* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers.  The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped.  However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".

ASTERISK-27651

Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977

ASTERISK-27651

Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34

Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068

Change-Id: I3c7106ff77009754725cee790eadf5da44154ab6

* Changed to create ami_event string only when the given blob is not
json_null().
* Fixed bad expression.

ASTERISK-27621

Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f

ASTERISK-27652 #close

Change-Id: I78a0d38bfd8d0d82830f3d53da04872d6b67284d

Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8

ast_str_append_event_header() could potentially leak and corrupt memory if
the ast_str needed to expand to add the AMI event header.

* Fixed to return error if the ast_str_append() failed.

Change-Id: I92f36b855540743b208d76e274152ee2d758176d

* Made not allocate memory if the channel snapshot is an internal channel.

* Free memory earlier when no longer needed.

Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38

Verified nothing in the testsuite lists this module as a dependency.

Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813

Currently in app_confbridge if someone mutes a channel while that channel
is talking, the talk detection code is suspended while the channel is
muted.  As far an an external observer is concerned, the muted channel's
talk status is still "talking" even though the channel is not contributing
audio to the conference bridge.  When the channel is later unmuted, it
takes the usual 'dsp_silence_threshold' option time to clear the talking
status even though the channel may have stopped talking while the channel
was muted.

* In bridge_softmix.c, clear the talking status and report talking stopped
if the channel was talking when the channel is muted.  When the channel is
unmuted and the channel is still talking then report the channel as
talking since it is contributing audio to the bridge again.

ASTERISK-27647

Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e

The dsp_talking_threshold does not represent time in milliseconds.  It
represents the average magnitude per sample in the audio packets.  This is
what the DSP uses to determine if a packet is silence or talking/noise.

Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443

Need to include signal.h to define pthread_kill() and SIGURG.

Change-Id: I10ae3aa4bf8e7386ac29ade78c0f2caed8e674fa

pjproject does not have a function to reverse pjsip_inv_usage_init.
This means we need to ignore any calls to the functions once shutdown is
final.

ASTERISK-27571 #close

Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d

Create ast_atomic macro's to provide a consistent interface to the
common functionality of __atomic and __sync built-in functions.

ASTERISK-27619

Change-Id: Ieba3f81832a0e25c5725ea067e5d6f742d33eb5b

In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped.  This same
process is now also applied to inbound subscriptions.

Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.

To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.

ASTERISK-27612
Reported by: Ross Beer

Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36

res_pjsip_registrar_expire remains as an empty module for now.

Change-Id: Ib93698938bae548d2199cb542f3692d1a171239f

The sample modules.conf explicitly loaded res_musiconhold.so.  This is
redundent as autoload=yes is already set.  It causes warnings if
res_musiconhold.so was not installed and results in an unexpected load
if the admin disables autoload without remembering to remove the
res_musiconhold load statement.

Also remove reference to unknown module pbx_gtkconsole.

Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a