From 3854faf2d75bb08c922f2314aa020f84bb168eb4 Mon Sep 17 00:00:00 2001
From: "Kevin P. Fleming" <kpfleming@digium.com>
Date: Fri, 13 Feb 2009 13:41:52 +0000
Subject: [PATCH] document G.722.1/.1C support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 CHANGES | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/CHANGES b/CHANGES
index e8fb6146a4..74e07b6407 100644
--- a/CHANGES
+++ b/CHANGES
@@ -48,6 +48,7 @@ SIP Changes
    first INVITE is generated - SIPRemoveHeader()
  * Channel variables set with setvar= in a device configuration is now 
    set both for inbound and outbound calls.
+ * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
 
 Skinny Changes
 --------------
@@ -135,6 +136,8 @@ Miscellaneous
  * The contrib/scripts/ directory now has a script called sip_nat_settings that will
    give you the correct output for an asterisk box behind nat. It will give you the
    externhost and localnet settings.
+ * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
+   can connect calls in passthrough mode, as well as record and play back files.
 
 Asterisk Manager Interface
 --------------------------
-- 
GitLab