When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".

ASTERISK-28841

Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13

Allow voice volume to be multiplied or divided by a floating point number.

ASTERISK-28813

Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c

Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437

Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7
(cherry picked from commit c9cd6812)

When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.

This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.

ASTERISK-28787

Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f

A pure blacklist is not good enough, we need a whitelist mechanism as
well, and the simplest way to do that is to re-use existing ACL
infrastructure.

This makes it simpler to blacklist say an entire block (/24) except a
smaller block (eg, a /29 or even a /32).  Normally you'd need to
recursively split the block, so if you want to blacklist a /24 except
for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28.  I
feel that having an ACL instead of a blacklist only is clearer.

Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

This change introduce a CLI command for the RTP to display the general
configuration.

In the first step add the follow fields of the configurations:
  - rtpstart
  - rtpend
  - dtmftimeout
  - rtpchecksum
  - strictrtp
  - learning_min_sequential
  - icesupport

Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f

Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f

When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.

Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.

ASTERISK-28755 #close

Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f

The optional synchronization behavior created in
64906c4c is now the default for
MixMonitor.

* Add a new flag 'n' that allows for this behavior to be turned off

* Add a notice when the 'S' option is used indicating that it is no
  longer necessary

Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4

When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.

* If the user wants to write to a wav49 file, make sure that it is
  reflected properly in MIXMONITOR_FILENAME.

* Add a note to the documentation describing this behavior.

* Add a note in main/file.c indicating that app_mixmonitor needs to be
  changed if the logic in build_filename was changed.

ASTERISK-24798 #close
Reported by: xrobau

Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c

Although the wiki page for the new CHANGES and UPGRADE scheme
states that the files must have the ".txt" suffix, the READMEs
didn't.

Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05
(cherry picked from commit 7416703f)

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

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 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

ASTERISK-28695 #close
Reported by: Kevin Flyn

Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8

This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.

A description of the protocol can be found on the above referenced
GitHub page.  A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.

ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.

ASTERISK-28484 #close

Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5

We allow for 'maxredirs' to be set, but this value is ignored when
followlocation is not enabled which, by default, it is not.

ASTERISK-17491 #close
Reported by: candrews

Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a

The QueueMemberPause AMI event includes two fields that return the
reason a member was paused.

* In release branches, deprecate Reason in favor of PausedReason.
* In master, remove the Reason field entirely.

ASTERISK-28349 #close
Reported by: Niksa Baldun

Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296

Adds source port matching support when IP matching is used:

  [example]
  type = identify
  match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444

If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.

ASTERISK-28639 #close
Reported by: Mitch Claborn

Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92

Dialplan has to be careful about passing an empty device list or empty
positions in the list.  As a result, dialplan has to check for these
conditions before using ChanIsAvail.  Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.

* Made tolerate empty positions in the device list.

* Simplified the code and eliminated some unnecessary indention.

ASTERISK-28638

Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3

* Made BridgeAdd not hangup the call if there is a problem.
* Reduced message level from warning to verbose for normal exception
cases.
* Added a loop safety check to BridgeAdd.
* Made BridgeAdd set BRIDGERESULT with the status when dialplan is
resumed.

Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426

Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Dial.  Simplify dialplan by making Dial
handle these conditions gracefully.

* Made tolerate empty positions in the dialed device list.

* Reduced some message log levels from notice to verbose.

ASTERISK-28638

Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9

Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Page.  Simplify dialplan by making Page
handle these conditions gracefully.

* Made tolerate empty positions in the paged device list.

* Reduced some warnings associated with the 's' option to verbose
messages.  The warning level for those messages really serves no purpose
as that is why the 's' option exists.

ASTERISK-28638

Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3

This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.

ASTERISK-28629

Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc

ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee

A previous patch:

Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.

This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).

This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.

The default is disabled keeping with the old behavior.

ASTERISK-28660

Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a

add missing argument "rtt" and "status" to the documentation

The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.

This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)

ASTERISK-28626

Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d

This reverts commit 7e3015d7.

Reason for revert: Regression in XML validation.

validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)

As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.

Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f

If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.

Action: Originate
Channel: Local/1111
Application: Originate
Data: Local/2222,app,System,touch /tmp/owned

If the "system" authorization isn't set, we now block the
Originate app as well as the System, Exec, etc. apps.

ASTERISK-28580
Reported by: Eliel Sardañons

Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa

add missing argument "rtt" and "status" to the documentation

ASTERISK-28626
Change-Id: I8419e4c8203e411b87d93dc395acdbcf7526dedf

When user wants to send json data, the default Content-Type header
is incorect (application/x-www-form-urlencoded). This patch allows
to set any custom headers so the Content-Type header can be
overriden. User can set multiple headers by multiple calls of
curlopt(). This approach is not consistent with other parameters,
but is more readable in dialplan than one call with multiple
headers.

ASTERISK-28613

Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9

This reverts commit fd2e8d0d.

Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.

ASTERISK-28567 #close

Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41

Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis

ASTERISK-28542 #close

Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8

Added "like" support for 'core show taskprocessors'. Now you
can specify a specific set of taskprocessors (or just one) by
adding the keyword "like" to the above command, followed by
your search criteria.

Change-Id: I021e740201e9ba487204b5451e46feb0e3222464

Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.

Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa

Added two new CLI commands to reset stats for taskprocessors. You can
reset stats for a single, specific taskprocessor ('core reset
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
('core reset taskprocessors'). These commands will reset the counter for
the number of tasks processed as well as the max queue size.

Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d

This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.

ASTERISK-28533

Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492

This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.

Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.

ASTERISK-28512

Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2

This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.

ASTERISK-17808

Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce