Update CHANGES and UPGRADE.txt for 20.1.0

CHANGES

@@ -12,6 +12,112 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * The AOCMessage action can now be used to generate AOC-S messages.
+
+Add support for named capture agent.
+------------------
+ * A name for the capture agent can now be specified
+   using the capture_name option which, if specified,
+   will be sent to the HEP server.
+
+app_if
+------------------
+ * Adds the If, ElseIf, Else, EndIf, and ExitIf applications
+   for conditional execution of a block of code.
+
+app_mixmonitor
+------------------
+ * The d option for MixMonitor now allows deleting
+   the original recording when MixMonitor exits,
+   which can be useful when MixMonitor copies it
+   somewhere else before exiting.
+
+ * Adds the c option to use the real Caller ID on
+   the channel in voicemail recordings as opposed
+   to the Connected Line.
+
+app_voicemail
+------------------
+ * The voicemail user option attachextrecs can
+   now be set to control whether external recordings
+   trigger voicemail email notifications.
+
+cdr
+------------------
+ * Two new options have been added which allow
+   bridging and dial state changes to be ignored
+   in CDRs, which can be useful if a single CDR
+   is desired for a channel.
+
+chan_dahdi
+------------------
+ * FXO channels (FXS signaled) that don't use callerid or
+   distinctive ring detection can now be configured
+   to enter the dialplan immediately using immediate=yes,
+   instead of waiting for at least one ring.
+
+pbx_builtins
+------------------
+ * It is now possible to not wait for media on
+   a channel when answering it using Answer,
+   by specifying the i option.
+
+res_pjsip
+------------------
+ * Added options "security_negotiation" and "security_mechanisms" to pjsip
+   endpoints and registrations. "security_negotiation" can be set to "no" (default)
+   or "mediasec", and "security_mechanisms" can be a list of comma-separated
+   security_mechanisms in the form defined by RFC 3329 section 2.2.
+
+ * A new option named "all_codecs_on_empty_reinvite" has been added to the
+   global section. When this option is enabled, on reception of a re-INVITE
+   without SDP, Asterisk will send an SDP offer in the 200 OK response containing
+   all configured codecs on the endpoint, instead of simply those that have
+   already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
+   The default value is "off".
+
+res_pjsip_aoc
+------------------
+ * Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
+   A new endpoint option, send_aoc, controls this.
+
+res_pjsip_header_funcs
+------------------
+ * The new PJSIP_HEADER_PARAM function now fully supports both
+   URI and header parameters. Both reading and writing
+   parameters are supported.
+
+res_pjsip_logger
+------------------
+ * SIP messages can now be filtered by SIP request method
+   (INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
+   SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
+   allowing for more granular debugging to be done
+   in the CLI. This applies to requests but not responses.
+
+res_pjsip_notify
+------------------
+ * Allows using the config options in pjsip_notify.conf
+   from AMI actions as with the existing CLI commands.
+
+res_tonedetect
+------------------
+ * The TONE_DETECT function now supports
+   detection of audible ringback tone
+   using the p option.
+
+xmldocs
+------------------
+ * The XML documentation can now be reloaded without restarting
+   Asterisk, which makes it possible to load new modules that
+   enforce documentation without restarting Asterisk.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
 ------------------------------------------------------------------------------

UPGRADE.txt

@@ -18,6 +18,19 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
+------------------------------------------------------------------------------
+
+AMI (Asterisk Manager Interface)
+------------------
+ * Previously, GetConfig and UpdateConfig were able to access files outside of
+   the Asterisk configuration directory. Now this access is put behind the
+   live_dangerously configuration option in asterisk.conf, which is disabled by
+   default. If access to configuration files outside of the Asterisk configuation
+   directory is required via AMI, then the live_dangerously configuration option
+   must be set to yes.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
 ------------------------------------------------------------------------------

doc/CHANGES-staging/answer.txt

@@ -1,5 +0,0 @@
-Subject: pbx_builtins
-
-It is now possible to not wait for media on
-a channel when answering it using Answer,
-by specifying the i option.

doc/CHANGES-staging/app_if.txt

@@ -1,4 +0,0 @@
-Subject: app_if
-
-Adds the If, ElseIf, Else, EndIf, and ExitIf applications
-for conditional execution of a block of code.

doc/CHANGES-staging/app_mixmonitor_clid.txt

@@ -1,5 +0,0 @@
-Subject: app_mixmonitor
-
-Adds the c option to use the real Caller ID on
-the channel in voicemail recordings as opposed
-to the Connected Line.

doc/CHANGES-staging/app_mixmonitor_delete.txt

@@ -1,6 +0,0 @@
-Subject: app_mixmonitor
-
-The d option for MixMonitor now allows deleting
-the original recording when MixMonitor exits,
-which can be useful when MixMonitor copies it
-somewhere else before exiting.

doc/CHANGES-staging/app_voicemail_attachext.txt

@@ -1,5 +0,0 @@
-Subject: app_voicemail
-
-The voicemail user option attachextrecs can
-now be set to control whether external recordings
-trigger voicemail email notifications.

doc/CHANGES-staging/cdr_ignore.txt

@@ -1,6 +0,0 @@
-Subject: cdr
-
-Two new options have been added which allow
-bridging and dial state changes to be ignored
-in CDRs, which can be useful if a single CDR
-is desired for a channel.

doc/CHANGES-staging/fxo_immediate.txt

@@ -1,6 +0,0 @@
-Subject: chan_dahdi
-
-FXO channels (FXS signaled) that don't use callerid or
-distinctive ring detection can now be configured
-to enter the dialplan immediately using immediate=yes,
-instead of waiting for at least one ring.

doc/CHANGES-staging/manager_aoc.txt

@@ -1,3 +0,0 @@
-Subject: AMI
-
-The AOCMessage action can now be used to generate AOC-S messages.

doc/CHANGES-staging/res_hep.txt

@@ -1,5 +0,0 @@
-Subject: Add support for named capture agent.
-
-A name for the capture agent can now be specified
-using the capture_name option which, if specified,
-will be sent to the HEP server.

doc/CHANGES-staging/res_pjsip_all_codecs_on_empty_reinvite_option.txt

@@ -1,8 +0,0 @@
-Subject: res_pjsip
-
-A new option named "all_codecs_on_empty_reinvite" has been added to the
-global section. When this option is enabled, on reception of a re-INVITE
-without SDP, Asterisk will send an SDP offer in the 200 OK response containing
-all configured codecs on the endpoint, instead of simply those that have
-already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
-The default value is "off".

doc/CHANGES-staging/res_pjsip_aoc.txt

@@ -1,4 +0,0 @@
-Subject: res_pjsip_aoc
-
-Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
-A new endpoint option, send_aoc, controls this.

doc/CHANGES-staging/res_pjsip_logger_method.txt

@@ -1,7 +0,0 @@
-Subject: res_pjsip_logger
-
-SIP messages can now be filtered by SIP request method
-(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
-SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
-allowing for more granular debugging to be done
-in the CLI. This applies to requests but not responses.

doc/CHANGES-staging/res_pjsip_notify_options.txt

@@ -1,4 +0,0 @@
-Subject: res_pjsip_notify
-
-Allows using the config options in pjsip_notify.conf
-from AMI actions as with the existing CLI commands.

doc/CHANGES-staging/res_pjsip_parameters.txt

@@ -1,5 +0,0 @@
-Subject: res_pjsip_header_funcs
-
-The new PJSIP_HEADER_PARAM function now fully supports both
-URI and header parameters. Both reading and writing
-parameters are supported.

doc/CHANGES-staging/res_pjsip_rfc3329.txt

@@ -1,6 +0,0 @@
-Subject: res_pjsip
-
-Added options "security_negotiation" and "security_mechanisms" to pjsip
-endpoints and registrations. "security_negotiation" can be set to "no" (default)
-or "mediasec", and "security_mechanisms" can be a list of comma-separated
-security_mechanisms in the form defined by RFC 3329 section 2.2.

doc/CHANGES-staging/res_tonedetect_ring.txt

@@ -1,5 +0,0 @@
-Subject: res_tonedetect
-
-The TONE_DETECT function now supports
-detection of audible ringback tone
-using the p option.

doc/CHANGES-staging/xmldoc.txt

@@ -1,5 +0,0 @@
-Subject: xmldocs
-
-The XML documentation can now be reloaded without restarting
-Asterisk, which makes it possible to load new modules that
-enforce documentation without restarting Asterisk.

doc/UPGRADE-staging/manager_config_live_dangerously.txt

@@ -1,8 +0,0 @@
-Subject: AMI (Asterisk Manager Interface)
-
-Previously, GetConfig and UpdateConfig were able to access files outside of
-the Asterisk configuration directory. Now this access is put behind the
-live_dangerously configuration option in asterisk.conf, which is disabled by
-default. If access to configuration files outside of the Asterisk configuation
-directory is required via AMI, then the live_dangerously configuration option
-must be set to yes.