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Update CHANGES and UPGRADE.txt for 16.26.0

This commit is contained in:
Asterisk Development Team 2022-05-05 09:12:06 -05:00
parent e5dc9e0979
commit 1e3acba443
14 changed files with 90 additions and 73 deletions
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78
CHANGES
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@ -12,6 +12,84 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.25.3 to Asterisk 16.26.0 ----------
------------------------------------------------------------------------------
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_mf
------------------
* Adds an option to ReceiveMF to cap the
number of digits read at a user-specified
maximum.
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.
func_odbc
------------------
* A SQL_ESC_BACKSLASHES dialplan function has been added which
escapes backslashes. Usage of this is dependent on whether the
database in use can use backslashes to escape ticks or not. If
it can, then usage of this prevents a broken SQL query depending
on how the SQL query is constructed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.25.1 to Asterisk 16.25.2 ----------
------------------------------------------------------------------------------

12
UPGRADE.txt
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@ -18,6 +18,18 @@
===
===========================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.25.3 to Asterisk 16.26.0 ----------
------------------------------------------------------------------------------
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 16.24.0 to Asterisk 16.25.0 ----------
------------------------------------------------------------------------------

8
doc/CHANGES-staging/app_confbridge_hear_join.txt
View File

@ -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_mf_maxdigits.txt
View File

@ -1,5 +0,0 @@
Subject: app_mf
Adds an option to ReceiveMF to cap the
number of digits read at a user-specified
maximum.

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

@ -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
View File

@ -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
View File

@ -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
View File

@ -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
View File

@ -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
View File

@ -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
View File

@ -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
View File

@ -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/CHANGES-staging/func_odbc_esc_backslashes.txt
View File

@ -1,7 +0,0 @@
Subject: func_odbc
A SQL_ESC_BACKSLASHES dialplan function has been added which
escapes backslashes. Usage of this is dependent on whether the
database in use can use backslashes to escape ticks or not. If
it can, then usage of this prevents a broken SQL query depending
on how the SQL query is constructed.

7
doc/UPGRADE-staging/res_pjsip_async_operations.txt
View File

@ -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.