Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5

Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.

This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.

ASTERISK-28780

Change-Id: I8044c06daa06f0f16607788c596f55623be26f58

This is a generic jenkinsfile to build Asterisk and optionally
perform one or more of the following:
 * Publish the API docs to the wiki
 * Run the Unit tests
 * Run Testsuite Tests

This job can be triggered manually from Jenkins or be triggered
automatically on a schedule based on a cron string.

Change-Id: Id9d22a778a1916b666e0e700af2b9f1bacda0852

bridge_p2p_rtp_write will forward rtp to the bridged rtp instance
without modifying the ssrc.  However, it is not updating the SSRC
in the bridged rtp.  Thus, when SSRC packets are generated, they
have the correct SSRC for the sender.

ASTERISK-28773 #close

Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478

If ICE support is enabled but not negotiated, the rtp->ice structure is
not being destroyed. This leads to Asterisk waiting for ICE to complete
instead of immediately starting the DTLS handshake, resulting in the
call leg having no RTP.

ASTERISK-28769 #close

Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27

If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.

The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.

ASTERISK-28759

Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec

A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.

Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.

ASTERISK-26711 #close
Reported by: Vitold

Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0

These tones come from http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

ASTERISK-23407

Change-Id: I48e2285f1e5bb29b3335f762006f66c423d0fcb8

This change introduce a CLI command for the RTP to display the general
configuration.

In the first step add the follow fields of the configurations:
  - rtpstart
  - rtpend
  - dtmftimeout
  - rtpchecksum
  - strictrtp
  - learning_min_sequential
  - icesupport

Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f

The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in
65b83815 such that we would never
actually make a DNS TXT query as described.

This patch restores the documented behavior.

ASTERISK-19460 #close
Reported by: George Joseph

Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376

ASTERISK-28766 #close

Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1

Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f

The receive buffer will now grow if we end up flushing the
receive queue after not receiving the expected packet in time.
This is done in hopes that if this is encountered again the
extra buffer size will allow more time to pass and any missing
packets to be received.

The send buffer will now grow if we are asked for packets and
can't find them. This is done in hopes that the packets are
from the past and have simply been expired. If so then in
the future with the extra buffer space the packets should be
available.

Sequence number cycling has been handled so that the
correct sequence number is calculated and used in
various places, including for sorting packets and
for determining if a packet is old or not.

NACK sending is now more aggressive. If a substantial number
of missing sequence numbers are added a NACK will be sent
immediately. Afterwards once the receive buffer reaches 25%
a single NACK is sent. If the buffer continues to grow and
reaches 50% or greater a NACK will be sent for each received
future packet to aggressively ask the remote endpoint to
retransmit.

ASTERISK-28764

Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41

When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.

Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.

ASTERISK-28755 #close

Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f

When building check_expr2 with ASLR PIE hardening enabled the linker
fails. This is resolved by adding the regular compiler flags when
building the object files from ast_expr2f.c and ast_expr2.c.

Note: The STANDALONE define is removed because it is already defined in
_ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives
'--enable-dev-mode'.

Also, a Makefile variable "CROSS_COMPILING" is added so that the
build system doesn't try to run check_expr2 when cross-compiling,
because that will fail the build as will.

ASTERISK-28685 #close

Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915

Update the state of remote_hold immediately on receipt of remote
SDP so that the information is available when building the SDP
answer

ASTERISK-28754 #close

Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f

Change-Id: Icba97905e331812f129e5966e91a59b104c7a748

There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.

Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e

There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.

Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113

Warnings without this:

res_config_mysql.c: In function 'update2_mysql':
res_config_mysql.c:741:15: warning: format '%llu' expects argument of type
    'long long unsigned int', but argument 6 has type 'my_ulonglong'
    {aka 'long unsigned int'} [-Wformat=]
ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n",
    numrows, tablename);

(reformatted for readability within line-wrap)

Change-Id: I2af4d419a37c1a7eeee750cf9ae4a9a2b3a37fd3