<span class="gmail_quote"></span>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.
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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>My asterisk setup.<br>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.<br><br>Thanks & Regards<br>