I'm using the script from http://lists.iptel.org/pipermail/serusers/2004-February/005996.html (the first ser.cfg). I'm trying to figure out where to include my SIP outgoing calls to my Asterisk PSTN server. For example, let's say I want to transfer a SIP call for a local XXX-XXXX number to pstn.xyz.com port 5050. How would I set that up? Additionally, I'm rather confused as to which section I would put a dialplan into for say connecting to FWD users, etc. Again, like before if I wanted to send a certain call to a certain termination point, let's say 407-xxx-xxxx, I want to send that to one IP, and 954-xxx-xxxx I want to sent to a different one. Where are those statements supposed to be? Thanks a lot... _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers