ASTERISK-29244

Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5

Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.

ASTERISK-29260
Reported by: Alexander Traud

Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458

If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext.  You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.

* Added logic to iax2_call() and authenticate_reply() to print
  a warning and hanhup the call if encryption is requested and
  there's no secret or auth method.  This prevents the crash.

* Added the ability to specify a default "auth" in the [general]
  section of iax.conf.

ASTERISK-29624
Reported by: N A

Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

As described in the issue, /tmp is not a suitable location for a
large amount of cached media files, since most distributions make
/tmp a RAM-based tmpfs mount with limited capacity.

I opted for a location that can be configured separately, as opposed
to using a subdirectory of spooldir, given the different storage
profile (transient files vs files that might stay there indefinitely).

This commit just makes the cache directory configurable, and changes
the default location from /tmp to /var/cache/asterisk.

ASTERISK-29143

Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d

When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.

ASTERISK-29118 #close

Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb

Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a

Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:

  dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
  stun, stun_packet

These debug categories can be enable/disable via an Asterisk CLI command.

While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).

ASTERISK-29054 #close

Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849

app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge.  To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second.  The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".

Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a

ASTERISK-29083 #close

Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003

This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function
is called on a channel prior to answering a warning is issued and the
function returns unsuccessful.

ASTERISK-28878 #close

Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb

Added a new log formatter called "plain" that always prints
file, function and line number if available (even for verbose
messages) and never prints color control characters.  It also
doesn't apply any special formatting for verbose messages.
Most suitable for file output but can be used for other channels
as well.

You use it in logger.conf like so:
debug => [plain]debug
console => [plain]error,warning,debug,notice,pjsip_history
messages => [plain]warning,error,verbose

Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d

Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.

Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.

Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.

Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1

If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.

(Actually, commit ed615afb already
partially fixed it for shared_lastcall.)

ASTERISK-28951

Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6

Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.

This causes unexpected rport handle at the other end.

Added option for disable this behaviour in the pjsip.conf.

This is a system option, but working as a gloabl option.

ASTERISK-28959

Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc

This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.

Note, the AMI version has been bumped for this change.

ASTERISK-28945 #close

Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb

The Streams API becomes the home for the core ACN capabilities.
These include...

 * Parsing and formatting of codec negotation preferences.
 * Resolving pending streams and topologies with those configured
   using configured preferences.
 * Utility functions for creating string representations of
   streams, topologies, and negotiation preferences.

For codec negotiation preferences:
 * Added ast_stream_codec_prefs_parse() which takes a string
   representation of codec negotiation preferences, which
   may come from a pjsip endpoint for example, and populates
   a ast_stream_codec_negotiation_prefs structure.
 * Added ast_stream_codec_prefs_to_str() which does the reverse.
 * Added many functions to parse individual parameter name
   and value strings to their respectrive enum values, and the
   reverse.

For streams:
 * Added ast_stream_create_resolved() which takes a "live" stream
   and resolves it with a configured stream and the negotiation
   preferences to create a new stream.
 * Added ast_stream_to_str() which create a string representation
   of a stream suitable for debug or display purposes.

For topology:
 * Added ast_stream_topology_create_resolved() which takes a "live"
   topology and resolves it, stream by stream, with a configured
   topology stream and the negotiation preferences to create a new
   topology.
 * Added ast_stream_topology_to_str() which create a string
   representation of a topology suitable for debug or display
   purposes.
 * Renamed ast_format_caps_from_topology() to
   ast_stream_topology_get_formats() to be more consistent with
   the existing ast_stream_get_formats().

Additional changes:
 * A new function ast_format_cap_append_names() appends the results
   to the ast_str buffer instead of replacing buffer contents.

Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56

The AMI action and CLI command did not take into account the properties
of full backend caching. This resulted in an expired object remaining
removed until a full backend update occurred, instead of having the
object updated when needed.

This change makes it so that the AMI action and CLI command for object
expire will now fail instead of putting the cache into an undesired
state. If full backend caching is enabled then only operations
which act on the entire cache are available.

ASTERISK-28942

Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4

The PJSIP packet logger now has the following CLI commands:

pjsip set logger pcap <filename>

When used this will create a pcap file containing the incoming
and outgoing SIP packets, in unencrypted form.

pjsip set logger verbose <on / off>

This allows you to toggle logging to verbose on and off.

pjsip set logger host <IP/subnet mask> add

This allows you to add an additional IP address or subnet
mask to logging, allowing you to log multiple instead of
just a single IP address or all traffic.

The normal "pjsip set logger host" CLI command has also been
expanded to allow subnet masks as well.

ASTERISK-28895

Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e

This change adds the same variable functionality that
is available for originating a channel to the create
call. Now when creating a channel you can specify
dialplan variables to set instead of having to do another
API call.

ASTERISK-28896

Change-Id: If13997ba818136d7c070585504fc4164378aa992

There are a lot of moving parts in this patch, but the focus of it is on
the verification of the signature using a public key located at the
public key URL provided in the JSON payload. First, we check the
database to see if we have already downloaded the key. If so, check to
see if it has expired. If it has, redownload from the URL. If we don't
have an entry in the database, just go ahead and download the public
key. The expiration is tested each time we download the file. After
that, read the public key from the file and use it to verify the
signature. All sanity checking is done when the payload is first
received, so the verification is complete once this point is reached.

The XML has also been added since a new config option was added to
general (curl_timeout). The maximum amount of time to wait for a
download can be configured through this option, with a low value by
default.

Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c

Some places in Asterisk did not treat the formats on a stream
as immutable when they are.

The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.

An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.

ASTERISK-28846

Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe

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