Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ...
sip.conf extract :
[gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw
extensions.conf extract (from the contact [tlsgw]) :
exten => 57228047,Dial(SIP/cs001,40,tr) ...
Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found
Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context. If this is your default context, Asterisk doesn't connect the incoming call with gw001.
You have host=dynamic - is the gateway registred with Asterisk at all?
/O _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
