OpenSSL has 2 levels or error processing.  It's possible for the
top layer to return SSL_ERROR_SYSCALL but the lower layer return
no error, in which case processing should continue.  Only the top
layer was being examined though so connections were being torn
down when they didn't need to be.  This patch adds the examination
of the lower level codes, and if they return no errors, allows
processing to continue.

ASTERISK-27001
Reported-by: Ian Gilmour
patches:
	pjproject-2.6.patch submitted by Ian Gilmour (license 6889)

Updated-by: George Joseph and Sauw Ming (Teluu)

Merged to upstream pjproject on 7/27/2017 (commit 5631)

Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

Change-Id: Ia578ede1a55b21014581793992a429441903278b

Case scenario with Applications ARI:

* Once you subscribe to deviceState with Applications REST API, it will be
added into subscription pool.

* When you unsubscribe it will remove from the device_state_subscription
hash table but not from the subscription pool.

* When you subscribe again, it will add it to pool again.

* Now you will have two subscriptions and you will receive same event
twice.

This fix should now remove deviceState subscription from pool and it
should fix unsubscribe on deviceState.

ASTERISK-27130 #close

Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4

This change adds VP9 as a known codec and creates a cached
"vp9" media format for use.

Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc

The following testsuite voicemail tests were failing to re-enter the
mailbox after the first login attempt.

tests/apps/voicemail/authenticate_invalid_mailbox
tests/apps/voicemail/authenticate_invalid_password

The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt.  Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.

* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.

Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8

A few minor problems were found in comments in format.h.  This patch fixes them.

Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94

The seconds and minutes files have always existed in the base language
directory of the Core package. So say.c has always been calling the wrong
location (under digits/) for those two files and in the case of second and
minute they didn't exist in the Core packages at all.

The 1.6 sounds release moves the second and minute files into Core from
Extra for the languages that already had them. A future release will include
the second and minute files for languages that didn't already have them.

This patch just changes all the target locations for second, seconds,
minute, and minutes that were under the digits subdir to be under the root of
sounds instead. Which is where the sounds will be for some languages after 1.6
sounds and for all languages after a future release.

ASTERISK-25810 #close

Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702
Reported-by: Nicolas Riendeau

Incrementing version for the Extra sounds release. 1.5.1 Extra sounds
removes two prompts that were moved into the Core packages in the 1.6 Core
sounds release.

ASTERISK-27142 #close

Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7

Ready for next major version

Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0

Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5

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

This change does a few things to improve packet loss and renegotiation:

1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.

2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.

3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.

4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.

5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.

ASTERISK-27143

Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45

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