Ready for next major version

Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0

AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0

Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago

Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7

Also add new corosync packages to install_prereq.

Reported by Travis Ryan in #asterisk-dev

Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db

The maximum packet size for PJSIP has been increased to handle the
multiple streams being added for WebRTC.

Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3

When a participant leaves a bridge while operating in SFU mode
their respective stream on every other participant needs to be
removed. Leaving the stream out of the new topology results in
every stream after it being moved and reordered. This causes
problems with clients. Instead simply mark the stream as removed
which leaves it in place in the SDP and doesn't reorder or touch
any other streams.

ASTERISK-27136

Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1

Streams are never truly removed in SDP, they still occupy
a location within the SDP. This location can be reused by
another stream if it so chooses.

This change takes advantage of this such that if a new stream
is needing to be added for a new participant any removed streams
are instead replaced first. This reduces the size of the SDP
and the number of streams.

ASTERISK-27134

Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d

This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.

ASTERISK-27133

Change-Id: I6200f611ede709602ee9b89501720c29545ed68b

This commit fixes two possible scenarios:

* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.

ASTERISK-27123 #close

Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625

On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.

This commit fixes it by making sure we check for existing subscription
first.

This was verified with 'core show taskprocessors' CLI command.

ASTERISK-27122 #close

Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9

This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:

  rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
  media_encryption=dtls
  dtls_verify=fingerprint
  dtls_setup=actpass

When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.

Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.

ASTERISK-27119 #close

Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd

Added necessary lines to make the en_NZ language set selectable and to get
core sounds 1.6 pulled down.

ASTERISK-26807 #close
ASTERISK-25816 #close
ASTERISK-26274 #close

Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4

This adds support for parsing timelen values from config files.  This
includes support for all flags which apply to PARSE_INT32.  Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.

Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.

Testing is included.

ASTERISK-27117 #close

Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554

BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.

This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.

For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.

ASTERISK-27118

Change-Id: I96c0920b9f9aca7382256484765a239017973c11

Currently when rtp is paused, no packets are written to the
recorded audio file, causing the silence to be skipped and recording
not properly time aligned.  The read handler as been adapted to
return a silence frame of the correct size.

ASTERISK-27128 #close

Change-Id: I2d7f60650457860b9c70907b14426756b058a844

arm the t38 webhook always, so we can correctly reject a
T38 negotiation request when t38 is disabled on a channel

Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d

This adds a parameter to ast_waitfordigit_full which can be used to only
stop waiting when certain expected digits are received.  Any unexpected
DTMF digits are simply ignored.

This also creates a new dialplan application WaitDigit.

ASTERISK-27129 #close

Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9

In say_date_generic the timezonename parameter is passed but never
used. Fix it by passing it to the ast_localtime function.

ASTERISK-27124

Change-Id: I63106b8db10426d417d7275f22554a616e92fae4

ASTERISK-27127 #close
Reported by: HZMI8gkCvPpom0tM

Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789