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Update CHANGES and UPGRADE.txt for 20.2.0

This commit is contained in:
Asterisk Development Team 2023-03-02 11:37:42 -05:00
parent ceda5a9859
commit 93813c9dca
13 changed files with 79 additions and 60 deletions
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66
CHANGES
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@ -12,6 +12,72 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
------------------------------------------------------------------------------
app_broadcast
------------------
* A Broadcast application is now available which allows
for asynchronous one-to-many and many-to-one channel audio.
app_directory
------------------
* A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.
app_read
------------------
* A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.
app_senddtmf
------------------
* A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.
app_signal
------------------
* Adds Signal and WaitForSignal applications
which can be used for signaling or as a
simple message queue in the dialplan.
func_json
------------------
* Additional parsing capabilities have been added to the
JSON_DECODE function, including support for arrays
and recursive indexing.
res_phoneprov
------------------
* On multihomed Asterisk servers with dynamic SERVER template variables,
reloading this module is no longer required when re-provisioning your
phone to another interface address (e.g. when moving between VLANs.)
res_pjsip_rfc3326
------------------
* Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
addition to currently supported Q.850). The first header found will be used to set
the HANGUPCAUSE variable.
res_pjsip_session
------------------
* The overlap_context option now allows explicitly
specifying a context to use for overlap dialing matches.
res_rtp_asterisk
------------------
* This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is available using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------

13
UPGRADE.txt
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@ -18,6 +18,19 @@
===
===========================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
------------------------------------------------------------------------------
app_playback
------------------
* In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------

4
doc/CHANGES-staging/app_broadcast.txt
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@ -1,4 +0,0 @@
Subject: app_broadcast
A Broadcast application is now available which allows
for asynchronous one-to-many and many-to-one channel audio.

5
doc/CHANGES-staging/app_directory_skip_call.txt
View File

@ -1,5 +0,0 @@
Subject: app_directory
A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.

5
doc/CHANGES-staging/app_read_return_terminator.txt
View File

@ -1,5 +0,0 @@
Subject: app_read
A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.

5
doc/CHANGES-staging/app_senddtmf_answer.txt
View File

@ -1,5 +0,0 @@
Subject: app_senddtmf
A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.

5
doc/CHANGES-staging/app_signal.txt
View File

@ -1,5 +0,0 @@
Subject: app_signal
Adds Signal and WaitForSignal applications
which can be used for signaling or as a
simple message queue in the dialplan.

5
doc/CHANGES-staging/func_json_additions.txt
View File

@ -1,5 +0,0 @@
Subject: func_json
Additional parsing capabilities have been added to the
JSON_DECODE function, including support for arrays
and recursive indexing.

5
doc/CHANGES-staging/res_phoneprov_multihomed_server.txt
View File

@ -1,5 +0,0 @@
Subject: res_phoneprov
On multihomed Asterisk servers with dynamic SERVER template variables,
reloading this module is no longer required when re-provisioning your
phone to another interface address (e.g. when moving between VLANs.)

4
doc/CHANGES-staging/res_pjsip_session_overlap.txt
View File

@ -1,4 +0,0 @@
Subject: res_pjsip_session
The overlap_context option now allows explicitly
specifying a context to use for overlap dialing matches.

9
doc/CHANGES-staging/res_rtp_asterisk.txt
View File

@ -1,9 +0,0 @@
Subject: res_rtp_asterisk
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is available using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

5
doc/CHANGES-staging/res_rtp_rfc3326_sip.txt
View File

@ -1,5 +0,0 @@
Subject: res_pjsip_rfc3326
Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
addition to currently supported Q.850). The first header found will be used to set
the HANGUPCAUSE variable.

8
doc/UPGRADE-staging/app_playback_playbackstatus.txt
View File

@ -1,8 +0,0 @@
Subject: app_playback
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.