Update CHANGES and UPGRADE.txt for 19.5.0
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Asterisk Development Team
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101
CHANGES
101
CHANGES
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@ -12,6 +12,107 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------
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------------------------------------------------------------------------------
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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_queue
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------------------
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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_voicemail
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------------------
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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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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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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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chan_pjsip
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------------------
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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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cli
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------------------
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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_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_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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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_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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stasis_channels
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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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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 19.3.1 to Asterisk 19.3.2 ------------
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------------------------------------------------------------------------------
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12
UPGRADE.txt
12
UPGRADE.txt
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@ -18,6 +18,18 @@
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===
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===========================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------
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------------------------------------------------------------------------------
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res_pjsip
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------------------
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* The 'async_operations' setting on transports is no longer
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obeyed and instead is always set to 1. This is due to the
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functionality not being applicable to Asterisk and causing
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excess unnecessary memory usage. This setting will now be
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ignored but can also be removed from the configuration file.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 19.2.0 to Asterisk 19.3.0 ------------
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------------------------------------------------------------------------------
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@ -1,7 +0,0 @@
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Subject: app_confbridge
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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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@ -1,8 +0,0 @@
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Subject: app_confbridge
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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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@ -1,5 +0,0 @@
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Subject: app_queue
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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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@ -1,5 +0,0 @@
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Subject: app_voicemail
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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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@ -1,7 +0,0 @@
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Subject: ari
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Subject: stasis_channels
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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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@ -1,8 +0,0 @@
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Subject: chan_dahdi
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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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@ -1,8 +0,0 @@
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Subject: chan_pjsip
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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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@ -1,4 +0,0 @@
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Subject: chan_pjsip
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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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@ -1,6 +0,0 @@
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Subject: chan_sip
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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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@ -1,5 +0,0 @@
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Subject: cli
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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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@ -1,6 +0,0 @@
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Subject: func_db
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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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@ -1,4 +0,0 @@
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Subject: func_evalexten
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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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@ -1,5 +0,0 @@
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Subject: res_agi
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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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@ -1,4 +0,0 @@
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Subject: res_parking
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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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@ -1,7 +0,0 @@
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Subject: res_pjsip
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The 'async_operations' setting on transports is no longer
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obeyed and instead is always set to 1. This is due to the
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functionality not being applicable to Asterisk and causing
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excess unnecessary memory usage. This setting will now be
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ignored but can also be removed from the configuration file.
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