How to make incoming call trunks

All I’d like to implement an incoming call trunks between server A and B.
could someone help me :’(

Consultancy requests should go on the Biz and Jobs forum.

Do you want an analogue, circuit switched digital, or VoIP one. For the first two, do you already have some hardware, and if so what. For the latter, are both servers Asterisk, if so, do you want to use IAX rather than SIP (assuming that you don’t want H.323, etc?

Have you already read and if so, where did you get stuck?

I have 2 asterisk serveur and want to use SIP

The correct way of doing the things that people get wrong are:

  • Use type = peer (on the local devices, as well);
  • Use static addresses;
  • Do not include a register line.

Everything else should be straightforward for anyone who is capable of surviving without paying for support. They are just like mutual ITSP connections with DID, except that you have much more control of the environment.