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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 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
messages to console clients. It's up to the Asterisk remote consoles
to format verbose messages. Asterisk remote consoles previously did
not load dateformat from logger.conf.
Previously there was a non-configurable and hard-coded "%b %e %T"
dateformat that would be used no matter what on all verbose console
messages printed on remote consoles.
Example:
logger.conf
dateformat=%F %T.%3q
# asterisk -rvvv -T
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
[Mar 19 09:55:43] -- Goto (dialExten,s,1)
Given the following example configuration in logger.conf, Asterisk log
files and the console, will log verbose messages using the given
timestamp. Now ensuring that all remote console messages are logged
with the same dateformat as other log streams.
---
[general]
dateformat=%F %T.%3q
[logfiles]
console => notice,warning,error,verbose
full => notice,warning,error,debug,verbose
---
Now we have a globally-defined dateformat that will be used
consistently across the Asterisk main console, remote consoles, and
log files.
Now we have consistent logging:
# asterisk -rvvv -T
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
[2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
res_pjsip
------------------
* PJSIP transports can now be partially reloaded safely. This allows the
local_net and external_* options to be updated without restarting Asterisk.
* PJSIP endpoints can now be configured to skip authentication when
handling OPTIONS requests by setting the allow_unauthenticated_options
configuration property to 'yes.'
* PJSIP support of registrations of endpoints in multidomain
scenarios, where the endpoint contains the domain info
in pjsip.conf.
res_pjsip_dialog_info_body_generator
------------------
* PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
remote elements by iterating through ringing channels and inserting
that info into NOTIFY packet sent to the endpoint.
res_pjsip_messaging
------------------
* Implemented the new "to" parameter of the MessageSend()
dialplan application. This allows a user to specify
a complete SIP "To" header separate from the Request URI.
We now also accept a destination in the same format
as Dial()... PJSIP/number@endpoint
res_rtp_asterisk
------------------
* By default Asterisk reports the PJSIP version in all
STUN packets it sends.
This behaviour may not be desired in a production
environment and can now be disabled by setting the
stun_software_attribute option to 'no' in rtp.conf.
res_srtp
------------------
* SRTP replay protection has been added to res_srtp and
a new configuration option "srtpreplayprotection" has
been added to the rtp.conf config file. For security
reasons, the default setting is "yes". Buggy clients
may not handle this correctly which could result in
no, or one way, audio and Asterisk error messages like
"replay check failed".
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------
Core
------------------
* The Streams API becomes the home for the core ACN capabilities.
These include...
* Parsing and formatting of codec negotiation 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 respective 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.
app_bridgeaddchan
------------------
* The BridgeAdd application now behaves more like the Bridge application.
The application now sets the BRIDGERESULT channel variable to indicate
what happened when the channel resumes in dialplan. This is instead of
hanging up the channel on failure conditions.
res_pjsip
------------------
* Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
have been added to res_pjsip endpoints that specify the preferred order
of codecs to use between those received/sent in an SDP offer and those
set in the endpoint configuration.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* You can now specify an optional 'Content-Type' as an argument for the Asterisk
SendText manager action.
ARI
------------------
* A new parameter 'inhibitConnectedLineUpdates' is now available in the
'bridges.addChannel' call. This prevents the identity of the newly connected
channel from being presented to other bridge members.
ARI Channels
------------------
* The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server. Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.
Core
------------------
* H.265/HEVC is now a supported video codec and it can be used by
specifying "h265" in the allow line.
Please note however, that handling of the additional SDP parameters
described in RFC 7798 section 7.2 is not yet supported.
Features
------------------
* Adds support for 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 referenced wiki page. A
short talk about the reasons and implementation can be found on YouTube at
the link provided.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
Messaging
------------------
* 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".
STIR/SHAKEN
------------------
* STIR/SHAKEN support has been added to Asterisk. Configuration is done in
stir_shaken.conf. There is a sample configuration file to help you get
started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
to yes on the endpoint configuration object. This will add an Identity
header on outgoing INVITEs, and check for an Identity header on incoming
INVITEs. This option has been added to Alembic as well.
The information received on an incoming INVITE can be checked using the
STIR_SHAKEN dialplan function. There are two variations:
STIR_SHAKEN(count)
STIR_SHAKEN(0, verify_result)
The first variation will tell you how many STIR/SHAKEN results are on the
channel. The second fetches information for a specific result. The first
parameter is the index, followed by what information you want to retrieve.
The available options are 'verify_result', 'identity', and 'attestation'.
app_chanisavail
------------------
* The ChanIsAvail application now tolerates empty positions in the supplied
device list. Dialplan can now be simplified by not having to check for
empty positions in the device list.
app_confbridge
------------------
* A new bridge profile option, maximum_sample_rate, has been added which sets
a maximum sample rate that the bridge will be mixed at. This allows the bridge
to move below the maximum sample rate as needed but caps it at the maximum.
* A new option, "text_messaging", has been added to the user profile
which allows control over whether text messaging is enabled or
disabled for a user. If enabled (the default) text messages
will be sent to the user. If disabled no text messages will be
sent to the user.
app_dial
------------------
* The Dial application now tolerates empty positions in the supplied
destination list. Dialplan can now be simplified by not having to check
for empty positions in the destination list. If there are no endpoints to
dial then DIALSTATUS is set to CHANUNAVAIL.
app_mixmonitor
------------------
* An option 'S' has been added to MixMonitor. If used in combination with
the r() and/or t() options, if a frame is available to write to one of
those files but not the other, a frame of silence if written to the file
that does not have an audio frame. This should prevent the two files
from "drifting" when mixed after the fact.
* If the 'filename' argument to MixMonitor() ended with '.wav49,'
Asterisk would silently convert the extension to '.WAV' when opening
the file for writing. This caused the MIXMONITOR_FILENAME variable to
reference the wrong file. The MIXMONITOR_FILENAME variable will now
reflect the name of the file that Asterisk actually used instead of
the filename that was passed to the application.
app_page
------------------
* The Page application now tolerates empty positions in the supplied
destination list. Dialplan can now be simplified by not having to check
for empty positions in the destination list.
app_voicemail
------------------
* A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
the Asterisk voicemail directory on startup. Some users that store their
voicemails on network storage devices experienced slow startup times due to the
relative expense of traversing the voicemail directory structure looking for
orphaned lock files. This feature has now been removed.
Users who require the lock files to be removed at startup should modify their
startup scripts to do so before starting the asterisk process.
chan_pjsip
------------------
* A new dialplan function, PJSIP_MOH_PASSTHROUGH, has been added to chan_pjsip. This
allows the behaviour of the moh_passthrough endpoint option to be read or changed
in the dialplan. This allows control on a per-call basis.
chan_rtp
------------------
* The UnicastRTP channel driver provided by chan_rtp now accepts
"<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
The first AAAA (preferred) or A record resolved will be used as the destination.
The lookup is synchronous so beware of possible dialplan delays if you specify a
hostname.
func_curl
------------------
* A new parameter, httpheader, has been added to CURLOPT function. This parameter
allows to set custom http headers for subsequent calls off CURL function.
Any setting of headers will replace the default curl headers
(e.g. "Content-type: application/x-www-form-urlencoded")
* A new option, followlocation, can now be enabled with the CURLOPT()
dialplan function. Setting this will instruct cURL to follow 3xx
redirects, which it does not by default.
func_jitterbuffer
------------------
* The JITTERBUFFER dialplan function now has an option to enable video synchronization
support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
the video is buffered according to the size of the audio jitterbuffer and is
synchronized to the audio.
func_volume
------------------
* Accept decimal number as argument.
http
------------------
* You can now disable the /httpstatus page served by Asterisk's built-in
HTTP server by setting 'enable_status' to 'no' in http.conf.
minmemfree
------------------
* The 'minmemfree' configuration option now counts memory allocated to
the filesystem cache as "free" because it is memory that is available
to the process.
res_ari_channels
------------------
* When creating a channel in ARI using the create call
you can now specify dialplan variables to be set as part
of the same operation.
res_musiconhold
------------------
* 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.
* A new mode - playlist - has been added to res_musiconhold. This mode allows the
user to specify the files (or URLs) to play explicitly by putting them directly
in musiconhold.conf.
res_pjsip
------------------
* Added a new PJSIP system setting called disable_rport.
Default is no to keep support working as before.
If it is false (default) it adds the 'rport' parameter in the outgoing request message.
If it is true it does not add the 'rport' parameter in the outgoing request message.
This is a system option, but working as a global option.
res_pjsip_endpoint_identifier_ip
------------------
* In 'type = identify' sections, the addresses specified for the 'match'
clause can now include a port number. For IP addresses, the port is
provided by including a colon after the address, followed by the
desired port number. If supplied, the netmask should follow the port
number. To specify a port for IPv6 addresses, the address itself must
be enclosed in brackets to be parsed correctly.
res_pjsip_logger
------------------
* 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 console <on / off>
This allows you to toggle logging to console 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.
res_pjsip_session
------------------
* When placing an outgoing call to a PJSIP endpoint the intent
of any requested formats will now be respected. If only an audio
format is requested (such as ulaw) but the underlying endpoint
does not support the format the resulting SDP will still only
contain an audio stream, and not any additional streams such as
video.
* Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
have been added to res_pjsip endpoints that specify the preferred order
of codecs to use between those received/sent in an SDP offer and those
set in the endpoint configuration.
res_rtp_asterisk
------------------
* This change include a new cli command 'rtp show settings'
The command display by general settings of rtp configuration. For this
point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
strictrtp, learning_min_sequential and icesupport.
* The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
an ACL mechanism.
As such six now options are now available:
ice_deny
ice_permit
ice_acl
stun_deny
stun_permit
stun_acl
These options have their obvious meanings as used elsewhere.
Backwards compatibility was maintained by adding {stun,ice}_blacklist as
aliases for {stun,ice}_deny.
res_sorcery_memory_cache
------------------
* The SorceryMemoryCacheExpireObject AMI action and CLI
command allow expiring of a specific object within the
sorcery memory cache. This is done by removing the
object from the cache with the expectation that the
cache will then re-populate the object when it is next
needed.
For full backend caching this does not occur. The cache
won't repopulate until an entire refresh is done resulting
in the possibility that objects are missing until that
time.
The AMI action and CLI command will now not allow
expiring of an object if the cache is configured as a
full backend cache. Instead you must use either the
SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
AMI actions or their associated CLI commands.
taskprocessor.c
------------------
* 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.
* 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.
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 17.0.0 --------------------------
------------------------------------------------------------------------------
Bridging
------------------
* The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
A topic pool is now used for individual bridge topics.
The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.
Channels
------------------
* The core no longer uses the stasis cache for channels snapshots.
The following APIs are no longer available:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now returns an ao2_container of ast_channel_snapshots rather than a
container of stasis_messages therefore you can't call stasis_cache
functions on it.
The ast_channel_topic_all() function now returns a normal topic,
not a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
chan_sip
------------------
* The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
------------------------------------------------------------------------------
AttendedTransfer
------------------
* A new application, this will queue up attended transfer to the given extension.
BlindTransfer
------------------
* A new application, this will redirect all channels currently
bridged to the caller channel to the specified destination.
ConfBridge
------------------
* Add "average_all", "highest_all", and "lowest_all" values for
the remb_behavior option. These values operate on a bridge
level instead of a per-source level. This means that a single
REMB value is calculated and sent to every sender, instead of
a REMB value that is unique for the specific sender..
Dial
------------------
* Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
milliseconds between creation of the dialing channel and receiving the first
RINGING signal
Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
the PROGRESS signal. Shorter of these two times should be equivalent to
the PDD (Post Dial Delay) value
Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
versions of DIALEDTIME and ANSWEREDTIME
RTP/ICE
------------------
* You can now indicate that you'd like an ice_host_candidate's local address
to be published as well as the mapped address. See the sample rtp.conf
for more information.
ReadExten
------------------
* Add 'p' option to stop reading extension if user presses '#' key.
pbx_dundi
------------------
* The DUNDi PBX module now supports IPv4/IPv6 dual binding.
res_pjsip
------------------
* Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
res_rtp_asterisk
------------------
* DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
allows larger certificates to be used for the DTLS negotiation. By default this value
is 1200.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
------------------------------------------------------------------------------
ARI
------------------
* Application event filtering is now supported. An application can now specify
an "allowed" and/or "disallowed" list(s) of event types. Only those types
indicated in the "allowed" list are sent to the application. Conversely, any
types defined in the "disallowed" list are not sent to the application. Note
that if a type is specified in both lists "disallowed" takes precedence.
* A new REST API call has been added: 'move'. It follows the format
'channels/{channelId}/move' and can be used to move channels from one application
to another without needing to exit back into the dialplan. An application must be
specified, but the passing a list of arguments to the new application is optional.
An example call would look like this:
client.channels.move(channelId=chan.id, app='ari-example', appArgs='a,b,c')
If the channel was inside of a bridge when switching applications, it will
remain there. If the application specified cannot be moved to, then the channel
will remain in the current application and an event will be triggered named
"ApplicationMoveFailed", which will provide the destination application's name
and the channel information.
res_pjsip
------------------
* A new configuration parameter "taskprocessor_overload_trigger" has been
added to the pjsip.conf "globals" section. The distributor currently stops
accepting new requests when any taskprocessor overload is triggered. The
new option allows you to completely disable overload detection (NOT
RECOMMENDED), keep the current behavior, or trigger only on pjsip
taskprocessor overloads.
chan_pjsip
------------------
* A new configuration parameter 'ignore_183_without_sdp' has been added
to the pjsip.conf "endpoints" section. If enabled, will make chan_pjsip
discard 183s that do not contain an SDP body, which can resolve no
ringback tone issues as well as making the behavior match chan_sip.
MWI
------------------
* A new module "res_mwi_devstate" has been added that allows subscriptions
to voicemail boxes using "presence" events. This allows common BLF keys
to act as voicemail waiting indicators.
app_queue
------------------
* Added the ability to set the wrapuptime per-member using the AddQueueMember
application.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------
------------------------------------------------------------------------------
ARI
------------------
* Whenever an ARI application is started, a context will be created for it
automatically as long as one does not already exist, following the format
'stasis-<app_name>'. Two extensions are also added to this context: a match-all
extension, and the 'h' extension. Any phone that registers under this context
will place all calls to the corresponding Stasis application.
res_pjsip
------------------
* Added "send_contact_status_on_update_registration" global configuration option
to enable sending AMI ContactStatus event when a device refreshes its registration.
Core
------------------
* Reworked the media indexer so it doesn't cache the index. Testing revealed
that the cache added no benefit but that it could consume excessive memory.
Two new index related functions were created: ast_sounds_get_index_for_file()
and ast_media_index_update_for_file() which restrict index updating to
specific sound files. The original ast_sounds_get_index() and
ast_media_index_update() calls are still available but since they no longer
cache the results internally, developers should re-use an index they may
already have instead of calling ast_sounds_get_index() repeatedly. If
information for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().
Features
------------------
* Before Asterisk 12, when using the automon or automixmon features defined
in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
both channels, indicating the filename of the recording.
When bridging was overhauled in Asterisk 12, the behavior was changed such
that the variable was only set on the peer channel and not on the channel
that initiated the automon or automixmon.
The previous behavior has been restored so both channels receive the
channel variable when one of these features is invoked.
app_voicemail
------------------
* You can now specify a special context with the "aliasescontext" parameter
in voicemail.conf which will allow you to create aliases for physical
mailboxes.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
------------------------------------------------------------------------------
pbx_config
------------------
* pbx_config will now find and process multiple 'globals' sections from
extensions.conf. Variables are processed in the order they are found
and duplicate variables overwrite the previous value.
chan_pjsip
------------------
* New dialplan function PJSIP_PARSE_URI added to parse an URI and return
a specified part of the URI.
Core
------------------
* ast_bt_get_symbols() now returns a vector of strings instead of an
array of strings. This must be freed with ast_bt_free_symbols.
res_pjsip
------------------
* New options 'trust_connected_line' and 'send_connected_line' have been
added to the endpoint. The option 'trust_connected_line' is to control
if connected line updates are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
res_rtp_asterisk
------------------
* The existing strictrtp option in rtp.conf has a new choice availabe, called
'seqno', which behaves the same way as setting strictrtp to 'yes', but will
ignore the time interval during learning so that bursts of packets can still
trigger learning our source.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
------------------------------------------------------------------------------
app_fax
------------------
* The app_fax module is now deprecated, users should migrate to the
replacement module res_fax.
app_originate
------------------
* An 'a' option has been added to the Originate dialplan application which
will execute the originate in an asynchronous fashion. If set then the
application will return immediately without waiting for the originated
channel to answer.
Build System
------------------
* MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built
with MALLOC_DEBUG can now successfully load binary modules built without
MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer
need to have a special build with it enabled.
* Asterisk now depends on libjansson >= 2.11. If this version is not
available on your distro you can use `./configure --with-jansson-bundled`.
app_macro
------------------
* The app_macro module is now deprecated and by default it is no longer
built. Users should migrate to app_stack (Gosub). A warning is logged
the first time any Macro is used.
app_setcallerid
------------------
* The app_setcallerid module has been removed. The CALLERID dialplan function
should be used instead.
chan_sip
------------------
* New function SIP_HEADERS() enumerates all headers in the incoming INVITE.
* The variable GET_TRANSFERRER_DATA set in the peer channel causes matching
headers be retrieved from the REFER message and made accessible to the
dialplan in the hash TRANSFER_DATA.
chan_dahdi
------------------
* Timeouts for reading digits from analog phones are now configurable in
chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.
AMI
------------------
* The ContactStatus and Status fields for the manager events ContactStatus
and ContactStatusDetail are now set to "NonQualified" when a contact exists
but has not been qualified.
* The "Newexten" event is now part of the "dialplan" class. The documentation
for Asterisk 15 already specified this, but the implementation was actually
using the "call" class instead.
ARI
------------------
* The ContactInfo event's contact_status field is now set to "NonQualified"
when a contact exists but has not been qualified.
app_queue
------------------