When listing the applications the apps lock was incorrectly
locked twice instead of being locked and then unlocked.

ASTERISK-28302

Change-Id: If7d064592a9e88c0f1049214c50e02be6dabf79e

When processing SSRC attributes we were iterating through
all of them, even though we only need to know the remote
SSRC once. This was problematic because some browsers group
SSRCs together on a stream, and due to our negotiation only
end up using the first one. Since we set the second one as
the remote SSRC we would drop the received media from them
instead of allowing it through.

In the future this may be extended to allow SSRC groups
and to use information from the attributes.

Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270

The current settings AST_PBX_MAX_STACK is 128 entries which is
too low for some FreePBX installations with complex parking
arrangements.  Increased to 512 if LOW_MEMORY is not defined.

ASTERISK-28300

Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299

This change provides an easier mechanism to determine which
subscribers are subscribed to a topic. Using this you can
inspect the specific subscribers for further details.

Change-Id: I8deea21703cd5c5357b85593b46c3eaf24e18c0c

Event type filtering is now enabled, and configurable per application. An app is
now able to specify which events are sent to the application by configuring an
allowed and/or disallowed list(s). This can be done by issuing the following:

PUT /applications/{applicationName}/eventFilter

And then enumerating the allowed/disallowed event types as a body parameter.

ASTERISK-28106

Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b

Some tests require Asterisk to execute scripts which
are stored in /tmp. When mount is used for tmpfs there
is no ability to allow scripts to be executed from
that location.

This change switches to using tmpfs which can be told
to allow executables to be run from /tmp.

Change-Id: I0e598ca2b76af1f7f2d29f0da7b1731a214a291a

Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a
little easier to retrieve a string field from the JSON object.

Also added an 'ast_strings_equal' function that safely checks (checks for NULLs)
for equality between two strings.

Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b

p2p_write updates txformat but doesn't require a smoother.  If a smoother
was created by another bridge type the smoother could fall out of date causing
one way audio issues.  To prevent this the smoother is now destroyed on the
start of native bridge.

ASTERISK-28284 #close

Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6

Currently, the pjsip show channelstats cli does not show channel's
stats after hits the invalid channel info. This makes hard to use
this cli. Changed to keep iterate after hits the invalid channel
info.

ASTERISK-28292

Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c

Currently, the Asterisk's pjsip_session module does not keeping the
rtcp's stats info after it was removed. But by adding the results
vector and keeping it until session is destroying, it can give more
useful information for other modules.

ASTERISK-28253

Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5

This change makes it so that even if non-code changes
occur (such as commit message changing) unit tests
will still be run and result in a verification.

ASTERISK-28251

Change-Id: I6491fff7c93e5d5cd8e41054486968bf66c4f608

A previous patch attempt to mitigate blocked threads on transport shutdown for
a given contact. It was thought that a second lock could be avoided by checking
the 'removing' flag on the transport monitor twice (once before and once after
the normal named aor locking). However as with usual threading issues if the
timing was right the original problem still occured.

This patch adds locking around the first 'removing' flag check and set, thus
nullifying the secondary check, so it was removed.

ASTERISK-28213

Change-Id: Iaa8e36e5311789549b76d8de42dfcea96013b2ed

When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.

This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.

This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.

ASTERISK-28277

Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6

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

Change-Id: Ib39052a745040f75eb635f15a042da15b20e22ab

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

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

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