originated.

This change fixes this by no longer reinitializing the CDR when
transitioning out of the dialed pending state to the single state.

ASTERISK-28181

Change-Id: I921bc04064b6cff1deb2eea56a94d86489561cdc

Split destroy_hint method to separate hint removal and extension hint
state changed callback, the latter now called via stasis.
This avoids deadlock between res_parking reload that is removing the
parking lot and the related hint and subscribe requests coming for the
same parking lot.

ASTERISK-28173

Change-Id: I5b03c3455b3b12b6f83cea4cc34f4b4b20444f7e

Added ARI resource.
GET /ari/asterisk/ping : It returns "pong" message with timestamp
and asterisk id. It would be useful for simple heath check.

Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29

At AstriCon, there was a strong desire for the ability to completely
bypass dialplan when using ARI. This is possible through the automatic
creation of a context and a couple of extensions whenever an application
is started.

For example, if you have an application named 'ari-example', a context
named 'stasis-ari-example' will be automatically created whenever this
application is started as long as one does not already exist. Two
extensions (a match-all extension for Stasis and a 'h' extension) are
created within this context. Any endpoint that registers to Asterisk
within this context will send all calls to the corresponding Stasis
application. When the application is destroyed, the context is removed.

ASTERISK-28104 #close

Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac

On OpenSuse Leap, libjansson.a is installed in
third-party/jansson/dest/lib64 instead of lib (which is where
the top-level makeopts looks).  This causes a link failure.

* Updated jansson/Makefile to add an explicit --libdir to force
  the installation to third-party/jansson/dest/lib.

ASTERISK-28271
Reported by: David Wilcox

Change-Id: Ibf8af75e5da13562105fcc39ed898c6ef0b5a5f3

The context specified by 'regcontext' was not being created, so when Asterisk
attempted to later dynamically add an extension it would fail. This patch now
creates the context if a 'regcontext' is specified.

ASTERISK-28238

Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265

Testing revealed that the cache added no benefit but that it could
consume excessive memory.

Two new index related functions were created:
ast_sounds_get_index_for_file() and ast_media_index_update_for_file()
which restrict index updating to specific sound files.

The original ast_sounds_get_index() and ast_media_index_update()
calls are still available but since they no longer cache the results
internally, developers should re-use an index they may already have
instead of calling ast_sounds_get_index() repeatedly.  If information
for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().

The media_index directory scan code was elimininated in favor of
using the existing ast_file_read_dirs() function.

Since there's no more cache, ast_sounds_index_init now only
registers the sounds cli commands instead of generating the
initial index and subscribing to stasis format register/unregister
messages.

"sounds" is no longer a valid target for the "module reload"
command.

Both the sounds cli commands and the sounds ari resources were
refactored to only call ast_sounds_get_index() once per invocation
and to use ast_sounds_get_index_for_file() when a specific sound
file is requested.

Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d

The option value "sdp" for some of the settings was removed a while back,
however the sample conf was not updated.

This patch removes any wording with regards to the old "sdp" option value,
and adjusts the defaults to what they are now.

ASTERISK-28263

Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445

Added support for the seek function in format_g726
so playback can start from anywhere.
Before the fix, playback of g726 files
always started from the beginning.

ASTERISK-28246

Change-Id: I626235bc4642df1479050d3d06828412603a9b40

Bundled pjproject and jansson must be configured with the host and build
parameters provided to the configure script.
Autotools do not permit to check for the existence of local header files, so
the control of hrirs.h must not be done when cross-compiling.

ASTERISK-28250

Change-Id: If0a76e52a87d4ab82b7d4c72d27d8759ca931880

Control frames (PING / PONG / CLOSE) can be received in the middle of a
fragmented message. In order to ensure they do not interfere with the
reassembly buffer, we exit early and do not return the payload to the
caller.

ASTERISK-28257 #close

Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc

To avoid the stream name collide if there're more than one video track
in one client. If client has multi video tracks, the name of ast_stream
which represents each video track may be the same. Use the MSID:LABEL
here because it's identifiable.

ASTERISK-28196 #close
Reported-by: xiemchen

Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b

The type value extracted from stasis message data in channel_hangup_handler_cb
isn't compared against the valid values "run", "pop" and "push". Thus the
manager events HangupHandlerPush, HangupHandlerPop and HangupHandlerRun are
never thrown.

This regression was introduced by ASTERISK_21462.

ASTERISK-28252

Change-Id: I9956e35e18da1873113644df1ddc3c7cd37bf524

A bug in GCC causes TEST_CEL to return failure under the following
conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline.  The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()

Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7

You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes.  These can be used anywhere
the mailbox is specified.

Example:
[general]
aliasescontext = myaliases

[default]
1234 = yadayada

[myaliases]
4321@devices = 1234@default

Now you can use 4321@devices to refer to the 1234@default mailbox.

This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.

Change-Id: I395b9205c91523a334fe971be0d1de4522067b04

When a reliable transport is shutdown it's possible for the pjsip registrar
resource shutdown handler to get called multiple times. If this happens and one
of the threads is taking "too long" (slow database call for instance) then the
others get blocked waiting to delete.

Since it only takes one to delete the contact then the other threads should be
able to continue on if one of the threads is currently "deleting". This patch
makes it so now when a thread enters the shutdown handler it checks to see if a
thread is currently already "deleting". If so, then the thread does not attempt
to get the lock, and instead continues on thus avoiding the blockage.

ASTERISK-28213 #close

Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a

Fixed #2172: Avoid double reference counter decrements in
timer in the scenario of race condition between
pj_timer_heap_cancel() and pj_timer_heap_poll().

Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8

The transport management code that checks for idle connections keeps a
reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by
default). Because of this, if the transport is closed before this
timeout, the idle checking code will keep the transport from actually
being shutdown until the timeout expires.

Rather than passing the AO2 object to the scheduler task, we just pass
its key and look it up when it is time to potentially close the idle
connection. The other transport management code handles cleaning up
everything else for us.

Additionally, because we use the address of the transport when
generating its name, we concatenate an incrementing ID to the end of the
name to guarantee uniqueness.

Related to ASTERISK~28231

Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb

If the Monitor is started with the i option the read_stream will be
NULL. One code path in channel.c checks if write_stream is set but than
uses read_stream instead causing a segfault.

ASTERISK-28249

Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525

Previously both AMI and ARI used a default route on
their stasis message router to handle some of the
messages for publishing out their respective
connection. This caused messages to be given to
their subscription that could not be formatted
into AMI or JSON.

This change adds an API call to the stasis message
router which allows a default route to be set as well
as formatters that the default route is expecting.
This allows both AMI and ARI to specify that their
default route only wants messages of their given
formatter. By doing so stasis can more intelligently
filter at publishing time so that they do not receive
messages which will not be turned into AMI or JSON.

ASTERISK-28244

Change-Id: I65272819a53ce99f869181d1d370da559a7d1703

Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5

This ensures that Asterisk responds properly to frames received from a
client with opcode 8 (CLOSE) by echoing back the status code in its own
CLOSE frame.

Handling of the CLOSE opcode is moved up with the rest of the opcodes so
that unmasking gets applied. The payload is no longer returned to the
caller, but neither ARI nor the chan_sip nor pjsip made use of the
payload, which is a good thing since it was masked.

ASTERISK-28231 #close

Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf