Adds a new option to SendDTMF() which will answer the specified
channel if it is not already up. If no channel is specified, the
current channel will be answered instead.

ASTERISK-30422

Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085

Adds the Signal and WaitForSignal
applications, which can be used for inter-channel
signaling in the dialplan.

Signal supports sending a signal to other channels
listening for a signal of the same name, with an
optional data payload. The signal is received by
all channels waiting for that named signal.

ASTERISK-29810 #close

Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3

Adds support for arrays to JSON_DECODE by allowing the
user to print out entire arrays or index a particular
key or print the number of keys in a JSON array.

Additionally, adds support for recursively iterating a
JSON tree in a single function call, making it easier
to parse JSON results with multiple levels. A maximum
depth is imposed to prevent potentially blowing
the stack.

Also fixes a bug with the unit tests causing an empty
string to be printed instead of the actual test result.

ASTERISK-29913 #close

Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b

Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.

ASTERISK-30262 #close

Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db

In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b3) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.

Partial fix for ASTERISK~25661.

Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff

Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).

ASTERISK-30319 #close

Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645

-----------------

This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures.  The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.

ASTERISK-30391

-----------------

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038

This reverts commit d454801c.

Reason for revert: Issue when transcoding to/from g722

Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86

Adds a new application, Broadcast, which can be used for
one-to-many transmission and many-to-one reception of
channel audio in Asterisk. This is similar to ChanSpy,
except it is designed for multiple channel targets instead
of a single one. This can make certain kinds of audio
manipulation more efficient and streamlined. New kinds
of audio injection impossible with ChanSpy are also made
possible.

ASTERISK-30180 #close

Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

ASTERISK-30280

Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a

ASTERISK-21502

Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08

chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).

This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.

* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
  INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)

The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.

The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.

Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.

ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>

Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333

msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.

This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.

ASTERISK-30283 #close

Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7

Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.

ASTERISK-30322 #close

Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b

Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.

ASTERISK-29497

Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49

Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.

ASTERISK-30150 #close

Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428

The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).

This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.

ASTERISK-30289 #close

Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07

MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.

ASTERISK-30286 #close

Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c

Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.

ASTERISK-30176

Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404

The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.

This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.

ASTERISK-30308 #close

Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4

Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5

Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.

This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.

ASTERISK-30284 #close

Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e

Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.

This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.

According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP

The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.

ASTERISK-30193 #close

Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148

The PJSIP notify CLI commands allow for using
"options" configured in pjsip_notify.conf.

This allows these same options to be used in
AMI actions as well.

Additionally, as part of this improvement,
some repetitive common code is refactored.

ASTERISK-30263 #close

Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5

Expands the pjsip logger to support the ability to filter
by SIP message method. This can make certain types of SIP debugging
easier by only logging messages of particular method(s).

ASTERISK-30146 #close

Co-authored-by: Sean Bright <sean@seanbright.com>
Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f

Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.

This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.

These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.

Current default behavior remains unchanged.

ASTERISK-30091 #close

Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832

Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.

ASTERISK-30254 #close

Change-Id: Ie2329ff245248768367d26749c285fbe823f6414

This patch adds support for mediasec SIP headers and SDP attributes.
These are defined in RFC 3329, 3GPP TS 24.229 and
draft-dawes-sipcore-mediasec-parameter. The new features are
implemented so that a backbone for RFC 3329 is present to streamline
future work on RFC 3329.

With this patch, Asterisk can communicate with Deutsche Telekom trunks
which require these fields.

ASTERISK-30032

Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2

Adds the n "no answer" option to the Bridge application
so that answer supervision can not automatically
be provided when Bridge is executed.

Additionally, a mechanism (dialplan variable)
is added to prevent bridge targets (typically the
target of a masquerade) from answering the channel
when they enter the bridge.

ASTERISK-30223 #close

Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d

Adds the n option to not answer the channel when calling
BridgeWait, so the application can be used without
forcing answer supervision.

ASTERISK-30216 #close

Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a

Adds an option that will play an audio file
to the party while AMD is running on the
channel, so the called party does not just
hear silence.

ASTERISK-30179 #close

Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6

Adds the EXPORT function, which allows write
access to variables and functions on other
channels.

ASTERISK-29432 #close

Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf

This patch adds a new option to the 100rel parameter for pjsip
endpoints called "peer_supported". When an endpoint with this option
receives an incoming request and the request indicated support for the
100rel extension, then Asterisk will send 1xx responses reliably. If
the request did not indicate 100rel support, Asterisk sends 1xx
responses normally.

ASTERISK-30158

Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad

Adds TRIM, LTRIM, and RTRIM, which can be used
for trimming leading and trailing whitespace
from strings.

ASTERISK-30222 #close

Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554

Adding user=phone to local-side uri's when user_eq_phone=yes is set for
an endpoint. Previously this would only add the header to the To and R-URI.

ASTERISK-30178

Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600

This change adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.

ASTERISK-30135

Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08

ASTERISK-30046

Change-Id: Ie77e0648f8b0b1c2159fb24662d1989cfd4cc36d