[Serusers] SIP to SIP calls & Asterisk

[Alexey N. Kovyrin @ Home]

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?

-- 
/Scoundrel