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.
My asterisk setup.
Asterisk server is in one network and my 2 grandstreams are in 2 different networks and behind the firewall and natting enabled, all of them are connected thru the internet.
Thanks & Regards
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