I want to setup Asterisk for Phone to IP calls, but not using static routing which could be done easily in extensions.conf. I want dynamic mapping
I had a digium E1 card. All i want is when a user dials my E1, I get the CLI of this user and query some database (mysql e.g) and get a SIP address of another user or multiple user. And finally send an Invite to this SIP address. Can any one tell me where should i start, i mean in which files I have to modify the code. And what is the best way to do this.
Thanx for your response and the code sample.
Helped me a lot. At least it gave a clear direction where to go and what to do. But since im a C programmer I’ll to try to convert this thing into C. and if I coudn’t I’ll let you know ( or most probably use this as such ). Thanking you once again.