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document G.722.1/.1C support

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Kevin P. Fleming
2009-02-13 13:41:52 +00:00
parent 2a53f2ec98
commit 3854faf2d7
1 changed files with 3 additions and 0 deletions
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@@ -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
--------------------------