As described in issue ASTERISK-26282 the AEL parser creates macros with
extension '~~s~~'.  app_macro searches only for extension 's' so the
created extension cannot be found.  with this patch app_macro searches for
both extensions and performs the right extension.

ASTERISK-26282 #close

Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb

Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth
is not supported in IAX2 protocol. Please refer to section 8.6.13 of
RFC 5456.

But plaintext auth is still supported by Asterisk implementation of IAX2.
This support should be dropped.

Patch, based on asterisk-dev discussion, adds deprecation warning on
startup if 'auth' is set to 'plaintext', changes default values of
'auth' from 'md5, plaintext' to 'md5'.

Patch is safe in terms of backwards compatibility, will work even if
remote peers have auth=plaintext and we have defaults.

auth=plaintext setting will remain deprecated in Asterisk 14 and 15,
and IAX2 plaintext support will be removed in Asterisk 16.

ASTERISK-22820 #close

Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf

ast_multicast_rtp_create_options now checks for NULL or empty options

Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362

NOTE: This patch was submitted earlier and reverted because of a failing
test. The test has been patched so that it adjusts for the changes here,
so this is being resubmitted for review.

One feature that confbridge has is the ability to play sounds to all
participants in the conference. Prior to this commit, the algorithm for
this was as follows:

* Grab the playback lock
* Push the conference announcer channel into the bridge
* Play back the sound
* Pull the conference announcer channel from the bridge
* Release the playback lock

The issue here is that the act of adding the playback channel to the
bridge and removing it for each announcement is expensive. Amongst the
expenses:

* The announcer channel is imparted into the bridge, meaning a new
  thread is spun up for each playback.
* When the announcer is added or removed from the bridge, it results
  in the BRIDGEPEER channel variable being set on all channels in the
  bridge. This requires keeping the bridge locked and locking each
  individual channel in order to set it.
* There's also just the general overhead of adding the channel and
  removing it from the bridge. The bridge potentially has to reconfigure
  every single time

With this commit, the paradigm for playing back announcements has
shifted.

* The announcer channel is now added to the bridge when the conference
  is allocated, and it is hung up when the conference is destroyed.
* A taskprocessor is used to queue playbacks onto the announcer channel.
  This keeps the behavior from before where playbacks do not overlap.
* The announcer channel is no longer placed into the bridge as
  departable. Since we are not constantly removing the channel from
  the bridge, it is safe to add the channel using an independent thread
  and simply hang the channel up when it is time for the conference to
  be destroyed.

The use of the taskprocessor for playbacks opens up the interesting
possibility of having asynchronous announcements played. In this commit,
however, the behavior is still exactly the same as it previously was.

ASTERISK-26289
Reported by Mark Michelson

Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0

This reverts commit 5aa87730.

Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491

The compilation failed for devmode
--enable DONT_OPTIMIZE
--enable BETTER_BACKTRACES
--enable DO_CRASH
--enable TEST_FRAMEWORK

res_pjsip/pjsip_configuration.c: In function dtls_handler:
res_pjsip/pjsip_configuration.c:974:20: error:
back may be used uninitialized in this function [-Werror=maybe-uninitialized]
int size = strlen(front);
           ^
cc1: all warnings being treated as errors

Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580

When res_odbc_transaction depended on res_odbc, it got the generic_odbc
headers and libs implicitly. Now that it no longer depends on res_odbc,
its dependency on generic_odbc must be explicit.

Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911

PJProject supports a lot of platforms even Windows, some with different defaults
when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS,
"/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows.

Because of this, if configured with just an IPv6 address/transport, PJProject
listens to both IPv4 and IPv6. However, this is not supported by the PJProject
team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP,
incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only
server which accepts incoming connections on IPv4.

If you try to configure two transports, one with IPv4 and one with IPv6 on the
same interface, as expected by the PJProject team, the IPv4 transport is not
able to bind because the IPv6 transport listens to both already.

One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide.
Then, you are able to configure two transports, one for each IP version on the
same interface. That way, you get a server which works with IPv4 clients and
IPv6 clients at the same time over the same interface.

Here, this change sets this parameter directly within PJProject to match the
expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack
servers out of the box like in chan_sip. This change was accepted by the
PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected
to arrive in the next version, PJProject 2.6.0. Until then, this change is
incorporated in the bundled PJProject of Asterisk.

ASTERISK-26309

Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae

updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs

Change-Id: I279335a2625261a8492206c37219698f42591c2e
(cherry picked from commit 6f448f32)

When using the migration script sip_to_pjsip.py, cert_file was not migrated to
pjsip.conf. A previous change regarding this contained a copy/paste error.

ASTERISK-22374

Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b

When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL
SSLv23_method. This was documented incorrectly in the file sip.conf.sample.

SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method
enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that
function should have been called 'secure_method' or 'automatic_method' back in
the 90s.

Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if
you face a server which has problems like not falling back to TLSv1.0
automatically.

ASTERISK-24425

Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3

Historically, Asterisk has always specified annexb=no for the g729 format.
However, when using res_pjsip no format attribute was specified. This patch
makes it so the SDP now contains a format attribute line with annexb=no.

Note, that this means only g729a is negotiated. Even for pass through support.
According to rfc7261 the type of annex used (a or b) is dependent upon the
answerer. However, Asterisk being a back to back user agent makes this tricky
to support at this time, thus we only allow annex 'a' for now.

ASTERISK-26228 #close
patches:
  res_format_attr_g729.c submitted by Jason Parker (license 4993)

Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0

Given resource paths did not have 'json' substituted in for the '{format}'. For
some auto generated documentation/comment strings it resulted in something like
the following:

"... REST handler for /api-docs/sounds.{format}"

This patch makes sure the resource api's path is properly substituted.

ASTERISK-25472 #close

Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23

The MODULEINFO dependencies between these 2 modules was reversed.
res_odbc should depend on res_odbc_transaction, not the other way
around.

ASTERISK-25984 #close

Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f

A recent update had a copy/paste error where the unused variable 'val' was
being passed to the set_value function instead of the 'method' value itself.

This patch passes in the right variable.

ASTERISK-22374

Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06