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@ -12,6 +12,278 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
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------------------------------------------------------------------------------
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AMI Flash event
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------------------
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* Hook flash events are now exposed as AMI events.
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Add variable support to Originate
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------------------
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* The Originate application now allows
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variables to be set on the new channel
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through a new option.
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Core
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------------------
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* Added debug logging categories that allow a user to output debug information
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based on a specified category. This lets the user limit, and filter debug
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output to data relevant to a particular context, or topic. For instance the
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following categories are now available for debug logging purposes:
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dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
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These debug categories can be enable/disable via an Asterisk CLI command:
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core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
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core set debug category off [<category> [<category>] ...]
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If no sub-level is associated all debug statements for a given category are
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output. If a sub-level is given then only those statements assigned a value
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at or below the associated sub-level are output.
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* The location where the media cache stores its temporary files
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is no longer hardcoded to /tmp but can now be configured separately
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via the astcachedir config variable in asterisk.conf.
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The default location for astcachedir is now /var/cache/asterisk
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instead of /tmp, please make sure to manually cleanup and/or
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migrate the temporary files in /tmp after upgrading.
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MessageSend
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------------------
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* The MessageSend dialplan application now takes an
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optional third argument that can set the message's
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"To" field on outgoing messages. It's an alternative
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to using the MESSAGE(to) dialplan function.
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To prevent confusion with the first argument, currently
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named "to", it's been renamed to "destination".
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Its function, creating the request URI, hasn't changed.
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The online documentation has also been enhanced to
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explain the behavior.
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Despite the changes in this commit, there should be
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no impact to current users of MessageSend.
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New ConfKick application
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------------------
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* Adds a ConfKick() application, which allows
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a specific channel, all users, or all non-admin
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users to be kicked from a conference bridge.
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New Reload application
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------------------
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* Adds an application to reload modules
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PlaybackFinished has a new error state
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------------------
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* The PlaybackFinished event now has a new state "failed"
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that is used when the sound file was not played due to an error.
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Before the state on PlaybackFinished was always "done".
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In case of multiple sound files to be played,
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the PlaybackFinished is sent only once in the end of the list,
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even in case of error.
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WaitForCondition application
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------------------
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* This application provides a way to halt
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dialplan execution until a provided
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condition evaluates to true.
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app_confbridge
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------------------
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* app_confbridge now has the ability to force the estimated bitrate on an SFU
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bridge. To use it, set a bridge profile's remb_behavior to "force" and
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set remb_estimated_bitrate to a rate in bits per second. The
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remb_estimated_bitrate parameter is ignored if remb_behavior is something
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other than "force".
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app_confbridge answer supervision control
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------------------
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* app_confbridge now provides a user option to prevent
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answer supervision if the channel hasn't been
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answered yet. To use it, set a user profile's
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answer_channel option to no.
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app_dial announcement option
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------------------
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* The A option for Dial now supports
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playing audio to the caller as well
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as the called party.
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app_mixmonitor
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------------------
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* app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
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MixMonitorMute when the channel monitoring is started, stopped and muted (or
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unmuted) respectively.
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app_voicemail
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------------------
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* The VoiceMail application can now be configured to send greetings and
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instructions via early media and only answering the channel when it is
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time for the caller to record their message. This behavior can be
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activated by passing the new 'e' option to VoiceMail.
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* You can now customize the "beep" tone or omit it entirely.
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chan_iax2
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------------------
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* You can now specify a default "auth" method in the
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[general] section of iax.conf
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chan_pjsip
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------------------
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* The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
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returns unsuccessful if it's used on a channel prior to answering.
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chan_pjsip, app_transfer
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------------------
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* Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
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transfers can pass a protocol specific error code.
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Example, in SIP 3xx-6xx represent any SIP specific error received when
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performing a REFER.
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func_math: Three new dialplan functions
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------------------
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* Introduce three new functions, MIN, MAX, and ABS, which can be used to
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obtain the minimum or maximum of up to two integers or absolute value.
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func_odbc
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------------------
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* Introduce an ARGC variable for func_odbc functions, along with a minargs
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per-function configuration option.
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minargs enables enforcing of minimum count of arguments to pass to
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func_odbc, so if you're unconditionally using ARG1 through ARG4 then
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this should be set to 4. func_odbc will generate an error in this case,
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so for example
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[FOO]
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minargs = 4
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and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
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potentially leaked ARG4 from Gosub().
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ARGC is needed if you're using optional argument, to verify whether or
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not an argument has been passed, else it's possible to use a leaked ARGn
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from Gosub (app_stack). So now you can safely do
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${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
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func_volume now can be read
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------------------
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* The VOLUME function can now also be used
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to read existing values previously set.
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logger
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------------------
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* Added a new log formatter called "plain" that always prints
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file, function and line number if available (even for verbose
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messages) and never prints color control characters. Most
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suitable for file output but can be used for other channels
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as well.
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You use it in logger.conf like so:
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debug => [plain]debug
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console => [plain]error,warning,debug,notice,pjsip_history
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messages => [plain]warning,error,verbose
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* The dateformat option in logger.conf will now control the remote
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console (asterisk -r -T) timestamp format. Previously, dateformat only
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controlled the formatting of the timestamp going to log files and the
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main console (asterisk -c) but only for non-verbose messages.
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Internally, Asterisk does not send the logging timestamp with verbose
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messages to console clients. It's up to the Asterisk remote consoles
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to format verbose messages. Asterisk remote consoles previously did
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not load dateformat from logger.conf.
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Previously there was a non-configurable and hard-coded "%b %e %T"
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dateformat that would be used no matter what on all verbose console
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messages printed on remote consoles.
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Example:
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logger.conf
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dateformat=%F %T.%3q
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# asterisk -rvvv -T
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[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
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[Mar 19 09:55:43] -- Goto (dialExten,s,1)
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Given the following example configuration in logger.conf, Asterisk log
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files and the console, will log verbose messages using the given
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timestamp. Now ensuring that all remote console messages are logged
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with the same dateformat as other log streams.
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---
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[general]
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dateformat=%F %T.%3q
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[logfiles]
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console => notice,warning,error,verbose
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full => notice,warning,error,debug,verbose
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---
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Now we have a globally-defined dateformat that will be used
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consistently across the Asterisk main console, remote consoles, and
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log files.
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Now we have consistent logging:
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# asterisk -rvvv -T
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[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
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[2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
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res_pjsip
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------------------
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* PJSIP transports can now be partially reloaded safely. This allows the
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local_net and external_* options to be updated without restarting Asterisk.
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* PJSIP endpoints can now be configured to skip authentication when
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handling OPTIONS requests by setting the allow_unauthenticated_options
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configuration property to 'yes.'
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* PJSIP support of registrations of endpoints in multidomain
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scenarios, where the endpoint contains the domain info
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in pjsip_endpoint.conf
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res_pjsip_dialog_info_body_generator
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------------------
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* PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
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remote elements by iterating through ringing channels and inserting
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that info into NOTIFY packet sent to the endpoint.
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res_pjsip_messaging
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------------------
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* Implemented the new "to" parameter of the MessageSend()
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dialplan application. This allows a user to specify
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a complete SIP "To" header separate from the Request URI.
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We now also accept a destination in the same format
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as Dial()... PJSIP/number@endpoint
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res_rtp_asterisk
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------------------
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* By default Asterisk reports the PJSIP version in all
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STUN packets it sends.
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This behaviour may not be desired in a production
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environment and can now be disabled by setting the
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stun_software_attribute option to 'no' in rtp.conf.
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res_srtp
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------------------
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* SRTP replay protection has been added to res_srtp and
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a new configuration option "srtpreplayprotection" has
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been added to the rtp.conf config file. For security
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reasons, the default setting is "yes". Buggy clients
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may not handle this correctly which could result in
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no, or one way, audio and Asterisk error messages like
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"replay check failed".
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------------------------------------------------------------------------------
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--- New functionality introduced in Asterisk 18.0.0 --------------------------
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------------------------------------------------------------------------------
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Asterisk Development Team
4 years ago