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