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==============================================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
=== doc/CHANGES-staging/README.md FOR MORE DETAILS.
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
------------------------------------------------------------------------------
app_broadcast
------------------
* A Broadcast application is now available which allows
for asynchronous one-to-many and many-to-one channel audio.
app_directory
------------------
* A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.
app_read
------------------
* A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.
app_senddtmf
------------------
* A new option has been added 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.
app_signal
------------------
* Adds Signal and WaitForSignal applications
which can be used for signaling or as a
simple message queue in the dialplan.
func_json
------------------
* Additional parsing capabilities have been added to the
JSON_DECODE function, including support for arrays
and recursive indexing.
res_phoneprov
------------------
* On multihomed Asterisk servers with dynamic SERVER template variables,
reloading this module is no longer required when re-provisioning your
phone to another interface address (e.g. when moving between VLANs.)
res_pjsip_rfc3326
------------------
* Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
addition to currently supported Q.850). The first header found will be used to set
the HANGUPCAUSE variable.
res_pjsip_session
------------------
* The overlap_context option now allows explicitly
specifying a context to use for overlap dialing matches.
res_rtp_asterisk
------------------
* This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is available 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
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* The AOCMessage action can now be used to generate AOC-S messages.
Add support for named capture agent.
------------------
* A name for the capture agent can now be specified
using the capture_name option which, if specified,
will be sent to the HEP server.
app_if
------------------
* Adds the If, ElseIf, Else, EndIf, and ExitIf applications
for conditional execution of a block of code.
app_mixmonitor
------------------
* The d option for MixMonitor now allows deleting
the original recording when MixMonitor exits,
which can be useful when MixMonitor copies it
somewhere else before exiting.
* Adds the c option to use the real Caller ID on
the channel in voicemail recordings as opposed
to the Connected Line.
app_voicemail
------------------
* The voicemail user option attachextrecs can
now be set to control whether external recordings
trigger voicemail email notifications.
cdr
------------------
* Two new options have been added which allow
bridging and dial state changes to be ignored
in CDRs, which can be useful if a single CDR
is desired for a channel.
chan_dahdi
------------------
* FXO channels (FXS signaled) that don't use callerid or
distinctive ring detection can now be configured
to enter the dialplan immediately using immediate=yes,
instead of waiting for at least one ring.
pbx_builtins
------------------
* It is now possible to not wait for media on
a channel when answering it using Answer,
by specifying the i option.
res_pjsip
------------------
* Added options "security_negotiation" and "security_mechanisms" to pjsip
endpoints and registrations. "security_negotiation" can be set to "no" (default)
or "mediasec", and "security_mechanisms" can be a list of comma-separated
security_mechanisms in the form defined by RFC 3329 section 2.2.
* A new option named "all_codecs_on_empty_reinvite" has been added to the
global section. When this option is enabled, on reception of a re-INVITE
without SDP, Asterisk will send an SDP offer in the 200 OK response containing
all configured codecs on the endpoint, instead of simply those that have
already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
The default value is "off".
res_pjsip_aoc
------------------
* Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
A new endpoint option, send_aoc, controls this.
res_pjsip_header_funcs
------------------
* The new PJSIP_HEADER_PARAM function now fully supports both
URI and header parameters. Both reading and writing
parameters are supported.
res_pjsip_logger
------------------
* SIP messages can now be filtered by SIP request method
(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
allowing for more granular debugging to be done
in the CLI. This applies to requests but not responses.
res_pjsip_notify
------------------
* Allows using the config options in pjsip_notify.conf
from AMI actions as with the existing CLI commands.
res_tonedetect
------------------
* The TONE_DETECT function now supports
detection of audible ringback tone
using the p option.
xmldocs
------------------
* The XML documentation can now be reloaded without restarting
Asterisk, which makes it possible to load new modules that
enforce documentation without restarting Asterisk.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
New EXPORT function
------------------
* A new function, EXPORT, allows writing variables
and functions on other channels, the complement
of the IMPORT function.
app_amd
------------------
* An audio file to play during AMD processing can
now be specified to the AMD application or configured
in the amd.conf configuration file.
app_bridgewait
------------------
* Adds the n option to not answer the channel when
the BridgeWait application is called.
features
------------------
* The Bridge application now has the n "no answer" option
that can be used to prevent the channel from being
automatically answered prior to bridging.
func_strings
------------------
* Three new functions, TRIM, LTRIM, and RTRIM, are
now available for trimming leading and trailing
whitespace.
res_pjsip
------------------
* A new option named "peer_supported" has been added to the endpoint option
100rel. When set to this option, Asterisk sends provisional responses
reliably if the peer supports it. If the peer does not support reliable
provisional responses, Asterisk sends them normally.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
Transfer feature
------------------
* The following capabilities have been added to the
transfer feature:
- The transfer initiation announcement prompt can
now be customized in features.conf.
- The TRANSFER_EXTEN variable now can be set on the
transferer's channel in order to allow the transfer
function to automatically attempt to go to the extension
contained in this variable, if it exists. The transfer
context behavior is not changed (TRANSFER_CONTEXT is used
if it exists; otherwise the default context is used).
app_confbridge
------------------
* Adds the end_marked_any option which can be used
to kick users from a conference after any
marked user leaves (including marked users).
db
------------------
* The DBPrefixGet AMI action now allows retrieving
all of the DB keys beginning with a particular
prefix.
locks
------------------
* A new AMI event, DeadlockStart, is now available
when Asterisk is compiled with DETECT_DEADLOCKS,
and can indicate that a deadlock has occured.
res_geolocation
------------------
* * Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
Added 4 built-in profiles:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
The profiles are empty except for having their precedence
set.
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles. This is
mutually exclusive with setting location_reference on the
profile.
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.
Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.
res_musiconhold_answeredonly
------------------
* This change adds an option, answeredonly, that will prevent music
on hold on channels that are not answered.
res_pjsip
------------------
* TLS transports in res_pjsip can now reload their TLS certificate
and private key files, provided the filename of them has not
changed.
Applications
------------------
* added support for Danish syntax, playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
* added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
Channel-agnostic MF support
------------------
* A SendMF application and PlayMF manager
application are now included to send
arbitrary standard R1 MF tones on the
current channel or another specified channel.
Core
------------------
* Bundled PJProject Build
The build process has been updated to make pjproject troubleshooting
and development easier. See third-party/pjproject/README-hacking.md or
https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
for more info.
Handle non-standard Meter metric type safely
------------------
* A meter_support flag has been introduced that defaults to true to maintain current behaviour.
If disabled, a counter metric type will be used instead wherever a meter metric type was used,
the counter will have a "_meter" suffix appended to the metric name.
MessageSend
------------------
* The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.
ToneScan application
------------------
* A new application, ToneScan, allows for
synchronous detection of call progress
signals such as dial tone, busy tone,
Special Information Tones, and modems.
ami
------------------
* An AMI event now exists for "Wink".
* AMI events can now be globally disabled using
the disabledevents [general] setting.
app_confbridge
------------------
* Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio file.
* Adds the CONFBRIDGE_CHANNELS function which can
be used to retrieve a list of channels in a ConfBridge,
optionally filtered by a particular category. This
list can then be used with functions like SHIFT, POP,
UNSHIFT, etc.
app_dtmfstore
------------------
* New application which collects digits
dialed and stores them into
a specified variable.
app_mf
------------------
* Adds MF receiver and sender applications to support
the R1 MF signaling protocol, including integration
with the Dial application.
* Adds an option to ReceiveMF to cap the
number of digits read at a user-specified
maximum.
app_milliwatt
------------------
* The Milliwatt application's existing behavior is
incorrect in that it plays a constant tone, which
is not how digital milliwatt test lines actually
work.
An option is added so that a proper milliwatt test
tone can be provided, including a 1 second silent
interval every 10 seconds. However, for compatability
reasons, the default behavior remains unchanged.
app_morsecode
------------------
* Extends the Morsecode application by adding support for
American Morse code and adds a configurable option
for the frequency used in off intervals.
app_originate
------------------
* Codecs can now be specified for dialplan-originated
calls, as with call files and the manager action.
By default, only the slin codec is now used, instead
of all the slin* codecs.
app_playback
------------------
* A new option 'mix' is added to the Playback application that
will play by filename and say.conf. It will look on the format of the
name, if it is like say format it will play with say.conf if not it
will play the file name.
app_queue
------------------
* Reload behavior in app_queue has been changed so
queue and agent stats are not reset during full
app_queue module reloads. The queue reset stats
CLI command may still be used to reset stats while
Asterisk is running.
* Add field to save the time value when a member enter a queue.
Shows this time in seconds using 'queue show' command and the
field LoginTime for responses for AMI the events.
The output for the CLI command `queue show` is changed by added a
extra data field for the information of the time login time for each
member.
* added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
* Load queues and members from Realtime for
AMI actions: QueuePause, QueueStatus and QueueSummary,
Applications: PauseQueueMember and UnpauseQueueMember.
* Added a new AMI action: QueueWithdrawCaller
This AMI action makes it possible to withdraw a caller from a queue
back to the dialplan. The call will be signaled to leave the queue
whenever it can, hence, it not guaranteed that the call will leave
the queue.
Optional custom data can be passed in the request, in the WithdrawInfo
parameter. If the call successfully withdrawn the queue,
it can be retrieved using the QUEUE_WITHDRAW_INFO variable.
This can be useful for certain uses, such as dispatching the call
to a specific extension.
* The m option now allows an override music on hold
class to be specified for the Queue application
within the dialplan.
app_queue.c
------------------
* Allow multiple files to be streamed for agent announcement.
app_queues
------------------
* adding support for playing the correct en/et for nordic languages
* Don't play sound_thanks if there is no leading hold_time message
When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience"
app_read
------------------
* A new option allows the digit '#' to be read literally,
rather than used exclusively as the input terminator
character.
app_sendtext
------------------
* A ReceiveText application has been added that can be
used in conjunction with the SendText application.
app_voicemail
------------------
* Add a new 'S' option to VoiceMail which prevents the instructions
(vm-intro) from being played if a busy/unavailable/temporary greeting
from the voicemail user is played. This is similar to the existing 's'
option except that instructions will still be played if no user
greeting is available.
* added support for Danish syntax, playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
* The r option has been added, which prevents deletion
of messages from VoiceMailMain, which can be
useful for shared mailboxes.
apps
------------------
* A new option 'mix' is added to the Playback application that
will play by filename and say.conf. It will look on the format of the
name, if it is like say format it will play with say.conf if not it
will play the file name.
ari
------------------
* Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
to ARI channel resources as 'protocol_id'.
ASTERISK-30027
ast_coredumper
------------------
* New options:
--pid=<asterisk_pid>
Allows specification of an Asterisk instance when trying to
and the script can't determine it itself.
--libdir=<system library directory>
Allows specification of a non-standard installation directory
containing the Asterisk modules.
--(no-)rename
Renames the coredump and the output files with readable
timestamps. This is the default.
Removed unneeded or confusing options:
--append-coredumps
--conffile
--no-default-search
--tarball-uniqueid
Changed Variables:
COREDUMPS is now just "/tmp/core!(*.txt)"
DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
Changed behavior:
If you use 'running' or 'RUNNING' you no longer need to specify
'--no-default-search' to ignore existing coredumps.
cdr
------------------
* A new CDR option, channeldefaultenabled, allows controlling
whether CDR is enabled or disabled by default on
newly created channels. The default behavior remains
unchanged from previous versions of Asterisk (new
channels will have CDR enabled, as long as CDR is
enabled globally).
chan_dahdi
------------------
* Previously, cadences were appended on dahdi restart,
rather than reloaded. This prevented cadences from
being updated and maxed out the available cadences
if reloaded multiple times. This behavior is fixed
so that reloading cadences is idempotent and cadences
can actually be reloaded.
* A POLARITY function is now available that allows
getting or setting the polarity on a channel
from the dialplan.
chan_iax2
------------------
* ANI2 (OLI) is now transmitted over IAX2 calls
as an information element.
* Both a secret and an outkey may be specified at dial time,
since encryption is possible with RSA authentication.
chan_pjsip
------------------
* Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do.
Add ability to read header by pattern using PJSIP_HEADER().
* added global config option "allow_sending_180_after_183"
Allow Asterisk to send 180 Ringing to an endpoint
after 183 Session Progress has been send.
If disabled Asterisk will instead send only a
183 Session Progress to the endpoint.
* Hook flash events can now be sent on a PJSIP channel
if requested to do so.
chan_sip
------------------
* Session timers get removed on UPDATE
Fix if Asterisk receives a SIP REFER with Session-Timers UAC
that Asterisk maintains Session-Timers when sending UPDATE request
chan_sip.c
------------------
* resolve issue with pickup on device that uses "183" and not "180"
channel_internal_api
------------------
* CHANNEL(lastcontext) and CHANNEL(lastexten)
are now available for use in the dialplan.
cli
------------------
* The "module refresh" command has been added,
which allows unloading and then loading a
module with a single command.
* A new CLI command 'dialplan eval function' has been
added which allows users to test the behavior of
dialplan function calls directly from the CLI.
func_channel
------------------
* Adds the CHANNEL_EXISTS function to check for the existence
of a channel by name or unique ID.
func_db
------------------
* The function DB_KEYCOUNT has been added, which
returns the cardinality of the keys at a specified
prefix in AstDB, i.e. the number of keys at a
given prefix.
func_env.c
------------------
* Two new functions, DIRNAME and BASENAME, are now
included which allow users to obtain the directory
or the base filename of any file.
func_evalexten
------------------
* This adds the EVAL_EXTEN function which may be
used to evaluate data at dialplan extensions.
func_framedrop
------------------
* New function to selectively drop specified frames
in either direction on a channel.
func_json
------------------
* The JSON_DECODE dialplan function can now be used
to parse JSON strings, such as in conjunction with
CURL for using API responses.
func_odbc
------------------
* A SQL_ESC_BACKSLASHES dialplan function has been added which
escapes backslashes. Usage of this is dependent on whether the
database in use can use backslashes to escape ticks or not. If
it can, then usage of this prevents a broken SQL query depending
on how the SQL query is constructed.
func_scramble
------------------
* Adds an audio scrambler function that may be used to
distort voice audio on a channel as a privacy
enhancement.
func_strings
------------------
* A new STRBETWEEN function is now included which
allows a substring to be inserted between characters
in a string. This is particularly useful for transforming
dial strings, such as adding pauses between digits
for a string of digits that are sent to another channel.
func_vmcount
------------------
* Multiple mailboxes may now be specified instead of just one.
logger
------------------
* Added the ability to define custom log levels in logger.conf
and use them in the Log dialplan application. Also adds a
logger show levels CLI command.
res_agi
------------------
* Agi command 'exec' can now be enabled
to evaluate dialplan functions and variables
by setting the variable AGIEXECFULL to yes.
res_cliexec
------------------
* A new CLI command, dialplan exec application, has
been added which allows dialplan applications to be
executed at the CLI, useful for some quick testing
without needing to write dialplan.
res_fax_spandsp
------------------
* Adds support for spandsp 3.0.0.
res_geolocation
------------------
* Added res_geolocation which creates the core capabilities
to manipulate Geolocation information on SIP INVITEs.
res_parking
------------------
* An m option to Park and ParkAndAnnounce now allows
specifying a music on hold class override.
res_pjproject
------------------
* In pjproject.conf you can now map pjproject log levels
to the Asterisk TRACE log level. The default mappings
have therefore changed so that only pjproject levels
3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
to TRACE. Previously 3, 4, 5, and 6 were all mapped to
DEBUG.
res_pjsip
------------------
* A new transport option 'allow_wildcard_certs' has been added that when it
and 'verify_server' are both set to 'yes', enables verification against
wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS
for TLS transport types. Names must start with the wildcard. Partial wildcards,
e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only
match against a single level meaning '*.example.com' matches 'foo.example.com',
but not 'foo.bar.example.com'.
res_pjsip_geolocation
------------------
* Added res_pjsip_geolocation which gives chan_pjsip
the ability to use the core geolocation capabilities.
res_pjsip_header_funcs
------------------
* Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request.
Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request.
res_pjsip_pubsub
------------------
* A new resource_list option, resource_display_name, indicates
whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.
* The Resource List Subscriptions (RLS) is dynamic now.
The asterisk now updates current subscriptions to reflect the changes
to the list on subscription refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.
res_pjsip_registrar
------------------
* Adds new PJSIP AOR option remove_unavailable to either
remove unavailable contacts when a REGISTER exceeds
max_contacts when remove_existing is disabled, or
prioritize unavailable contacts over other existing
contacts when remove_existing is enabled.
res_pjsip_t38
------------------
* In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.
res_rtp_asterisk
------------------
* When the address of the STUN server (stunaddr) is a name resolved via DNS, the
stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
expires. This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
res_tonedetect
------------------
* Arbitrary tone detection is now available through a
WaitForTone application (blocking) and a TONE_DETECT
function (non-blocking).
say.c
------------------
* Adds SAYFILES function to retrieve the file names that would
be played by corresponding Say applications, such as
SayDigits, SayAlpha, etc.
Additionally adds SayMoney and SayOrdinal applications.
stasis_channels
------------------
* Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
to ARI channel resources as 'protocol_id'.
ASTERISK-30027
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
------------------------------------------------------------------------------
AMI Flash event
------------------
* Hook flash events are now exposed as AMI events.
Add variable support to Originate
------------------
* The Originate application now allows
variables to be set on the new channel
through a new option.
Core
------------------
* 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:
core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
core set debug category off [<category> [<category>] ...]
If no sub-level is associated all debug statements for a given category are
output. If a sub-level is given then only those statements assigned a value
at or below the associated sub-level are output.
* The location where the media cache stores its temporary files
is no longer hardcoded to /tmp but can now be configured separately
via the astcachedir config variable in asterisk.conf.
The default location for astcachedir is now /var/cache/asterisk
instead of /tmp, please make sure to manually cleanup and/or
migrate the temporary files in /tmp after upgrading.
MessageSend
------------------
* The MessageSend dialplan application now takes an
optional third argument that can set the message's
"To" field on outgoing messages. It's an alternative
to using the MESSAGE(to) dialplan function.
To prevent confusion with the first argument, currently
named "to", it's been renamed to "destination".
Its function, creating the request URI, hasn't changed.
The online documentation has also been enhanced to
explain the behavior.
Despite the changes in this commit, there should be
no impact to current users of MessageSend.
New ConfKick application
------------------
* Adds a ConfKick() application, which allows
a specific channel, all users, or all non-admin
users to be kicked from a conference bridge.
New Reload application
------------------
* Adds an application to reload modules
PlaybackFinished has a new error state
------------------
* The PlaybackFinished event now has a new state "failed"
that is used when the sound file was not played due to an error.
Before the state on PlaybackFinished was always "done".
In case of multiple sound files to be played,
the PlaybackFinished is sent only once in the end of the list,
even in case of error.
WaitForCondition application
------------------
* This application provides a way to halt
dialplan execution until a provided
condition evaluates to true.
app_confbridge
------------------
* app_confbridge now has the ability to force 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".
app_confbridge answer supervision control
------------------
* app_confbridge now provides a user option to prevent
answer supervision if the channel hasn't been
answered yet. To use it, set a user profile's
answer_channel option to no.
app_dial announcement option
------------------
* The A option for Dial now supports
playing audio to the caller as well
as the called party.
app_mixmonitor
------------------
* app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
MixMonitorMute when the channel monitoring is started, stopped and muted (or
unmuted) respectively.
app_voicemail
------------------
* The VoiceMail application can now be configured to send greetings and
instructions via early media and only answering the channel when it is
time for the caller to record their message. This behavior can be
activated by passing the new 'e' option to VoiceMail.
* You can now customize the "beep" tone or omit it entirely.
chan_iax2
------------------
* You can now specify a default "auth" method in the
[general] section of iax.conf
chan_pjsip
------------------
* The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
returns unsuccessful if it's used on a channel prior to answering.
chan_pjsip, app_transfer
------------------
* Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
transfers can pass a protocol specific error code.
Example, in SIP 3xx-6xx represent any SIP specific error received when
performing a REFER.
func_math: Three new dialplan functions
------------------
* Introduce three new functions, MIN, MAX, and ABS, which can be used to
obtain the minimum or maximum of up to two integers or absolute value.
func_odbc
------------------
* Introduce an ARGC variable for func_odbc functions, along with a minargs
per-function configuration option.
minargs enables enforcing of minimum count of arguments to pass to
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
this should be set to 4. func_odbc will generate an error in this case,
so for example
[FOO]
minargs = 4
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
potentially leaked ARG4 from Gosub().
ARGC is needed if you're using optional argument, to verify whether or
not an argument has been passed, else it's possible to use a leaked ARGn
from Gosub (app_stack). So now you can safely do
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
func_volume now can be read
------------------
* The VOLUME function can now also be used
to read existing values previously set.
logger
------------------
* 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. 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
* The dateformat option in logger.conf will now control the remote
console (asterisk -r -T) timestamp format. Previously, dateformat only
controlled the formatting of the timestamp going to log files and the
main console (asterisk -c) but only for non-verbose messages.
Internally, Asterisk does not send the logging timestamp with verbose