document G.722.1/.1C support

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175512 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 3 insertions, 0 deletions

diff --git a/CHANGES b/CHANGES
index e8fb6146a..74e07b640 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
 --------------------------