Update CHANGES and UPGRADE.txt for 18.12.0

CHANGES

@@ -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 ----------
 ------------------------------------------------------------------------------

UPGRADE.txt

@@ -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 ----------
 ------------------------------------------------------------------------------

doc/CHANGES-staging/app_confbridge_hear_join.txt

@@ -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.

doc/CHANGES-staging/app_queue_music.txt

@@ -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.

doc/CHANGES-staging/chan_dahdi_cadences.txt

@@ -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.

doc/CHANGES-staging/chan_pjsip_180_sdp.txt

@@ -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.

doc/CHANGES-staging/chan_pjsip_flash.txt

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

doc/CHANGES-staging/chan_sip_session-timer_on_update.txt

@@ -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
-

doc/CHANGES-staging/cli_eval_function.txt

@@ -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.

doc/CHANGES-staging/func_db.txt

@@ -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.

doc/CHANGES-staging/func_evalexten.txt

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

doc/UPGRADE-staging/res_pjsip_async_operations.txt

@@ -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.