In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use.  To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes"  In Asterisk versions <18, the default
is "no" preserving existing behavior.  Beginning with
Asterisk 18, the option will default to "yes".

NOTE:  This change does not affect UserEvents or the ARI
TextMessageReceived events.

* Added the "hide_messaging_ami_events" option to asterisk.conf.

* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
  the "Message/ast_msg_queue" channel if the option is set in
  asterisk.conf.  This suppresses the reporting of the events.

Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b

Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.

ASTERISK-28714

Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a

ASTERISK-26082 #close
Reported by: Alex

Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb

Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.

This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.

ASTERISK-28423 #close

Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27

The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.

This patch makes it so the cleanup code is executed post 'update'.

ASTERISK-28679 #close

Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b

We need to wait for the message sending callback to finish to know if
we succeeded or failed.

ASTERISK-25421 #close
Reported by:  Dmitriy Serov

Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059

Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).

If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.

This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.

Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.

(An alternative fix would be to set ignoresdpversion=yes on the peer.)

ASTERISK-28686

Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee

If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.

We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.

ASTERISK-28139 #close
Reported by: Paul Brooks

Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3

When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.

* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.

NOTE:  This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.

ASTERISK-28677
Reported by: Maciej Michno

Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca

Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.

We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.

Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.

Additionally:

* Change 'enablestatic' to 'enable_static' but keep the former for
  backwards compatibility.
* Improve some internal variable names

ASTERISK-28710 #close

Change-Id: I647510f796473793b1d3ce1beb32659813be69e1

The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX
to match behavior of "no timeout" defined in comment.

ASTERISK-28702 #close

Change-Id: I4ea015986e061374385dba247b272f7aac60bf11

ASTERISK-24484 #close
Reported by: Dan Jenkins

Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec

SILK @ 24kHz is not shown in the 'core show translation' output because of an
off-by-one-error. Discovered while looking into ASTERISK~19871.

ASTERISK-28706
Reported by: Sean Bright

Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510

Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84

* The MailboxExists dialplan application was deprecated on 2006-09-26
  in Asterisk 1.6.0 (commit ec83b111)

* The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in
  Asterisk 11.0.0 (commit fd64bb66)

Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f

In af90afd9, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.

Whitespace only, no functional change.

ASTERISK~23324
Reported by: Kevin McCoy

Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87

ASTERISK-28695 #close
Reported by: Kevin Flyn

Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8

lws2sws() does not stop trying to handle header continuation lines
even after all headers have been found. This is problematic if the
first character of a SIP message body is a space or tab character, so
we update to recognize the end of the message header.

ASTERISK-28693 #close
Reported by: Frank Matano

Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df

ast_store_realtime() is not NULL tolerant, so we need to initialize
the field values we pass to it to the empty string to avoid a crash.

ASTERISK-23739 #close
Reported by: Stas Kobzar

Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c