* say.c Changed 'digits/and' to 'vm-and' for en_GB

ASTERISK-26598 #close

Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe

* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension.  Ran into this when
developing a testsuite test.  The AMI event ExtensionStatus came out with
the hint header value containing garbage.  The AMI event PresenceStatus
also had the same issue.

* manager.c:action_extensionstate() no need to completely initialize the
hint[].  Only initialize the first element.

* pbx.c:ast_add_hint() Remove unnecessary assignment.

* chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
about the return value of ast_get_hint() there.

Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12

* Removed the AST_CHAN_TP_MULTISTREAM tech property.  We now rely
  on read_stream being set to indicate a multi stream channel.
* Added ast_channel_is_multistream convenience function.
* Fixed issue where stream and default_stream weren't being set on
  a frame retrieved from the queue.
* Now testing for NULL being returned from the driver's read or
  read_stream callback.
* Fixed issue where the dropnondefault code was crashing on a
  NULL f.
* Now enforcing that if either read_stream or write_stream are
  set when ast_channel_tech_set is called that BOTH are set.
* Added the unit tests.

ASTERISK-26816

Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2

res_config_pgsql should match the behavior of other realtime backend
drivers so that queue_log can disable adaptive logging.

ASTERISK-25628 #close
Reported by: Dmitry Wagin

Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372

This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.

Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed

In the event that a cache file is removed out from under us, we should
treat the cache entry as stale and force a refresh.

ASTERISK-26774 #close
Reported by: Igor Gamayunov

Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52

The find_table() functions NULL or a locked table pointer. We are
not consistently calling release_table() in failure paths.

Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544

Use the description of useragent from sip.conf here.

ASTERISK-26825 #close

Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755

When a subscription was being recreated and the endpoint wasn't
found, we were trying to unref the endpoint.  This was causing
FRACKs.  Removed the unref.

ASTERISK-26823 #close

Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164

This change fixes an assumption in res_pjsip that a contact will
always have a status. There is a race condition where this is
not true and would crash. The status will now be unknown when
this situation occurs.

ASTERISK-26623 #close

Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5

Outbound registration now subscribes to network change events
published by res_stun_monitor and refreshes all registrations
when an event happens.

The 'pjsip send (un)register' CLI commands were updated to accept
'*all' as an argument to operate on all registrations.

The 'PJSIP(Un)Register' AMI commands were also updated to
accept '*all'.

ASTERISK-26808 #close

Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25

... and clean them both up on uninstall.

We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64.  Now we need
to clean up the remnants in /usr/lib.

* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
  contains 'lib64' and libasterisk* shared libraries or modules
  are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
  to 'lib'.

* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
  DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.

ASTERISK-26705

Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f

The bridge_native_rtp module did not properly handle the case where
a smart bridge operation occurs while a channel is suspended. In this
scenario the module would incorrectly set up local or remote RTP
bridging despite the media having to flow through Asterisk. The remote
endpoint would see two media streams and experience wonky audio.

The module has been changed so that it ensures both channels are
not suspended when performing the native RTP bridging and this
requirement has been documented in the bridge technology.

ASTERISK-26781

Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

This change updates the documentation for the outbound_proxy option
to ensure it is consistently stated that a full SIP URI must be
provided for the option.

The res_pjsip_outbound_registration module has also been changed so
that the provided outbound_proxy value is checked to ensure it is a
URI and if not an error is output stating so.

ASTERISK-26782

Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593

This change introduces an ast_read_stream function and callback in
the channel technology which allows reading frames from all streams
and not just the default streams.

The stream number has also been added to frames. This is to allow the
case where frames are queued onto the channel instead of being read
directly from the driver.

This change does impose a restriction on reading though: a chain of
frames can only contain frames from the same stream.

ASTERISK-26816

Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce

 * Removed all 2.5.5 functional patches.
 * Updated usages of pj_release_pool to be "safe".
 * Updated configure options to disable webrtc.
 * Updated config_site.h to disable webrtc in pjmedia.
 * Added Richard Mudgett's recent resolver patches.

Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.  To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.

 * Fedora has a /usr/share/config.site that automatically tells
   autoconf to use /usr/lib64 but CentOS does not. This logic was
   copied to configure.ac and modified so systems like Ubuntu,
   which still use /usr/lib for 64-bit systems, aren't affected.

Now that we have them in the correct directory...

In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...

  - The linker cache must be rebuilt including the directory
    where the libasterisk* libraries were installed.  Only root
    can rebuild the cache.  This was busted.
  - We have to link the asterisk binary with an rpath pointing
    to the directrory where the libasterisk* libraries were
    installed.  This makes things very complicated and will happen
    over the collective dead bodies of everyone who's had to
    package a distribution with an rpath.
  - Finally, you can start asterisk with LD_LIBRARY_PATH set to the
    directrory where the libasterisk* libraries were installed.

There are no other options. So...

 * The invokation of ldconfig has been moved from main/Makefile
   to ASTTOPDIR/Makefile, the options have been removed, and
   DESTDIR/ASTLIBDIR appended.  If you aren't root, you will be
   warned after the "Asterisk Installation Compete" banner that
   you must re-run 'make install' as root, manually run
   'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
   LD_LIBRARY_PATH.

ASTERISK-26705

Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982

* A missing AST_LIST_UNLOCK() in find_table()

* The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were
  not consistently locking before calling it.

* There were a handful of other places where pgsqlConn was accessed
  directly without appropriate locking.

Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed

This change adds an ast_write_stream function which allows
writing a frame to a specific media stream. It also moves
ast_write() to using this underneath by writing media
frames provided to it to the default streams of the channel.
Existing functionality (such as audiohooks, framehooks, etc)
are limited to being applied to the default stream only.

Unit tests have also been added which test the behavior of
both non-multistream and multistream channels to confirm that
the write() and write_stream() callbacks are invoked
appropriately.

ASTERISK-26793

Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c

Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.

ASTERISK-26292

Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf