From: Kevin P. Fleming <kpfleming@digium.com>
Date: Fri, 13 Feb 2009 13:41:52 +0000 (+0000)
Subject: document G.722.1/.1C support
X-Git-Tag: 1.6.2.0-beta1~235
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=3854faf2d75bb08c922f2314aa020f84bb168eb4;p=thirdparty%2Fasterisk.git

document G.722.1/.1C support



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

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
 --------------------------