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asterisk/CHANGES
Asterisk Development Team a818b05ca1 Update CHANGES and UPGRADE.txt for 20.0.0
2022-07-20 05:44:50 -05:00

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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 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
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
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
------------------
* Added the ability to set the wrapuptime in the configuration of member.
When set the wrapuptime on the member is used instead of the wrapuptime
defined for the queue itself.
* Added predial handler support for caller and callee channels with the
B and b options respectively. This is similar to the predial support
in app_dial.
res_config_sqlite
------------------
* The res_config_sqlite module is now deprecated, users should migrate to the
replacement module res_config_sqlite3.
res_monitor
------------------
* The res_monitor module is now deprecated, users should migrate to the
replacement module app_mixmonitor.
res_pjsip
------------------
* A new AMI action, PJSIPShowAors, has been added which displays information
about all configured PJSIP AORs.
* A new AMI action, PJSIPShowAuths, has been added which displays information
about all configured PJSIP Auths.
* A new AMI action, PJSIPShowContacts, has been added which displays information
about all configured PJSIP Contacts.
res_pjsip_registrar_expire
------------------
* The res_pjsip_registrar_expire module has been removed. The functionality has
been moved into res_pjsip_registrar.
func_audiohookinherit
------------------
* The func_audiohookinherit module has been removed. Due to architectural changes
in Asterisk 12, audiohook inheritance is performed automatically and this
function now lacks function.
cdr_syslog
------------------
* The cdr_syslog module is now deprecated and by default it is no longer
built.
cdr_sqlite
------------------
* The cdr_sqlite module has been removed. Users should move to using the
cdr_sqlite3_custom module instead.
format_jpeg
------------------
* The format_jpeg module has been removed.
pbx_dundi
------------------
* DUNDi now supports IPv6
Core:
------------------
* libedit is no longer available as an embedded library and must be provided
by the system.
* The STATIC_BUILD functionality has been removed as it has not been maintained
and has not worked in quite some time.
* The module loader now enforces inter-module dependencies. This ensures that
a module is not started before another it depends on, even if preload is used.
If a dependency is not available or fails to startup this will block any
dependants from startup.
* Parts of the Asterisk core which can load configuration from realtime are now
built-in modules. It is no longer necessary to preload realtime drivers as
they are always initialized before the built-in modules.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------
------------------------------------------------------------------------------
res_pjsip
------------------
* A new option 'suppress_q850_reason_headers' has been added to the endpoint
object. Some devices can't accept multiple Reason headers and get confused
when both 'SIP' and 'Q.850' Reason headers are received. This option allows
the 'Q.850' Reason header to be suppressed. The default value is 'no'.
res_pjsip_endpoint_identifier_ip
------------------
* Added regex support to the identify section match_header option. You
specify a regex instead of an explicit string by surrounding the header
value with slashes:
match_header = SIPHeader: /regex/
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------
------------------------------------------------------------------------------
Core
------------------
* Core bridging and, more specifically, bridge_softmix have been enhanced to
relay received frames of type TEXT or TEXT_DATA to all participants in a
softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to
take advantage of this so when res_pjsip_messaging receives an in-dialog
MESSAGE message from a user in a conference call, it's relayed to all
other participants in the call.
app_sendtext
------------------
* Support Enhanced Messaging. SendText now accepts new channel variables
that can be used to override the To and From display names and set the
Content-Type of a message. Since you can now set Content-Type, other
text/* content types are now valid.
app_confbridge
------------------
* ConfbridgeList now shows talking status. This utilizes the same voice
detection as the ConfbridgeTalking event, so bridges must be configured
with "talk_detection_events=yes" for this flag to have meaning.
* ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc. In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.
res_pjsip
------------------
* Two new options have been added to the system and endpoint objects to
control whether, on outbound calls, Asterisk will accept updated SDP answers
during the initial INVITE transaction when 100rel is not in effect.
This usually happens when the INVITE is forked to multiple UASs and more
than one sends an SDP answer or when a single UAS needs to change a media
port to switch from custom ringback to the actual media destination.
The 'follow_early_media_forked' option sets whether Asterisk will accept
the updated SDP when the To tag on the subsequent response is different than
that on the the previous response. This usually occurs in the forked INVITE
scenario. The default value is "yes" which is the current behavior.
The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the
updated SDP when the To tag on the subsequent response is the same as that
on the previous response. This can occur when a UAS needs to switch media
ports from custom ringback to the final media path. The default value is
"no" which is the current behavior.
These options have to be enabled system-wide in the system config section
of pjsip.conf as well as on individual endpoints that require the
functionality.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------
------------------------------------------------------------------------------
Core
------------------
* A new configuration option "genericplc_on_equal_codecs" was added to the
"plc" section of codecs.conf to allow generic packet loss concealment even
if no transcoding was originally needed. Transcoding via SLIN is forced
in this case.
res_pjproject
------------------
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool contents
are used after free and who freed it.
res_pjsip_notify
------------------
* Extend the PJSIPNotify AMI command to send an in-dialog notify on a
channel.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------
------------------------------------------------------------------------------
Core
------------------
* During dialplan reload log messages are produced for each context,
extension and include. These messages are no longer printed by the
verbose loggers, they are now only logged as debug messages.
app_confbridge
------------------
* Added the Muted header to the ConfbridgeJoin AMI event to indicate the
participant's starting mute status.
* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers. The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped. However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".
app_followme
------------------
* Added a new prompt, connecting-prompt, which will be played
(if configured) to the "winner" callee before connecting the call.
res_pjsip
------------------
* Users who are matching endpoints by SIP header need to reevaluate their
global "endpoint_identifier_order" option in light of the "ip" endpoint
identifier method split into the "ip" and "header" endpoint identifier
methods.
* The pjsip_transport_event feature introduced in 15.1.0 has been refactored.
Any external modules that may have used that feature (highly unlikely) will
need to be changed as the API has been altered slightly.
res_pjsip_endpoint_identifier_ip
------------------
* The endpoint identifier "ip" method previously recognized endpoints either
by IP address or a matching SIP header. The "ip" endpoint identifier method
is now split into the "ip" and "header" endpoint identifier methods. The
"ip" endpoint identifier method only matches by IP address and the "header"
endpoint identifier method only matches by SIP header. The split allows the
user to control the relative priority of the IP address and the SIP header
identification methods in the global "endpoint_identifier_order" option.
e.g., If you have two type=identify sections where one matches by IP address
for endpoint alice and the other matches by SIP header for endpoint bob then
you can now predict which endpoint is matched when a request comes in that
matches both.
res_pjsip_pubsub
------------------
* In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions. Since this
required the addition of a new column to the ps_subscription_persistence
realtime table, users who store their subscriptions in a database will
need to run the "alembic upgrade head" process to add the column to
the schema.
res_pjsip_transport_management
------------------
* Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed. This way the features will always
be available if res_pjsip is loaded.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------
------------------------------------------------------------------------------
Core
------------------
* Added the "cache_media_frames" option to asterisk.conf. Disabling the option
helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled because
the cache code does not exist.
chan_sip
------------------
* Calls to invalid extensions are now reported as an ACL failure security event
"no_extension_match".
res_rtp_asterisk
------------------
* The X.509 certificate used for DTLS negotiation can now be automatically
generated. This is supported by res_pjsip by specifying
"dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you
would set "dtlsautogeneratecert = yes" either in the [general] section of
sip.conf or on a specific peer.
res_pjsip
------------------
* The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
being matched based only on IP address. To ensure no behavior change the
default has been changed to "username,ip".
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.0.0 to Asterisk 15.1.0 ------------
------------------------------------------------------------------------------
res_pjsip
------------------
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
AMI
------------------
* Added a new CancelAtxfer action that cancels an attended transfer.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
------------------------------------------------------------------------------
app_queue
------------------
* PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has
been defined.
* A new option, "announce-position-only-up," has been added that, when set to
yes, causes position announcements to only be played when the caller's
queue position has improved since the last time that we announced their
position. This default is no.
Build System
------------------
* '--with-pjproject-bundled' is now the default when running ./configure
It can be disabled with '--without-pjproject-bundled'.
* A '--with-download-cache' option is now available which is equivalent to
setting '--with-sounds-cache' and '--with-externals-cache' to the same
value. The download cache can also be set via the AST_DOWNLOAD_CACHE
environment variable.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------
------------------------------------------------------------------------------
res_pjsip
------------------
* The "external_media_address" on transports is now resolved using dnsmgr and
when dnsmgr refreshes are enabled will be automatically updated with the new
IP address of a given hostname.
* A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive
unsolicited MWI NOTIFY requests and make them available to other modules via
the stasis message bus.
res_musiconhold
------------------
* By default, when res_musiconhold reloads or unloads, it sends a HUP signal
to custom applications (and all descendants), waits 100ms, then sends a
TERM signal, waits 100ms, then finally sends a KILL signal. An application
which is interacting with an external device and/or spawns children of its
own may not be able to exit cleanly in the default times, expecially if sent
a KILL signal, or if it's children are getting signals directly from
res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
class option can be used to set the number of milliseconds res_musiconhold
waits before escalating kill signals, with the default being the current
100ms. To control to whom the signals are sent, the "kill_method"
class option can be set to "process_group" (the default, existing behavior),
which sends signals to the application and its descendants directly, or
"process" which sends signals only to the application itself.
* New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
of a channel on a per-call basis.
res_xmpp
-----------------
* OAuth 2.0 authentication is now supported when contacting Google. Follow the
instructions in xmpp.conf.sample to retrieve and configure the necessary
tokens.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
------------------------------------------------------------------------------
app_voicemail
------------------
* A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.
Default: no
res_pjsip
------------------
* A new endpoint option "refer_blind_progress" was added to turn off notifying
the progress details on Blind Transfer. If this option is not set then
the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
On default is enabled.
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
* A new endpoint option "notify_early_inuse_ringing" was added to control
whether to notify dialog-info state 'early' or 'confirmed' on Ringing
when already INUSE.
* The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
mode works similar to 'auto' except uses DTMF INFO as fallback instead of
INBAND.
res_agi
------------------
* The EAGI() application will now look for a dialplan variable named
EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that
EAGI provides. If not specified, it will continue to use the default signed
linear (slin).
chan_pjsip
------------------
* When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan
function any contact which is considered unreachable due to qualify being
enabled will no longer be called.
* The asymmetric_rtp_codec option now also controls whether chan_pjsip will
send media as-is without transcoding if the codec has been negotiated in the
SDP. If set to "no" then Asterisk will only ever send the preferred codec
from the SDP, unless the remote side sends a different codec and we will
switch to match.
Build System
------------------
* Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
to pass arbitrary options to the bundled pjproject configure.
* Automatically set the bundled pjproject configure --host and --build
options to match those supplied for the asterisk configure.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
------------------------------------------------------------------------------
res_rtp_asterisk
------------------
* Added the stun_blacklist option to rtp.conf. Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to find
the external IP address. Attempting to send the STUN packet needlessly
delays processing incoming and outgoing SIP INVITEs because we will wait
for a response that can never come until we give up on the response.
Multiple subnets may be listed.
Logging
-------------------
* Added logger_queue_limit to the configuration options.
All log messages go to a queue serviced by a single thread
which does all the IO. This setting controls how big that
queue can get (and therefore how much memory is allocated)
before new messages are discarded.
The default is 1000.
res_pjsip_config_wizard
------------------
* Two new parameters have been added to the pjsip config wizard.
Setting 'sends_line_with_registrations' to true will cause the wizard
to skip the creation of an identify object to match incoming requests
to the endpoint and instead add the line and endpoint parameters to
the outbound registration object.
Setting 'outbound_proxy' is a shortcut for adding individual
endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
parameters.
res_hep_rtcp
------------------
* If the 'call-id' value is specified for the uuid_type option and a
chan_sip channel is used the resulting HEP traffic will now contain the
SIP Call-ID instead of the Asterisk channel name.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
------------------------------------------------------------------------------
Build System
------------------
* LOW_MEMORY no longer has an effect on Asterisk ABI. Symbols that were
previously suppressed by LOW_MEMORY are now replaced by stub functions.
Asterisk built with LOW_MEMORY can now successfully load binary modules
built without LOW_MEMORY and vice versa.
* RADIUS backends for CEL and CDR can now also be built using the radcli
client library, in addition to the existing support for building them
using either freeradius or radiusclient-ng.
Core
------------------
* ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources
which use mtx_prof must now manually declare and initialize the variable.
chan_sip
------------------
* If an offer is received with optional SRTP (a media stream with RTP/AVP but
which contains a crypto line) chan_sip will now accept it and enable SRTP.
If you would like to do optional SRTP on outbound you will need to create
a dialplan that dials with it enabled initially and if it fails fall back to
without.
res_pjsip
------------------
* Added endpoint configuration parameter "preferred_codec_only".
This allow asterisk response to a SIP invite with the single most
preferred codec rather than advertising all joint codec capabilities.
This limits the other side's codec choice to exactly what we prefer.
cdr_radius
------------------
* To fix a memory leak the syslog channel is now empty if it has not been set
and used by a syslog channel in the logger.
cel_radius
------------------
* To fix a memory leak the syslog channel is now empty if it has not been set
and used by a syslog channel in the logger.
RTP
------------------
* New setting "rtp_pt_dynamic = 35" in asterisk.conf:
Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32
formats. To avoid the message "No Dynamic RTP mapping available", the range
was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However,
when you use more than 32 formats and calls are not accepted by a remote
implementation, please report this and go back to rtp_pt_dynamic = 96.
* A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set
to "yes" RTP dynamic payload types are assigned dynamically per RTP instance.
When set to "no" RTP dynamic payload types are globally initialized to pre-
designated numbers and function similar to static payload types.
app_originate
------------------
* Added support to gosub predial routines on both original channel and on the
created channel using options parameter (like app_dial) B() and b(). This
allows for adding variables to newly created channel or, e.g. setting callerid.
CLI Commands
------------------
* 'dialplan show' output will now show [config_file:line_number] instead of
[registrar] when that information is available. Currently only extensions
registered by pbx_config when loading/reloading will use this format.
app_queue
------------------
* Add 'QueueUpdate' application which can be used to track outbound calls
using app_queue.
pbx_spool
------------------
* Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that
attempt-specific behavior is possible. This is a 1-based number that
simply increases by 1 for each attempt.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now
contains a new optional parameter, 'MatchHeader', mapping to the new
configuration option 'match_header' for the corresponding 'identify' object.
It should be noted that since 'match_header' takes in a key: value pair, the
event parameter will contain a ':' as well.
app_record
------------------
* Added new 'u' option to Record() application which prevents Asterisk from
truncating silence from the end of recorded files.
res_pjsip_outbound_registration
------------------
* Outbound registrations are now refreshed when res_stun_monitor detects
a network change event has happened.
The 'pjsip send (un)register' CLI commands were updated to accept '*all'
as an argument to operate on all registrations.
The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'.
app_voicemail
------------------
* The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
'vm-newuser' configuration options in voicemail.conf.
* Added 'fromstring' field to the voicemail boxes. If set, it will override
the global 'fromstring' field on a per-mailbox basis.
func_channel
------------------
* Added CHANNEL(callid) to retrieve the call log tag associated with the
channel. e.g., [C-00000000] Dialplan now has access to the call log
search key associated with the channel so it can be saved in case there
is a problem with the call.
res_pjsip
------------------
* A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with this
option set to 'yes', the transport name will be saved and used for
subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
saved as a contact uri parameter named 'x-ast-txp' and will display with
the contact uri in CLI, AMI, and ARI output. On the outgoing request,
if a transport wasn't explicitly set on the endpoint AND the request URI
is not a hostname, the saved transport will be used and the 'x-ast-txp'
parameter stripped from the outgoing packet. To facilitate recreation of
subscriptions on asterisk restart, a new column 'contact_uri' needed to be
added to the ps_subcsription_persistence table. Since new columns were
added to both transport and subscription_persistence, an alembic upgrade
should be run to bring the database tables up to date.
* A new option, allow_overlap, has been added to endpoints which allows
overlap dialing functionality to be enabled or disabled. The option defaults
to enabled.
res_pjsip_transport_websocket
------------------
* Removed non-secure websocket support. Firefox and Chrome have not allowed
non-secure websockets for quite some time so this shouldn't be an issue
for people. Attempting to use a non-secure websocket may or may not work
when Asterisk attempts to send SIP requests to do something like initiate
call hangup.
res_pjsip_endpoint_identifier_ip
------------------
* A new option has been added to the 'identify' configuration object,
'match_header'. The 'match_header' attribute should contain a SIP
header: value pair that, When set, will cause inbound requests that contain
the matching SIP header/value pair to be associated with the corresponding
endpoint. This option is cumulative with the 'match' option, so that if
either option matches the request, the request is associated with the
endpoint.
In a future release, this module will be renamed to something more
appropriate, as it now matches inbound requests on more than just IP
address.
res_rtp_asterisk
-----------------
* The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
Data and Control Packets on a Single Port." So far, the only channel driver
that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
a PJSIP endpoint in pjsip.conf to enable the feature.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
------------------------------------------------------------------------------
res_pjproject
------------------
* Added new CLI command "pjproject set log level". The new command allows
the maximum PJPROJECT log levels to be adjusted dynamically and
independently from the set debug logging level like many other similar
module debug logging commands.
* Added new companion CLI command "pjproject show log level" to allow the
user to see the current maximum pjproject logging level.
* Added new pjproject.conf startup section "log_level' option to set the
initial maximum PJPROJECT logging level.
res_pjsip_outbound_registration
------------------
* Statsd no longer logs redundant status PJSIP.registrations.state changes
for internal state transitions that don't change the reported public status
state.
res_pjsip_registrar
------------------
* The PJSIPShowRegistrationInboundContactStatuses AMI command has been added
to return ContactStatusDetail events as opposed to
PJSIPShowRegistrationsInbound which just a dumps every defined AOR.
res_pjsip
------------------
* Six existing contact fields have been added to the end of the
ContactStatusDetail AMI event:
ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and
QualifyTimeout. Existing fields have not been disturbed.
res_pjsip_endpoint_identifier_ip
------------------
* SRV lookups can now be done on provided hostnames to determine additional
source IP addresses for requests. This is configurable using the
"srv_lookups" option on the identify and defaults to "yes".
ARI
------------------
* The 'ari set debug' command has been enhanced to accept 'all' as an
application name. This allows dumping of all apps even if an app
hasn't registered yet.
* 'ari set debug' now displays requests and responses as well as events.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.1.0 to Asterisk 14.2.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* Events that reference a bridge may now contain two new optional fields:
- 'BridgeVideoSourceMode': the video source mode for the bridge.
Can be one of 'none', 'talker', or 'single'.
- 'BridgeVideoSource': the unique ID of the channel that is the video
source in this bridge, if one exists.
* A new event, BridgeVideoSourceUpdate, has been added with a class
authorization of CALL. The event is raised when the video source changes
in a multi-party mixing bridge.
ARI
------------------
* The bridges resource now exposes two new operations:
- POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a
multi-party mixing bridge
- DELETE /bridges/{bridgeId}/videoSource: Remove the set video source,
reverting to talk detection for the video source
* The bridge model in any returned response or event now contains the following
optional fields:
- video_mode: the video source mode for the bridge. Can be one of 'none',
'talker', or 'single'.
- video_source_id: the unique ID of the channel that is the video source
in this bridge, if one exists.
* A new event, BridgeVideoSourceChanged, has been added for bridges.
Applications subscribed to a bridge will receive this event when the source
of video changes in a mixing bridge.
* The ARI major version has been bumped. There are not any known breaking changes
in ARI. The major version has been bumped because otherwise we can end up with
overlapping version numbers between different Asterisk versions. Now each major
version of Asterisk will bring with it a change in the major version of ARI.
The ARI version in Asterisk 14 is now 2.0.0.
res_pjsip
------------------
* Automatic dual stack support is now implemented. Depending on DNS resolution
and the transport used for sending a message the SIP signaling and SDP will
be updated with the correct IP address and protocol version. This means that
the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The
res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
that messages are updated with the correct address information in all cases.
chan_pjsip
------------------
* The default behavior for RTP codecs has been changed. The sending codec will
now match the receiving codec. This can be turned off and behavior reverted
to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
option is set then the sending and received codec are allowed to differ.
CLI Commands
------------------
* Three new CLI commands have been added for ARI:
- ari show apps:
Displays a listing of all registered ARI applications.
- ari show app <name>:
Display detailed information about a registered ARI application.
- ari set debug <name> <on|off>:
Enable/disable debugging of an ARI application. When debugged, verbose
information will be sent to the Asterisk CLI.
Queue
------------------
* A new dialplan variable, ABANDONED, is set when the call is not answered
by an agent.
res_ari
------------------
* The configuration file ari.conf now supports a channelvars option, which
specifies a list of channel variables to include in each channel-oriented
ARI event.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ------------
------------------------------------------------------------------------------
Build System
------------------
* The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
codec_siren14 binary modules hosted at downloads.digium.com can now be
automatically downloaded and installed during the Asterisk install
process. If selected in menuselect, when 'make install' is run, the
script will check the downloads site for a new version and download
and install it if needed. The '--with-externals-cache' option to
./configure can be used to specify a location to cache the latest
tarballs so they don't have to be re-downloaded for every install.
app_voicemail
------------------
* Added "tps_queue_high" and "tps_queue_low" options.
The options can modify the taskprocessor alert levels for this module.
Additional information can be found in the sample configuration file at
config/samples/voicemail.conf.sample.
res_pjsip_mwi
------------------
* Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
options to tune taskprocessor alert levels.
* Added "mwi_disable_initial_unsolicited" global configuration option
to disable sending unsolicited MWI to all endpoints on startup.
Additional information can be found in the sample configuration file at
config/samples/pjsip.conf.sample.
chan_pjsip
------------------
* A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
invoked, a re-INVITE or UPDATE request will be sent immediately to the
endpoint underlying the channel. When used in combination with the existing
dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
channel to be re-negotiated and updated after session set up.
res_pjsip
------------------
* A new endpoint configuration parameter 'contact_user' has been added which
when set will override the default user set on Contact headers in outgoing
requests.
* If you are using a sorcery realtime backend to store global res_pjsip
options (ps_globals table) then you now have to do a res_pjsip reload for
changes to these options to take effect. If you are using pjsip.conf to
configure these options then you already had to do a reload after making
changes.
* Added "ignore_uri_user_options" global configuration option for
compatibility with an ITSP that sends URI user field options. When enabled
the user field is truncated at the first semicolon.
Example:
URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
The user field is "1235557890;phone-context=national"
Which is truncated to this: "1235557890"
Note: The caller-id and redirecting number strings obtained from incoming
SIP URI user fields are now always truncated at the first semicolon.
res_rtp_asterisk
------------------
* An option, ice_blacklist, has been added which allows certain subnets to be
excluded from local ICE candidates.
app_confbridge
------------------
* Some sounds played into the bridge are played asynchronously. This, for
instance, allows a channel to immediately exit the ConfBridge without having
to wait for a leave announcement to play.
app_dial
------------------
* Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
when another channel answers the call. The default of ANSWERED_ELSEWHERE
is unchanged.
res_ari
------------------
* ARI events will all now include a new field in the root of the JSON message,
'asterisk_id'. This will be the unique ID for the Asterisk system
transmitting the event. The value can be overridden using the 'entityid'
setting in asterisk.conf.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
------------------------------------------------------------------------------
AMI
-----------------
* A new event, "DialState" has been added. This is similar to "DialBegin" and
"DialEnd" in that it tracks the state of a dialed call. The difference is that
this indicates some intermediate state change in the dial attempt, such as
"RINGING", "PROGRESS", or "PROCEEDING".
ARI
-----------------
* A new ARI method has been added to the channels resource. "create" allows for
you to create a new channel and place that channel into a Stasis application.
This is similar to origination except that the specified channel is not
dialed. This allows for an application writer to create a channel, perform
manipulations on it, and then delay dialing the channel until later.
* To complement the "create" method, a "dial" method has been added to the
channels resource in order to place a call to a created channel.
* All operations that initiate playback of media on a resource now support
a list of media URIs. The list of URIs are played in the order they are
presented to the resource. A new event, "PlaybackContinuing", is raised when
a media URI finishes but before the next media URI starts. When a list is
played, the "Playback" model will contain the optional attribute
"next_media_uri", which specifies the next media URI in the list to be played
back to the resource. The "PlaybackFinished" event is raised when all media
URIs are done.
* Stored recordings now allow for the media associated with a stored recording
to be retrieved. The new route, GET /recordings/stored/{name}/file, will
transmit the raw media file to the requester as binary.
* "Dial" events have been modified to not only be sent when dialing begins and ends.
They now are also sent for intermediate states, such as "RINGING", "PROGRESS", and
"PROCEEDING".
Applications
------------------
BridgeAdd
------------------
* A new application in Asterisk, this will join the calling channel
to an existing bridge containing the named channel prefix.
ChanSpy
------------------
* Added the 'l' option, which forces ChanSpy's audiohook to use a long queue
to store the audio frames. This option is useful if audio loss is
experienced when using ChanSpy, but may introduce some delay in the audio
feed on the listening channel.
Codecs
------------------
* Added format attribute negotiation for the iLBC audio codec. Format attribute
negotiation is provided by the res_format_attr_ilbc module. iLBC 20 is the
default now. Falls back to iLBC 30, when the remote party requests this.
ConfBridge
------------------
* Added the ability to pass options to MixMonitor when recording is used with
ConfBridge. This includes the addition of the following configuration
parameters for the 'bridge' object:
- record_file_timestamp: whether or not to append the start time to the
recorded file name
- record_options: the options to pass to the MixMonitor application
- record_command: a command to execute when recording is finished
Note that these options may also be with the CONFBRIDGE function.
ControlPlayback
------------------
* Remote files can now be retrieved and played back. See the Playback
dialplan application for more details.
FollowMe
------------------
* It is now possible to disable the prompt from a callee by setting
'enable_callee_prompt = no' in followme.conf.
Playback
------------------
* Remote files can now be retrieved and played back via the Playback and other
media playback dialplan applications. This is done by directly providing
the URL to play to the dialplan application:
same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav)
Note that unlike 'normal' media files, the entire URI to the file must be
provided, including the file extension. Currently, on HTTP and HTTPS URI
schemes are supported.
Queue
-------------------
* Added field ReasonPause on QueueMemberStatus if set when paused, the reason
the queue member was paused.
* Added field LastPause on QueueMemberStatus for time when started the last
pause for a queue member.
* Show the time when started the last pause for queue member on CLI for command
'queue show'.
SMS
------------------
* Added the 'n' option, which prevents the SMS from being written to the log
file. This is needed for those countries with privacy laws that require
providers to not log SMS content.
Channel Drivers
------------------
chan_dahdi
------------------
* The CALLERID(ani2) value for incoming calls is now populated in featdmf
signaling mode. The information was previously discarded.
* Added the force_restart_unavailable_chans compatibility option. When
enabled it causes Asterisk to restart the ISDN B channel if an outgoing
call receives cause 44 (Requested channel not available).
chan_iax2
------------------
* The iax.conf forcejitterbuffer option has been removed. It is now always
forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer
on a channel it will be on the channel.
* A new configuration parameters, 'calltokenexpiration', has been added that
controls the duration before a call token expires. Default duration is 10
seconds. Setting this to a higher value may help in lagged networks or those
experiencing high packet loss.
* Plaintext auth mode is deprecated and removed from possible default modes.
chan_rtp (was chan_multicast_rtp)
------------------
* Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.
* The format for dialing a unicast RTP channel is:
UnicastRTP/<destination-addr>[/[<options>]]
Where <destination-addr> is something like '127.0.0.1:5060'.
Where <options> are in standard Asterisk flag options format:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
* New options were added for a multicast RTP channel. The format for
dialing a multicast RTP channel is:
MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
Where <type> can be either 'basic' or 'linksys'.
Where <destination-addr> is something like '224.0.0.3:5060'.
Where <control-addr> is something like '127.0.0.1:5060'.
Where <options> are in standard Asterisk flag options format:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
i(<address>) - Specify the interface address from which multicast RTP
is sent.
l(<enable>) - Set whether packets are looped back to the sender. The
enable value can be 0 to set looping to off and non-zero to set
looping on.
t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
chan_sip
------------------
* New 'rtpbindaddr' global setting. This allows a user to define which
ipaddress to bind the rtpengine to. For example, chan_sip might bind
to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
* DTLS related configuration options can now be set at a general level.
Enabling DTLS support, though, requires enabling it at the user
or peer level.
* Added the possibility to set the From: header through the the SIP dial
string (populating the fromuser/fromdomain fields), complementing the
[!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
NOTE: This is again separated by an exclamation mark, so the To: header may
not contain one of those.
* Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
Previously Asterisk dropped calls only with UDP transports. However with
longer international calls via TCP, the SIP channel might break, because
all hops on the Internet route must stay online (have not a single power
outage, for example). Therefore with Session-Timers enabled (which are
enabled at default), you might see additional dropped calls. Consequently
please, consider to go for session-timers=refuse in your sip.conf.
chan_pjsip
------------------
* New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
to the request URI and From URI if the user is determined to be a phone
number.
* New 'moh_passthrough' endpoint setting. This will pass hold and unhold
requests through using SIP re-invites with sendonly and sendrecv accordingly.
* Added the pjsip.conf system type disable_tcp_switch option. The option
allows the user to disable switching from UDP to TCP transports described
by RFC 3261 section 18.1.1.
* New 'line' and 'endpoint' options added on outbound registrations. This
allows some identifying information to be added to the Contact of the
outbound registration. If this information is present on messages received
from the remote server the message will automatically be associated with the
configured endpoint on the outbound registration.
Core
------------------
* The core of Asterisk uses a message bus called "Stasis" to distribute
information to internal components. For performance reasons, the message
distribution was modified to make use of a thread pool instead of a
dedicated thread per consumer in certain cases. The initial settings for
the thread pool can now be configured in 'stasis.conf'.
* A new core DNS API has been implemented which provides a common interface
for DNS functionality. Modules that use this functionality will require that
a DNS resolver module is loaded and available.
* Modified processing of command-line options to first parse only what
is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
the remaining options are processed. The -X option now applies to
asterisk.conf only. To enable #exec for other config files you must
set execincludes=yes in asterisk.conf. Any other option set on the
command-line will now override the equivalent setting from asterisk.conf.
* The TLS core in Asterisk now supports X.509 certificate subject alternative
names. This way one X.509 certificate can be used for hosts that can be
reached under multiple DNS names or for multiple hosts.
* The Asterisk logging system now supports JSON structured logging. Log
channels specified in logger.conf or added dynamically via CLI commands now
support an optional specifier prior to their levels that determines their
formatting. To set a log channel to format its entries as JSON, a formatter
of '[json]' can be set, e.g.,
full => [json]debug,verbose,notice,warning,error
* The core now supports a 'media cache', which stores temporary media files
retrieved from external sources. CLI commands have been added to manipulate
and display the cached files, including:
- 'media cache show <all>' - show all cached media files, or details about
one particular cached media file
- 'media cache refresh <item>' - force a refresh of a particular media file
in the cache
- 'media cache delete <item>' - remove an item from the cache
- 'media cache create <uri>' - retrieve a URI and store it in the cache
* The ability for device state hints to be automatically created as a result of
device state changes now exists in the PBX. This functionality is referred to
as "autohints" and is configurable in extensions.conf by placing "autohints=yes"
in the context. If enabled a device state hint will be automatically created
with the name of the device.
* If Asterisk is built with systemd support, and run under systemd, it will
notify systemd of its state using sd_notify. Use 'Type=notify' in
asterisk.service.
Functions
------------------
* The func_odbc global option "single_db_connection" default value has been
changed to 'no'.
Formats
------------------
* New module format_ogg_speex added which supports Speex codec inside
Ogg containers (filename extension .spx).
CHANNEL
------------------
* Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
the hold status of a channel.
CURL
------------------
* The CURL function now supports a write option, which will save the retrieved
file to a location on disk. As an example:
same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav)
will save 'foo.wav' to /tmp.
DTMF Features
------------------
* The transferdialattempts default value has been changed from 1 to 3. The
transferinvalidsound has been changed from "pbx-invalid" to
"privacy-incorrect". These were changed to make DTMF transfers be more
user-friendly by default.
Resources
------------------
res_http_media_cache
------------------
* A backend for the core media cache, this module retrieves media files from
a remote HTTP(S) server and stores them in the core media cache for later
playback.
res_musiconhold
------------------
* Added sort=randstart to the sort options. It sorts the files by name and
then chooses the first file to play at random.
* Added preferchannelclass=no option to prefer the application-passed class
over the channel-set musicclass. This allows separate hold-music from
application (e.g. Queue or Dial) specified music.
res_resolver_unbound
------------------
* Added a res_resolver_unbound module which uses the libunbound resolver library
to perform DNS resolution. This module requires the libunbound library to be
installed in order to be used.
res_pjsip
------------------
* A new SIP resolver using the core DNS API has been implemented. This relies on
external SIP resolver support in PJSIP which is only available as of PJSIP
2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
will be used instead. The new SIP resolver provides NAPTR support, improved
SRV support, and AAAA record support.
res_pjsip_info_empty
--------------------
* A new module that can respond to empty Content-Type INFO packets during call.
Some SBCs will terminate a call if their empty INFO packets are not responded
to within a predefined time.
res_pjsip_outbound_registration
-------------------------------
* A new 'fatal_retry_interval' option has been added to outbound registration.
When set (default is zero), and upon receiving a failure response to an
outbound registration, registration is retried at the given interval up to
'max_retries'.
res_pjsip_outbound_publish
------------------
* Added a new multi_user option that when set to 'yes' allows a given configuration
to be used for multiple users.
CEL Backends
------------------
cel_pgsql
------------------
* Added a new option, 'usegmtime', which causes timestamps in CEL events
to be logged in GMT.
* Added support to set schema where located the table cel. This settings is
configurable for cel_pgsql via the 'schema' in configuration file
cel_pgsql.conf.
CDR Backends
------------------
cdr_adaptive_odbc
------------------
* Added the ability to set the character to quote identifiers. This
allows adding the character at the start and end of table and column
names. This setting is configurable for cdr_adaptive_odbc via the
quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
cdr_odbc
------------------
* Added a new configuration option, "newcdrcolumns", which enables use of the
post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
cdr_csv
------------------
* Added a new configuration option, "newcdrcolumns", which enables use of the
post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
------------------------------------------------------------------------------
chan_dahdi
------------------
* Added "faxdetect_timeout" option.
The option determines how many seconds into a call before faxdetect
is disabled for the call. Setting the value to zero disables the timeout.
res_pjsip
------------------
* Added "fax_detect_timeout" to endpoint.
The option determines how many seconds into a call before fax_detect
is disabled for the call. Setting the value to zero disables the timeout.
* Added "subscribe_context" to endpoint.
If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
res_rtp_asterisk
------------------
* The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
Enabling PFS is attempted by default, and is dependent on the configuration
of the module using TLS.
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in sip.conf, for example:
dtlscipher=AES128-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, e.g., sip.conf dtlsprivatekey. For example:
openssl dhparam -out ./dh.pem 2048
- Because clients expect the server to prefer PFS, and because OpenSSL sorts
its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
Consider re-ordering your cipher suites in the respective configuration
file. For example:
dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
which forces PFS and requires at least DTLS 1.2.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
------------------------------------------------------------------------------
Core
------------------
* A channel variable FORWARDERNAME is now set which indicates which channel
was responsible for a forwarding requests received on dial attempt.
func_odbc
------------------
* Added new global option "single_db_connection".
Enabling this option func_odbc will use a single database connection per DSN.
This option is enabled by default.
res_fax
------------------
* Added FAXMODE variable to let dialplan know what fax transport was used.
FAXMODE variable is set to either "audio" or "T38".
res_pjsip
------------------
* Added "via_addr", "via_port", "call_id" to contacts.
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered endpoint.
Added new fields ViaAddress,CallID to AMI event ContactStatus
* Endpoint IP Access Controls
Added new configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
* Added "reg_server" to contacts.
If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.
* When starting Asterisk, received traffic will now be ignored until Asterisk
has loaded all modules and is fully booted.
res_hep
------------------
* Added a new option, 'uuid_type', that sets the preferred source of the Homer
correlation UUID. The valid options are:
- call-id: Use the PJSIP SIP Call-ID header value
- channel: Use the Asterisk channel name
The default value is 'call-id'. In the event that a HEP module cannot find a
valid value using the specified 'uuid_type', the module may fallback to a
more readily available source for the correlation UUID.
res_odbc
------------------
* A new option has been added, 'max_connections', which sets the maximum number
of concurrent connections to the database. This option defaults to 1 which
returns the behavior to that of Asterisk 13.7 and prior.
app_confbridge
------------------
* Added a bridge profile option called regcontext that allows you to
dynamically register the conference bridge name as an extension into
the specified context. This allows tracking down conferences on multi-
server installations via alternate means (DUNDI for example). By default
this feature is not used.
Codecs
------------------
* Added the associated format name to 'core show codecs'.
res_ari_channels
------------------
* Added 'formats' to channel create/originate to allow setting the allowed
formats for a channel when no originator channel is available. Especially
useful for Local channel creation where no other format information is
available. 'core show codecs' can now be used to look up suitable format
names.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------
------------------------------------------------------------------------------
res_parking:
- The dynamic parking lot creation channel variables PARKINGDYNAMIC,
PARKINGDYNCONTEXT, PARKINGDYNEXTEN, and PARKINGDYNPOS are now looked
for in the parker's channel instead of the parked channel. This is only
of significance if the parker uses blind transfer or the DTMF one-step
parking feature. You need to use the double underscore '__' inheritance
for these variables. The indefinite inheritance is also recommended
for the PARKINGEXTEN variable.
res_pjsip
------------------
* Added new global option (disable_multi_domain) to pjsip.
Disabling Multi Domain can improve realtime performace by reducing
number of database requsts.
chan_pjsip
------------------
* Added 'pjsip show channelstats' CLI command.
res_pjsip_outbound_publish
------------------
* Added support for setting the transport used on outbound publish
using the transport configuration option.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
------------------------------------------------------------------------------
res_pjsip_caller_id
------------------
* Per RFC3325, the 'From' header is now anonymized on outgoing calls when
caller id presentation is prohibited.
res_pjsip_config_wizard
------------------
* A new command (pjsip export config_wizard primitives) has been added that
will export all the pjsip objects it created to the console or a file
suitable for reuse in a pjsip.conf file.
Build System
------------------
* To help insure that Asterisk is compiled and run with the same known
version of pjproject, a new option (--with-pjproject-bundled) has been
added to ./configure. When specified, the version of pjproject specified
in third-party/versions.mak will be downloaded and configured. When you
make Asterisk, the build process will also automatically build pjproject
and Asterisk will be statically linked to it. Once a particular version
of pjproject is configured and built, it won't be configured or built
again unless you run a 'make distclean'.
To facilitate testing, when 'make install' is run, the pjsua and pjsystest
utilities and the pjproject python bindings will be installed in
ASTDATADIR/third-party/pjproject.
The default behavior remains building with the shared pjproject
installation, if any.
app_confbridge
------------------
* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
* Added Muted header to AMI ConfbridgeListRooms action response list events
to indicate the muted conference state.
* Added Muted column to CLI "confbridge list" output to indicate the muted
conference state and made the locked column a yes/no value instead of a
locked/unlocked value.
REDIRECTING(reason)
------------------
* The REDIRECTING(reason) value is now treated consistently between
chan_sip and chan_pjsip.
Both channel drivers match incoming reason values with values documented
by REDIRECTING(reason) and values documented by RFC5806 regardless of
whether they are quoted or not. RFC5806 values are mapped to the
equivalent REDIRECTING(reason) documented value and is set in
REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
quoted string version ('"unconditional"') is converted to
REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
with 'cfu' instead of any of the aliases.
The incoming 480 response reason text supported by chan_sip checks for
known reason values and if not matched then puts quotes around the reason
string and assigns that to REDIRECTING(reason).
Both channel drivers send outgoing known REDIRECTING(reason) values as the
unquoted RFC5806 equivalent. User custom values are either sent as is or
with added quotes if SIP doesn't allow a character within the value as
part of a RFC3261 Section 25.1 token. Note that there are still
limitations on what characters can be put in a custom user value. e.g.,
embedding quotes in the middle of the reason string is just going to cause
you grief.
* Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
'cfu' value.
res_pjproject
------------------
* This module is the successor of res_pjsip_log_forwarder. As well as
handling the log forwarding (which now displays as 'pjproject:0' instead
of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
This displays the compiled-in options of the pjproject installation
Asterisk is currently running against.
* Another feature of this module is the ability to map pjproject log levels
to Asterisk log levels, or to suppress the pjproject log messages
altogether. Many of the messages emitted by pjproject itself are the result
of errors which Asterisk will ultimately handle so the messages can be
misleading or just noise. A new config file (pjproject.conf) has been added
to configure the mapping and a new CLI command (pjproject show log mappings)
has been added to display the mappings currently in use.
res_pjsip
------------------
* Transports are now reloadable. In testing, no in-progress calls were
disrupted if the ip address or port weren't changed, but the possibility
still exists. To make sure there are no unintentional drops, a new option
'allow_reload', which defaults to 'no' has been added to transport. If
left at the default, changes to the particular transport will be ignored.
If set to 'yes', changes (if any) will be applied.
* Added new global option (regcontext) to pjsip. When set, Asterisk will
dynamically create and destroy a NoOp priority 1 extension
for a given endpoint who registers or unregisters with us.
* Endpoints and aors can now be identified by the username and realm in an
incoming Authorization header. To use this feature, add "auth_username"
to your endpoint's "identify_by" list. You can combine "auth_username"
and the original "username" to test both the From/To and Authorization
headers. For endpoints, the order is controlled by the global
"endpoint_identifier_order" setting. For matching aors to an endpoint
for inbound registration, the order is controlled by this option.
* In conjunction with the "auth_username" change, 3 new options have been
added to the global configuration object that control how many unidentified
requests over a certain period from the same IP address can be received
before a security alert is generated. A new CLI command
"pjsip show unidentified_requests" will list the current candidates.
res_pjsip_history
------------------
* A new module, res_pjsip_history, has been added that provides SIP history
viewing/filtering from the CLI. The module is intended to be used on systems
with busy SIP traffic, where existing forms of viewing SIP messages - such
as the res_pjsip_logger - may be inadequate. The module provides two new
CLI commands:
- 'pjsip set history {on|off|clear}' - this enables/disables SIP history
capturing, as well as clears an existing history capture. Note that SIP
packets captured are stored in memory until cleared. As a result, the
history capture should only be used for debugging/viewing purposes, and
should *NOT* be left permanently enabled on a system.
- 'pjsip show history' - displays the captured SIP history. When invoked
with no options, the entire captured history is displayed. Two options
are available:
-- 'entry <num>' - display a detailed view of a single SIP message in
the history
-- 'where ...' - filter the history based on some expression. For more
information on filtering, view the current CLI help for the
'pjsip show history' command.
Voicemail
------------------
* app_voicemail and res_mwi_external can now be built together. The default
remains to build app_voicemail and not res_mwi_external but if they are
both built, the load order will cause res_mwi_external to load first and
app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
modules.conf to force app_voicemail to be the voicemail provider.
res_pjsip_sdp_rtp
------------------
* A new option (bind_rtp_to_media_address) has been added to endpoint which
will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
media_address as well as using it in the SDP. If set, RTP packets will now
originate from the media address instead of the operating system's "primary"
ip address.
res_rtp_asterisk
------------------
* A new configuration section - ice_host_candidates - has been added to
rtp.conf, allowing automatically discovered ICE host candidates to be
overriden. This allows an Asterisk server behind a 1:1 NAT to send its
external IP as a host candidate rather than relying on STUN to discover it.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
------------------------------------------------------------------------------
Codecs
------------------
* Added format attribute negotiation for the VP8 video codec. Format attribute
negotiation is provided by the res_format_attr_vp8 module.
ConfBridge
------------------
* A new "timeout" user profile option has been added. This configures the number
of seconds that a participant may stay in the ConfBridge after joining. When
the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
is set to "TIMEOUT" on the channel.
chan_sip
------------------
* The websockets_enabled option has been added to the general section of
sip.conf. The option is enabled by default to match the previous behavior.
The option should be disabled when using res_pjsip_transport_websockets to
ensure chan_sip will not conflict with PJSIP websockets.
Dialplan Functions
------------------
* The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
While support for the events was added in Asterisk 13.4.0, the function
accidentally never made it in. That function is now present, and will cause
the 'hold' raised by a channel to be intercepted and converted into an
event instead.
res_pjsip_outbound_registration
-------------------------------
* If res_statsd is loaded and a StatsD server is configured, basic statistics
regarding the state of outbound registrations will now be emitted. This
includes:
- A GAUGE statistic for the overall number of outbound registrations, i.e.:
PJSIP.registrations.count
- A GAUGE statistic for the overall number of outbound registrations in a
particular state, e.g.:
PJSIP.registrations.state.Registered
res_pjsip
------------------
* The ability to use "like" has been added to the pjsip list and show
CLI commands. For instance: CLI> pjsip list endpoints like abc
* If res_statsd is loaded and a StatsD server is configured, basic statistics
regarding the state of PJSIP contacts will now be emitted. This includes:
- A GAUGE statistic for the overall number of contacts in a particular
state, e.g.:
PJSIP.contacts.states.Reachable
- A TIMER statistic for the RTT time for each qualified contact, e.g.:
PJSIP.contacts.alice@@127.0.0.1:5061.rtt
res_sorcery_memory_cache
------------------------
* A new caching strategy, full_backend_cache, has been added which caches
all stored objects in the backend. When enabled all objects will be
expired or go stale according to the configuration. As well when enabled
all retrieval operations will be performed against the cache instead of
the backend.
func_callerid
-------------------
* CALLERID(pres) is now documented as a valid alternative to setting both
CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
like chan_sip, don't make a distinction between the two: they take the
least public value from name-pres and num-pres. By using CALLERID(pres)
for reading and writing, you touch the same combined value in the dialplan.
The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
REDIRECTING(to-pres) and REDIRECTING(from-pres).
res_endpoint_stats
-------------------
* A new module that emits StatsD statistics regarding Asterisk endpoints.
This includes a total count of the number of endpoints, the count of the
number of endpoints in the technology agnostic state of the endpoint -
online or offline - as well as the number of channels associated with each
endpoint. These are recorded as three different GAUGE statistics:
- endpoints.count
- endpoints.state.{unknown|offline|online}
- endpoints.{tech}.{resource}.channels
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
------------------------------------------------------------------------------
Dialplan Functions
------------------
* The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
extraction option when using with the 'pjsip' signalling option. It will
return the SIP Call-ID associated with the INVITE request that established
the PJSIP channel.
ARI
------------------
* Two new endpoint related events are now available: PeerStatusChange and
ContactStatusChange. In particular, these events are useful when subscribing
to all event sources, as they provide additional endpoint related
information beyond the addition/removal of channels from an endpoint.
* Added the ability to subscribe to all ARI events in Asterisk, regardless
of whether the application 'controls' the resource. This is useful for
scenarios where an ARI application merely wants to observe the system,
as opposed to control it. There are two ways to accomplish this:
(1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
has been added that, when present and True, will subscribe all
specified applications to all ARI event sources in Asterisk.
(2) Via the applications resource. An ARI client can, at any time, subscribe
to all resources in an event source merely by not providing an explicit
resource. For example, subscribing to an event source of 'channels:'
as opposed to 'channels:12345' will subscribe the application to all
channels.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* A new ContactStatus event has been added that reflects res_pjsip contact
lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
* Added the Linkedid header to the common channel headers listed for each
channel in AMI events.
ARI
------------------
* A new feature has been added that enables the retrieval of modules and
module information through an HTTP request. Information on a single module
can be also be retrieved. Individual modules can be loaded to Asterisk, as
well as unloaded and reloaded.
* A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
This resource allows for push configuration of sorcery derived objects
within Asterisk. The resource supports creation, retrieval, updating, and
deletion. Sorcery derived objects that are manipulated by this resource
must have a sorcery wizard that supports the desired operations.
* A new feature has been added that allows for the rotation of log channels
through HTTP requests.
res_pjsip
------------------
* A new 'g726_non_standard' endpoint option has been added that, when set to
'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
is AAL2 packed on the channel.
* A new 'rtp_keepalive' endpoint option has been added. This option specifies
an interval, in seconds, at which we will send RTP comfort noise packets to
the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
* New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
These options specify the amount of time, in seconds, that Asterisk will wait
before terminating the call due to lack of received RTP. These are identical
to chan_sip's rtptimeout and rtpholdtimeout options.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
------------------------------------------------------------------------------
chan_pjsip
------------------
* New 'rpid_immediate' option to control if connected line update information
goes to the caller immediately or waits for another reason to send the
connected line information update. See the online option documentation for
more information. Defaults to 'no' as setting it to 'yes' can result in
many unnecessary messages being sent to the caller.
* The configuration setting 'progressinband' now defaults to 'no', which
matches the actual behavior of previous versions.
res_pjsip
------------------
* A new CLI command has been added: "pjsip show settings", which shows
both the global and system configuration settings.
* A new aor option has been added: "qualify_timeout", which sets the timeout
in seconds for a qualify. The default is 3 seconds. This overrides the
hard coded 32 seconds in pjproject.
* Endpoint status will now change to "Unreachable" when all contacts are
unavailable. When any contact becomes available, the endpoint will status
will change back to "Reachable".
* A new global option has been added: "max_initial_qualify_time", which
sets the maximum amount of time from startup that qualifies should be
attempted on all contacts.
res_ari_channels
------------------
* Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
events data model. These events are raised when a channel indicates a hold
or unhold, respectively.
func_holdintercept
------------------
* A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
placed on a channel, intercepts hold/unhold indications signalled by the
channel and prevents them from moving on to other channels in a bridge with
the hold initiator. Instead, AMI or ARI events are raised indicating that
the channel wanted to place someone on hold. This allows external
applications to implement their own custom hold/unhold logic.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
------------------------------------------------------------------------------
chan_pjsip/app_transfer
------------------
* The Transfer application, when used with chan_pjsip, now supports using
a PJSIP endpoint as the transfer destination. This is in addition to
explicitly specifying a SIP URI to transfer to.
res_ari_channels
------------------
* The ARI /channels resource now supports a new operation, 'redirect'. The
redirect operation will perform a technology and state specific redirection
on the channel to a specified endpoint or destination. In the case of SIP
technologies, this is either a 302 Redirect response to an on-going INVITE
dialog or a SIP REFER request.
res_pjsip
------------------
* A new 'endpoint_identifier_order' option has been added that allows one to
set the order by which endpoint identifiers are processed and checked. This
option is specified under the 'global' type configuration section.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
------------------------------------------------------------------------------
* New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
allow examining PJSIP AORs or contacts from the dialplan.
res_pjsip_outbound_registration
------------------
* The 'pjsip send unregister' command now stops further registrations.
* A new command 'pjsip send register' has been added which allows you to
start or restart periodic registration. It can be used after a
'send unregister' or after a 401 permanent error.
res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targeted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
res_fax
-----------
* The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
and is now configurable via the 't38timeout' configuration option in
res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
The default remains at 5000 milliseconds.
PJSIP Transports
----------
* The ca_list_path transport parameter has been added for TLS transports. This
option behaves similarly to the old sip.conf option "tlscapath". In order to
use this, you must be using PJProject version 2.4 or higher.
ARI
------------------
* The Originate operation now takes in an originator channel. The linked ID of
this originator channel is applied to the newly originated outgoing channel.
If using CEL this allows an association to be established between the two so
it can be recognized that the originator is dialing the originated channel.
* "language" (the default spoken language for the channel) is now included in
the standard channel state output for suitable events.
* The POST channels/{id} operation and the POST channels/{id}/continue operation
now have a new "label" parameter. This allows for origination or continuation
to a labeled priority in the dialplan instead of requiring a specific priority
number. The ARI version has been bumped to 1.7.0 as a result.
AMI
------------------
* "Language" (the default spoken language for the channel) is now included in
the standard channel state output for suitable events.
* AMI actions that return a list of events have been made to return consistent
headers for the action response event starting the list and the list complete
event. The AMI version has been bumped to 2.7.0 as a result.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* Event NewConnectedLine is emitted when the connected line information on
a channel changes.
ARI
------------------
* Event ChannelConnectedLine is emitted when the connected line information
on a channel changes.
Core Transfers
-----------------
The features.conf general section has three new configurable options:
* transferdialattempts
* transferretrysound
* transferinvalidsound
For more information on what these options do, see the Asterisk wiki:
https://wiki.asterisk.org/wiki/x/W4fAAQ
Channel Drivers
------------------
chan_pjsip
------------------
* New 'media_encryption_optimistic' endpoint setting. This will use SRTP
when possible but does not consider lack of it a failure.
res_pjsip_endpoint_identifer_ip
------------------
* New CLI commands have been added: "pjsip show identif(y|ies)", which lists
all configured PJSIP identify objects
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
------------------------------------------------------------------------------
Overview
------------------
Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
the focus of development for this release of Asterisk was on improving the
usability and features developed in the previous Standard release, Asterisk 12.
Beyond a general refinement of end user features, development focussed heavily
on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
new features include:
* Asterisk security events are now provided via AMI, allowing end users to
monitor their Asterisk system in real time for security related issues.
* External control of Message Waiting Indicators (MWI) through both AMI and ARI.
* Reception/transmission of out of call text messages using any supported
channel driver/protocol stack through ARI.
* Resource List Server support in the PJSIP stack, providing subscriptions to
lists of resources and batched delivery of NOTIFY requests.
* Inter-Asterisk distributed device state and mailbox state using the PJSIP
stack.
It is important to note that Asterisk 13 is built on the architecture developed
during the previous Standard release, Asterisk 12. Users upgrading to
Asterisk 13 should read about the new features in Asterisk 12 later in this file
(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
UPGRADE-12.txt delivered with this release. In particular, users upgrading to
Asterisk 13 from a release prior to Asterisk 12 should read the specifications
on AMI, CDRs, and CEL on the Asterisk wiki:
* AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
* CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
* CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
Many new featuers in Asterisk 13 were introduced in point releases of
Asterisk 12. Following this section - which documents the changes from all
versions of Asterisk 12 to Asterisk 13 - users should examine the new features
that were introduced in the point releases of Asterisk 12, as they are also
included in Asterisk 13.
Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
delivered with this release.
Build System
------------------
* Sample config files have been moved from configs/ to a sub-folder of that
directory, samples.
* The menuselect utility has been pulled into the Asterisk repository. As a
result, the libxml2 development library is now a required dependency for
Asterisk.
* A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
counted objects will emit additional debug information to the refs log file
located in the standard Asterisk log file directory. This log file is useful
in tracking down object leaks and other reference counting issues. Prior to
this version, this option was only available by modifying the source code
directly. This change also includes a new script, refcounter.py, in the
contrib folder that will process the refs log file. Note that this replaces
the refcounter utility that could be built from the utils directory.
Applications
------------------
DahdiBarge
------------------
* This module was deprecated and has been removed. Users of app_dahdibarge
should use ChanSpy instead.
MixMonitor
------------------
* New options to play a beep when starting a recording and stopping a recording
have been added. The option "p" will play a beep to the channel that starts
the recording. The option "P" will play a beep to the channel that stops the
recording.
Queue
------------------
* Queue rules can now be stored in a database table, queue_rules. Unlike other
RealTime tables, the queue_rules table is only examined on module load or
module reload. A new general setting has been added to queuerules.conf,
'realtime_rules', which, when set to 'yes', will cause app_queue to look in
RealTime for additional queue rules to parse. Note that both the file and
the database can be used as a provide of queue rules when 'realtime_rules'
is set to 'yes'.
When app_queue is reloaded, all rules are re-parsed and loaded into memory.
There is no caching of RealTime queue rules.
ReadFile
------------------
* This module was deprecated and has been removed. Users of app_readfile
should use func_env's FILE function instead.
Say
------------------
* The 'say' family of dialplan applications now support the Japanese
language. The 'language' parameter in say.conf now recognizes a setting of
'ja', which will enable Japanese language specific mechanisms for playing
back numbers, dates, and other items.
* Counting, enumeration and dates now supports Icelandic grammar with the
'language' parameter set to 'is'.
SayCountPL
------------------
* This module was deprecated and has been removed. Users of app_saycountpl
should use the Say family of applications.
SetMusicOnHold
------------------
* The SetMusicOnHold dialplan application was deprecated and has been removed.
Users of the application should use the CHANNEL function's musicclass
setting instead.
WaitMusicOnHold
------------------
* The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a duration
parameter instead.
VoiceMail
------------------
* VoiceMail and VoiceMailMain now support the Japanese language. The
'language' parameter in voicemail.conf now recognizes a setting of 'ja',
which will enable prompts to be played back using a Japanese grammatical
structure. Additional prompts are necessary for this functionality,
including:
- jb-arimasu: there is
- jb-arimasen: there is not
- jb-oshitekudasai: please press
- jb-ni: article ni
- jb-ga: article ga
- jb-wa: article wa
- jb-wo: article wo
* Add the ability to specify multiple email addresses in configuration,
separated by a |.
CDR Backends
------------------
cdr_sqlite
-----------------
* This module was deprecated and has been removed. Users of cdr_sqlite
should use cdr_sqlite3_custom.
cdr_pgsql
------------------
* Added the ability to support PostgreSQL application_name on connections.
This allows PostgreSQL to display the configured name in the
pg_stat_activity view and CSV log entries. This setting is configurable
for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
CEL Backends
------------------
cel_pgsql
------------------
* Added the ability to support PostgreSQL application_name on connections.
This allows PostgreSQL to display the configured name in the
pg_stat_activity view and CSV log entries. This setting is configurable
for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
Channel Drivers
------------------
chan_dahdi
------------------
* SS7 support now requires libss7 v2.0 or later.
* Added SS7 support for connected line and redirecting.
* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
chan_gtalk
------------------
* This module was deprecated and has been removed. Users of chan_gtalk
should use chan_motif.
chan_h323
------------------
* This module was deprecated and has been removed. Users of chan_h323
should use chan_ooh323.
chan_jingle
------------------
* This module was deprecated and has been removed. Users of chan_jingle
should use chan_motif.
chan_pjsip
------------------
* Added the CLI command 'pjsip list ciphers' so a user can know what
OpenSSL names are available on their system for the pjsip.conf cipher
option.
chan_sip
------------------
* The SIPPEER dialplan function no longer supports using a colon as a
delimiter for parameters. The parameters for the function should be
delimited using a comma.
* The SIPCHANINFO dialplan function was deprecated and has been removed. Users
of the function should use the CHANNEL function instead.
Core
------------------
Account Codes
------------------
* Added functional peeraccount support. Except for Queue, the
accountcode propagation is now consistently propagated to outgoing
channels before dialing. The channel accountcode can change from its
original non-empty value on channel creation for the following specific
reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
originate method that can specify an accountcode value. Three, the
calling channel propagates its peeraccount or accountcode to the
outgoing channel's accountcode before dialing. The change has two
visible effects. One, local channels now cross accountcode and
peeraccount across the special bridge between the ;1 and ;2 channels
just like channels between normal bridges. Two, the
CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
set the accountcode on the outgoing channel(s).
For Queue, an outgoing channel's non-empty accountcode will not change
unless explicitly set by CHANNEL(accountcode). The change has three
visible effects. One, local channels now cross accountcode and
peeraccount across the special bridge between the ;1 and ;2 channels
just like channels between normal bridges. Two, the queue member will
get an accountcode if it doesn't have one and one is available from the
calling channel's peeraccount. Three, accountcode propagation includes
local channel members where the accountcodes are propagated early
enough to be available on the ;2 channel.
AMI
------------------
* New DeviceStateChanged and PresenceStateChanged AMI events have been added.
These events are emitted whenever a device state or presence state change
occurs. The events are controlled by res_manager_device_state.so and
res_manager_presence_state.so. If the high frequency of these events is
problematic for you, do not load these modules.
* Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
work in basically the same way as the 'dialplan add extension' and
'dialplan remove extension' CLI commands respectively.
* New AMI action LoggerRotate reloads and rotates logger in the same manner
as CLI command 'logger rotate'
* New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
functionality of CLI commands 'fax show sessions', 'fax show session',
and fax show stats' respectively.
* New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
enable manager control over PRI debugging levels and file output.
* AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
endpoint as long as a default outbound endpoint is set. This also applies
to the equivalent CLI command (pjsip send notify)
* The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
that give information on Asterisk's attempts to qualify the endpoint.
* The DialEnd event will now contain a Forward header if the dial is ending
due to the call being forwarded. The contents of the Forward header is the
extension in the number to which the call is being forwarded.
CEL
------------------
* The "bridge_technology" extra field key has been added to BRIDGE_ENTER
and BRIDGE_EXIT events.
Features
------------------
* Channel variables are now substituted in arguments passed to applications
run by using dynamic features.
TLS
------------------
* The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
Enabling PFS is attempted by default, and is dependent on the configuration
of the module using TLS.
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in sip.conf, for example:
tlscipher=AES128-SHA:DES-CBC3-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, e.g., sip.conf tlsprivatekey. For example, the
default dh2048.pem - see
http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
- Because clients expect the server to prefer PFS, and because OpenSSL sorts
its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
Consider re-ordering your cipher suites in the respective configuration
file. For example:
tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
will use PFS when offered by the client. Clients which do not offer PFS
fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
Functions
------------------
JACK_HOOK
------------------
* The JACK_HOOK function now supports audio with a sample rate higher than
8kHz.
Resources
------------------
res_config_pgsql
------------------
* Added the ability to support PostgreSQL application_name on connections.
This allows PostgreSQL to display the configured name in the
pg_stat_activity view and CSV log entries. This setting is configurable
for res_config_pgsql via the dbappname configuration setting in
res_pgsql.conf.
res_pjsip_outbound_publish
------------------
* A new module, res_pjsip_outbound_publish provides the mechanisms for sending
PUBLISH requests for specific event packages to another SIP User Agent.
res_pjsip_pubsub
------------------
* The publish/subscribe core module has been updated to support RFC 4662
Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
Resource lists are configured in pjsip.conf under a new object type,
resource_list. Resource lists can contain either message-summary or presence
events, and can be composed of specific resources that provide the event or
other resource lists.
* Inbound publication support is provided by a new object, inbound-publication.
This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
resource. Which events are accepted is constructed dynamically; see
res_pjsip_publish_asterisk for more information.
res_pjsip_publish_asterisk
------------------
* A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
Asterisk information to other Asterisk servers. This module is intended only
for Asterisk to Asterisk exchanges of information. Currently, this includes
both mailbox state and device state information.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
------------------------------------------------------------------------------
ARI
------------------
* Stored recordings now support a new operation, copy. This will take an
existing stored recording and copy it to a new location in the recordings
directory.
* LiveRecording objects now have three additional fields that can be reported
in a RecordingFinished ARI event:
- total_duration: the duration of the recording
- talking_duration: optional. The duration of talking detected in the
recording. This is only available if max_silence_seconds was specified
when the recording was started.
- silence_duration: optional. The duration of silence detected in the
recording. This is only available if max_silence_seconds was specified
when the recording was started.
Note that all duration values are reported in seconds.
* Users of ARI can now send and receive out of call text messages. Messages
can be sent directly to a particular endpoint, or can be sent to the
endpoints resource directly and inferred from the URI scheme. Text
messages are passed to ARI clients as TextMessageReceived events. ARI
clients can choose to receive text messages by subscribing to the particular
endpoint technology or endpoints that they are interested in.
* The applications resource now supports subscriptions to all endpoints of
a particular channel technology. For example, subscribing to an eventSource
of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
res_pjsip
------------------
* The endpoint configuration object now supports 'accountcode'. Any channel
created for an endpoint with this setting will have its accountcode set
to the specified value.
res_hep_rtcp
------------------
* A new module, res_hep_rtcp, has been added that will forward RTCP call
statistics to a HEP capture server. See res_hep for more information.
Functions
------------------
* Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
unconditionally inherited through masquerades. As a side benefit, more
than one audiohook of a given type may persist through a masquerade now.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
------------------------------------------------------------------------------
AgentRequest
------------------
* Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
connect with an incoming caller after being alerted to the presence
of the incoming caller. The most likely reason this would happen is
the agent did not acknowledge the call in time.
AMI
------------------
* New events have been added for the TALK_DETECT function. When the function
is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
emitted to connected AMI clients indicating the start/stop of talking on
the channel.
ARI
------------------
* New event models have been aded for the TALK_DETECT function. When the
function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
events will be emitted to connected WebSockets subscribed to the channel,
indicating the start/stop of talking on the channel.
Functions
------------------
* A new function, TALK_DETECT, has been added. When set on a channel, this
fucntion causes events indicating the starting/stoping of talking on said
channel to be emitted to both AMI and ARI clients.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
------------------------------------------------------------------------------
ARI
------------------
* A new Playback URI 'tone' has been added. Tones are specified either as
an indication name (e.g. 'tone:busy') from indications.conf or as a tone
pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
URIs in that they must be stopped manually and will continue to occupy
a channel's ARI control queue until they are stopped. They also can not
be rewound or fastforwarded.
* User events can now be generated from ARI. Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots. An application must be specified which will receive
the event message (other applications can subscribe to it). The message
will also be delivered via AMI provided a channel is attached. Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.
chan_sip
-----------
* SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
fields for prohibited callingpres information. Values are legacy, no, and
yes. By default, legacy is used.
trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
headers are appended to outbound SIP messages just as they are with
allowed callingpres values, but data about the remote party's identity is
anonymized.
When sendrpid=rpid, only the remote party's domain is anonymized.
trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
headers are not sent.
trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
party information in tact even for prohibited callingpres information.
In the case of PAI, a Privacy: id header will be appended for prohibited
calling information to communicate that the private information should
not be relayed to untrusted parties.
res_parking
------------------
* Manager action 'Park' now takes an additional argument 'AnnounceChannel'
which can be used to announce the parked call's location to an arbitrary
channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
parties in a one to one bridge, 'TimeoutChannel' is treated as having
parked 'Channel' like with the Park Call DTMF feature and will receive
announcements prior to being hung up.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
------------------------------------------------------------------------------
Record
------------------
* Record application now has an option 'o' which allows 0 to act as an exit
key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
ChanSpy
--------------------------
* ChanSpy now accepts a channel uniqueid or a fully specified channel name
as the chanprefix parameter if the 'u' option is specified.
ConfBridge
--------------------------
* CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
conference user menus.
* CONFBRIDGE dialplan function is now capable of removing dynamic conference
menus, bridge settings, and user settings that have been applied by the
CONFBRIDGE dialplan function.
* The ConfBridge dialplan application now sets a channel variable,
CONFBRIDGE_RESULT, upon exiting. This variable can be used to determine
how a channel exited the conference.
* Added conference user option 'announce_join_leave_review'. This option
implies 'announce_join_leave' with the added effect that the user will
be asked if they want to confirm or re-record the recording of their
name when entering the conference
Directory
--------------------------
* At exit, the Directory application now sets a channel variable
DIRECTORY_RESULT to one of the following based on the reason for exiting:
OPERATOR user requested operator by pressing '0' for operator
ASSISTANT user requested assistant by pressing '*' for assistant
TIMEOUT user pressed nothing and Directory stopped waiting
HANGUP user's channel hung up
SELECTED user selected a user from the directory and is routed
USEREXIT user pressed '#' from the selection prompt to exit
FAILED directory failed in a way that wasn't accounted for. Dang.
Monitor
------------------
* Monitor() - A new option, B(), has been added that will turn on a periodic
beep while the call is being recorded.
MusicOnHold
--------------------------
* MusicOnHold streams (all modes other than "files") now support wide band
audio too.
Page
--------------------------
* Added options 'b' and 'B' to apply predial handlers for outgoing calls
and for the channel executing Page respectively.
PickupChan
--------------------------
* PickupChan now accepts channel uniqueids of channels to pickup.
Say
--------------------------
* If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
to 'true' (case insensitive), then any Say application (SayNumber,
SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
anticipate DTMF. If DTMF is received, these applications will behave like
the background application and jump to the received extension once a match
is established or after a short period of inactivity.
MixMonitor
-------------------------
* A new function, MIXMONITOR, has been added to allow access to individual
instances of MixMonitor on a channel.
* A new option, B(), has been added that will turn on a periodic beep while the
call is being recorded.
Channel Drivers
-------------------------
chan_sip
-------------------------
* TEL URI support for inbound INVITE requests has been added. chan_sip will
now handle TEL schemes in the Request and From URIs. The phone-context in
the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
the inbound channel.
Core
------------------
* Exposed sorcery-based configuration files like pjsip.conf to dialplans via
the new AST_SORCERY diaplan function.
* Core Show Locks output now includes Thread/LWP ID if the platform
supports this feature.
* New "logger add channel" and "logger remove channel" CLI commands have
been added to allow creation and deletion of dynamic logger channels
without configuration changes. These dynamic logger channels will only
exist until the next restart of asterisk.
ARI
------------------
* The live recording object on recording events now contains a target_uri
field which contains the URI of what is being recorded.
* The bridge type used when creating a bridge is now a comma separated list of
bridge properties. Valid options are: mixing, holding, dtmf_events, and
proxy_media.
* A channelId can now be provided when creating a channel, either in the
uri (POST channels/my-channel-id) or as query parameter. A local channel
will suffix the second channel id with ';2' unless provided as query
parameter otherChannelId.
* A bridgeId can now be provided when creating a bridge, either in the uri
(POST bridges/my-bridge-id) or as a query parameter.
* A playbackId can be provided when starting a playback, either in the uri
(POST channels/my-channel-id/play/my-playback-id /
POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
* A snoop channel can be started with a snoopId, in the uri or query.
AMI
------------------
* Originate now takes optional parameters ChannelId and OtherChannelId,
used to set the UniqueId on creation. The other id is assigned to the
second channel when dialing LOCAL, or defaults to appending ;2 if only
the single Id is given.
* The Mixmonitor action now has a "Command" header that can be used to
indicate a post-process command to run once recording finishes.
RealTime
------------------
* A new set of Alembic scripts has been added for CDR tables. This will create
a 'cdr' table with the default schema that Asterisk expects.
Functions
------------------
* A new function was added: PERIODIC_HOOK. This allows running a periodic
dialplan hook on a channel. Any audio generated by this hook will be
injected into the call.
Resources
------------------
res_hep
------------------
* A new module, res_hep, has been added, that acts as a generic packet
capture agent for the Homer Encapsulation Protocol (HEP) version 3.
It can be configured via hep.conf. Other modules can use res_hep to send
message traffic to a HEP capture server.
res_hep_pjsip
------------------
* A new module, res_hep_pjsip, has been added that will forward PJSIP
message traffic to a HEP capture server. See res_hep for more
information.
res_pjsip
------------------
* transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
* Added the following new CLI commands:
- "pjsip show contacts" - list all current PJSIP contacts.
- "pjsip show contact" - show specific information about a current PJSIP
contact.
- "pjsip show channel" - show detailed information about a PJSIP channel.
res_pjsip_multihomed
------------------
* A new module, res_pjsip_multihomed handles situations where the system
Asterisk is running out has multiple interfaces. res_pjsip_multihomed
determines which interface should be used during message sending.
res_pjsip_pidf_digium_body_supplement
------------------
* A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
request body formatting for presence support in Digium phones.
res_pjsip_send_to_voicemail
------------------
* A new module, res_pjsip_send_to_voicemail allows for REFER requests with
particular headers to transfer a PJSIP channel directly to a particular
extension that has VoiceMail. This is intended to be used with Digium
phones that support this feature.
res_pjsip_outbound_registration
------------------
* A new CLI command has been added: "pjsip show registrations", which lists
all configured PJSIP registrations
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* Added a new module that provides AMI control over MWI within Asterisk,
res_mwi_external_ami. Note that this module depends on res_mwi_external;
for more information on enabling this module, see res_mwi_external.
This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
the MWIGet/MWIGetComplete events.
* The DialStatus field in the DialEnd event can now contain additional
statuses that convey how the dial operation terminated. This includes
ABORT, CONTINUE, and GOTO.
* AMI will now emit security events. A new class authorization has been
added in manager.conf for the security events, 'security'. The new events
are:
- FailedACL - raised when a request violates an ACL check
- InvalidAccountID - raised when a request fails an authentication
check due to an invalid account ID
- SessionLimit - raised when a request fails due to exceeding the
number of allowed concurrent sessions for a service
- MemoryLimit - raised when a request fails due to an internal memory
allocation failure
- LoadAverageLimit - raised when a request fails because a configured
load average limit has been reached
- RequestNotAllowed - raised when a request is not allowed by
the service
- AuthMethodNotAllowed - raised when a request used an authentication
method not allowed by the service
- RequestBadFormat - raised when a request is received with bad formatting
- SuccessfulAuth - raised when a request successfully authenticates
- UnexpectedAddress - raised when a request has a different source address
then what is expected for a session already in progress with a service
- ChallengeResponseFailed - raised when a request's attempt to authenticate
has been challenged, and the request failed the authentication challenge
- InvalidPassword - raised when a request provides an invalid password
during an authentication attempt
- ChallengeSent - raised when an Asterisk service send an authentication
challenge to a request
- InvalidTransport - raised when a request attempts to use a transport not
allowed by the Asterisk service
* Bridge related events now have two additional fields: BridgeName and
BridgeCreator. BridgeName is a descriptive name for the bridge;
BridgeCreator is the name of the entity that created the bridge. This
affects the following events: ConfbridgeStart, ConfbridgeEnd,
ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
ARI
------------------
* The Bridge data model now contains the additional fields 'name' and
'creator'. The 'name' field conveys a descriptive name for the bridge;
the 'creator' field conveys the name of the entity that created the bridge.
This affects all responses to HTTP requests that return a Bridge data model
as well as all event derived data models that contain a Bridge data model.
The POST /bridges operation may now optionally specify a name to give to
the bridge being created.
* Added a new ARI resource 'mailboxes' which allows the creation and
modification of mailboxes managed by external MWI. Modules res_mwi_external
and res_stasis_mailbox must be enabled to use this resource. For more
information on external MWI control, see res_mwi_external.
* Added new events for externally initiated transfers. The event
BridgeBlindTransfer is now raised when a channel initiates a blind transfer
of a bridge in the ARI controlled application to the dialplan; the
BridgeAttendedTransfer event is raised when a channel initiates an
attended transfer of a bridge in the ARI controlled application to the
dialplan.
* Channel variables may now be specified as a body parameter to the
POST /channels operation. The 'variables' key in the JSON is interpreted
as a sequence of key/value pairs that will be added to the created channel
as channel variables. Other parameters in the JSON body are treated as
query parameters of the same name.
HTTP
------------------
* Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
automatically handled by the HTTP server if a request is received with a
Transfer-Encoding type of "chunked".
res_pjsip
------------------
* Path support has been added with the 'support_path' option in registration
and aor sections.
* A 'debug' option has been added to the globals section that will allow
sip messages to be logged.
* A 'set_var' option has been added to endpoints that will automatically
set the desired variable(s) on a channel created for that endpoint.
* Several new tables and columns have been added to the realtime schema for
the res_pjsip related modules. See the UPGRADE.txt notes for updating
the database schema.
res_mwi_external
------------------
* A new module, res_mwi_external, has been added to Asterisk. This module
acts as a base framework that other modules can build on top of to allow
an external system to control MWI within Asterisk. For implementations
that make use of res_mwi_external, see res_mwi_external_ami and
res_ari_mailboxes. Note that res_mwi_external conflicts with other modules
that may produce MWI themselves, such as app_voicemail. res_mwi_external
and other modules that depend on it cannot be built or loaded with
app_voicemail present.
res_pjsip
------------------
* DNS functionality will now automatically be enabled if the system configured
nameservers can be retrieved. If the system configured nameservers can not be
retrieved the functionality will resort to using system resolution. Functionality
such as SRV records and failover will not be available if system resolution
is in use.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
------------------------------------------------------------------------------
Overview
------------------
Asterisk 12 is a standard release of the Asterisk project. As such, the
focus of development for this release was on core architectural changes and
major new features. This includes:
* A more flexible bridging core based on the Bridging API
* A new internal message bus, Stasis
* Major standardization and consistency improvements to AMI
* Addition of the Asterisk RESTful Interface (ARI)
* A new SIP channel driver, chan_pjsip
In addition, as the vast majority of bridging in Asterisk was migrated to the
Bridging API used by ConfBridge, major changes were made to most of the
interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
Specifications have been written for the affected interfaces. These
specifications are available on the Asterisk wiki:
* AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
* CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
* CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
It is *highly* recommended that anyone migrating to Asterisk 12 read the
information regarding its release both in this file and in the accompanying
UPGRADE.txt file. More detailed information on the major changes can be found
on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
Build System
------------------
* Added build option DISABLE_INLINE. This option can be used to work around a
bug in gcc. For more information, see
http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
* Removed the CHANNEL_TRACE development mode build option. Certain aspects of
the CHANNEL_TRACE build option were incompatible with the new bridging
architecture.
* Asterisk now optionally uses libxslt to improve XML documentation generation
and maintainability. If libxslt is not available on the system, some XML
documentation will be incomplete.
* Asterisk now depends on libjansson. If a package of libjansson is not
available on your distro, please see http://www.digip.org/jansson/.
* Asterisk now depends on libuuid and, optionally, uriparser. It is
recommended that you install uriparser, even if it is optional.
* The new SIP stack and channel driver uses a particular version of PJSIP.
Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
configuring and installing PJSIP for usage with Asterisk.
* Optional API was re-implemented to be more portable, and no longer requires
weak reference support from the compiler. The build option OPTIONAL_API may
be disabled to disable Optional API support.
Applications
------------------
AgentLogin
------------------
* Along with AgentRequest, this application has been modified to be a
replacement for chan_agent. The act of a channel calling the AgentLogin
application places the channel into a pool of agents that can be
requested by the AgentRequest application. Note that this application, as
well as all other agent related functionality, is now provided by the
app_agent_pool module. See chan_agent and AgentRequest for more information.
* This application no longer performs agent authentication. If authentication
is desired, the dialplan needs to perform this function using the
Authenticate or VMAuthenticate application or through an AGI script before
running AgentLogin.
* If this application is called and the agent is already logged in, the
dialplan will continue execution with the AGENT_STATUS channel variable set
to ALREADY_LOGGED_IN.
* The agents.conf schema has changed. Rather than specifying agents on a
single line in comma delineated fashion, each agent is defined in a separate
context. This allows agents to use the power of context templates in their
definition.
* A number of parameters from agents.conf have been removed. This includes
maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
urlprefix, and savecallsin. These options were obsoleted by the move from
a channel driver model to the bridging/application model provided by
app_agent_pool.
AgentRequest
------------------
* A new application, this will request a logged in agent from the pool and
bridge the requested channel with the channel calling this application.
Logged in agents are those channels that called the AgentLogin application.
If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
application will be set with an appropriate error value.
AgentMonitorOutgoing
------------------
* This application has been removed. It was a holdover from when
AgentCallbackLogin was removed.
AlarmReceiver
------------------
* Added support for additional Ademco DTMF signalling formats, including
Express 4+1, Express 4+2, High Speed and Super Fast.
* Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
call time, in milliseconds, to run the application.
* Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
maximum number of times to retry the call.
* Added a new configuration option answait. If set, the AlarmReceiver
application will wait the number of milliseconds specified by answait
after the channel has answered. Valid values range between 500
milliseconds and 10000 milliseconds.
* Added configuration option no_group_meta. If enabled, grouping of metadata
information in the AlarmReceiver log file will be skipped.
Answer
------------------
* It is now no longer possible to bypass updating the CDR on the channel
when answering. CDRs reflect the state of the channel and will always
reflect the time they were Answered.
BridgeWait
------------------
* A new application in Asterisk, this will place the calling channel
into a holding bridge, optionally entertaining them with some form of
media. Channels participating in a holding bridge do not interact with
other channels in the same holding bridge. Optionally, however, a channel
may join as an announcer. Any media passed from an announcer channel is
played to all channels in the holding bridge. Channels leave a holding
bridge either when an optional timer expires, or via the ChannelRedirect
application or AMI Redirect action.
ConfBridge
------------------
* All participants in a bridge can now be kicked out of a conference room
by specifying the channel parameter as 'all' in the ConfBridge kick CLI
command, i.e., 'confbridge kick <conference> all'
* CLI output for the 'confbridge list' command has been improved. When
displaying information about a particular bridge, flags will now be shown
for the participating users indicating properties of that user.
* The ConfbridgeList event now contains the following fields: WaitMarked,
EndMarked, and Waiting. This displays additional properties about the
user's profile, as well as whether or not the user is waiting for a
Marked user to enter the conference.
* Added a new option for conference recording, record_file_append. If enabled,
when the recording is stopped and then re-started, the existing recording
will be used and appended to.
* ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in confbridge.conf
or by the dialplan function CONFBRIDGE(bridge,language)=en.
ControlPlayback
------------------
* The channel variable CPLAYBACKSTATUS may now return the value
'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
such as AMI. See the AMI action ControlPlayback for more information.
Directory
------------------
* Added the 'a' option, which allows the caller to enter in an additional
alias for the user in the directory. This option must be used in conjunction
with the 'f', 'l', or 'b' options. Note that the alias for a user can be
specified in voicemail.conf.
DumpChan
------------------
* The output of DumpChan no longer includes the DirectBridge or IndirectBridge
fields. Instead, if a channel is in a bridge, it includes a BridgeID field
containing the unique ID of the bridge that the channel happens to be in.
ForkCDR
------------------
* ForkCDR no longer automatically resets the forked CDR. See the 'r' option
for more information.
* Variables are no longer purged from the original CDR. See the 'v' option for
more information.
* The 'A' option has been removed. The Answer time on a CDR is never updated
once set.
* The 'd' option has been removed. The disposition on a CDR is a function of
the state of the channel and cannot be altered.
* The 'D' option has been removed. Who the Party B is on a CDR is a function
of the state of the respective channels involved in the CDR and cannot be
altered.
* The 'r' option has been changed. Previously, ForkCDR always reset the CDR
such that the start time and, if applicable, the answer time was updated.
Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
'r' option now triggers the Reset, setting the start time (and answer time
if applicable) to the current time. Note that the 'a' option still sets
the answer time to the current time if the channel was already answered.
* The 's' option has been removed. A variable can be set on the original CDR
if desired using the CDR function, and removed from a forked CDR using the
same function.
* The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
longer applies in the CDR engine.
* The 'v' option now prevents the copy of the variables from the original CDR
to the forked CDR. Previously the variables were always copied but were
removed from the original. This was changed as removing variables from a CDR
can have unintended side effects - this option allows the user to prevent
propagation of variables from the original to the forked without modifying
the original.
MeetMe
-------------------
* Added the 'n' option to MeetMe to prevent application of the DENOISE
function to a channel joining a conference. Some channel drivers that vary
the number of audio samples in a voice frame will experience significant
quality problems if a denoiser is attached to the channel; this option gives
them the ability to remove the denoiser without having to unload func_speex.
MixMonitor
------------------
* The 'b' option now includes conferences as well as sounds played to the
participants.
* The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
running during a transfer. If a MixMonitor is started on a channel,
the MixMonitor will continue to record the audio passing through the
channel even in the presence of transfers.
NoCDR
------------------
* The NoCDR application is deprecated. Please use the CDR_PROP function to
disable CDRs.
* While the NoCDR application will prevent CDRs for a channel from being
propagated to registered CDR backends, it will not prevent that data from
being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
function that enables CDRs on a channel will restore those records that have
not yet been finalized.
ParkAndAnnounce
-------------------
* The app_parkandannounce module has been removed. The application
ParkAndAnnounce is now provided by the res_parking module. See the
res_parking changes for more information.
Queue
-------------------
* Added queue available hint. The hint can be added to the dialplan using the
following syntax: exten,hint,Queue:{queue_name}_avail
For example, if the name of the queue is 'markq':
exten => 8501,hint,Queue:markq_avail
This will report 'InUse' if there are no logged in agents or no free agents.
It will report 'Idle' when an agent is free.
* Queues now support a hint for member paused state. The hint uses the form
'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
are the name of the queue and the name of the member to subscribe to,
respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
Members will show as In Use when paused.
* The configuration options eventwhencalled and eventmemberstatus have been
removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
sent. The "Variable" fields will also no longer exist on the Agent* events.
These events can be filtered out from a connected AMI client using the
eventfilter setting in manager.conf.
* The queue log now differentiates between blind and attended transfers. A
blind transfer will result in a BLINDTRANSFER message with the destination
context and extension. An attended transfer will result in an
ATTENDEDTRANSFER message. This message will indicate the method by which
the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
for running an application on a bridge or channel, or "LINK" for linking
two bridges together with local channels. The queue log will also now detect
externally initiated blind and attended transfers and record the transfer
status accordingly.
* When performing queue pause/unpause on an interface without specifying an
individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
least one member of any queue exists for that interface.
* Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
for realtime queue log entries.
ResetCDR
------------------
* The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
CDRs when they were previously disabled on a channel.
* The 'w' and 'a' options have been removed. Dispatching CDRs to registered
backends occurs on an as-needed basis in order to preserve linkedid
propagation and other needed behavior.
SayAlphaCase
------------------
* A new application, this is similar to SayAlpha except that it supports
case sensitive playback of the specified characters. For example,
SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
SetAMAFlags
------------------
* This application is deprecated in favor of CHANNEL(amaflags).
SendDTMF
------------------
* The SendDTMF application will now accept 'W' as valid input. This will cause
the application to delay one second while streaming DTMF.
Stasis
------------------
* A new application in Asterisk 12, this hands control of the channel calling
the application over to an external system. Currently, external systems
manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
UserEvent
------------------
* UserEvent will now handle duplicate keys by overwriting the previous value
assigned to the key.
* In addition to AMI, UserEvent invocations will now be distributed to any
interested Stasis applications.
VoiceMail
------------------
* Mailboxes defined by app_voicemail MUST be referenced by the rest of the
system as mailbox@context. The rest of the system cannot add @default
to mailbox identifiers for app_voicemail that do not specify a context
any longer. It is a mailbox identifier format that should only be
interpreted by app_voicemail.
* The voicemail.conf configuration file now has an 'alias' configuration
parameter for use with the Directory application. The voicemail realtime
database table schema has also been updated with an 'alias' column.
Codecs
------------------
* Pass through support has been added for both VP8 and Opus.
* Added format attribute negotiation for the Opus codec. Format attribute
negotiation is provided by the res_format_attr_opus module.
Core
------------------
* Masquerades as an operation inside Asterisk have been effectively hidden
by the migration to the Bridging API. As such, many 'quirks' of Asterisk
no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
dropping of frame/audio hooks, and other internal implementation details
that users had to deal with. This fundamental change has large implications
throughout the changes documented for this version. For more information
about the new core architecture of Asterisk, please see the Asterisk wiki.
* Multiple parties in a bridge may now be transferred. If a participant in a
multi-party bridge initiates a blind transfer, a Local channel will be used
to execute the dialplan location that the transferer sent the parties to. If
a participant in a multi-party bridge initiates an attended transfer,
several options are possible. If the attended transfer results in a transfer
to an application, a Local channel is used. If the attended transfer results
in a transfer to another channel, the resulting channels will be merged into
a single bridge.
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
driver specific. If the channel variable is set on the transferrer channel,
the sound will be played to the target of an attended transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers in
a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
listed. Any more peers in the bridge will not be included in the list.
BRIDGEPEER is not valid in holding bridges like parking since those channels
do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* A channel variable ATTENDEDTRANSFER is now set which indicates which channel
was responsible for an attended transfer in a similar fashion to
BLINDTRANSFER.
* Modules using the Configuration Framework or Sorcery must have XML
configuration documentation. This configuration documentation is included
with the rest of Asterisk's XML documentation, and is accessible via CLI
commands. See the CLI changes for more information.
AMI (Asterisk Manager Interface)
------------------
* Major changes were made to both the syntax as well as the semantics of the
AMI protocol. In particular, AMI events have been substantially improved
in this version of Asterisk. For more information, please see the AMI
specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
* AMI events that reference a particular channel or bridge will now always
contain a standard set of fields. When multiple channels or bridges are
referenced in an event, fields for at least some subset of the channels
and bridges in the event will be prefixed with a descriptive name to avoid
name collisions. See the AMI event documentation on the Asterisk wiki for
more information.
* The CLI command 'manager show commands' no longer truncates command names
longer than 15 characters and no longer shows authorization requirement
for commands. 'manager show command' now displays the privileges needed
for using a given manager command instead.
* The SIPshowpeer action will now include a 'SubscribeContext' field for a
peer in its response if the peer has a subscribe context set.
* The SIPqualifypeer action now acknowledges the request once it has
established that the request is against a known peer. It also issues a new
event, 'SIPQualifyPeerDone', once the qualify action has been completed.
* The PlayDTMF action now supports an optional 'Duration' parameter. This
specifies the duration of the digit to be played, in milliseconds.
* Added VoicemailRefresh action to allow an external entity to trigger mailbox
updates when changes occur instead of requiring the use of pollmailboxes.
* Added a new action 'ControlPlayback'. The ControlPlayback action allows an
AMI client to manipulate audio currently being played back on a channel. The
supported operations depend on the application being used to send audio to
the channel. When the audio playback was initiated using the ControlPlayback
application or CONTROL STREAM FILE AGI command, the audio can be paused,
stopped, restarted, reversed, or skipped forward. When initiated by other
mechanisms (such as the Playback application), the audio can be stopped,
reversed, or skipped forward.
* Channel related events now contain a snapshot of channel state, adding new
fields to many of these events.
* The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
in a future release. Please use the common 'Exten' field instead.
* The AMI event 'UserEvent' from app_userevent now contains the channel state
fields. The channel state fields will come before the body fields.
* The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
'UnParkedCall' have changed significantly in the new res_parking module.
The 'Channel' and 'From' headers are gone. For the channel that was parked
or is coming out of parking, a 'Parkee' channel snapshot is issued and it
has a number of fields associated with it. The old 'Channel' header relayed
the same data as the new 'ParkeeChannel' header.
The 'From' field was ambiguous and changed meaning depending on the event.
for most of these, it was the name of the channel that parked the call
(the 'Parker'). There is no longer a header that provides this channel name,
however the 'ParkerDialString' will contain a dialstring to redial the
device that parked the call.
On UnParkedCall events, the 'From' header would instead represent the
channel responsible for retrieving the parkee. It receives a channel
snapshot labeled 'Retriever'. The 'from' field is is replaced with
'RetrieverChannel'.
Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
* The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
fashion has changed the field names 'StartExten' and 'StopExten' to
'StartSpace' and 'StopSpace' respectively.
* The deprecated use of | (pipe) as a separator in the channelvars setting in
manager.conf has been removed.
* Channel Variables conveyed with a channel no longer contain the name of the
channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
ChanVariable: bar=baz. When multiple channels are present in a single AMI
event, the various ChanVariable fields will contain a suffix that specifies
which channel they correspond to.
* The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
event always conveys the AMI event for a particular channel.
* All 'Reload' events have been consolidated into a single event type. This
event will always contain a Module field specifying the name of the module
and a Status field denoting the result of the reload. All modules now issue
this event when being reloaded.
* The 'ModuleLoadReport' event has been removed. Most AMI connections would
fail to receive this event due to being connected after modules have loaded.
AMI connections that want to know when Asterisk is ready should listen for
the 'FullyBooted' event.
* app_fax now sends the same send fax/receive fax events as res_fax. The
'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
now the 'ReceiveFAX' event.
* The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
'MusicOnHoldStop'. The sub type field has been removed.
* The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
carrier for another protocol.
* The Bridge Manager action's 'Playtone' header now accepts more fine-grained
options. 'Channel1' and 'Channel2' may be specified in order to play a tone
to the specific channel. 'Both' may be specified to play a tone to both
channels. The old 'yes' option is still accepted as a way of playing the
tone to Channel2 only.
* The AMI 'Status' response event to the AMI Status action replaces the
'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
indicate what bridge the channel is currently in.
* The AMI 'Hold' event has been moved out of individual channel drivers, into
core, and is now two events: 'Hold' and 'Unhold'. The status field has been
removed.
* The AMI events in app_queue have been made more consistent with each other.
Events that reference channels (QueueCaller* and Agent*) will show
information about each channel. The (infamous) 'Join' and 'Leave' AMI
events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
* The 'MCID' AMI event now publishes a channel snapshot when available and
its non-channel-snapshot parameters now use either the "MCallerID" or
'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
parameters in the channel snapshot.
* The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
'AgentLogin' and 'AgentLogoff' respectively.
* The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
* 'ChannelUpdate' events have been removed.
* All AMI events now contain a 'SystemName' field, if available.
* Local channel optimization is now conveyed in two events:
'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
when the Local channel driver begins attempting to optimize itself out of
the media path; the End event is sent after the channel halves have
successfully optimized themselves out of the media path.
* Local channel information in events is now prefixed with 'LocalOne' and
'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
and 'LocalOptimizationEnd' events.
* The option 'allowmultiplelogin' can now be set or overriden in a particular
account. When set in the general context, it will act as the default
setting for defined accounts.
* The 'BridgeAction' event was removed. It technically added no value, as the
Bridge Action already receives confirmation of the bridge through a
successful completion Event.
* The 'BridgeExec' events were removed. These events duplicated the events that
occur in the Bridging API, and are conveyed now through BridgeCreate,
BridgeEnter, and BridgeLeave events.
* The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
previous versions. They now report all SR/RR packets sent/received, and
have been restructured to better reflect the data sent in a SR/RR. In
particular, the event structure now supports multiple report blocks.
* Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
raised when a blind transfer/attended transfer completes successfully.
They contain information about the transfer that just completed, including
the location of the transfered channel.
* Added a 'security' class to AMI which outputs the required fields for
security messages similar to the log messages from res_security_log
* The AMI event 'ExtensionStatus' now contains a 'StatusText' field
that describes the status value in a human readable string.
CDR (Call Detail Records)
------------------
* Significant changes have been made to the behavior of CDRs. The CDR engine
was effectively rewritten and built on the Stasis message bus. For a full
definition of CDR behavior in Asterisk 12, please read the specification
on the Asterisk wiki (wiki.asterisk.org).
* CDRs will now be created between all participants in a bridge. For each
pair of channels in a bridge, a CDR is created to represent the path of
communication between those two endpoints. This lets an end user choose who
to bill for what during bridge operations with multiple parties.
* The duration, billsec, start, answer, and end times now reflect the times
associated with the current CDR for the channel, as opposed to a cumulative
measurement of all CDRs for that channel.
* When a CDR is dispatched, user defined CDR variables from both parties are
included in the resulting CDR. If both parties have the same variable, only
the Party A value is provided.
* Added a new option to cdr.conf, 'debug'. When enabled, significantly more
information regarding the CDR engine is logged as verbose messages. This
option should only be used if the behavior of the CDR engine needs to be
debugged.
* Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
normally configured in cdr.conf.
* Added CLI command 'cdr show active {channel}'. When {channel} is not
specified, this command provides a summary of the channels with CDR
information and their statistics. When {channel} is specified, it shows
detailed information about all records associated with {channel}.
CEL (Channel Event Logging)
------------------
* CEL has undergone significant rework in Asterisk 12, and is now built on the
Stasis message bus. Please see the specification for CEL on the Asterisk
wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
information.
* The 'extra' field of all CEL events that use it now consists of a JSON blob
with key/value pairs which are defined in the Asterisk 12 CEL documentation.
* BLINDTRANSFER events now report the transferee bridge unique
identifier, extension, and context in a JSON blob as the extra string
instead of the transferee channel name as the peer.
* ATTENDEDTRANSFER events now report the peer as NULL and additional
information in the 'extra' string as a JSON blob. For transfers that occur
between two bridged channels, the 'extra' JSON blob contains the primary
bridge unique identifier, the secondary channel name, and the secondary
bridge unique identifier. For transfers that occur between a bridged channel
and a channel running an app, the 'extra' JSON blob contains the primary
bridge unique identifier, the secondary channel name, and the app name.
* LOCAL_OPTIMIZE events have been added to convey local channel
optimizations with the record occurring for the semi-one channel and
the semi-two channel name in the peer field.
* BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
regardless of whether or not that bridge happens to contain multiple
parties.
CLI
-------------------
* When compiled with '--enable-dev-mode', the astobj2 library will now add
several CLI commands that allow for inspection of ao2 containers that
register themselves with astobj2. The CLI commands are 'astobj2 container
dump', 'astobj2 container stats', and 'astobj2 container check'.
* Added specific CLI commands for bridge inspection. This includes 'bridge
show all', which lists all bridges in the system, and 'bridge show {id}',
which provides specific information about a bridge.
* Added CLI command 'bridge destroy'. This will destroy the specified bridge,
ejecting the channels currently in the bridge. If the channels cannot
continue in the dialplan or application that put them in the bridge, they
will be hung up.
* Added command 'bridge kick'. This will eject a single channel from a bridge.
* Added commands to inspect and manipulate the registered bridge technologies.
This include 'bridge technology show', which lists the registered bridge
technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
which controls whether or not a registered bridge technology can be used
during smart bridge operations. If a technology is suspended, it will not
be used when a bridge technology is picked for channels; when unsuspended,
it can be used again.
* The command 'config show help {module} {type} {option}' will show
configuration documentation for modules with XML configuration
documentation. When {module}, {type}, and {option} are omitted, a listing
of all modules with registered documentation is displayed. When {module}
is specified, a listing of all configuration types for that module is
displayed, along with their synopsis. When {module} and {type} are
specified, a listing of all configuration options for that type are
displayed along with their synopsis. When {module}, {type}, and {option}
are specified, detailed information for that configuration option is
displayed.
* Added 'core show sounds' and 'core show sound' CLI commands. These display
a listing of all installed media sounds available on the system and
detailed information about a sound, respectively.
* 'xmldoc dump' has been added. This CLI command will dump the XML
documentation DOM as a string to the specified file. The Asterisk core
will populate certain XML elements pulled from the source files with
additional run-time information; this command lets a user produce the
XML documentation with all information.
Features
-------------------
* Parking has been pulled from core and placed into a separate module called
res_parking. See Parking changes below for more details. Configuration for
parking should now be performed in res_parking.conf. Configuration for
parking in features.conf is now unsupported.
* Core attended transfers now have several new options. While performing an
attended transfer, the transferer now has the following options:
- *1 - cancel the attended transfer (configurable via atxferabort)
- *2 - complete the attended transfer, dropping out of the call
(configurable via atxfercomplete)
- *3 - complete the attended transfer, but stay in the call. This will turn
the call into a multi-party bridge (configurable via atxferthreeway)
- *4 - swap to the other party. Once an attended transfer has begun, this
options may be used multiple times (configurable via atxferswap)
* For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
must be on the channel initiating the transfer to have any effect.
* The BRIDGE_FEATURES channel variable would previously only set features for
the calling party and would set this feature regardless of whether the
feature was in caps or in lowercase. Use of a caps feature for a letter
will now apply the feature to the calling party while use of a lowercase
letter will apply that feature to the called party.
* Add support for automixmon to the BRIDGE_FEATURES channel variable.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
only on the channel executing the dynamic feature. Executing a dynamic
feature on the bridge peer in a multi-party bridge will execute it on all
peers of the activating channel.
* You can now have the settings for a channel updated using the FEATURE()
and FEATUREMAP() functions inherited to child channels by setting
FEATURE(inherit)=yes.
* automixmon now supports additional channel variables from automon including:
TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
and TOUCH_MIXMONITOR_MESSAGE_STOP
* A new general features.conf option 'recordingfailsound' has been added which
allowssetting a failure sound for a user tries to invoke a recording feature
such as automon or automixmon and it fails.
* It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
features.c for atxferdropcall=no to work properly. This option now just
works.
Logging
-------------------
* Added log rotation strategy 'none'. If set, no log rotation strategy will
be used. Given that this can cause the Asterisk log files to grow quickly,
this option should only be used if an external mechanism for log management
is preferred.
Realtime
------------------
* Dynamic realtime tables for SIP Users can now include a 'path' field. This
will store the path information for that peer when it registers. Realtime
tables can also use the 'supportpath' field to enable Path header support.
* LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
objectIdentifier. This maps to the supportpath option in sip.conf.
Sorcery
------------------
* Sorcery is a new data abstraction and object persistence API in Asterisk. It
provides modules a useful abstraction on top of the many storage mechanisms
in Asterisk, including the Asterisk Database, static configuration files,
static Realtime, and dynamic Realtime. It also provides a caching service.
Users can configure a hierarchy of data storage layers for specific modules
in sorcery.conf.
* All future modules which utilize Sorcery for object persistence must have a
column named "id" within their schema when using the Sorcery realtime module.
This column must be able to contain a string of up to 128 characters in length.
Security Events Framework
------------------
* Security Event timestamps now use ISO 8601 formatted date/time instead of
the "seconds-microseconds" format that it was using previously.
Stasis Message Bus
------------------
* The Stasis message bus is a publish/subscribe message bus internal to
Asterisk. Many services in Asterisk are built on the Stasis message bus,
including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
Stasis can be configured in stasis.conf. Note that these parameters operate
at a very low level in Asterisk, and generally will not require changes.
Channel Drivers
------------------
* When a channel driver is configured to enable jiterbuffers, they are now
applied unconditionally when a channel joins a bridge. If a jitterbuffer
is already set for that channel when it enters, such as by the JITTERBUFFER
function, then the existing jitterbuffer will be used and the one set by
the channel driver will not be applied.
chan_agent
------------------
* chan_agent has been removed and replaced with AgentLogin and AgentRequest
dialplan applications provided by the app_agent_pool module. Agents are
connected with callers using the new AgentRequest dialplan application.
The Agents:<agent-id> device state is available to monitor the status of an
agent. See agents.conf.sample for valid configuration options.
* The updatecdr option has been removed. Altering the names of channels on a
CDR is not supported - the name of the channel is the name of the channel,
and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
has also been removed, for the same reason.
* The endcall and enddtmf configuration options are removed. Use the
dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
channel before calling AgentLogin.
chan_bridge
------------------
* chan_bridge has been removed. Its functionality has been incorporated
directly into the ConfBridge application itself.
chan_dahdi
------------------
* Added the CLI command 'pri destroy span'. This will destroy the D-channel
of the specified span and its B-channels. Note that this command should
only be used if you understand the risks it entails.
* The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
A range of channels can be specified to be destroyed. Note that this command
should only be used if you understand the risks it entails.
* Added the CLI command 'dahdi create channels'. A range of channels can be
specified to be created, or the keyword 'new' can be used to add channels
not yet created.
* The script specified by the chan_dahdi.conf mwimonitornotify option now gets
the exact configured mailbox name. For app_voicemail mailboxes this is
mailbox@context.
* Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
chan_iax2
------------------
* IPv6 support has been added. We are now able to bind to and
communicate using IPv6 addresses.
chan_local
------------------
* The /b option has been removed.
* chan_local moved into the system core and is no longer a loadable module.
chan_mobile
------------------
* Added general support for busy detection.
* Added ECAM command support for Sony Ericsson phones.
chan_pjsip
------------------
* A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
SIP stack. A collection of resource modules provides the bulk of the SIP
functionality. For more information on the new SIP channel driver, see
https://wiki.asterisk.org/wiki/x/JYGLAQ
chan_sip
------------------
* Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
using the 'supportpath' setting, either on a global basis or on a peer basis.
This setting enables Asterisk to route outgoing out-of-dialog requests via a
set of proxies by using a pre-loaded route-set defined by the Path headers in
the REGISTER request. See Realtime updates for more configuration information.
* The SIP_CODEC family of variables may now specify more than one codec. Each
codec must be separated by a comma. The first codec specified is the
preferred codec for the offer. This allows a dialplan writer to specify both
audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
* The 'callevents' parameter has been removed. Hold AMI events are now raised
in the core, and can be filtered out using the 'eventfilter' parameter
in manager.conf.
* Added 'ignore_requested_pref'. When enabled, this will use the preferred
codecs configured for a peer instead of the requested codec.
* The option "register_retry_403" has been added to chan_sip to work around
servers that are known to erroneously send 403 in response to valid
REGISTER requests and allows Asterisk to continue attepmting to connect.
chan_skinny
------------------
* Added the 'immeddialkey' parameter. If set, when the user presses the
configured key the already entered number will be immediately dialed. This
is useful when the dialplan allows for variable length pattern matching.
Valid options are '*' and '#'.
* Added the 'callfwdtimeout' parameter. This configures the amount of time (in
milliseconds) before a call forward is considered to not be answered.
* The 'serviceurl' parameter allows Service URLs to be attached to line
buttons.
Functions
------------------
AGENT
------------------
* The password option has been disabled, as the AgentLogin application no
longer provides authentication.
AUDIOHOOK_INHERIT
------------------
* Due to changes in the Asterisk core, this function is no longer needed to
preserve a MixMonitor on a channel during transfer operations and dialplan
execution. It is effectively obsolete.
CDR (function)
------------------
* The 'amaflags' and 'accountcode' attributes for the CDR function are
deprecated. Use the CHANNEL function instead to access these attributes.
* The 'l' option has been removed. When reading a CDR attribute, the most
recent record is always used. When writing a CDR attribute, all non-finalized
CDRs are updated.
* The 'r' option has been removed, for the same reason as the 'l' option.
* The 's' option has been removed, as LOCKED semantics no longer exist in the
CDR engine.
CDR_PROP
------------------
* A new function CDR_PROP has been added. This function lets you set properties
on a channel's active CDRs. This function is write-only. Properties accept
boolean values to set/clear them on the channel's CDRs. Valid properties
include:
- 'party_a' - make this channel the preferred Party A in any CDR between two
channels. If two channels have this property set, the creation time of the
channel is used to determine who is Party A. Note that dialed channels are
never Party A in a CDR.
- 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
application when set to True, and analogous to the 'e' option in ResetCDR
when set to False.
CHANNEL
------------------
* Added the argument 'dtmf_features'. This sets the DTMF features that will be
enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
application.
* Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
string, i.e., [[context],extension],priority. If set on a channel, if a
channel leaves a bridge but is not hung up it will resume dialplan execution
at that location.
JITTERBUFFER
------------------
* JITTERBUFFER now accepts an argument of 'disabled' which can be used
to remove jitterbuffers previously set on a channel with JITTERBUFFER.
The value of this setting is ignored when disabled is used for the argument.
PJSIP_DIAL_CONTACTS
------------------
* A new function provided by chan_pjsip, this function can be used in
conjunction with the Dial application to construct a dial string that will
dial all contacts on an Address of Record associated with a chan_pjsip
endpoint.
PJSIP_MEDIA_OFFER
------------------
* Provided by chan_pjsip, this function sets the codecs to be offered on the
outbound channel prior to dialing.
REDIRECTING
------------------
* Redirecting reasons can now be set to arbitrary strings. This means
that the REDIRECTING dialplan function can be used to set the redirecting
reason to any string. It also allows for custom strings to be read as the
redirecting reason from SIP Diversion headers.
SPEECH_ENGINE
------------------
* The SPEECH_ENGINE function now supports read operations. When read from, it
will return the current value of the requested attribute.
VMCOUNT:
------------------
* Mailboxes defined by app_voicemail MUST be referenced by the rest of the
system as mailbox@context. The rest of the system cannot add @default
to mailbox identifiers for app_voicemail that do not specify a context
any longer. It is a mailbox identifier format that should only be
interpreted by app_voicemail.
Resources
------------------
res_agi (Asterisk Gateway Interface)
------------------
* The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
* The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
and AsyncAGIEnd.
* The CONTROL STREAM FILE command now accepts an offsetms parameter. This
will start the playback of the audio at the position specified. It will
also return the final position of the file in 'endpos'.
* The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
channel variable if the user stopped the file playback or if a remote
entity stopped the playback. If neither stopped the playback, it will
indicate the overall success/failure of the playback. If stopped early,
the final offset of the file will be set in the CPLAYBACKOFFSET channel
variable.
* The SAY ALPHA command now accepts an additional parameter to control
whether it specifies the case of uppercase, lowercase, or all letters to
provide functionality similar to SayAlphaCase.
res_ari (Asterisk RESTful Interface) (and others)
------------------
* The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
control telephony primitives in Asterisk by remote client. This includes
channels, bridges, endpoints, media, and other fundamental concepts. Users
of ARI can develop their own communications applications, controlling
multiple channels using an HTTP RESTful interface and receiving JSON events
about the objects via a WebSocket connection. ARI can be configured in
Asterisk via ari.conf. For more information on ARI, see
https://wiki.asterisk.org/wiki/x/0YCLAQ
res_parking
-------------------
* Parking has been extracted from the Asterisk core as a loadable module,
res_parking. Configuration for parking is now provided by res_parking.conf.
Configuration through features.conf is no longer supported.
* res_parking uses the configuration framework. If an invalid configuration is
supplied, res_parking will fail to load or fail to reload. Previously,
invalid configurations would generally be accepted, with certain errors
resulting in individually disabled parking lots.
* Parked calls are now placed in bridges. While this is largely an
architectural change, it does have implications on how channels in a parking
lot are viewed. For example, commands that display channels in bridges will
now also display the channels in a parking lot.
* The order of arguments for the new parking applications have been modified.
Timeout and return context/exten/priority are now implemented as options,
while the name of the parking lot is now the first parameter. See the
application documentation for Park, ParkedCall, and ParkAndAnnounce for more
in-depth information as well as syntax.
* Extensions are by default no longer automatically created in the dialplan to
park calls or pickup parked calls. Generation of dialplan extensions can be
enabled using the 'parkext' configuration option.
* ADSI functionality for parking is no longer supported. The 'adsipark'
configuration option has been removed as a result.
* The PARKINGSLOT channel variable has been deprecated in favor of
PARKING_SPACE to match the naming scheme of the new system.
* PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
channel even when the configuration option 'comebactoorigin' is enabled.
* A new CLI command 'parking show' has been added. This allows a user to
inspect the parking lots that are currently in use.
'parking show <parkinglot>' will also show the parked calls in a specific
parking lot.
* The CLI command 'parkedcalls' is now deprecated in favor of
'parking show <parkinglot>'.
* The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
can be used to get a list of parked calls for a specific parking lot.
* The AMI command 'Park' field 'Channel2' has been deprecated and replaced
with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
longer a required argument.
* The ParkAndAnnounce application is now provided through res_parking instead
of through the separate app_parkandannounce module.
* ParkAndAnnounce will no longer go to the next position in dialplan on timeout
by default. Instead, it will follow the timeout rules of the parking lot. The
old behavior can be reproduced by using the 'c' option.
* Dynamic parking lots will now fail to be created under the following
conditions:
- if the parking lot specified by PARKINGDYNAMIC does not exist
- if they require exclusive park and parkedcall extensions which overlap
with existing parking lots.
* Dynamic parking lots will be cleared on reload for dynamic parking lots that
currently contain no calls. Dynamic parking lots containing parked calls
will persist through the reloads without alteration.
* If 'parkext_exclusive' is set for a parking lot and that extension is
already in use when that parking lot tries to register it, this is now
considered a parking system configuration error. Configurations which do
this will be rejected.
* Added channel variable PARKER_FLAT. This contains the name of the extension
that would be used if 'comebacktoorigin' is enabled. This can be useful when
comebacktoorigin is disabled, but the dialplan or an external control
mechanism wants to use the extension in the park-dial context that was
generated to re-dial the parker on timeout.
res_pjsip (and many others)
------------------
* A large number of resource modules make up the SIP stack based on pjsip.
The chan_pjsip channel driver users these resource modules to provide
various SIP functionality in Asterisk. The majority of configuration for
these modules is performed in pjsip.conf. Other modules may use their
own configuration files.
* Added 'set_var' option for an endpoint. For each variable specified that
variable gets set upon creation of a channel involving the endpoint.
res_rtp_asterisk
------------------
* ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
them, an Asterisk-specific version of PJSIP needs to be installed.
Tarballs are available from https://github.com/asterisk/pjproject/tags/.
res_statsd/res_chan_stats
------------------
* A new resource module, res_statsd, has been added, which acts as a statsd
client. This module allows Asterisk to publish statistics to a statsd
server. In conjunction with res_chan_stats, it will publish statistics about
channels to the statsd server. It can be configured via res_statsd.conf.
res_xmpp
------------------
* Device state for XMPP buddies is now available using the following format:
XMPP/<client name>/<buddy address>
If any resource is available the device state is considered to be not in use.
If no resources exist or all are unavailable the device state is considered
to be unavailable.
Scripts
------------------
Realtime/Database Scripts
------------------
* Asterisk previously included example db schemas in the contrib/realtime/
directory of the source tree. This has been replaced by a set of database
migrations using the Alembic framework. This allows you to use alembic to
initialize the database for you. It will also serve as a database migration
tool when upgrading Asterisk in the future.
See contrib/ast-db-manage/README.md for more details.
sip_to_res_pjsip.py
-------------------
* A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
This python script will convert an existing sip.conf file to a
pjsip.conf file, for use with the chan_pjsip channel driver. This script
is meant to be an aid in converting an existing chan_sip configuration to
a chan_pjsip configuration, but it is expected that configuration beyond
what the script provides will be needed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
Build System
-------------------
* The Asterisk build system will now build and install a shared library
(libasteriskssl.so) used to wrap various initialization and shutdown functions
from the libssl and libcrypto libraries provided by OpenSSL. This is done so
that Asterisk can ensure that these functions do *not* get called by any
modules that are loaded into Asterisk, since they should only be called once
in any single process. If desired, this feature can be disabled by supplying
the "--disable-asteriskssl" option to the configure script.
* A new make target, 'full', has been added to the Makefile. This performs
the same compilation actions as make all, but will also scan the entirety of
each source file for documentation. This option is needed to generate AMI
event documentation. Note that your system must have Python in order for
this make target to succeed.
* The optimization portion of the build system has been reworked to avoid
broken builds on certain architectures. All architecture-specific
optimization has been removed in favor of using -march=native to allow gcc
to detect the environment in which it is running when possible. This can
be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
* BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
* Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
previously parsed the header file to obtain the version of Asterisk, you
will now have to go through Asterisk to get the version information.
Applications
-------------------
Bridge
-------------------
* Added 'F()' option. Similar to the dial option, this can be supplied with
arguments indicating where the callee should go after the caller is hung up,
or without options specified, the priority after the Queue will be used.
ConfBridge
-------------------
* Added menu action admin_toggle_mute_participants. This will mute / unmute
all non-admin participants on a conference. The confbridge configuration
file also allows for the default sounds played to all conference users when
this occurs to be overriden using sound_participants_unmuted and
sound_participants_muted.
* Added menu action participant_count. This will playback the number of
current participants in a conference.
* Added announcement configuration option to user profile. If set the sound
file will be played to the user, and only the user, upon joining the
conference bridge.
* Added record_file_append option that defaults to "yes", but if set to no
will create a new file between each start/stop recording.
Dial
-------------------
* Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
channels respectively before the callee channels are called.
ExternalIVR
-------------------
* Added support for IPv6.
* Add interrupt ('I') command to ExternalIVR. Sending this command from an
external process will cause the current playlist to be cleared, including
stopping any audio file that is currently playing. This is useful when you
want to interrupt audio playback only when specific DTMF is entered by the
caller.
FollowMe
-------------------
* A new option, 'I' has been added to app_followme. By setting this option,
Asterisk will not update the caller with connected line changes when they
occur. This is similar to app_dial and app_queue.
* The 'N' option is now ignored if the call is already answered.
* Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
and caller channels respectively before the callee channels are called.
* The winning FollowMe outgoing call is now put on hold if the caller put it on
hold.
MixMonitor
------------------
* MixMonitor hooks now have IDs associated with them which can be used to
assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
will allow storage of the MixMonitor ID in a channel variable. StopMixmonitor
now accepts that ID as an argument.
* Added 'm' option, which stores a copy of the recording as a voicemail in the
indicated mailboxes.
MySQL
-------------------
* The connect action in app_mysql now allows you to specify a port number to
connect to. This is useful if you run a MySQL server on a non-standard
port number.
OSP Applications
-------------------
* Increased the default number of allowed destinations from 5 to 12.
Page
-------------------
* The app_page application now no longer depends on DAHDI or app_meetme. It
has been re-architected to use app_confbridge internally.
Queue
-------------------
* Added queue options autopausebusy and autopauseunavail for automatically
pausing a queue member when their device reports busy or congestion.
* The 'ignorebusy' option for queue members has been deprecated in favor of
the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
per interface basis. Individual ringinuse values can now be set in
queues.conf via an argument to member definitions. Lastly, the queue
'ringinuse' setting now only determines defaults for the per member
'ringinuse' setting and does not override per member settings like it does
in earlier versions.
* Added 'F()' option. Similar to the dial option, this can be supplied with
arguments indicating where the callee should go after the caller is hung up,
or without options specified, the priority after the Queue will be used.
* Added new option log_member_name_as_agent, which will cause the membername to
be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
state_interface has been set.
* Add queue monitoring hints. exten => 8501,hint,Queue:markq.
* App_queue will now play periodic announcements for the caller that
holds the first position in the queue while waiting for answer.
SayUnixTime
------------------
* Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
when receiving DTMF. Use the 'j' option to enable extension jumping. Also
changed arguments to SayUnixTime so that every option is truly optional even
when using multiple options (so that j option could be used without having to
manually specify timezone and format) There are other benefits, e.g., format
can now be used without specifying time zone as well.
Voicemail
------------------
* Addition of the VM_INFO function - see Function changes.
* The imapserver, imapport, and imapflags configuration options can now be
overriden on a user by user basis.
* When voicemail plays a message's envelope with saycid set to yes, when
reaching the caller id field it will play a recording of a file with the same
base name as the sender's callerid if there is a similarly named file in
<astspooldir>/recordings/callerids/
* Voicemails now contains a unique message identifier "msg_id", which is stored
in the message envelope with the sound files. IMAP backends will now store
the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
backends will store the message identifier in a "msg_id" column. See
UPGRADE.txt for more information.
* Added VoiceMailPlayMsg application. This application will play a single
voicemail message from a mailbox. The result of the application, SUCCESS or
FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
Functions
------------------
* Hangup handlers can be attached to channels using the CHANNEL() function.
Hangup handlers will run when the channel is hung up similar to the h
extension. The hangup_handler_push option will push a GoSub compatible
location in the dialplan onto the channel's hangup handler stack. The
hangup_handler_pop option will remove the last added location, and optionally
replace it with a new GoSub compatible location. The hangup_handler_wipe
option will remove all locations on the stack, and optionally add a new
location.
* The expression parser now recognizes the ABS() absolute value function,
which will convert negative floating point values to positive values.
* FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
control of faxdetect.
* Addition of the VM_INFO function that can be used to retrieve voicemail
user information, such as the email address and full name.
The MAILBOX_EXISTS dialplan function has been deprecated in favour of
VM_INFO.
* The REDIRECTING function now supports the redirecting original party id
and reason.
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details.
* MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
instead of simply the uri. This is the format that MessageSend() can use
in the from parameter for outgoing SIP messages.
* Added the PRESENCE_STATE function. This allows retrieving presence state
information from any presence state provider. It also allows setting
presence state information from a CustomPresence presence state provider.
See AMI/CLI changes for related commands.
* Added the AMI_CLIENT function to make manager account attributes available
to the dialplan. It currently supports returning the current number of
active sessions for a given account.
* Added support for private party ID information to CALLERID, CONNECTEDLINE,
and the REDIRECTING functions.
Channel Drivers
------------------
chan_local
------------------
* Added a manager event "LocalBridge" for local channel call bridges between
the two pseudo-channels created.
chan_dahdi
------------------
* Added dialtone_detect option for analog ports to disconnect incoming
calls when dialtone is detected.
* Added option colp_send to send ISDN connected line information. Allowed
settings are block, to not send any connected line information; connect, to
send connected line information on initial connect; and update, to send
information on any update during a call. Default is update.
* Add options namedcallgroup and namedpickupgroup to support installations
where a higher number of groups (>64) is required.
* Added support to use private party ID information with PRI calls.
chan_motif
------------------
* A new channel driver named chan_motif has been added which provides support for
Google Talk and Jingle in a single channel driver. This new channel driver includes
support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
hold, unhold, and ringing notification. It is also compliant with the current Jingle
specification, current Google Jingle specification, and the original Google Talk
protocol.
chan_ooh323
------------------
* Added NAT support for RTP. Setting in config is 'nat', which can be set
globally and overriden on a peer by peer basis.
* Direct media functionality has been added. Options in config are:
directmedia (directrtp) and directrtpsetup (earlydirect)
* ChannelUpdate events now contain a CallRef header.
chan_sip
------------------
* Asterisk will no longer substitute CID number for CID name in the display
name field if CID number exists without a CID name. This change improves
compatibility with certain device features such as Avaya IP500's directory
lookup service.
* A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
created using that setting to not be removed during SIP reload.
* Added settings recordonfeature and recordofffeature. When receiving an INFO
request with a "Record:" header, this will turn the requested feature on/off.
Allowed values are 'automon', 'automixmon', and blank to disable. Note that
dynamic features must be enabled and configured properly on the requesting
channel for this to function properly.
* Add support to realtime for the 'callbackextension' option.
* When multiple peers exist with the same address, but differing
callbackextension options, incoming requests that are matched by address
will be matched to the peer with the matching callbackextension if it is
available.
* Two new NAT options, auto_force_rport and auto_comedia, have been added
which set the force_rport and comedia options automatically if Asterisk
detects that an incoming SIP request crossed a NAT after being sent by
the remote endpoint.
* The default global nat setting in sip.conf has been changed from force_rport
to auto_force_rport.
* NAT settings are now a combinable list of options. The equivalent of the
deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
* Adds an option send_diversion which can be disabled to prevent
diversion headers from automatically being added to INVITE requests.
* Add support for lightweight NAT keepalive. If enabled a blank packet will
be sent to the remote host at a given interval to keep the NAT mapping open.
This can be enabled using the keepalive configuration option.
* Add option 'tonezone' to specify country code for indications. This option
can be set both globally and overridden for specific peers.
* The SIP Security Events Framework now supports IPv6.
* Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
between multiple user agents. When set, for directmedia reinvites,
Asterisk will not send an immediate reinvite on an incoming call leg. This
option is useful when peered with another SIP user agent that is known to
send immediate direct media reinvites upon call establishment.
* Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
as the transport.
* Add options subminexpiry and submaxexpiry to set limits of subscription
timer independently from registration timer settings. The setting of the
registration timer limits still is done by options minexpiry, maxexpiry
and defaultexpiry. For backwards compatibility the setting of minexpiry
and maxexpiry also is used to configure the subscription timer limits if
subminexpiry and submaxexpiry are not set in sip.conf.
* Set registration timer limits to default values when reloading sip
configuration and values are not set by configuration.
* Add options namedcallgroup and namedpickupgroup to support installations
where a higher number of groups (>64) is required.
* When a MESSAGE request is received, the address the request was received from
is now saved in the SIP_RECVADDR variable.
* Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
the ANI2/OLI information is set on the channel, which can be retrieved using
the CALLERID function.
* Peers can now be configured to support negotiation of ICE candidates using
the setting icesupport. See res_rtp_asterisk changes for more information.
* Added support for format attribute negotiation. See the Codecs changes for
more information.
* Extra headers specified with SIPAddHeader are sent with the REFER message
when using Transfer application. See refer_addheaders in sip.conf.sample.
* Added support to use private party ID information with calls.
* Adds an option discard_remote_hold_retrieval that when set stops telling
the peer to start music on hold.
chan_skinny
------------------
* Added skinny version 17 protocol support.
chan_unistim
--------------------
* Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
* Modified option 'date_format' to allow options to display date in 31Jan and Jan31
formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
as per the UNISTIM protocol.
* Fixed issues with dialtone not matching indications.conf and mute stopping rx
as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
* Added ability to use multiple lines for a single phone. This allows multiple
calls to occur on a single phone, using callwaiting and switching between calls.
* Added option 'sharpdial' allowing end dialing by pressing # key
* Added option 'interdigit_timer' to control phone dial timeout
* Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
* Added global 'debug' option, that enables debug in channel driver
* Added ability to translate on-screen menu in multiple languages. Tested on
Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
menu of phone
* In addition to English added French and Russian languages for on-screen menus
* Reworked dialing number input: added dialing by timeout, immediate dial on
on dialplan compare, phone number length now not limited by screen size
* Added ability to pickup a call using features.conf defined value and
on-screen key
chan_mISDN:
------------------
* Add options namedcallgroup and namedpickupgroup to support installations
where a higher number of groups (>64) is required.
* Added support to use private party ID information with calls.
Core
------------------
* The minimum DTMF duration can now be configured in asterisk.conf
as "mindtmfduration". The default value is (as before) set to 80 ms.
(previously it was only available in source code)
* Named ACLs can now be specified in acl.conf and used in configurations that
use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
used to specify an ACL, a similar form of 'acl' will add a named ACL to the
working ACL. In addition, some CLI commands have been added to provide
show information and allow for module reloading - see CLI Changes.
* Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
items (separated by commas), and items in the rule can be negated by prefixing
them with '!'. This simplifies Asterisk Realtime configurations, since it is no
longer necessray to control the order that the 'permit' and 'deny' columns are
returned from queries.
* DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
be used within the dynamic weight attribute when specifying a mapping.
* CEL backends can now be configured to show "USER_DEFINED" in the EventName
header, instead of putting the user defined event name there. When enabled
the UserDefType header is added for user defined events. This feature is
enabled with the setting show_user_defined.
* Macro has been deprecated in favor of GoSub. For redirecting and connected
line purposes use the following variables instead of their macro equivalents:
REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
cc_callback_macro in channel configurations.
* Asterisk can now use a system-provided NetBSD editline library (libedit) if it
is available.
* Call files now support the "early_media" option to connect with an outgoing
extension when early media is received.
* Added support to use private party ID information with calls.
AGI
------------------
* A new channel variable, AGIEXITONHANGUP, has been added which allows
Asterisk to behave like it did in Asterisk 1.4 and earlier where the
AGI application would exit immediately after a channel hangup is detected.
* IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
are resolved and each address is attempted in turn until one succeeds or
all fail.
AMI (Asterisk Manager Interface)
------------------
* The originate action now has an option "EarlyMedia" that enables the
call to bridge when we get early media in the call. Previously,
early media was disregarded always when originating calls using AMI.
* Added setvar= option to manager accounts (much like sip.conf)
* Originate now generates an error response if the extension given is not found
in the dialplan
* MixMonitor will now show IDs associated with the mixmonitor upon creating
them if the i(variable) option is used. StopMixMonitor will accept
MixMonitorID as an option to close specific MixMonitors.
* The SIPshowpeer manager action response field "SIP-Forcerport" has been
updated to include information about peers configured with
nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
returned if auto_force_rport is not enabled.
* Added SIPpeerstatus manager command which will generate PeerStatus events
similar to the existing PeerStatus events found in chan_sip on demand.
* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
* Support for IPv6 addresses has been added.
* AMI Events can now be documented in the Asterisk source. Note that AMI event
documentation is only generated when Asterisk is compiled using 'make full'.
See the CLI section for commands to display AMI event information.
* The AMI Hangup event now includes the AccountCode header so you can easily
correlate with AMI Newchannel events.
* The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
the StateInterface of the queue member.
* Added AMI event SessionTimeout in the Call category that is issued when a
call is terminated due to either RTP stream inactivity or SIP session timer
expiration.
* CEL events can now contain a user defined header UserDefType. See core
changes for more information.
* OOH323 ChannelUpdate events now contain a CallRef header.
* Added PresenceState command. This command will report the presence state for
the given presence provider.
* Added Parkinglots command. This will list all parking lots as a series of
AMI Parkinglot events.
* Added MessageSend command. This behaves in the same manner as the
MessageSend application, and is a technolgoy agnostic mechanism to send out
of call text messages.
* Added "message" class authorization. This grants an account permission to
send out of call messages. Write-only.
CLI
-------------------
* The "dialplan add include" command has been modified to create context a context
if one does not already exist. For instance, "dialplan add include foo into bar"
will create context "bar" if it does not already exist.
* A "dialplan remove context" command has been added to remove a context from
the dialplan
* The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
filenames of all running mixmonitors on a channel.
* The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
numeric instead of 0, 1, or 2.
* "stun show status" will show a table describing how the STUN client is
behaving.
* "acl show [named acl]" will show information regarding a Named ACL. The
acl module can be reloaded with "reload acl".
* Added CLI command to display AMI event information - "manager show events",
which shows a list of all known and documented AMI events, and "manager show
event [event name]", which shows detail information about a specific AMI
event.
* The result of the CLI command "queue show" now includes the state interface
information of the queue member.
* The command "core set verbose" will now set a separate level of logging for
each remote console without affecting any other console.
* Added command "cdr show pgsql status" to check connection status
* "sip show channel" will now display the complete route set.
* Added "presencestate list" command. This command will list all custom
presence states that have been set by using the PRESENCE_STATE dialplan
function.
* Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
command. This changes a custom presence to a new state.
Codecs
-------------------
* Codec lists may now be modified by the '!' character, to allow succinct
specification of a list of codecs allowed and disallowed, without the
requirement to use two different keywords. For example, to specify all
codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
* Add support for parsing SDP attributes, generating SDP attributes, and
passing it through. This support includes codecs such as H.263, H.264, SILK,
and CELT. You are able to set up a call and have attribute information pass.
This should help considerably with video calls.
* The iLBC codec can now use a system-provided iLBC library if one is installed,
just like the GSM codec.
DUNDi changes
-------------
* Added CLI commands dundi show hints and dundi show cache which will list DUNDi
'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
Logging
-------------------
* Asterisk version and build information is now logged at the beginning of a
log file.
* Threads belonging to a particular call are now linked with callids which get
added to any log messages produced by those threads. Log messages can now be
easily identified as involved with a certain call by looking at their call id.
Call ids may also be attached to log messages for just about any case where
it can be determined to be related to a particular call.
* Each logging destination and console now have an independent notion of the
current verbosity level. Logger.conf now allows an optional argument to
the 'verbose' specifier, indicating the level of verbosity sent to that
particular logging destination. Additionally, remote consoles now each
have their own verbosity level. The command 'core set verbose' will now set
a separate level for each remote console without affecting any other
console.
Music On Hold
-------------------
* Added 'announcement' option which will play at the start of MOH and between
songs in modes of MOH that can detect transitions between songs (eg.
files, mp3, etc).
Parking
-------------------
* New per parking lot options: comebackcontext and comebackdialtime. See
configs/features.conf.sample for more details.
* Channel variable PARKER is now set when comebacktoorigin is disabled in
a parking lot.
* Channel variable PARKEDCALL is now set with the name of the parking lot
when a timeout occurs.
CDRs
-------------------
CDR Postgresql Driver
-------------------
* Added command "cdr show pgsql status" to check connection status
CDR Adaptive ODBC Driver
-------------------
* Added schema option for databases that support specifying a schema.
Resource Modules
-------------------
Calendars
-------------------
* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
CALENDAR_WRITE has completed successfully.
res_rtp_asterisk
-------------------
* A new option, 'probation' has been added to rtp.conf
RTP in strictrtp mode can now require more than 1 packet to exit learning
mode with a new source (and by default requires 4). The probation option
allows the user to change the required number of packets in sequence to any
desired value. Use a value of 1 to essentially restore the old behavior.
Also, with strictrtp on, Asterisk will now drop all packets until learning
mode has successfully exited. These changes are based on how pjmedia handles
media sources and source changes.
* Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
enabled or disabled using the icesupport setting. A variety of other
settings have been introduced to configure STUN/TURN connections.
res_corosync
-------------------
* A new module, res_corosync, has been introduced. This module uses the
Corosync cluster engineer (http://www.corosync.org) to allow a local cluster
of Asterisk servers to both Message Waiting Indication (MWI) and/or
Device State (presence) information. This module is very similar to, and
is a replacement for the res_ais module that was in previous releases of
Asterisk.
res_xmpp
-------------------
* This module adds a cleaned up, drop-in replacement for res_jabber called
res_xmpp. This provides the same externally facing functionality but is
implemented differently internally. res_jabber has been deprecated in favor
of res_xmpp; please see the UPGRADE.txt file for more information.
Scripts
-------------------
* The safe_asterisk script has been updated to allow several of its parameters
to be set from environment variables. This also enables a custom run
directory of Asterisk to be specified, instead of defaulting to /tmp.
* The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
its value to determine the directory to assume is the top-level directory of
the source tree. If the variable is not set, it defaults to the current
behavior and uses the current working directory.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
------------------------------------------------------------------------------
Text Messaging
--------------
* Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in
jabber.conf and sip.conf to allow enabling these features.
-> jabber.conf: see the "sendtodialplan" and "context" options.
-> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
and "outofcall_message_context" options.
The MESSAGE() dialplan function and MessageSend() application have been
added to go along with this functionality. More detailed usage information
can be found on the Asterisk wiki (http://wiki.asterisk.org/).
* If real-time text support (T.140) is negotiated, it will be preferred for
sending text via the SendText application. For example, via SIP, messages
that were once sent via the SIP MESSAGE request would be sent via RTP if
T.140 text is negotiated for a call.
Parking
-------
* parkedmusicclass can now be set for non-default parking lots.
Asterisk Manager Interface
--------------------------
* PeerStatus now includes Address and Port.
* Added Hold events for when the remote party puts the call on and off hold
for chan_dahdi ISDN channels.
* Added new action MeetmeListRooms to list active conferences (shows same
data as "meetme list" at the CLI).
* DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
Description field that is set by 'description' in the channel configuration
file.
* Added Uniqueid header to UserEvent.
* Added new action FilterAdd to control event filters for the current session.
This requires the system permission and uses the same filter syntax as
filters that can be defined in manager.conf
* The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
versions had some instances of the event converted, but others were left
as-is. All Unlink events should now be converted to Bridge events. The AMI
protocol version number was incremented to 1.2 as a result of this change.
Asterisk HTTP Server
--------------------------
* The HTTP Server can bind to IPv6 addresses.
chan_dahdi
--------------------------
* Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
with busydetect. usage example: busypattern=200,200,200,600
CLI Changes
--------------------------
* New 'gtalk show settings' command showing the current settings loaded from
gtalk.conf.
* The 'logger reload' command now supports an optional argument, specifying an
alternate configuration file to use.
* 'dialplan add extension' command will now automatically create a context if
the specified context does not exist with a message indicated it did so.
* 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
Description field which can be populated with 'description' in the channel
configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
CDR
--------------------------
* The filter option in cdr_adaptive_odbc now supports negating the argument,
thus allowing records which do NOT match the specified filter.
* Added ability to log CONGESTION calls to CDR
CODECS
--------------------------
* Ability to define custom SILK formats in codecs.conf.
* Addition of speex32 audio format with translation.
* CELT codec pass-through support and ability to define
custom CELT formats in codecs.conf.
* Ability to read raw signed linear files with sample rates
ranging from 8khz - 192khz. The new file extensions introduced
are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
* Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
Skinny, H.323, etc) can still only support the following codecs:
Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
Video: h261, h263, h263p, h264, mpeg4
Image: jpeg, png
Text: red, t140
ConfBridge
--------------------------
* New highly optimized and customizable ConfBridge application capable of
mixing audio at sample rates ranging from 8khz-96khz.
* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
and bridge profiles on a channel.
* CONFBRIDGE_INFO dialplan function capable of retrieving information
about a conference such as locked status and number of parties, admins,
and marked users.
* Addition of video_mode option in confbridge.conf for adding video support
into a bridge profile.
* Addition of the follow_talker video_mode in confbridge.conf. This video
mode dynamically switches the video feed to always display the loudest talker
supplying video in the conference.
Dialplan Variables
------------------
* Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
variables from asterisk.conf.
Dialplan Functions
------------------
* Addition of the JITTERBUFFER dialplan function. This function allows
for jitterbuffering to occur on the read side of a channel. By using
this function conference applications such as ConfBridge and MeetMe can
have the rx streams jitterbuffered before conference mixing occurs.
* Added DB_KEYS, which lists the next set of keys in the Asterisk database
hierarchy.
* Added STRREPLACE function. This function let's the user search a variable
for a given string to replace with another string as many times as the
user specifies or just throughout the whole string.
* Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
* Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
* Added extensions to chan_ooh323 in function CHANNEL()
libpri channel driver (chan_dahdi) DAHDI changes
--------------------------
* Added moh_signaling option to specify what to do when the channel's bridged
peer puts the ISDN channel on hold.
* Added display_send and display_receive options to control how the display ie
is handled. To send display text from the dialplan use the SendText()
application when the option is enabled.
* Added mcid_send option to allow sending a MCID request on a span.
Calendaring
--------------------------
* Added setvar option to calendar.conf to allow setting channel variables on
notification channels.
* Added "calendar show types" CLI command to list registered calendar
connectors.
MixMonitor
--------------------------
* Added two new options, r and t with file name arguments to record
single direction (unmixed) audio recording separate from the bidirectional
(mixed) recording. The mixed file name argument is optional now as long
as at least one recording option is used.
FollowMe
--------------------------
* Added a new option, l, which will disable local call optimization for
channels involved with the FollowMe thread. Use this option to improve
compatability for a FollowMe call with certain dialplan apps, options, and
functions.
Meetme
--------------------------
* Added option "k" that will automatically close the conference when there's
only one person left when a user exits the conference.
CEL
--------------------------
* cel_pgsql now supports the 'extra' column for data added using the
CELGenUserEvent() application.
pbx_lua
--------------------------
* Support for defining hints has been added to pbx_lua. See the 'hints' table
in the sample extensions.lua file for syntax details.
* Applications that perform jumps in the dialplan such as Goto will now
execute properly. When pbx_lua detects that the context, extension, or
priority we are executing on has changed it will immediately return control
to the asterisk PBX engine. Currently the engine cannot detect a Goto to
the priority after the currently executing priority.
* An autoservice is now started by default for pbx_lua channels. It can be
stopped and restarted using the autoservice_stop() and autoservice_start()
functions.
res_fax
--------------------------
* The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
into a FAXStatus event with an 'Operation' header that will be either
'send', 'receive', and 'gateway'.
* T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
feature will handle converting a fax call between an audio T.30 fax terminal
and an IFP T.38 fax terminal.
SIP Changes
-----------
* Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
* Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
* SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
Queue changes
-------------
* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
Applications
------------
* Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
a MeetMe conference
* Added 'k' option to MeetMe to automatically kill the conference when there's only
one participant left (much like a normal call bridge)
* Added extra argument to Originate to set timeout.
Asterisk Database
-----------------
* The internal Asterisk database has been switched from Berkeley DB 1.86 to
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
utility in the UTILS section of menuselect. If an existing astdb is found and no
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
convert an existing astdb to the SQLite3 version automatically at runtime.
Asterisk Modules
----------------
* Modules marked as deprecated are no longer marked as building by default. Enabling
these modules is still available via menuselect.
IAX2 Changes
------------
* authdebug is now disabled by default. To enable this functionality again
set authdebug = yes in iax.conf.
RTP Changes
-----------
* The rtp.conf setting "strictrtp" is now enabled by default. In previous
releases it was disabled.
PBX Core
--------
* The PBX core previously made a call with a non-existing extension test for
extension s@default and jump there if the extension existed.
This was a bad default behaviour and violated the principle of least surprise.
It has therefore been changed in this release. It may affect some
applications and configurations that rely on this behaviour. Most channel
drivers have avoided this for many releases by testing whether the extension
called exists before starting the PBX and generating a local error.
This behaviour still exists and works as before.
Extension "s" is used when no extension is given in a channel driver,
like immediate answer in DAHDI or calling to a domain with no user part
in a SIP uri.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
------------------------------------------------------------------------------
SIP Changes
-----------
* Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
now defaults to force_rport. It is very important that phones requiring nat=no be
specifically set as such instead of relying on the default setting. If at all
possible, all devices should have nat settings configured in the general section as
opposed to configuring nat per-device.
* Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
* Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
to be used for the outgoing call. It must be one of the codecs configured
for the device.
* Added tlsprivatekey option to sip.conf. This allows a separate .pem file
to be used for holding a private key. If tlsprivatekey is not specified,
tlscertfile is searched for both public and private key.
* Added tlsclientmethod option to sip.conf. This allows the protocol for
outbound client connections to be specified.
* The sendrpid parameter has been expanded to include the options
'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
header to be sent (equivalent to setting sendrpid=yes) and setting
sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
* The 'ignoresdpversion' behavior has been made automatic when the SDP received
is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
since the call will fail if Asterisk does not process the incoming SDP, Asterisk
will accept the SDP even if the SDP version number is not properly incremented,
but will generate a warning in the log indicating that the SIP peer that sent
the SDP should have the 'ignoresdpversion' option set.
* The 'nat' option has now been been changed to have yes, no, force_rport, and
comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
remote side requests it and disables symmetric RTP support. Setting it to
force_rport forces RFC 3581 behavior and disables symmetric RTP support.
Setting it to comedia enables RFC 3581 behavior if the remote side requests it
and enables symmetric RTP support.
* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
response. This permits the master channel to know how each channel dialled
in a multi-channel setup resolved in an individual way. This carries a
performance penalty and can be disabled in sip.conf using the
'storesipcause' option.
* Added 'externtcpport' and 'externtlsport' options to allow custom port
configuration for the externip and externhost options when tcp or tls is used.
* Added support for message body (stored in content variable) to SIP NOTIFY message
accessible via AMI and CLI.
* Added 'media_address' configuration option which can be used to explicitly specify
the IP address to use in the SDP for media (audio, video, and text) streams.
* Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
received.
* Added 'use_q850_reason' configuration option for generating and parsing
if available Reason: Q.850;cause=<cause code> header. It is implemented
in some gateways for better passing PRI/SS7 cause codes via SIP.
* When dialing SIP peers, a new component may be added to the end of the dialstring
to indicate that a specific remote IP address or host should be used when dialing
the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
* SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
ability to selectively force bridged channels to also be encrypted is also
implemented. Branching in the dialplan can be done based on whether or not
a channel has secure media and/or signaling.
* Added directmediapermit/directmediadeny to limit which peers can send direct media
to each other
* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
Charge messages to snom phones.
* Added support for G.719 media streams.
* Added support for 16khz signed linear media streams.
* SIP is now able to bind to and communicate with IPv6 addresses. In addition,
RTP has been outfitted with the same abilities.
* Added support for setting the Max-Forwards: header in SIP requests. Setting is
available in device configurations as well as in the dial plan.
* Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_sip.
* Addition of the 'auth_options_requests' option for turning on and off
authentication for OPTIONS requests in chan_sip.
Configuration files
-------------------
* Add #tryinclude statement for config files. This provides the same
functionality as the #include statement however an asterisk module will
still load if the filename does not exist. Using the #include statement
Asterisk will not allow the module to load.
IAX2 Changes
-----------
* Added rtsavesysname option into iax.conf to allow the systname to be saved
on realtime updates.
* Added the ability for chan_iax2 to inform the dialplan whether or not
encryption is being used. This interoperates with the SIP SRTP implementation
so that a secure SIP call can be bridged to a secure IAX call when the
dialplan requires bridged channels to be "secure".
* Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_iax.
MGCP Changes
------------
* Added ability to preset channel variables on indicated lines with the setvar
configuration option. Also, clearvars=all resets the list of variables back
to none.
* PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
See configs/res_pktccops.conf for more information.
XMPP Google Talk/Jingle changes
-------------------------------
* Added the externip option to gtalk.conf.
* Added the stunaddr option to gtalk.conf which allows for the automatic
retrieval of the external ip from a stun server.
Applications
------------
* Added 'p' option to PickupChan() to allow for picking up channel by the first
match to a partial channel name.
* Added .m3u support for Mp3Player application.
* Added progress option to the app_dial D() option. When progress DTMF is
present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
* Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
* Added the 'a' option to app_dial to answer the calling channel before any
announcements or macros are executed.
* Modified app_dial to set answertime when the called channel answers even if
the called channel hangs up during playback of an announcement.
* Modified app_dial 'r' option to support an additional parameter to play an
indication tone from indications.conf
* Added c() option to app_chanspy. This option allows custom DTMF to be set
to cycle through the next available channel. By default this is still '*'.
* Added x() option to app_chanspy. This option allows DTMF to be set to
exit the application.
* The Voicemail application has been improved to automatically ignore messages
that only contain silence.
* If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
associated mailbox(es) to be greetings-only.
* The ChanSpy application now has the 'S' option, which makes the application
automatically exit once it hits a point where no more channels are available
to spy on.
* The ChanSpy application also now has the 'E' option, which spies on a single
channel and exits when that channel hangs up.
* The MeetMe application now turns on the DENOISE() function by default, for
each participant. In our tests, this has significantly decreased background
noise (especially noisy data centers).
* Voicemail now permits storage of secrets in a separate file, located in the
spool directory of each individual user. The control for this is located in
the "passwordlocation" option in voicemail.conf. Please see the sample
configuration for more information.
* The ChanIsAvail application now exposes the returned cause code using a separate
variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
* Added 'd' option to app_followme. This option disables the "Please hold"
announcement.
* Added 'y' option to app_record. This option enables a mode where any DTMF digit
received will terminate recording.
* Voicemail now supports per mailbox settings for folders when using IMAP storage.
Previously the folder could only be set per context, but has now been extended
using the imapfolder option.
* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
* Voicemail now allows the pager date format to be specified separately from the
email date format.
* New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
to allow joining, leaving, and sending text to group chats.
* MeetMe has a new option 'G' to play an announcement before joining a conference.
* Page has a new option 'A(x)' which will playback an announcement simultaneously
to all paged phones (and optionally excluding the caller's one using the new
option 'n') before the call is bridged.
* The 'f' option to Dial has been augmented to take an optional argument. If no
argument is provided, the 'f' option works as it always has. If an argument is
provided, then the connected party information of all outgoing channels created
during the Dial will be set to the argument passed to the 'f' option.
* Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
Gosub on the peer.
* The OSP lookup application adds in/outbound network ID, optional security,
number portability, QoS reporting, destination IP port, custom info and service
type features.
* Added new application VMSayName that will play the recorded name of the voicemail
user if it exists, otherwise will play the mailbox number.
* Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
retrieve state for a particular bridge, where <name> is the conference name
* app_directory now allows exiting at any time using the operator or pound key.
* Voicemail now supports setting a locale per-mailbox.
* Two new applications are provided for declining counting phrases in multiple
languages. See the application notes for SayCountedNoun and SayCountedAdj for
more information.
* Voicemail now runs the externnotify script when pollmailboxes is activated and
notices a change.
* Voicemail now includes rdnis within msgXXXX.txt file.
* ExternalIVR now supports IPv6 addresses.
* Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
at https://wiki.asterisk.org/wiki/x/oQBB
* ParkedCall and Park can now specify the parking lot to use.
Dialplan Functions
------------------
* SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
over SRV records associated with a specific service. From the CLI, type
'core show function SRVQUERY' and 'core show function SRVRESULT' for more
details on how these may be used.
* PITCH_SHIFT dialplan function added. This function can be used to modify the
pitch of a channel's tx and rx audio streams.
* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
setting various connected line and redirecting party information.
* CALLERID and CONNECTEDLINE dialplan functions have been extended to
support ISDN subaddressing.
* The CHANNEL() function now supports the "name" and "checkhangup" options.
* For DAHDI channels, the CHANNEL() dialplan function now allows
the dialplan to request changes in the configuration of the active
echo canceller on the channel (if any), for the current call only.
The syntax is:
exten => s,n,Set(CHANNEL(echocan_mode)=off)
The possible values are:
on - normal mode (the echo canceller is actually reinitialized)
off - disabled
fax - FAX/data mode (NLP disabled if possible, otherwise completely
disabled)
voice - voice mode (returns from FAX mode, reverting the changes that
were made when FAX mode was requested)
* Added new dialplan function MASTER_CHANNEL(), which permits retrieving
and setting variables on the channel which created the current channel.
Administrators should take care to avoid naming conflicts, when multiple
channels are dialled at once, especially when used with the Local channel
construct (which all could set variables on the master channel). Usage
of the HASH() dialplan function, with the key set to the name of the slave
channel, is one approach that will avoid conflicts.
* Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
audio in a channel.
* func_odbc now allows multiple row results to be retrieved without using
mode=multirow. If rowlimit is set, then additional rows may be retrieved
from the same query by using the name of the function which retrieved the
first row as an argument to ODBC_FETCH().
* Added JABBER_RECEIVE, which permits receiving XMPP messages from the
dialplan. This function returns the content of the received message.
* Added REPLACE, which searches a given variable name for a set of characters,
then either replaces them with a single character or deletes them.
* Added PASSTHRU, which literally passes the same argument back as its return
value. The intent is to be able to use a literal string argument to
functions that currently require a variable name as an argument.
* HASH-associated variables now can be inherited across channel creation, by
prefixing the name of the hash at assignment with the appropriate number of
underscores, just like variables.
* GROUP_MATCH_COUNT has been improved to allow regex matching on category
* CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
whether or not channels that are bridged to the current channel will be
required to have secure signaling and/or media.
* CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
the current channel has secure signaling and/or media.
* For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
Returns "0" if there is a B channel associated with the call.
Returns "1" if no B channel is associated with the call. The call is either
on hold or is a call waiting call.
* Added option to dialplan function CDR(), the 'f' option
allows for high resolution times for billsec and duration fields.
* FILE() now supports line-mode and writing.
* Added FIELDNUM(), which returns the 1-based offset of a field in a list.
* FRAME_TRACE(), for tracking internal ast_frames on a channel.
Dialplan Variables
------------------
* Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
* Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
and is set when a dynamic feature is triggered.
* Added PARKINGLOT which can be used with parkeddynamic feature.conf option
to dynamically create a new parking lot matching the value this varible is
set to.
* Added PARKINGDYNAMIC which represents the template parkinglot defined in
features.conf that should be the base for dynamic parkinglots.
* Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
parkinglot should have.
* Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
parkinglot should have.
* Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
should have.
Queue changes
-------------
* Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
timeout has expired.
* Added 'R' option to app_queue. This option stops moh and indicates ringing
to the caller when an Agent's phone is ringing. This can be used to indicate
to the caller that their call is about to be picked up, which is nice when
one has been on hold for an extened period of time.
* A new config option, penaltymemberslimit, has been added to queues.conf.
When set this option will disregard penalty settings when a queue has too
few members.
* A new option, 'I' has been added to both app_queue and app_dial.
By setting this option, Asterisk will not update the caller with
connected line changes or redirecting party changes when they occur.
* A 'relative-periodic-announce' option has been added to queues.conf. When
enabled, this option will cause periodic announce times to be calculated
from the end of announcements rather than from the beginning.
* The autopause option in queues.conf can be passed a new value, "all." The
result is that if a member becomes auto-paused, he will be paused in all
queues for which he is a member, not just the queue that failed to reach
the member.
* Added dialplan function QUEUE_EXISTS to check if a queue exists
* The queue logger now allows events to optionally propagate to a file,
even when realtime logging is turned on. Additionally, realtime logging
supports sending the event arguments to 5 individual fields, although it
will fallback to the previous data definition, if the new table layout is
not found.
mISDN channel driver (chan_misdn) changes
----------------------------------------
* Added display_connected parameter to misdn.conf to put a display string
in the CONNECT message containing the connected name and/or number if
the presentation setting permits it.
* Added display_setup parameter to misdn.conf to put a display string
in the SETUP message containing the caller name and/or number if the
presentation setting permits it.
* Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
indicate the dialplan settings are to be obtained from the asterisk
channel.
* Made misdn.conf parameter callerid accept the "name" <number> format
used by the rest of the system.
* Made use the nationalprefix and internationalprefix misdn.conf
parameters to prefix any received number from the ISDN link if that
number has the corresponding Type-Of-Number. NOTE: This includes
comparing the incoming call's dialed number against the MSN list.
* Added the following new parameters: unknownprefix, netspecificprefix,
subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
received number from the ISDN link if that number has the corresponding
Type-Of-Number.
* Added new dialplan application misdn_command which permits controlling
the CCBS/CCNR functionality.
* Added new dialplan function mISDN_CC which permits retrieval of various
values from an active call completion record.
* For PTP, you should manually send the COLR of the redirected-to party
for an incomming redirected call if the incoming call could experience
further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
set the REDIRECTING(to-pres) to the COLR. A call has been redirected
if the REDIRECTING(from-num) is not empty.
* For outgoing PTP redirected calls, you now need to use the inhibit(i)
option on all of the REDIRECTING statements before dialing the
redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
and the REDIRECTING(from-xxx,i) values. The PTP call will update the
redirecting-to presentation (COLR) when it becomes available.
* Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
information.
thirdparty mISDN enhancements
-----------------------------
mISDN has been modified by Digium, Inc. to greatly expand facility message
support to allow:
* Enhanced COLP support for call diversion and transfer.
* CCBS/CCNR support.
The latest modified mISDN v1.1.x based version is available at:
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
Tagged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
libpri channel driver (chan_dahdi) DAHDI changes
-------------------------------------------
* The channel variable PRIREDIRECTREASON is now just a status variable
and it is also deprecated. Use the REDIRECTING(reason) dialplan function
to read and alter the reason.
* For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
redirected-to party for an incomming redirected call if the incoming call
could experience further redirects. Just set the
REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
to the COLR. A call has been redirected if the REDIRECTING(count) is not
zero.
* For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
use the inhibit(i) option on all of the REDIRECTING statements before
dialing the redirected-to party. You still have to set the
REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
will update the redirecting-to presentation (COLR) when it becomes available.
* Added the ability to ignore calls that are not in a Multiple Subscriber
Number (MSN) list for PTMP CPE interfaces.
* Added dynamic range compression support for dahdi channels. It is
configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
* Added support for ISDN calling and called subaddress with partial support
for connected line subaddress.
* Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added standard location to add options to chan_dahdi dialing:
Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication
* Added Reverse Charging Indication (Collect calls) send/receive option.
Send reverse charging in SETUP message with the chan_dahdi R dialing option.
Dial(DAHDI/g1/extension/R)
Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
(requires latest LibPRI)
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
(requires latest LibPRI)
* Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
to eliminate tromboned calls. A tromboned call goes out an interface and comes
back into the same interface. Tromboned calls happen because of call routing,
call deflection, call forwarding, and call transfer.
* Added the ability to send and receive ETSI Advice-Of-Charge messages.
* Added the ability to support call waiting calls. (The SETUP has no B channel
assigned.)
* Added Malicious Call ID (MCID) event to the AMI call event class.
* Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
Asterisk Manager Interface
--------------------------
* The Hangup action now accepts a Cause header which may be used to
set the channel's hangup cause.
* sslprivatekey option added to manager.conf and http.conf. Adds the ability
to specify a separate .pem file to hold a private key. By default sslcert
is used to hold both the public and private key.
* Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
for options containing the 'tls' prefix. For example, 'sslenable' is now
'tlsenable'. This has been done in effort to keep ssl and tls options consistent
across all .conf files. All affected sample.conf files have been modified to
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
* Added a MuteAudio AMI action for muting inbound and/or outbound audio
in a channel. (res_mutestream.so)
* The configuration file manager.conf now supports a channelvars option, which
specifies a list of channel variables to include in each channel-oriented
event.
* The redirect command now has new parameters ExtraContext, ExtraExtension,
and ExtraPriority to allow redirecting the second channel to a different
location than the first.
* Added new event "JabberStatus" in the Jabber module to monitor buddies
status.
* Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
in a MixMonitor recording.
* The 'iax2 show peers' output is now similar to the expected output of
'sip show peers'.
* Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
aoc event class.
* Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
AOC-E messages on a channel.
* A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
conform more closely to similar events.
* Added a new eventfilter option per user to allow whitelisting and blacklisting
of events.
* Added optional parkinglot variable for park command.
* Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
if CallerIDNum and CallerIDName headers are also present.
Channel Event Logging
---------------------
* A new interface, CEL, is introduced here. CEL logs single events, much like
the AMI, but it differs from the AMI in that it logs to db backends much
like CDR does; is based on the event subsystem introduced by Russell, and
can share in all its benefits; allows multiple backends to operate like CDR;
is specialized to event data that would be of concern to billing systems,
like CDR. Backends for logging and accounting calls have been produced,
but a new CDR backend is still in development.
CDR
---
* 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
* Multiple files and formats can now be specified in cdr_custom.conf.
* cdr_syslog has been added which allows CDRs to be written directly to syslog.
See configs/cdr_syslog.conf.sample for more information.
* A 'sequence' field has been added to CDRs which can be combined with
linkedid or uniqueid to uniquely identify a CDR.
* Handling of billsec and duration field has changed. If your table definition
specifies those fields as float,double or similar they will now be logged with
microsecond accuracy instead of a whole integer.
Calendaring for Asterisk
------------------------
* A new set of modules were added supporting calendar integration with Asterisk.
Dialplan functions for reading from and writing to calendars are included,
as well as the ability to execute dialplan logic upon calendar event notifications.
iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2003 support does not support forms-based authentication).
Call Completion Supplementary Services for Asterisk
---------------------------------------------------
* Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
DAHDI/ISDN supports call completion for the following switch types:
EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
See https://wiki.asterisk.org/wiki/x/2ABQ for details.
Multicast RTP Support
---------------------
* A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP
paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
Type can be either basic or linksys.
Destination is the IP address and port for the RTP packets.
Control address is specific to the linksys type and is used for sending the control
packets unique to them.
Security Events Framework
-------------------------
* Asterisk has a new C API for reporting security events. The module res_security_log
sends these events to the "security" logger level. Currently, AMI is the only
Asterisk component that reports security events. However, SIP support will be
coming soon. For more information on the security events framework, see the
"Asterisk Security Framework" section of the Asterisk wiki at
https://wiki.asterisk.org/wiki/x/wgBQ
* SIP support was added in Asterisk 10
* This API now supports IPv6 addresses
Fax
---
* A technology independent fax frontend (res_fax) has been added to Asterisk.
* A spandsp based fax backend (res_fax_spandsp) has been added.
* The app_fax module has been deprecated in favor of the res_fax module and
the new res_fax_spandsp backend.
* The SendFAX and ReceiveFAX applications now send their log messages to a
'fax' logger level, instead of to the generic logger levels. To see these
messages, the system's logger.conf file will need to direct the 'fax' logger
level to one or more destinations; the logger.conf.sample file includes an
example of how to do this. Note that if the 'fax' logger level is *not*
directed to at least one destination, log messages generated by these
applications will be lost, and that if the 'fax' logger level is directed to
the console, the 'core set verbose' and 'core set debug' CLI commands will
have no effect on whether the messages appear on the console or not.
Miscellaneous
-------------
* The transmit_silence_during_record option in asterisk.conf.sample has been removed.
Now, in order to enable transmitting silence during record the transmit_silence
option should be used. transmit_silence_during_record remains a valid option, but
defaults to the behavior of the transmit_silence option.
* Addition of the Unit Test Framework API for managing registration and execution
of unit tests with the purpose of verifying the operation of C functions.
* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
XMPP text messages to the remote JID.
* Modules.conf has a new option - "require" - that marks a module as critical for
the execution of Asterisk.
If one of the required modules fail to load, Asterisk will exit with a return
code set to 2.
* An 'X' option has been added to the asterisk application which enables #exec support.
This allows #exec to be used in asterisk.conf.
* jabber.conf supports a new option auth_policy that toggles auto user registration.
* A new lockconfdir option has been added to asterisk.conf to protect the
configuration directory (/etc/asterisk by default) during reloads.
* The parkeddynamic option has been added to features.conf to enable the creation
of dynamic parkinglots.
* chan_dahdi now supports reporting alarms over AMI either by channel or span via
the reportalarms config option.
* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* The Realtime dialplan switch now caches entries for 1 second. This provides a
significant increase in performance (about 3X) for installations using this switchtype.
* Distributed devicestate now supports the use of the XMPP protocol, in addition to
AIS. For more information, please see the Distributed Device State section of the
Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
* The addition of G.719 pass-through support.
* Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
during device configuration.
* The UNISTIM channel driver (chan_unistim) has been updated to support devices that
have less than 3 lines on the LCD.
* Realtime now supports database failover. See the sample extconfig.conf for details.
* The addition of improved translation path building for wideband codecs. Sample
rate changes during translation are now avoided unless absolutely necessary.
* The addition of the res_stun_monitor module for monitoring and reacting to network
changes while behind a NAT.
* DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
These allow support for any Administration. Default is AT&T values.
CLI Changes
-----------
* The 'core set debug' and 'core set verbose' commands, in previous versions, could
optionally accept a filename, to apply the setting only to the code generated from
that source file when Asterisk was built. However, there are some modules in Asterisk
that are composed of multiple source files, so this did not result in the behavior
that users expected. In this version, 'core set debug' and 'core set verbose'
can optionally accept *module* names instead (with or without the .so extension),
which applies the setting to the entire module specified, regardless of which source
files it was built from.
* New 'manager show settings' command showing the current settings loaded from
manager.conf.
* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
the channel hangup request to all channels.
* Added a "core reload" CLI command that executes a global reload of Asterisk.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
------------------------------------------------------------------------------
SIP Changes
-----------
* Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
Snom phones use this for call pickup of extensions that the phone is
subscribed to.
* Added support for setting the domain in the URI for caller of an
outbound call by using the SIPFROMDOMAIN channel variable.
* Added a new configuration option "remotesecret" for authentication to
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret and a
local secret for mutual authentication.
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
finer control over how many peers Asterisk will qualify and the gap between them
when all peers need to be qualified at the same time.
* Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
(either globally or for a specific peer), chan_sip will treat any SDP data
it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session
version received is different from the current SDP session version. This
option is required to interoperate with devices that have non-standard SDP
session version implementations (observed with Microsoft OCS). This option
is disabled by default.
* The parsing of register => lines in sip.conf has been modified to allow a port
to be present in the "user" portion. Please see the sip.conf.sample file for more
information
* Added support for subscribing to MWI on a remote server and making the status available
as a mailbox. Please see the sip.conf.sample file for more information.
* Added a function to remove SIP headers added in the dialplan before the
first INVITE is generated - SIPRemoveHeader()
* Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
IAX2 changes
------------
* Added immediate option to iax.conf
* Added forceencryption option to iax.conf
* Added Encryption and Trunk status to manager command "iaxpeers"
Skinny Changes
--------------
* The configuration file now holds separate sections for devices and lines.
Please have a look at configs/skinny.conf.sample and change your skinny.conf
accordingly.
DAHDI Changes
-------------
* chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
support for LibOpenR2. http://www.libopenr2.org/
* The UK option waitfordialtone has been added for use with BT analog
lines.
* Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
is used in conjunction with the 'faxdetect' configuration option. When
'faxbuffers' is used and fax tones are detected, the channel will dynamically
switch to the configured faxbuffers policy. For example, to use 6 buffers
and a 'full' buffer policy for a fax transmission, add:
faxbuffers=>6,full
The faxbuffers configuration will be in affect until the call is torn down.
* Added service message support for 4ESS/5ESS switches.
Dialplan Functions
------------------
* For DAHDI channels, the CHANNEL() dialplan function now
supports changing the channel's buffer policy (for the current
call only), using this syntax:
exten => s,n,Set(CHANNEL(buffers)=6,full)
This would change the channel to the 'full' buffer policy and
6 (six) buffers. Possible options for this setting are the same
as those in chan_dahdi.conf.
* Added a new dialplan function, CURLOPT, which permits setting various
options that may be useful with the CURL dialplan function, such as
cookies, proxies, connection timeouts, passwords, etc.
* Permit the syntax and synopsis fields of the corresponding dialplan
functions to be individually set from func_odbc.conf.
* Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
* func_odbc now may specify an insert query to execute, when the write query
affects 0 rows (usually indicating that no such row exists).
* Added a new dialplan function, LISTFILTER, which permits removing elements
from a set list, by name. Uses the same general syntax as the existing CUT
and FIELDQTY dialplan functions, which also manage lists.
* Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
obtaining realtime data from the dialplan.
* Added LOCAL_PEEK, which allows access to variables in any stack frame within
a subroutine when using the GoSub() and Return() applications.
* Added AUDIOHOOK_INHERIT. For information on its use, please see the output
of "core show function AUDIOHOOK_INHERIT" from the CLI
* Added AES_ENCRYPT. For information on its use, please see the output
of "core show function AES_ENCRYPT" from the CLI
* Added AES_DECRYPT. For information on its use, please see the output
of "core show function AES_DECRYPT" from the CLI
* func_odbc now supports database transactions across multiple queries.
Applications
------------
* Scheduled meetme conferences may now have their end times extended by
using MeetMeAdmin.
* app_authenticate now gives the ability to select a prompt other than
the default.
* app_directory now pays attention to the searchcontexts setting in
voicemail.conf and will look through all contexts, if no context is
specified in the initial argument.
* A new application, Originate, has been introduced, that allows asynchronous
call origination from the dialplan.
* Voicemail now permits setting the emailsubject and emailbody per mailbox,
in addition to the setting in the "general" context.
* Added ConfBridge dialplan application which does conference bridges without
DAHDI. For information on its use, please see the output of
"core show application ConfBridge" from the CLI.
Miscellaneous
-------------
* The Asterisk CLI has a new command, "channel redirect", which is similar in
operation to the AMI Redirect action.
* extensions.conf now allows you to use keyword "same" to define an extension
without actually specifying an extension. It uses exactly the same pattern
as previously used on the last "exten" line. For example:
exten => 123,1,NoOp(something)
same => n,SomethingElse()
* musiconhold.conf classes of type 'files' can now use relative directory paths,
which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
* All deprecated CLI commands are removed from the sourcecode. They are now handled
by the new clialiases module. See cli_aliases.conf.sample file.
* Times within timespecs are now accurate down to the minute. This is a change
from historical Asterisk, which only provided timespecs rounded to the nearest
even (read: evenly divisible by 2) minute mark.
* The realtime switch now supports an option flag, 'p', which disables searches for
pattern matches.
* In addition to a time range and date range, timespecs now accept a 5th optional
argument, timezone. This allows you to perform time checks on alternate
timezones, especially if those daylight savings time ranges vary from your
machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
includes.
* The contrib/scripts/ directory now has a script called sip_nat_settings that will
give you the correct output for an asterisk box behind nat. It will give you the
externhost and localnet settings.
* The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
can connect calls in passthrough mode, as well as record and play back files.
* Successful and unsuccessful call pickup can now be alerted through sounds, by
using pickupsound and pickupfailsound in features.conf.
* ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
instead of the /var/run/asterisk.pid where it used to be. This will make
installs as non-root easier to manage.
CDR
---
* The cdr.conf file must exist and be correctly programmed in order for CDR records to
be written; they will no longer be explicitly written.
Asterisk Manager Interface
--------------------------
* When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
a non-empty value) in your request. If you do this, any pending AMI events will
*not* be included in the response to your request as they would normally, but
will be left in the event queue for the next request you make to retrieve. For
some applications, this will allow you to guarantee that you will only see
events in responses to 'WaitEvent' actions, and can better know when to expect them.
To know whether the Asterisk server supports this header or not, your client can
inspect the first response back from the server to see if it includes this header:
Pragma: SuppressEvents
If this is included, the server supports event suppression.
* Added 4 new Actions to list skinny device(s) and line(s)
SKINNYdevices
SKINNYshowdevice
SKINNYlines
SKINNYshowline
LDAP Schema File Additions
--------------------------
* Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
* Added new Fields:
- AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
- AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
- AstAccountVideoSupport, AstAccountIgnoreSDPVersion
* Removed redundant IPaddr (there's already IPAddress)
- Gives more configuration Flags for SIP-Users available (tested)
- Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
without extensibleObject (which really should be the last resort); gives
also additional possibilities for LDAP-filter
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
------------------------------------------------------------------------------
Device State Handling
---------------------
* The event infrastructure in Asterisk got another big update to help support
distributed events. It currently supports distributed device state and
distributed Voicemail MWI (Message Waiting Indication). A new module has
been merged, res_ais, which facilitates communicating events between servers.
It uses the SAForum AIS (Service Availability Forum Application Interface
Specification) CLM (Cluster Management) and EVT (Event) services to maintain
a cluster of Asterisk servers, and to share events between them. For more
information on setting this up, refer to the Distributed Device State section
of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
Dialplan Functions
------------------
* Added a new dialplan function, AST_CONFIG(), which allows you to access
variables from an Asterisk configuration file.
* The JACK_HOOK function now has a c() option to supply a custom client name.
* Added two new dialplan functions from libspeex for audio gain control and
denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
rx directions of a channel from the dialplan.
* The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
based on other parameters. The default is still to search based on the
forwarding station ID. However, there are new options that allow you to search
based on the message desk terminal ID, or the message desk number.
* TIMEOUT() has been modified to be accurate down to the millisecond.
* ENUM*() functions now include the following new options:
- 'u' returns the full URI and does not strip off the URI-scheme.
- 's' triggers ISN specific rewriting
- 'i' looks for branches into an Infrastructure ENUM tree
- 'd' for a direct DNS lookup without any flipping of digits.
* TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
* CHANNEL() now has options for the maximum, minimum, and standard or normal
deviation of jitter, rtt, and loss for a call using chan_sip.
DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
* Channels can now be configured using named sections in chan_dahdi.conf, just
like other channel drivers, including the use of templates.
* The default for pridialplan has changed from 'national' to 'unknown'.
PBX Changes
-----------
* It is now possible to specify a pattern match as a hint. Once a phone subscribes
to something that matches the pattern a hint will be created using the contents
and variables evaluated.
* Dialplan matching has been extended to allow an extension to return to the
PBX core to wait for more digits. This is done by using the new dialplan
application called "Incomplete". This will permit a whole new level of
extension control, by giving the administrator more control over early
matches employing one of the short-circuit pattern match operators. Note
that custom applications can trigger this same behavior by returning the
special value AST_PBX_INCOMPLETE.
Application Changes
-------------------
* Directory now permits both first and last names to be matched at the same
time. In addition, the number of digits to enter of the name can be set in
the arguments to Directory; previously, you could enter only 3, regardless
of how many names are in your company. For large companies, this should be
quite helpful.
* Voicemail now permits a mailbox setting to wrap around from first to last
messages, if the "messagewrap" option is set to a true value.
* Voicemail now permits an external script to be run, for password validation.
The script should output "VALID" or "INVALID" on stdout, depending upon the
wish to validate or invalidate the password given. Arguments are:
"mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
more details
* Dial has a new option: F(context^extension^pri), which permits a callee to
continue in the dialplan, at the specified label, if the caller hangs up.
* ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
technology name (e.g. SIP, IAX, etc) of the channel being spied on.
* The Jack application now has a c() option to supply a custom client name.
* Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
like the pre-existing whisper mode, except that the spy can also talk to the
participant on the bridged channel as well.
* Chanspy has a new option, 'n', which will allow for the spied-on party's name
to be spoken instead of the channel name or number. For more information on the
use of this option, issue the command "core show application ChanSpy" from the
Asterisk CLI.
* Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
words, if using the 'd' option, it is not possible to enter a number to append to
the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
change to whisper mode, and pressing 6 will change to barge mode.
* ExternalIVR now takes several options that affect the way it performs, as
well as having several new commands. Please see the External IVR page on the Asterisk
wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
* Added ability to communicate over a TCP socket instead of forking a child process for the
ExternalIVR application.
* ChanIsAvail has a new option, 'a', which will return all available channels instead
of just the first one if you give the function more then one channel to check.
* PrivacyManager now takes an option where you can specify a context where the
given number will be matched. This way you have more control over who is allowed
and it stops the people who blindly enter 10 digits.
* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
* The Dial() application no longer copies the language used by the caller to the callee's
channel. If you desire for the caller's channel's language to be used for file playback
to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
* SendImage() no longer hangs up the channel on error; instead, it sets the
status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
'UNSUPPORTED'. This change makes SendImage() more consistent with other
applications.
* Park has a new option, 's', which silences the announcement of the parking space number.
* A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
invalid input and will be assumed to mean that no timeout is desired.
SIP Changes
-----------
* Added DNS manager support to registrations for peers referencing peer entries.
DNS manager runs in the background which allows DNS lookups to be run asynchronously
as well as periodically updating the IP address. These properties allow for
better performance as well as recovery in the event of an IP change.
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
load/reload of large numbers of peers/users by ~40x (for large lists of peers).
These changes also provide performance improvements for call setup and tear down.
* Added ability to specify registration expiry time on a per registration basis in
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
lost packets.
* Added t38pt_usertpsource option. See sip.conf.sample for details.
* Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
* 'sip show peers' and 'sip show users' display their entries sorted in
alphabetical order, as opposed to the order they were in, in the config
file or database.
* Videosupport now supports an additional option, "always", which always sets
up video RTP ports, even on clients that don't support it. This helps with
callfiles and certain transfers to ensure that if two video phones are
connected, they will always share video feeds.
IAX Changes
-----------
* Existing DNS manager lookups extended to check for SRV records.
* IAX2 encryption support has been improved to support periodic key rotation
within a call for enhanced security. The option "keyrotate" has been
provided to disable this functionality to preserve backwards compatibility
with older versions of IAX2 that do not support key rotation.
CLI Changes
-----------
* New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
data tree based on the given <path>.
* New CLI command "data show providers" that will display all the registered
callbacks.
* New CLI command, "config reload <file.conf>" which reloads any module that
references that particular configuration file. Also added "config list"
which shows which configuration files are in use.
* New CLI commands, "pri show version" and "ss7 show version" that will
display which version of libpri and libss7 are being used, respectively.
A new API call was added so trunk will now have to be compiled against
a versions of libpri and libss7 that have them or it will not know that
these libraries exist.
* The commands "core show globals", "core set global" and "core set chanvar" has
been deprecated in favor of the more semantically correct "dialplan show globals",
"dialplan set chanvar" and "dialplan set global".
* New CLI command "dialplan show chanvar" to list all variables associated
with a given channel.
DNS manager changes
-------------------
* Addresses managed by DNS manager now can check to see if there is a DNS
SRV record for a given domain and will use that hostname/port if present.
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
* The Status command now takes an optional list of variables to display
along with channel status.
* The QueueEntry event now also includes the channel's uniqueid
ODBC Changes
------------
* res_odbc no longer has a limit of 1023 total possible unshared connections,
as some people were running into this limit. This limit has been increased
to 4.2 billion.
Queue changes
-------------
* The TRANSFER queue log entry now includes the the caller's original
position in the transferred-from queue.
* A new configuration option, "timeoutpriority" has been added. Please see the section labeled
"QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
as well as an explanation about timeout options in general
* Added a new option - C - for forcing the "answered elsewhere" flag on
cancellation of calls in to members of the queue. This is to avoid the
call to a member of a queue having the call listed as a "missed call".
Realtime changes
----------------
* Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
adaptive capabilities. What this means in practical terms is that if your
realtime table lacks critical fields, Asterisk will now emit warnings to
that effect. Also, some of the realtime drivers have the ability (if
configured) to automatically add those columns to the table with the
correct type and length.
Miscellaneous
-------------
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
the 'setvar' option to cause a given audio file to be played upon completion
of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
Skinny channels only.
* You can now compile Asterisk against the Hoard Memory Allocator, see the
Hoard page on the Asterisk wiki for more information:
https://wiki.asterisk.org/wiki/x/pQBB
* Config file variables may now be appended to, by using the '+=' append
operator. This is most helpful when working with long SQL queries in
func_odbc.conf, as the queries no longer need to be specified on a single
line.
* CDR config file, cdr.conf, has an added option, "initiatedseconds",
which will add a second to the billsec when the ending
time is set, if the number in the microseconds field of the end time is
greater than the number of microseconds in the answer time. This allows
users to count the 'initiated' seconds in their billing records.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
------------------------------------------------------------------------------
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
* Manager has undergone a lot of changes, all of them documented
on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
* Manager version has changed to 1.1
* Added a new action 'CoreShowChannels' to list currently defined channels
and some information about them.
* Added a new action 'SIPshowregistry' to list SIP registrations.
* Added TLS support for the manager interface and HTTP server
* Added the URI redirect option for the built-in HTTP server
* The output of CallerID in Manager events is now more consistent.
CallerIDNum is used for number and CallerIDName for name.
* Enable https support for builtin web server.
See configs/http.conf.sample for details.
* Added a new action, GetConfigJSON, which can return the contents of an
Asterisk configuration file in JSON format. This is intended to help
improve the performance of AJAX applications using the manager interface
over HTTP.
* SIP and IAX manager events now use "ChannelType" in all cases where we
indicate channel driver. Previously, we used a mixture of "Channel"
and "ChannelDriver" headers.
* Added a "Bridge" action which allows you to bridge any two channels that
are currently active on the system.
* Added a "ListAllVoicemailUsers" action that allows you to get a list of all
the voicemail users setup.
* Added 'DBDel' and 'DBDelTree' manager commands.
* cdr_manager now reports events via the "cdr" level, separating it from
the very verbose "call" level.
* Manager users are now stored in memory. If you change the manager account
list (delete or add accounts) you need to reload manager.
* Added Masquerade manager event for when a masquerade happens between
two channels.
* Added "manager reload" command for the CLI
* Lots of commands that only provided information are now allowed under the
Reporting privilege, instead of only under Call or System.
* The IAX* commands now require either System or Reporting privilege, to
mirror the privileges of the SIP* commands.
* Added ability to retrieve list of categories in a config file.
* Added ability to retrieve the content of a particular category.
* Added ability to empty a context.
* Created new action to create a new file.
* Updated delete action to allow deletion by line number with respect to category.
* Added new action insert to add new variable to category at specified line.
* Updated action newcat to allow new category to be inserted in file above another
existing category.
* Added new event "JitterBufStats" in the IAX2 channel
* Originate now requires the Originate privilege and, if you want to call out
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.
* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
* New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
or manager show command Atxfer from the CLI
* New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
details or manager show command IAXregistry from the CLI
Dialplan functions
------------------
* Added the DEVICE_STATE() dialplan function which allows retrieving any device
state in the dialplan, as well as creating custom device states that are
controllable from the dialplan.
* Extend CALLERID() function with "pres" and "ton" parameters to
fetch string representation of calling number presentation indicator
and numeric representation of type of calling number value.
* MailboxExists converted to dialplan function
* A new option to Dial() for telling IP phones not to count the call
as "missed" when dial times out and cancels.
* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
held for any given channel. Also, locks are automatically freed when a
channel is hung up.
* Added HINT() dialplan function that allows retrieving hint information.
Hints are mappings between extensions and devices for the sake of
determining the state of an extension. This function can retrieve the list
of devices or the name associated with a hint.
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
of any extension.
* Added SYSINFO() dialplan function which allows retrieval of system information
* Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
the existence of a dialplan target.
* Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
upper and lower case, respectively.
* When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
ID for the call (not the Asterisk call ID or unique ID), provided that the
channel driver supports this. For SIP, you get the SIP call-ID for the
bridged channel which you can store in the CDR with a custom field.
CLI Changes
-----------
* Added CLI permissions, config file: cli_permissions.conf
default is to allow all commands for every local user/group.
Also this new feature added three new CLI commands:
- cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
- cli reload permissions
- cli show permissions
* New CLI command "core show hint" (usage: core show hint <exten>)
* New CLI command "core show settings"
* Added 'core show channels count' CLI command.
* Added the ability to set the core debug and verbose values on a per-file basis.
* Added 'queue pause member' and 'queue unpause member' CLI commands
* Ability to set process limits ("ulimit") without restarting Asterisk
* Enhanced "agi debug" to print the channel name as a prefix to the debug
output to make debugging on busy systems much easier.
* New CLI commands "dialplan set extenpatternmatching true/false"
* New CLI command: "core set chanvar" to set a channel variable from the CLI.
* Added an easy way to execute Asterisk CLI commands at startup. Any commands
listed in the startup_commands section of cli.conf will get executed.
* Added a CLI command, "devstate change", which allows you to set custom device
states from the func_devstate module that provides the DEVICE_STATE() function
and handling of the "Custom:" devices.
* New CLI command: "sip show sched" which shows all ast_sched entries for sip,
sorted into the different possible callbacks, with the number of entries
currently scheduled for each. Gives you a feel for how busy the sip channel
driver is.
* Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
* Cleanup another bunch of CLI commands. Now all modules follow the same schema.
(Done by lmadsen, junky and mvanbaak during the devcon 2008)
SIP changes
-----------
* Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
option is enabled, Asterisk will watch for a CNG tone in the incoming audio
for a received call. If it is detected, the channel will jump to the
'fax' extension in the dialplan.
* The default SIP useragent= identifier now includes the Asterisk version
* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from the Digest header
rather than from the From: field. This feature is considered experimental.
* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
* The "localmask" setting was removed in version 1.2 and the reminder about it
being removed is now also removed.
* A new option "busylevel" for setting a level of calls where asterisk reports
a device as busy, to separate it from call-limit. This value is also added
to the SIP_PEER dialplan function.
* A new realtime family called "sipregs" is now supported to store SIP registration
data. If this family is defined, "sippeers" will be used for configuration and
"sipregs" for registrations. If it's not defined, "sippeers" will be used for
registration data, as before.
* The SIPPEER function have new options for port address, call and pickup groups
* Added support for T.140 realtime text in SIP/RTP
* The "checkmwi" option has been removed from sip.conf, as it is no longer
required due to the restructuring of how MWI is handled. See the descriptions
in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
for more information.
* Added rtpdest option to CHANNEL() dialplan function.
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
* SIP now adds a header to the CANCEL if the call was answered by another phone
in the same dial command, or if the new c option in dial() is used.
* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
states it is not needed. For phones, however, that do require it the "registertrying" option
has been added so it can be enabled.
* A new option called "callcounter" (global/peer/user level) enables call counters needed
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
used to enable this functionality).
* New settings for timer T1 and timer B on a global level or per device. This makes it
possible to force timeout faster on non-responsive SIP servers. These settings are
considered advanced, so don't use them unless you have a problem.
* Added a dial string option to be able to set the To: header in an INVITE to any
SIP uri.
* Added a new global and per-peer option, qualifyfreq, which allows you to configure
the qualify frequency.
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
* Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
and configs/sip.conf.sample for more information on how it is used.
* Added a new configuration option "authfailureevents" that enables manager events when
a peer can't authenticate properly.
* Added DNS manager support to registrations for peers not referencing a peer entry.
IAX2 changes
------------
* Added the trunkmaxsize configuration option to chan_iax2.
* Added the srvlookup option to iax.conf
* Added support for OSP. The token is set and retrieved through the CHANNEL()
dialplan function.
XMPP Google Talk/Jingle changes
-------------------------------
* Added the bindaddr option to gtalk.conf.
Skinny changes
-------------
* Added skinny show device, skinny show line, and skinny show settings CLI commands.
* Proper codec support in chan_skinny.
* Added settings for IP and Ethernet QoS requests
MGCP changes
------------
* Added separate settings for media QoS in mgcp.conf
Console Channel Driver changes
------------------------------
* Added experimental support for video send & receive to chan_oss.
This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
a video source.
Phone channel changes (chan_phone)
----------------------------------
* Added G729 passthrough support to chan_phone for Sigma Designs boards.
H.323 channel Changes
---------------------
* H323 remote hold notification support added (by NOTIFY message
and/or H.450 supplementary service)
Local channel changes
---------------------
* The device state functionality in the Local channel driver has been updated
to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
to just UNKNOWN if the extension exists.
* Added jitterbuffer support for chan_local. This allows you to use the
generic jitterbuffer on incoming calls going to Asterisk applications.
For example, this would allow you to use a jitterbuffer for an incoming
SIP call to Voicemail by putting a Local channel in the middle. This
feature is enabled by using the 'j' option in the Dial string to the Local
channel in conjunction with the existing 'n' option for local channels.
* A 'b' option has been added which causes chan_local to return the actual channel
that is behind it when queried. This is useful for transfer scenarios as the
actual channel will be transferred, not the Local channel.
Agent channel changes
----------------------
* The ackcall and endcall options are now supplemented with options acceptdtmf
and enddtmf. These allow for the DTMF keypress to be configurable. The options
default to their old hard-coded values ('#' and '*' respectively) so this should
not break any existing agent installations.
DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
* SS7 support (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added dahdi show version CLI command.
* Added setvar support to chan_dahdi.conf channel entries.
* Added two new options: mwimonitor and mwimonitornotify. These options allow
you to enable MWI monitoring on FXO lines. When the MWI state changes,
the script specified in the mwimonitornotify option is executed. An internal
event indicating the new state of the mailbox is also generated, so that
the normal MWI facilities in Asterisk work as usual.
* Added signalling type 'auto', which attempts to use the same signalling type
for a channel as configured in DAHDI. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your chan_dahdi.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
* Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.
New Channel Drivers
-------------------
* Added a new channel driver, chan_unistim. See the Asterisk wiki at
https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
for details. This new channel driver allows you to use Nortel i2002,
i2004, and i2050 phones with Asterisk.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
work with Mac CoreAudio, but portaudio supports a number of other audio
interfaces, as well. Note that this channel driver requires v19 or higher
of portaudio; older versions have a different API.
DUNDi changes
-------------
* Added the ability to specify arguments to the Dial application when using
the DUNDi switch in the dialplan.
* Added the ability to set weights for responses dynamically. This can be
done using a global variable or a dialplan function. Using the SHELL()
function would allow you to have an external script set the weight for
each response.
* Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
functions will allow you to initiate a DUNDi query from the dialplan,
find out how many results there are, and access each one.
* Added the ability to specify a port for a dundi peer.
ENUM changes
------------
* Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
functions will allow you to initiate an ENUM lookup from the dialplan,
and Asterisk will cache the results. ENUMRESULT can be used to access
the results without doing multiple DNS queries.
Voicemail Changes
-----------------
* Added the ability to customize which sound files are used for some of the
prompts within the Voicemail application by changing them in voicemail.conf
* Added the ability for the "voicemail show users" CLI command to show users
configured by the dynamic realtime configuration method.
* MWI (Message Waiting Indication) handling has been significantly
restructured internally to Asterisk. It is now totally event based
instead of polling based. The voicemail application will notify other
modules that have subscribed to MWI events when something in the mailbox
changes.
This also means that if any other entity outside of Asterisk is changing
the contents of mailboxes, then the voicemail application still needs to
poll for changes. Examples of situations that would require this option
are web interfaces to voicemail or an email client in the case of using
IMAP storage. So, two new options have been added to voicemail.conf
to account for this: "pollmailboxes" and "pollfreq". See the sample
configuration file for details.
* Added "tw" language support
* Added support for storage of greetings using an IMAP server
* Added ability to customize forward, reverse, stop, and pause keys for message playback
* SMDI is now enabled in voicemail using the smdienable option.
* A "lockmode" option has been added to asterisk.conf to configure the file
locking method used for voicemail, and potentially other things in the
future. The default is the old behavior, lockfile. However, there is a
new method, "flock", that uses a different method for situations where the
lockfile will not work, such as on SMB/CIFS mounts.
* Added the ability to backup deleted messages, to ease recovery in the case
that a user accidentally deletes a message, and discovers that they need it.
* Reworked the SMDI interface in Asterisk. The new way to access SMDI information
is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
voicemail boxes. The SMDI interface can also poll for MWI changes when some
outside entity is modifying the state of the mailbox (such as IMAP storage or
a web interface of some kind).
* Added the support for marking messages as "urgent." There are two methods to accomplish
this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
the message as urgent after he has recorded a voicemail by following the voice instructions.
When listening to voicemails using VoiceMailMain urgent messages will be presented before other
messages
* Added "is" language support
Queue changes
-------------
* Added the general option 'shared_lastcall' so that member's wrapuptime may be
used across multiple queues.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
* Added keepstats option to queues.conf which will keep queue
statistics during a reload.
* setinterfacevar option in queues.conf also now sets a variable
called MEMBERNAME which contains the member's name.
* Added 'Strategy' field to manager event QueueParams which represents
the queue strategy in use.
* Added option to run macro when a queue member is connected to a caller,
see queues.conf.sample for details.
* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
does not count paused queue members as unavailable.
* Added min-announce-frequency option to queues.conf which allows you to control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
* Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
queue log.
* Added ability for non-realtime queues to have realtime members
* Added the "linear" strategy to queues.
* Added the "wrandom" strategy to queues.
* Added new channel variable QUEUE_MIN_PENALTY
* QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
rules in queuerules.conf. See configs/queuerules.conf.sample for details
* Added a new parameter for member definition, called state_interface. This may be
used so that a member may be called via one interface but have a different interface's
device state reported.
* Added new CLI and Manager commands relating to reloading queues. From the CLI, see
"queue reload", "queue reset stats". Also see "manager show command QueueReload" and
"manager show command QueueReset."
* New configuration option: randomperiodicannounce. If a list of periodic announcements is
specified by the periodic-announce option, then one will be chosen randomly when it is time
to play a periodic announcment
* New configuration options: announce-position now takes two more values in addition to "yes" and
"no." Two new options, "limit" and "more," are allowed. These are tied to another option,
announce-position-limit. By setting announce-position to "limit" callers will only have their
position announced if their position is less than what is specified by announce-position-limit.
If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
will be told that their are more than announce-position-limit callers waiting.
* Two new queue log events have been added. An ADDMEMBER event will be logged
when a realtime queue member is added and a REMOVEMEMBER event will be logged
when a realtime queue member is removed. Since there is no calling channel associated
with these events, the string "REALTIME" is placed where the channel's unique id
is typically placed.
* The configuration method for the "joinempty" and "leavewhenempty" options has
changed to a comma-separated list of methods of determining member availability
instead of vague terms such as "yes," "loose," "no," and "strict." These old four
values are still accepted for backwards-compatibility, though.
* The average talktime is now calculated on queues. This information is reported via the
CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
the queue.
MeetMe Changes
--------------
* The 'o' option to provide an optimization has been removed and its functionality
has been enabled by default.
* When a conference is created, the UNIQUEID of the channel that caused it to be
created is stored. Then, every channel that joins the conference will have the
MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
callers that come and go from long standing conferences.
* Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
except it does operations on a channel by name, instead of number in a conference.
This is a very useful feature in combination with the 'X' option to ChanSpy.
* Added 'C' option to Meetme which causes a caller to continue in the dialplan
when kicked out.
* Added new RealTime functionality to provide support for scheduled conferencing.
This includes optional messages to the caller if they attempt to join before
the schedule start time, or to allow the caller to join the conference early.
Also included is optional support for limiting the number of callers per
RealTime conference.
* Added the S() and L() options to the MeetMe application. These are pretty
much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
* Added the ability to do "meetme concise" with the "meetme" CLI command.
This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.
* Added MEETME_INFO dialplan function which provides a way to query
various properties of a Meetme conference.
* Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
and *84: record in-conf
Other Dialplan Application Changes
----------------------------------
* Argument support for Gosub application
* From the to-do lists: straighten out the app timeout args:
Wait() app now really does 0.3 seconds- was truncating arg to an int.
WaitExten() same as Wait().
Congestion() - Now takes floating pt. argument.
Busy() - now takes floating pt. argument.
Read() - timeout now can be floating pt.
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
* Added an optional timeout argument to the Page application.
* Added 'E', 'V', and 'P' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
calling channel to any other active channel on the system.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
* The Read application no longer exits the dialplan on error. Instead, it sets
READSTATUS to ERROR, which you can catch and handle separately.
* Added 'm' option to Directory, which lists out names, 8 at a time, instead
of asking for verification of each name, one at a time.
* Privacy() no longer uses privacy.conf, as all options are specifiable as
direct options to the app.
* AMD() has a new "maximum word length" option. "show application AMD" from the CLI
for more details
* GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
* The ChannelRedirect application no longer exits the dialplan if the given channel
does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
or NOCHANNEL if the given channel was not found.
* The silencethreshold setting that was previously configurable in multiple
applications is now settable globally via dsp.conf.
Music On Hold Changes
---------------------
* A new option, "digit", has been added for music on hold classes in
musiconhold.conf. If this is set for a music on hold class, a caller
listening to music on hold can press this digit to switch to listening
to this music on hold class.
* Support for realtime music on hold has been added.
* In conjunction with the realtime music on hold, a general section has
been added to musiconhold.conf, its sole variable is cachertclasses. If this
is set, then music on hold classes found in realtime will be cached in memory.
AEL Changes
-----------
* AEL upgraded to use the Gosub with Arguments instead
of Macro application, to hopefully reduce the problems
seen with the artificially low stack ceiling that
Macro bumps into. Macros can only call other Macros
to a depth of 7. Tests run using gosub, show depths
limited only by virtual memory. A small test demonstrated
recursive call depths of 100,000 without problems.
-- in addition to this, all apps that allowed a macro
to be called, as in Dial, queues, etc, are now allowing
a gosub call in similar fashion.
* AEL now generates LOCAL(argname) declarations when it
Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
etc. That makes the arguments local in scope. The user
can define their own local variables in macros, now,
by saying "local myvar=someval;" or using Set() in this
fashion: Set(LOCAL(myvar)=someval); ("local" is now
an AEL keyword).
* utils/conf2ael introduced. Will convert an extensions.conf
file into extensions.ael. Very crude and unfinished, but
will be improved as time goes by. Should be useful for a
first pass at conversion.
* aelparse will now read extensions.conf to see if a referenced
macro or context is there before issuing a warning.
* AEL parser sets a local channel variable ~~EXTEN~~, to
preserve the value of ${EXTEN} thru switch statements.
* New operator in $[...] expressions: the ~~ operator serves
as a concatenation operator. AT THE MOMENT, it is really only
necessary and useful in AEL, especially in if() expressions.
Operation: ${a} ~~ ${b| with force both a and b to strings, strip
any enclosing double-quotes, and evaluate to the value of a
concatenated with the value of b. For example if a is set to
"xyz" and b has the value "abc", then ${a} ~~ ${b| would
evaluate to xyzabc .
Call Features (res_features) Changes
------------------------------------
* Added the parkedcalltransfers option to features.conf
* Added parkedcallparking option to control one touch parking w/ parking
pickup
* Added parkedcallhangup option to control disconnect feature w/ parking
pickup
* Added parkedcallrecording option to control one-touch record w/ parking
pickup
* Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
parkedcalltransfers option support for multiple parking lots.
* Added BRIDGE_FEATURES variable to set available features for a channel
* The built-in method for doing attended transfers has been updated to
include some new options that allow you to have the transferee sent
back to the person that did the transfer if the transfer is not successful.
See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
in features.conf.sample.
* Added support for configuring named groups of custom call features in
features.conf. This means that features can be written a single time, and
then mapped into groups of features for different key mappings or easier
access control.
* Updated the ParkedCall application to allow you to not specify a parking
extension. If you don't specify a parking space to pick up, it will grab
the first one available.
* Added cli command 'features reload' to reload call features from features.conf
* Moved into core asterisk binary.
* Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
* Added the ability for custom parking lots to be configured with their own
parking extension with the parkext option.
Language Support Changes
------------------------
* Brazilian Portuguese (pt-BR) in VM, and say.c was added
* Added support for the Hungarian language for saying numbers, dates, and times.
AGI Changes
-----------
* Added SPEECH commands for speech recognition. A complete listing can be found
using agi show.
* If app_stack is loaded, GOSUB is a native AGI command that may be used to
invoke subroutines in the dialplan. Note that calling EXEC with Gosub
does not behave as expected; the native command needs to be used, instead.
* Added the ability to perform SRV lookups on fast AGI calls. To use this
feature, simply use hagi: instead of agi: as the protocol portion
of the URI parameter to the AGI function call in your dial plan. Also note
that specifying a port number in the AGI URI will disable SRV lookups,
even if you use the hagi: protocol.
* No longer support MSG_OOB flag on HANGUP.
Logger changes
--------------
* Added rotatestrategy option to logger.conf, along with two new options:
"timestamp" which will use the time to name the logger files instead of
sequence number; and "rotate", which rotates the names of the log files,
similar to the way syslog rotates files.
* Added exec_after_rotate option to logger.conf, which allows a system
command to be run after rotation. This is primarily useful with
rotatestrategy=rotate, to allow a limit on the number of log files kept
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
Call Detail Records
-------------------
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
* Added cdr_adaptive_odbc, a new module that adapts to the structure of your
backend database CDR table. Specifically, additional, non-standard
columns are supported, merely by setting the corresponding CDR variable in
your dialplan. In addition, you may alias any column to another name (for
example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
simply "alias src => ANI" in the configuration file). Records may be
posted to more than one backend, simply by specifying multiple categories
in the configuration file. And finally, you may filter which CDRs get
posted to each backend, by specifying a filter (which the record must
match) for the particular category. Filters are additive (meaning all
rules must match to post that CDR).
* The Postgres CDR module now supports some features of the cdr_adaptive_odbc
module. Specifically, you may add additional columns into the table and
they will be set, if you set the corresponding CDR variable name. Also,
if you omit columns in your database table, they will be silently skipped
(but a record will still be inserted, based on what columns remain). Note
that the other two features from cdr_adaptive_odbc (alias and filter) are
not currently supported.
* The ResetCDR application now has an 'e' option that re-enables a CDR if it
has been disabled using the NoCDR application.
Miscellaneous New Modules
-------------------------
* Added a new CDR module, cdr_sqlite3_custom.
* Added a new realtime configuration module, res_config_sqlite
* Added a new codec translation module, codec_resample, which re-samples
signed linear audio between 8 kHz and 16 kHz to help support wideband
codecs.
* Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
* Added a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
* Added a new module, res_config_curl, which permits using a HTTP POST url
to retrieve, create, update, and delete realtime information from a remote
web server. Note that this module requires func_curl.so to be loaded for
backend functionality.
* Added a new module, res_config_ldap, which permits the use of an LDAP
server for realtime data access.
* Added support for writing and running your dialplan in lua using the pbx_lua
module. See configs/extensions.lua.sample for examples of how to do this.
Miscellaneous
-------------
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented on the Asterisk wiki at
https://wiki.asterisk.org/wiki/x/EYBG
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
It allows you to configure a prefix for auto-monitor recordings.
* A new extension pattern matching algorithm, based on a trie, is introduced
here, that could noticeably speed up mid-sized to large dialplans.
It is NOT used by default, as duplicating the behaviour of the old pattern
matcher is still under development. A config file option, in extensions.conf,
in the [general] section, called "extenpatternmatchingnew", is by default
set to false; setting that to true will force the use of the new algorithm.
Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
specifying which socket to use to connect to the running Asterisk daemon
(-s)
* Performance enhancements to the sched facility, which is used in
the channel drivers, etc. Added hashtabs and doubly-linked lists
to speed up deletion; start at the beginning or end of list to
speed up insertion.
* Added Doubly-linked lists after the fashion of linkedlists.h. They are in
dlinkedlists.h. Doubly-linked lists feature fast deletion times.
Added regression tests to the tests/ dir, also.
* Added a refcount trace feature to astobj2 for those trying to balance
object creation, deletion; work, play; space and time. See the
notes in astobj2.h. Also, see utils/refcounter as well, as a
quick way to find unbalanced refcounts in what could be a sea
of objects that were balanced.
* Added logging to 'make update' command. See update.log
* Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
do not come from the remote party.
* Added the 'n' option to the SpeechBackground application to tell it to not
answer the channel if it has not already been answered.
* Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
turned on, via the CHANNEL(trace) dialplan function. Could be useful for
dialplan debugging.
* iLBC source code no longer included (see UPGRADE.txt for details)
* If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
deadlock is detected, a backtrace of the stack which led to the lock calls
will be output to the CLI.
* If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
the "core show locks" CLI command will give lock information output as well
as a backtrace of the stack which led to the lock calls.
* users.conf now sports an optional alternateexts property, which permits
allocation of additional extensions which will reach the specified user.
* A new option for the configure script, --enable-internal-poll, has been added
for use with systems which may have a buggy implementation of the poll system
call. If you notice odd behavior such as the CLI being unresponsive on remote
consoles, you may want to try using this option. This option is enabled by default
on Darwin systems since it is known that the Darwin poll() implementation has
odd issues.
Timer Changes
--------------------
* In addition to timing from DAHDI, there is a new timing module called
res_timing_timerfd. In order to use this, you must be running Linux with
a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
script will be able to tell if you have the requirements. From menuselect, select
res_timing_timerfd from the Resource Modules menu.
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