Added ARI resource for channel statistics.
GET /ari/channels/{channelId}/rtp_statistics : It returns given
channel's rtp statistics detail.

ASTERISK-28320

Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376

strtok() uses a static buffer, making it not thread safe.

Also add a #define to cause a compile failure if strtok is used.

Change-Id: Icce265153e1e65adafa8849334438ab6d190e541

Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857

Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:

client.channels.move(channelId, app, appArgs)

The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.

ASTERISK-28267 #close

Change-Id: I43d12b10045a98a8d42541889b85695be26f288a

While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.

Also cleaned up an adjacent error message for consistency.

ASTERISK-28168 #close

Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202

In Asterisk configuration, a multiline comment starts with ;-- as long as it is
not followed by another dash (i.e. ;--- is not a multiline comment).

ASTERISK-28323 #close

Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72

The res_pjsip_websocket module requires the res_http_websocket
module so ensure it is loaded. As well the res_pjsip_notify
module needs the pjsip_notify.conf configuration file so
ensure it is installed.

ASTERISK-28272

Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd

This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.

ASTERISK-28279

Change-Id: I460238c488eca4d216b9176576211cb03286e040

PJSIP assumes that these header names are not allocated, and does not
clone the name strings when reusing headers.

Block unload of res_pjsip_diversion until shutdown to ensure static
memory stays valid.

ASTERISK-28312 #close

Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9

Change-Id: Ieb332d018ae3f2fc82b9465381fde0f299af1611

This file was added to the Subversion repository on 2007-03-15 by
Russell Bryant, a Digium employee at the time.

ASTERISK-24173 #close

Change-Id: Ie866fa9d31d550467613d362b35b03c031ee594d

Rather than calling ast_odbc_find_table() in the prepare callback, call
it beforehand and pass it in to the callback to avoid the need for a
second connection.

ASTERISK-28166 #close

Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202

apply_negotiated_sdp_stream was returning a "1" when no joint
capabilities were found on an outgoing call instead of a "-1".
This indicated to res_pjsip_session that the handler DID handle
the sdp when in fact it didn't.  Without the appropriate setup,
a subsequent media frame coming in would have an invalid stream_num
and cause a seg fault when the stream was attempted to be retrieved.

apply_negotiated_sdp_stream now returns the correct "-1" and any
media is now discarded before it reaches the core stream processing.

ASTERISK-28260
Reported by: Sotiris Ganouris

Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f

This reverts commit d524ad52.

Reason for revert: This causes Contact and Via headers to have the wrong
transport address.

ASTERISK-28309 #close

Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8

If both send_registrations and send_auth are both set to yes,
outbound_auth/username must be set or we crash.

ASTERISK-27992 #close

Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d

When a contact was removed by the registrar it did not always check to see if
the circumstances involved a monitored reliable transport. For instance, if the
'remove_existing' option was set to 'true' then when existing contacts were
removed due to 'max_contacts' being reached, those existing contacts being
removed did not unregister the transport monitor.

Also, it was possible to add more than one monitor on a reliable transport for
a given aor and contact.

This patch makes it so all contact removals done by the registrar also remove
any associated transport monitors if necessary. It also makes it so duplicate
monitors cannot be added for a given transport.

ASTERISK-28213

Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a

The recent upgrade of Gerrit to 2.16 elimiated referencing a
repository in a way the jenkinsfiles were relying on so
the URL references were changed to a more consistent and supported
format.

Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e
(cherry picked from commit 5ce08457)

This module allows presence subscriptions to voicemail boxes.  This
allows common BLF keys to act as voicemail waiting indicators.

ASTERISK-28301

Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd

Delivery timeval in the smoother object will fall behind while a DTMF is
being generated.  This can eventually lead to invalid rtp timestamps.
To prevent this from happening the smoother needs to be reset after every
DTMF to keep the timing up to date.

ASTERISK-28303 #close

Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f