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@ -12,6 +12,72 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
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------------------------------------------------------------------------------
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app_broadcast
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------------------
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* A Broadcast application is now available which allows
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for asynchronous one-to-many and many-to-one channel audio.
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app_directory
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------------------
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* A new option 's' has been added to the Directory() application that
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will skip calling the extension and instead set the extension as
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DIRECTORY_EXTEN channel variable.
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app_read
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------------------
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* A new option 'e' has been added to allow Read() to return the
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terminator as the dialed digits in the case where only the terminator
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is entered.
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app_senddtmf
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------------------
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* A new option has been added to SendDTMF() which will answer the
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specified channel if it is not already up. If no channel is specified,
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the current channel will be answered instead.
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app_signal
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------------------
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* Adds Signal and WaitForSignal applications
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which can be used for signaling or as a
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simple message queue in the dialplan.
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func_json
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------------------
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* Additional parsing capabilities have been added to the
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JSON_DECODE function, including support for arrays
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and recursive indexing.
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res_phoneprov
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------------------
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* On multihomed Asterisk servers with dynamic SERVER template variables,
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reloading this module is no longer required when re-provisioning your
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phone to another interface address (e.g. when moving between VLANs.)
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res_pjsip_rfc3326
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------------------
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* Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
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addition to currently supported Q.850). The first header found will be used to set
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the HANGUPCAUSE variable.
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res_pjsip_session
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------------------
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* The overlap_context option now allows explicitly
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specifying a context to use for overlap dialing matches.
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res_rtp_asterisk
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------------------
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* This module has been updated to provide additional
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quality statistics in the form of an Asterisk
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Media Experience Score. The score is available using
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the same mechanisms you'd use to retrieve jitter, loss,
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and rtt statistics. For more information about the
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score and how to retrieve it, see
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https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
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------------------------------------------------------------------------------
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Asterisk Development Team
2 years ago