[asterisk-users] How to make RTP does not go thru asterisk server

[Anuj Jain]

Hi All
I am using trixbox asterisk 1.2
I have enabled canreinvite=yes and no  "tT" in the dialplan as it has been
described in the various forums.
Still the voice call goes thru the asterisk server.
How can i really make the call between 2 grandstream devices( i am using HT
488, HT286 and SIP extensions) after the initial handshake.

Thanks & Regards
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