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...