Patrick Baker wrote:
and how would they be billed... sip to sip would be billed thru ser, all zaptel channels thru asterisk??
Now, I have the same problems...
1) My users are logging in with login and password. And they have numbers for sip-calls (aliases). Number to name resolving is done by agi-script at asterisk part. How can I setup conversion at SER level?
2) Billing... hoq can i bill local calls? If I'll call external script for invite, what I'll need to do if BYE packet will be lost for some reason?