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Update CHANGES and UPGRADE.txt for 18.12.0

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18.12
Asterisk Development Team 3 years ago
parent 1e6991f95e
commit efca7f4e8d
12 changed files with 76 additions and 61 deletions
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64
CHANGES
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@ -12,6 +12,70 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.11.3 to Asterisk 18.12.0 ----------
------------------------------------------------------------------------------
app_confbridge
------------------
* Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio file.
app_queue
------------------
* The m option now allows an override music on hold
class to be specified for the Queue application
within the dialplan.
chan_dahdi
------------------
* Previously, cadences were appended on dahdi restart,
rather than reloaded. This prevented cadences from
being updated and maxed out the available cadences
if reloaded multiple times. This behavior is fixed
so that reloading cadences is idempotent and cadences
can actually be reloaded.
chan_pjsip
------------------
* added global config option "allow_sending_180_after_183"
Allow Asterisk to send 180 Ringing to an endpoint
after 183 Session Progress has been send.
If disabled Asterisk will instead send only a
183 Session Progress to the endpoint.
* Hook flash events can now be sent on a PJSIP channel
if requested to do so.
chan_sip
------------------
* Session timers get removed on UPDATE
Fix if Asterisk receives a SIP REFER with Session-Timers UAC
that Asterisk maintains Session-Timers when sending UPDATE request
cli
------------------
* A new CLI command 'dialplan eval function' has been
added which allows users to test the behavior of
dialplan function calls directly from the CLI.
func_db
------------------
* The function DB_KEYCOUNT has been added, which
returns the cardinality of the keys at a specified
prefix in AstDB, i.e. the number of keys at a
given prefix.
func_evalexten
------------------
* This adds the EVAL_EXTEN function which may be
used to evaluate data at dialplan extensions.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.11.1 to Asterisk 18.11.2 ----------
------------------------------------------------------------------------------

12
UPGRADE.txt
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@ -18,6 +18,18 @@
===
===========================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.11.3 to Asterisk 18.12.0 ----------
------------------------------------------------------------------------------
res_pjsip
------------------
* The 'async_operations' setting on transports is no longer
obeyed and instead is always set to 1. This is due to the
functionality not being applicable to Asterisk and causing
excess unnecessary memory usage. This setting will now be
ignored but can also be removed from the configuration file.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.10.0 to Asterisk 18.11.0 ----------
------------------------------------------------------------------------------

8
doc/CHANGES-staging/app_confbridge_hear_join.txt
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@ -1,8 +0,0 @@
Subject: app_confbridge
Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio file.

5
doc/CHANGES-staging/app_queue_music.txt
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@ -1,5 +0,0 @@
Subject: app_queue
The m option now allows an override music on hold
class to be specified for the Queue application
within the dialplan.

8
doc/CHANGES-staging/chan_dahdi_cadences.txt
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@ -1,8 +0,0 @@
Subject: chan_dahdi
Previously, cadences were appended on dahdi restart,
rather than reloaded. This prevented cadences from
being updated and maxed out the available cadences
if reloaded multiple times. This behavior is fixed
so that reloading cadences is idempotent and cadences
can actually be reloaded.

8
doc/CHANGES-staging/chan_pjsip_180_sdp.txt
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@ -1,8 +0,0 @@
Subject: chan_pjsip
added global config option "allow_sending_180_after_183"
Allow Asterisk to send 180 Ringing to an endpoint
after 183 Session Progress has been send.
If disabled Asterisk will instead send only a
183 Session Progress to the endpoint.

4
doc/CHANGES-staging/chan_pjsip_flash.txt
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@ -1,4 +0,0 @@
Subject: chan_pjsip
Hook flash events can now be sent on a PJSIP channel
if requested to do so.

6
doc/CHANGES-staging/chan_sip_session-timer_on_update.txt
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@ -1,6 +0,0 @@
Subject: chan_sip
Session timers get removed on UPDATE
Fix if Asterisk receives a SIP REFER with Session-Timers UAC
that Asterisk maintains Session-Timers when sending UPDATE request

5
doc/CHANGES-staging/cli_eval_function.txt
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@ -1,5 +0,0 @@
Subject: cli
A new CLI command 'dialplan eval function' has been
added which allows users to test the behavior of
dialplan function calls directly from the CLI.

6
doc/CHANGES-staging/func_db.txt
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@ -1,6 +0,0 @@
Subject: func_db
The function DB_KEYCOUNT has been added, which
returns the cardinality of the keys at a specified
prefix in AstDB, i.e. the number of keys at a
given prefix.

4
doc/CHANGES-staging/func_evalexten.txt
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@ -1,4 +0,0 @@
Subject: func_evalexten
This adds the EVAL_EXTEN function which may be
used to evaluate data at dialplan extensions.

7
doc/UPGRADE-staging/res_pjsip_async_operations.txt
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@ -1,7 +0,0 @@
Subject: res_pjsip
The 'async_operations' setting on transports is no longer
obeyed and instead is always set to 1. This is due to the
functionality not being applicable to Asterisk and causing
excess unnecessary memory usage. This setting will now be
ignored but can also be removed from the configuration file.
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