Change-Id: I1e4d37415f3034abe36496dc30209c2303e6af5c

The download_externals script wasn't getting the PJPROJECT_BUNDLED
environment variable passed down to it so it wasn't downloading
the appropriate variant of res_digium_phone.  This could cause
crashes in the DPMA.

Change-Id: I5daa9369c7af1fd556d892e89a85f279a2533425

Apply flag -fno-partial-inlining on default optimization if and only if
gcc version >= 8.2.1 (this is the current ver on Fedora and Ubuntu).
This is done to avoid a bug that causes arithmetic calculations to fail
if the following conditions are met:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. There must exist a certain combination of multithreading.
6. Optimization level -O2 and -O3
7. Flag -fpartial-inline activated (default when optimization level>=2)
The following link points to a similar gcc bug reported in 2015. This leads me
to believe the bug has regressed. Note I am not able to replicate this bug
in an environment other than Asterisk + Test Framework + Test_cel because the
multithreading combination that causes it seems to be unique. Therefore I
am temporarily abandoning any thoughts of reporting the new occurrence of this
bug to gcc.gnu.org.  https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65307

Change-Id: Ibd1afe60e0a38b88e85fdcd9b051004601c2f102

Currently, when the Asterisk calculates rtp statistics, it uses
sample_count as a unsigned integer parameter. This would be fine
for most of cases, but in case of large enough number of sample_count,
this might be causing the divide by zero error.

ASTERISK-28321

Change-Id: If7e0629abaceddd2166eb012456c53033ea26249

chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP.  This new flag allows chan_pjsip to have the same
behavior as chan_sip.

ASTERISK-28322 #close

Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a

Add a json_pack at startup that will fail if runtime links against a
library older than jansson-2.11.

Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a

ASTERISK-28328 #close

Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b

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