From 0afd37e3b500802dfd8b921d5c79f46fd2b0ef4d Mon Sep 17 00:00:00 2001
From: Asterisk Development Team <asteriskteam@digium.com>
Date: Thu, 11 Mar 2021 11:40:29 -0500
Subject: [PATCH] Update CHANGES and UPGRADE.txt for 18.3.0
---
CHANGES | 53 +++++++++++++++++++
UPGRADE.txt | 14 +++++
doc/CHANGES-staging/app_transferprotocol.txt | 6 ---
doc/CHANGES-staging/chan_iax2.txt | 4 --
.../func_odbc_ARGC_minargs.txt | 20 -------
.../mixmonitor_manager_events.txt | 5 --
.../srtp_replay_protection.txt | 9 ----
.../srtp_replay_protection.txt | 9 ----
8 files changed, 67 insertions(+), 53 deletions(-)
delete mode 100644 doc/CHANGES-staging/app_transferprotocol.txt
delete mode 100644 doc/CHANGES-staging/chan_iax2.txt
delete mode 100644 doc/CHANGES-staging/func_odbc_ARGC_minargs.txt
delete mode 100644 doc/CHANGES-staging/mixmonitor_manager_events.txt
delete mode 100644 doc/CHANGES-staging/srtp_replay_protection.txt
delete mode 100644 doc/UPGRADE-staging/srtp_replay_protection.txt
diff --git a/CHANGES b/CHANGES
index 178fb8a2be..460f6f5af1 100644
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,59 @@
===
==============================================================================
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
+------------------------------------------------------------------------------
+
+app_mixmonitor
+------------------
+ * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
+ MixMonitorMute when the channel monitoring is started, stopped and muted (or
+ unmuted) respectively.
+
+chan_iax2
+------------------
+ * You can now specify a default "auth" method in the
+ [general] section of iax.conf
+
+chan_pjsip, app_transfer
+------------------
+ * Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
+ transfers can pass a protocol specific error code.
+ Example, in SIP 3xx-6xx represent any SIP specific error received when
+ performing a REFER.
+
+func_odbc
+------------------
+ * Introduce an ARGC variable for func_odbc functions, along with a minargs
+ per-function configuration option.
+
+ minargs enables enforcing of minimum count of arguments to pass to
+ func_odbc, so if you're unconditionally using ARG1 through ARG4 then
+ this should be set to 4. func_odbc will generate an error in this case,
+ so for example
+
+ [FOO]
+ minargs = 4
+
+ and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
+ potentially leaked ARG4 from Gosub().
+
+ ARGC is needed if you're using optional argument, to verify whether or
+ not an argument has been passed, else it's possible to use a leaked ARGn
+ from Gosub (app_stack). So now you can safely do
+ ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
+
+res_srtp
+------------------
+ * SRTP replay protection has been added to res_srtp and
+ a new configuration option "srtpreplayprotection" has
+ been added to the rtp.conf config file. For security
+ reasons, the default setting is "yes". Buggy clients
+ may not handle this correctly which could result in
+ no, or one way, audio and Asterisk error messages like
+ "replay check failed".
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.1.0 to Asterisk 18.2.0 ------------
------------------------------------------------------------------------------
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 68261ae716..7db926f1b5 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -18,6 +18,20 @@
===
===========================================================
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
+------------------------------------------------------------------------------
+
+res_srtp
+------------------
+ * SRTP replay protection has been added to res_srtp and
+ a new configuration option "srtpreplayprotection" has
+ been added to the rtp.conf config file. For security
+ reasons, the default setting is "yes". Buggy clients
+ may not handle this correctly which could result in
+ no, or one way, audio and Asterisk error messages like
+ "replay check failed".
+
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------
diff --git a/doc/CHANGES-staging/app_transferprotocol.txt b/doc/CHANGES-staging/app_transferprotocol.txt
deleted file mode 100644
index 5d3521bbd4..0000000000
--- a/doc/CHANGES-staging/app_transferprotocol.txt
+++ /dev/null
@@ -1,6 +0,0 @@
-Subject: chan_pjsip, app_transfer
-
-Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
-transfers can pass a protocol specific error code.
-Example, in SIP 3xx-6xx represent any SIP specific error received when
-performing a REFER.
diff --git a/doc/CHANGES-staging/chan_iax2.txt b/doc/CHANGES-staging/chan_iax2.txt
deleted file mode 100644
index 4e1d844204..0000000000
--- a/doc/CHANGES-staging/chan_iax2.txt
+++ /dev/null
@@ -1,4 +0,0 @@
-Subject: chan_iax2
-
-You can now specify a default "auth" method in the
-[general] section of iax.conf
diff --git a/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt b/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt
deleted file mode 100644
index 0984b5022d..0000000000
--- a/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt
+++ /dev/null
@@ -1,20 +0,0 @@
-Subject: func_odbc
-
-Introduce an ARGC variable for func_odbc functions, along with a minargs
-per-function configuration option.
-
-minargs enables enforcing of minimum count of arguments to pass to
-func_odbc, so if you're unconditionally using ARG1 through ARG4 then
-this should be set to 4. func_odbc will generate an error in this case,
-so for example
-
-[FOO]
-minargs = 4
-
-and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
-potentially leaked ARG4 from Gosub().
-
-ARGC is needed if you're using optional argument, to verify whether or
-not an argument has been passed, else it's possible to use a leaked ARGn
-from Gosub (app_stack). So now you can safely do
-${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
diff --git a/doc/CHANGES-staging/mixmonitor_manager_events.txt b/doc/CHANGES-staging/mixmonitor_manager_events.txt
deleted file mode 100644
index 64b63e52e7..0000000000
--- a/doc/CHANGES-staging/mixmonitor_manager_events.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-Subject: app_mixmonitor
-
-app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
-MixMonitorMute when the channel monitoring is started, stopped and muted (or
-unmuted) respectively.
diff --git a/doc/CHANGES-staging/srtp_replay_protection.txt b/doc/CHANGES-staging/srtp_replay_protection.txt
deleted file mode 100644
index 945ddb5704..0000000000
--- a/doc/CHANGES-staging/srtp_replay_protection.txt
+++ /dev/null
@@ -1,9 +0,0 @@
-Subject: res_srtp
-
-SRTP replay protection has been added to res_srtp and
-a new configuration option "srtpreplayprotection" has
-been added to the rtp.conf config file. For security
-reasons, the default setting is "yes". Buggy clients
-may not handle this correctly which could result in
-no, or one way, audio and Asterisk error messages like
-"replay check failed".
diff --git a/doc/UPGRADE-staging/srtp_replay_protection.txt b/doc/UPGRADE-staging/srtp_replay_protection.txt
deleted file mode 100644
index 945ddb5704..0000000000
--- a/doc/UPGRADE-staging/srtp_replay_protection.txt
+++ /dev/null
@@ -1,9 +0,0 @@
-Subject: res_srtp
-
-SRTP replay protection has been added to res_srtp and
-a new configuration option "srtpreplayprotection" has
-been added to the rtp.conf config file. For security
-reasons, the default setting is "yes". Buggy clients
-may not handle this correctly which could result in
-no, or one way, audio and Asterisk error messages like
-"replay check failed".
--
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