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Update CHANGES and UPGRADE.txt for 19.0.0
This commit is contained in:
Asterisk Development Team
436
CHANGES
436
CHANGES
@@ -12,6 +12,442 @@
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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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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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* 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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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 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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* 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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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_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_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_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_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_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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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_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_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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* 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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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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* ANI2 (OLI) is now transmitted over IAX2 calls
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as an information element.
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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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* 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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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_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_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_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_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_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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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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* 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_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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* 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_pjsip_registrar
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------------------
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* Adds new PJSIP AOR option remove_unavailable to either
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remove unavailable contacts when a REGISTER exceeds
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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
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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
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certain other situations. This change extends both of these behaviors
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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
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endpoint/t38_bind_udptl_to_media_address.
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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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* When the address of the STUN server (stunaddr) is a name resolved via DNS, the
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stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
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expires. This allows the STUN server to change its IP address without having to
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reload the res_rtp_asterisk module.
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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
|
||||
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
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||||
"replay check failed".
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||||
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||||
res_tonedetect
|
||||
------------------
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||||
* Arbitrary tone detection is now available through a
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||||
WaitForTone application (blocking) and a TONE_DETECT
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function (non-blocking).
|
||||
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||||
say.c
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||||
------------------
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||||
* Adds SAYFILES function to retrieve the file names that would
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||||
be played by corresponding Say applications, such as
|
||||
SayDigits, SayAlpha, etc.
|
||||
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||||
Additionally adds SayMoney and SayOrdinal applications.
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||||
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||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 18.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
Reference in New Issue
Block a user