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@ -12,6 +12,112 @@
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===
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===
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==============================================================================
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.15.0 to Asterisk 18.16.0 ----------
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------------------------------------------------------------------------------
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AMI
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------------------
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* The AOCMessage action can now be used to generate AOC-S messages.
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Add support for named capture agent.
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------------------
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* A name for the capture agent can now be specified
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using the capture_name option which, if specified,
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will be sent to the HEP server.
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app_if
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------------------
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* Adds the If, ElseIf, Else, EndIf, and ExitIf applications
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for conditional execution of a block of code.
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app_mixmonitor
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------------------
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* The d option for MixMonitor now allows deleting
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the original recording when MixMonitor exits,
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which can be useful when MixMonitor copies it
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somewhere else before exiting.
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* Adds the c option to use the real Caller ID on
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the channel in voicemail recordings as opposed
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to the Connected Line.
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app_voicemail
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------------------
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* The voicemail user option attachextrecs can
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now be set to control whether external recordings
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trigger voicemail email notifications.
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cdr
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------------------
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* Two new options have been added which allow
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bridging and dial state changes to be ignored
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in CDRs, which can be useful if a single CDR
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is desired for a channel.
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chan_dahdi
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------------------
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* FXO channels (FXS signaled) that don't use callerid or
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distinctive ring detection can now be configured
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to enter the dialplan immediately using immediate=yes,
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instead of waiting for at least one ring.
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pbx_builtins
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------------------
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* It is now possible to not wait for media on
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a channel when answering it using Answer,
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by specifying the i option.
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res_pjsip
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------------------
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* Added options "security_negotiation" and "security_mechanisms" to pjsip
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endpoints and registrations. "security_negotiation" can be set to "no" (default)
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or "mediasec", and "security_mechanisms" can be a list of comma-separated
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security_mechanisms in the form defined by RFC 3329 section 2.2.
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* A new option named "all_codecs_on_empty_reinvite" has been added to the
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global section. When this option is enabled, on reception of a re-INVITE
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without SDP, Asterisk will send an SDP offer in the 200 OK response containing
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all configured codecs on the endpoint, instead of simply those that have
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already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
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The default value is "off".
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res_pjsip_aoc
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------------------
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* Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
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A new endpoint option, send_aoc, controls this.
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res_pjsip_header_funcs
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------------------
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* The new PJSIP_HEADER_PARAM function now fully supports both
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URI and header parameters. Both reading and writing
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parameters are supported.
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res_pjsip_logger
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------------------
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* SIP messages can now be filtered by SIP request method
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(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
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SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
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allowing for more granular debugging to be done
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in the CLI. This applies to requests but not responses.
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res_pjsip_notify
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------------------
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* Allows using the config options in pjsip_notify.conf
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from AMI actions as with the existing CLI commands.
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res_tonedetect
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------------------
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* The TONE_DETECT function now supports
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detection of audible ringback tone
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using the p option.
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xmldocs
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------------------
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* The XML documentation can now be reloaded without restarting
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Asterisk, which makes it possible to load new modules that
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enforce documentation without restarting Asterisk.
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.14.0 to Asterisk 18.15.0 ----------
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--- Functionality changes from Asterisk 18.14.0 to Asterisk 18.15.0 ----------
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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Asterisk Development Team
2 years ago