Update CHANGES and UPGRADE.txt for 20.0.0
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Asterisk Development Team
parent
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commit
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494
CHANGES
494
CHANGES
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@ -12,6 +12,500 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
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------------------------------------------------------------------------------
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Applications
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------------------
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* added support for Danish syntax, playing the correct plural sound file
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dependen on where you have 1 or multipe messages
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based on the existing SE/NO code
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* added that we set DIALEDPEERNUMBER on the outgoing channels
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so it is avalible in b(content^extension^line)
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this add the same behaviour as Dial
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Channel-agnostic MF support
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------------------
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* A SendMF application and PlayMF manager
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application are now included to send
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arbitrary standard R1 MF tones on the
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current channel or another specified channel.
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Core
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------------------
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* Bundled PJProject Build
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The build process has been updated to make pjproject troubleshooting
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and development easier. See third-party/pjproject/README-hacking.md or
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https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
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for more info.
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Handle non-standard Meter metric type safely
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------------------
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* A meter_support flag has been introduced that defaults to true to maintain current behaviour.
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If disabled, a counter metric type will be used instead wherever a meter metric type was used,
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the counter will have a "_meter" suffix appended to the metric name.
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MessageSend
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------------------
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* The MessageSend AMI action has been updated to allow the Destination
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and the To addresses to be provided separately. This brings the
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MessageSend manager command in line with the capabilities of the
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MessageSend dialplan application.
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ToneScan application
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------------------
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* A new application, ToneScan, allows for
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synchronous detection of call progress
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signals such as dial tone, busy tone,
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Special Information Tones, and modems.
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ami
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------------------
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* An AMI event now exists for "Wink".
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* AMI events can now be globally disabled using
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the disabledevents [general] setting.
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app_confbridge
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------------------
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* Added the hear_own_join_sound option to the confbridge user profile to
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control who hears the sound_join audio file. When set to 'yes' the user
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entering the conference and the participants already in the conference
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will hear the sound_join audio file. When set to 'no' the user entering
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the conference will not hear the sound_join audio file, but the
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participants already in the conference will hear the sound_join audio file.
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* Adds the CONFBRIDGE_CHANNELS function which can
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be used to retrieve a list of channels in a ConfBridge,
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optionally filtered by a particular category. This
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list can then be used with functions like SHIFT, POP,
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UNSHIFT, etc.
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app_dtmfstore
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------------------
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* New application which collects digits
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dialed and stores them into
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a specified variable.
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app_mf
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------------------
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* Adds MF receiver and sender applications to support
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the R1 MF signaling protocol, including integration
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with the Dial application.
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* Adds an option to ReceiveMF to cap the
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number of digits read at a user-specified
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maximum.
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app_milliwatt
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------------------
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* The Milliwatt application's existing behavior is
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incorrect in that it plays a constant tone, which
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is not how digital milliwatt test lines actually
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work.
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An option is added so that a proper milliwatt test
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tone can be provided, including a 1 second silent
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interval every 10 seconds. However, for compatability
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reasons, the default behavior remains unchanged.
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app_morsecode
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------------------
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* Extends the Morsecode application by adding support for
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American Morse code and adds a configurable option
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for the frequency used in off intervals.
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app_originate
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------------------
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* Codecs can now be specified for dialplan-originated
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calls, as with call files and the manager action.
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By default, only the slin codec is now used, instead
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of all the slin* codecs.
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app_playback
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------------------
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* A new option 'mix' is added to the Playback application that
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will play by filename and say.conf. It will look on the format of the
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name, if it is like say format it will play with say.conf if not it
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will play the file name.
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app_queue
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------------------
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* Reload behavior in app_queue has been changed so
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queue and agent stats are not reset during full
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app_queue module reloads. The queue reset stats
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CLI command may still be used to reset stats while
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Asterisk is running.
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* Add field to save the time value when a member enter a queue.
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Shows this time in seconds using 'queue show' command and the
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field LoginTime for responses for AMI the events.
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The output for the CLI command `queue show` is changed by added a
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extra data field for the information of the time login time for each
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member.
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* added that we set DIALEDPEERNUMBER on the outgoing channels
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so it is avalible in b(content^extension^line)
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this add the same behaviour as Dial
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* Load queues and members from Realtime for
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AMI actions: QueuePause, QueueStatus and QueueSummary,
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Applications: PauseQueueMember and UnpauseQueueMember.
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* Added a new AMI action: QueueWithdrawCaller
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This AMI action makes it possible to withdraw a caller from a queue
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back to the dialplan. The call will be signaled to leave the queue
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whenever it can, hence, it not guaranteed that the call will leave
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the queue.
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Optional custom data can be passed in the request, in the WithdrawInfo
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parameter. If the call successfully withdrawn the queue,
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it can be retrieved using the QUEUE_WITHDRAW_INFO variable.
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This can be useful for certain uses, such as dispatching the call
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to a specific extension.
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* The m option now allows an override music on hold
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class to be specified for the Queue application
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within the dialplan.
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app_queue.c
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------------------
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* Allow multiple files to be streamed for agent announcement.
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app_queues
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------------------
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* adding support for playing the correct en/et for nordic languages
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* Don't play sound_thanks if there is no leading hold_time message
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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"
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app_read
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------------------
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* A new option allows the digit '#' to be read literally,
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rather than used exclusively as the input terminator
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character.
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app_sendtext
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------------------
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* A ReceiveText application has been added that can be
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used in conjunction with the SendText application.
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app_voicemail
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------------------
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* Add a new 'S' option to VoiceMail which prevents the instructions
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(vm-intro) from being played if a busy/unavailable/temporary greeting
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from the voicemail user is played. This is similar to the existing 's'
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option except that instructions will still be played if no user
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greeting is available.
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* added support for Danish syntax, playing the correct plural sound file
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dependen on where you have 1 or multipe messages
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based on the existing SE/NO code
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* The r option has been added, which prevents deletion
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of messages from VoiceMailMain, which can be
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useful for shared mailboxes.
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apps
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------------------
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* A new option 'mix' is added to the Playback application that
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will play by filename and say.conf. It will look on the format of the
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name, if it is like say format it will play with say.conf if not it
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will play the file name.
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ari
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------------------
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* Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
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to ARI channel resources as 'protocol_id'.
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ASTERISK-30027
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ast_coredumper
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------------------
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* New options:
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--pid=<asterisk_pid>
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Allows specification of an Asterisk instance when trying to
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and the script can't determine it itself.
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--libdir=<system library directory>
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Allows specification of a non-standard installation directory
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containing the Asterisk modules.
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--(no-)rename
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Renames the coredump and the output files with readable
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timestamps. This is the default.
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Removed unneeded or confusing options:
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--append-coredumps
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--conffile
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--no-default-search
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--tarball-uniqueid
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Changed Variables:
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COREDUMPS is now just "/tmp/core!(*.txt)"
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DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
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Changed behavior:
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If you use 'running' or 'RUNNING' you no longer need to specify
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'--no-default-search' to ignore existing coredumps.
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cdr
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------------------
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* A new CDR option, channeldefaultenabled, allows controlling
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whether CDR is enabled or disabled by default on
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newly created channels. The default behavior remains
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unchanged from previous versions of Asterisk (new
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channels will have CDR enabled, as long as CDR is
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enabled globally).
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chan_dahdi
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------------------
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* Previously, cadences were appended on dahdi restart,
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rather than reloaded. This prevented cadences from
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being updated and maxed out the available cadences
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if reloaded multiple times. This behavior is fixed
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so that reloading cadences is idempotent and cadences
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can actually be reloaded.
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* A POLARITY function is now available that allows
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getting or setting the polarity on a channel
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from the dialplan.
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chan_iax2
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------------------
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* ANI2 (OLI) is now transmitted over IAX2 calls
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as an information element.
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* Both a secret and an outkey may be specified at dial time,
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since encryption is possible with RSA authentication.
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chan_pjsip
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------------------
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* Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do.
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Add ability to read header by pattern using PJSIP_HEADER().
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* added global config option "allow_sending_180_after_183"
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Allow Asterisk to send 180 Ringing to an endpoint
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after 183 Session Progress has been send.
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If disabled Asterisk will instead send only a
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183 Session Progress to the endpoint.
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* Hook flash events can now be sent on a PJSIP channel
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if requested to do so.
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chan_sip
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------------------
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* Session timers get removed on UPDATE
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Fix if Asterisk receives a SIP REFER with Session-Timers UAC
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that Asterisk maintains Session-Timers when sending UPDATE request
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chan_sip.c
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------------------
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* resolve issue with pickup on device that uses "183" and not "180"
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channel_internal_api
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------------------
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* CHANNEL(lastcontext) and CHANNEL(lastexten)
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are now available for use in the dialplan.
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cli
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------------------
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* The "module refresh" command has been added,
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which allows unloading and then loading a
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module with a single command.
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* A new CLI command 'dialplan eval function' has been
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added which allows users to test the behavior of
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dialplan function calls directly from the CLI.
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func_channel
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------------------
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* Adds the CHANNEL_EXISTS function to check for the existence
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of a channel by name or unique ID.
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func_db
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------------------
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* The function DB_KEYCOUNT has been added, which
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returns the cardinality of the keys at a specified
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prefix in AstDB, i.e. the number of keys at a
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given prefix.
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func_env.c
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------------------
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* Two new functions, DIRNAME and BASENAME, are now
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included which allow users to obtain the directory
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or the base filename of any file.
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func_evalexten
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------------------
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* This adds the EVAL_EXTEN function which may be
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used to evaluate data at dialplan extensions.
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func_framedrop
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------------------
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* New function to selectively drop specified frames
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in either direction on a channel.
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func_json
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------------------
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* The JSON_DECODE dialplan function can now be used
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to parse JSON strings, such as in conjunction with
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CURL for using API responses.
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func_odbc
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------------------
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* A SQL_ESC_BACKSLASHES dialplan function has been added which
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escapes backslashes. Usage of this is dependent on whether the
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database in use can use backslashes to escape ticks or not. If
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it can, then usage of this prevents a broken SQL query depending
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on how the SQL query is constructed.
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func_scramble
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------------------
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* Adds an audio scrambler function that may be used to
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distort voice audio on a channel as a privacy
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enhancement.
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func_strings
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------------------
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* A new STRBETWEEN function is now included which
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allows a substring to be inserted between characters
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in a string. This is particularly useful for transforming
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dial strings, such as adding pauses between digits
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for a string of digits that are sent to another channel.
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func_vmcount
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------------------
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* Multiple mailboxes may now be specified instead of just one.
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logger
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------------------
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* Added the ability to define custom log levels in logger.conf
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and use them in the Log dialplan application. Also adds a
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logger show levels CLI command.
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res_agi
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------------------
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* Agi command 'exec' can now be enabled
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to evaluate dialplan functions and variables
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by setting the variable AGIEXECFULL to yes.
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res_cliexec
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------------------
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* A new CLI command, dialplan exec application, has
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been added which allows dialplan applications to be
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executed at the CLI, useful for some quick testing
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without needing to write dialplan.
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res_fax_spandsp
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------------------
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* Adds support for spandsp 3.0.0.
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res_geolocation
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------------------
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* Added res_geolocation which creates the core capabilities
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to manipulate Geolocation information on SIP INVITEs.
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res_parking
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------------------
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* An m option to Park and ParkAndAnnounce now allows
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specifying a music on hold class override.
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res_pjproject
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------------------
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* In pjproject.conf you can now map pjproject log levels
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to the Asterisk TRACE log level. The default mappings
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have therefore changed so that only pjproject levels
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3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
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to TRACE. Previously 3, 4, 5, and 6 were all mapped to
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DEBUG.
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res_pjsip
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------------------
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* A new transport option 'allow_wildcard_certs' has been added that when it
|
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and 'verify_server' are both set to 'yes', enables verification against
|
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wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS
|
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for TLS transport types. Names must start with the wildcard. Partial wildcards,
|
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e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only
|
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match against a single level meaning '*.example.com' matches 'foo.example.com',
|
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but not 'foo.bar.example.com'.
|
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|
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res_pjsip_geolocation
|
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------------------
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* Added res_pjsip_geolocation which gives chan_pjsip
|
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the ability to use the core geolocation capabilities.
|
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res_pjsip_header_funcs
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------------------
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* Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request.
|
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|
||||
Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request.
|
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|
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res_pjsip_pubsub
|
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------------------
|
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* A new resource_list option, resource_display_name, indicates
|
||||
whether display name of resource or the resource name being
|
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provided for RLS entries.
|
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If this option is enabled, the Display Name will be provided.
|
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This option is disabled by default to remain the previous behavior.
|
||||
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
|
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will be set as the Display Name.
|
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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
|
||||
------------------
|
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* Adds new PJSIP AOR option remove_unavailable to either
|
||||
remove unavailable contacts when a REGISTER exceeds
|
||||
max_contacts when remove_existing is disabled, or
|
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prioritize unavailable contacts over other existing
|
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contacts when remove_existing is enabled.
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|
||||
res_pjsip_t38
|
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------------------
|
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* 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
|
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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
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||||
problems on these systems, introducing a new option
|
||||
endpoint/t38_bind_udptl_to_media_address.
|
||||
|
||||
res_rtp_asterisk
|
||||
------------------
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* 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 ------------
|
||||
------------------------------------------------------------------------------
|
||||
|
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|||
67
UPGRADE.txt
67
UPGRADE.txt
|
|
@ -18,6 +18,73 @@
|
|||
===
|
||||
===========================================================
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 20.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
res_monitor
|
||||
------------------
|
||||
* This module is no longer built by default in
|
||||
accordance with the Module Deprecation Policy.
|
||||
If you require this functionality you will need
|
||||
to enable it for building in menuselect. Note
|
||||
that in the future res_monitor will be removed.
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
AMI
|
||||
------------------
|
||||
* The XML Manager Event Interface (amxml) now generates attribute names
|
||||
that are compliant with the XML 1.1 specification. Previously, an
|
||||
attribute name that started with a digit would be rendered as-is, even
|
||||
though attribute names must not begin with a digit. We now prefix
|
||||
attribute names that start with a digit with an underscore ('_') to
|
||||
prevent XML validation failures.
|
||||
|
||||
STIR/SHAKEN
|
||||
------------------
|
||||
* The STIR/SHAKEN configuration option has been split into
|
||||
4 different choices: off, attest, verify, and on. Off and
|
||||
on behave the same way as before. Attest will only perform
|
||||
attestation on the endpoint, and verify will only perform
|
||||
verification on the endpoint.
|
||||
|
||||
chan_iax2
|
||||
------------------
|
||||
* Encryption is now supported for RSA authentication.
|
||||
|
||||
Currently, these auth configurations will cause a crash:
|
||||
auth = md5,rsa
|
||||
auth = plaintext,md5,rsa
|
||||
|
||||
With a patched peer, the following will cause a crash:
|
||||
auth = rsa
|
||||
auth = md5,rsa
|
||||
auth = plaintext,md5,rsa
|
||||
|
||||
If both the peer and user are patches, no crash occurs.
|
||||
Existing good configurations should continue to work.
|
||||
|
||||
res_http_media_cache
|
||||
------------------
|
||||
* When fetching a file for playback from a URL, Asterisk will now first
|
||||
use the value of the Content-Type header in the HTTP response to
|
||||
determine the format of the audio data, and only if it is unable to do
|
||||
that will it attempt to parse the URL and extract the extension from
|
||||
the path portion. Previously Asterisk would first look at the end of
|
||||
the URL, which may have included query string parameters or a URL
|
||||
fragment, which was error prone.
|
||||
|
||||
res_pjsip
|
||||
------------------
|
||||
* The 'async_operations' setting on transports is no longer
|
||||
obeyed and instead is always set to 1. This is due to the
|
||||
functionality not being applicable to Asterisk and causing
|
||||
excess unnecessary memory usage. This setting will now be
|
||||
ignored but can also be removed from the configuration file.
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 19.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
|
|
|||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: app_playback
|
||||
Subject: 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.
|
||||
|
|
@ -1,9 +0,0 @@
|
|||
Subject: 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'.
|
||||
|
|
@ -1,3 +0,0 @@
|
|||
Subject: ami
|
||||
|
||||
An AMI event now exists for "Wink".
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: app_confbridge
|
||||
|
||||
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.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: app_dtmfstore
|
||||
|
||||
New application which collects digits
|
||||
dialed and stores them into
|
||||
a specified variable.
|
||||
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_mf
|
||||
|
||||
Adds an option to ReceiveMF to cap the
|
||||
number of digits read at a user-specified
|
||||
maximum.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_mf
|
||||
|
||||
Adds MF receiver and sender applications to support
|
||||
the R1 MF signaling protocol, including integration
|
||||
with the Dial application.
|
||||
|
|
@ -1,11 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: app_queue.c
|
||||
|
||||
Allow multiple files to be streamed for agent announcement.
|
||||
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: app_queue
|
||||
Subject: Applications
|
||||
|
||||
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
|
||||
|
|
@ -1,9 +0,0 @@
|
|||
Subject: app_queue
|
||||
|
||||
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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_queue
|
||||
|
||||
The m option now allows an override music on hold
|
||||
class to be specified for the Queue application
|
||||
within the dialplan.
|
||||
|
|
@ -1,3 +0,0 @@
|
|||
Subject: app_queues
|
||||
|
||||
adding support for playing the correct en/et for nordic languages
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: app_queues
|
||||
|
||||
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"
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_read
|
||||
|
||||
A new option allows the digit '#' to be read literally,
|
||||
rather than used exclusively as the input terminator
|
||||
character.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: app_sendtext
|
||||
|
||||
A ReceiveText application has been added that can be
|
||||
used in conjunction with the SendText application.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: app_voicemail
|
||||
Subject: 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
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_voicemail
|
||||
|
||||
The r option has been added, which prevents deletion
|
||||
of messages from VoiceMailMain, which can be
|
||||
useful for shared mailboxes.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: ari
|
||||
Subject: 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
|
||||
|
|
@ -1,23 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: 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).
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: chan_dahdi
|
||||
|
||||
A POLARITY function is now available that allows
|
||||
getting or setting the polarity on a channel
|
||||
from the dialplan.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: chan_iax2
|
||||
|
||||
ANI2 (OLI) is now transmitted over IAX2 calls
|
||||
as an information element.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: chan_iax2
|
||||
|
||||
Both a secret and an outkey may be specified at dial time,
|
||||
since encryption is possible with RSA authentication.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: chan_pjsip
|
||||
|
||||
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.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: chan_pjsip
|
||||
|
||||
Hook flash events can now be sent on a PJSIP channel
|
||||
if requested to do so.
|
||||
|
|
@ -1,3 +0,0 @@
|
|||
Subject: chan_sip.c
|
||||
|
||||
resolve issue with pickup on device that uses "183" and not "180"
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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
|
||||
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: channel_internal_api
|
||||
|
||||
CHANNEL(lastcontext) and CHANNEL(lastexten)
|
||||
are now available for use in the dialplan.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: cli
|
||||
|
||||
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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: cli
|
||||
|
||||
The "module refresh" command has been added,
|
||||
which allows unloading and then loading a
|
||||
module with a single command.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: func_channel
|
||||
|
||||
Adds the CHANNEL_EXISTS function to check for the existence
|
||||
of a channel by name or unique ID.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: func_evalexten
|
||||
|
||||
This adds the EVAL_EXTEN function which may be
|
||||
used to evaluate data at dialplan extensions.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: func_framedrop
|
||||
|
||||
New function to selectively drop specified frames
|
||||
in either direction on a channel.
|
||||
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: func_scramble
|
||||
|
||||
Adds an audio scrambler function that may be used to
|
||||
distort voice audio on a channel as a privacy
|
||||
enhancement.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,3 +0,0 @@
|
|||
Subject: func_vmcount
|
||||
|
||||
Multiple mailboxes may now be specified instead of just one.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: app_queue
|
||||
|
||||
Load queues and members from Realtime for
|
||||
AMI actions: QueuePause, QueueStatus and QueueSummary,
|
||||
Applications: PauseQueueMember and UnpauseQueueMember.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: ami
|
||||
|
||||
AMI events can now be globally disabled using
|
||||
the disabledevents [general] setting.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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().
|
||||
|
|
@ -1,14 +0,0 @@
|
|||
Subject: app_queue
|
||||
|
||||
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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: res_agi
|
||||
|
||||
Agi command 'exec' can now be enabled
|
||||
to evaluate dialplan functions and variables
|
||||
by setting the variable AGIEXECFULL to yes.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,3 +0,0 @@
|
|||
Subject: res_fax_spandsp
|
||||
|
||||
Adds support for spandsp 3.0.0.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: res_geolocation
|
||||
|
||||
Added res_geolocation which creates the core capabilities
|
||||
to manipulate Geolocation information on SIP INVITEs.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: res_parking
|
||||
|
||||
An m option to Park and ParkAndAnnounce now allows
|
||||
specifying a music on hold class override.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,4 +0,0 @@
|
|||
Subject: res_pjsip_geolocation
|
||||
|
||||
Added res_pjsip_geolocation which gives chan_pjsip
|
||||
the ability to use the core geolocation capabilities.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,9 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,5 +0,0 @@
|
|||
Subject: res_tonedetect
|
||||
|
||||
Arbitrary tone detection is now available through a
|
||||
WaitForTone application (blocking) and a TONE_DETECT
|
||||
function (non-blocking).
|
||||
|
|
@ -1,10 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: res_pjsip_pubsub
|
||||
|
||||
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.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,6 +0,0 @@
|
|||
Subject: 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.
|
||||
|
|
@ -1,15 +0,0 @@
|
|||
Subject: chan_iax2
|
||||
|
||||
Encryption is now supported for RSA authentication.
|
||||
|
||||
Currently, these auth configurations will cause a crash:
|
||||
auth = md5,rsa
|
||||
auth = plaintext,md5,rsa
|
||||
|
||||
With a patched peer, the following will cause a crash:
|
||||
auth = rsa
|
||||
auth = md5,rsa
|
||||
auth = plaintext,md5,rsa
|
||||
|
||||
If both the peer and user are patches, no crash occurs.
|
||||
Existing good configurations should continue to work.
|
||||
|
|
@ -1,9 +0,0 @@
|
|||
Subject: res_http_media_cache
|
||||
|
||||
When fetching a file for playback from a URL, Asterisk will now first
|
||||
use the value of the Content-Type header in the HTTP response to
|
||||
determine the format of the audio data, and only if it is unable to do
|
||||
that will it attempt to parse the URL and extract the extension from
|
||||
the path portion. Previously Asterisk would first look at the end of
|
||||
the URL, which may have included query string parameters or a URL
|
||||
fragment, which was error prone.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: AMI
|
||||
|
||||
The XML Manager Event Interface (amxml) now generates attribute names
|
||||
that are compliant with the XML 1.1 specification. Previously, an
|
||||
attribute name that started with a digit would be rendered as-is, even
|
||||
though attribute names must not begin with a digit. We now prefix
|
||||
attribute names that start with a digit with an underscore ('_') to
|
||||
prevent XML validation failures.
|
||||
|
|
@ -1,8 +0,0 @@
|
|||
Subject: res_monitor
|
||||
Master-Only: True
|
||||
|
||||
This module is no longer built by default in
|
||||
accordance with the Module Deprecation Policy.
|
||||
If you require this functionality you will need
|
||||
to enable it for building in menuselect. Note
|
||||
that in the future res_monitor will be removed.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: res_pjsip
|
||||
|
||||
The 'async_operations' setting on transports is no longer
|
||||
obeyed and instead is always set to 1. This is due to the
|
||||
functionality not being applicable to Asterisk and causing
|
||||
excess unnecessary memory usage. This setting will now be
|
||||
ignored but can also be removed from the configuration file.
|
||||
|
|
@ -1,7 +0,0 @@
|
|||
Subject: STIR/SHAKEN
|
||||
|
||||
The STIR/SHAKEN configuration option has been split into
|
||||
4 different choices: off, attest, verify, and on. Off and
|
||||
on behave the same way as before. Attest will only perform
|
||||
attestation on the endpoint, and verify will only perform
|
||||
verification on the endpoint.
|
||||
Loading…
Reference in New Issue