Site-to-site sip in asterisk

Is it possible to implement site-to-site sip in asterisk?

Also is it possible to send calls to asterisk servers from provider without registering the peer?

Please suggest.

What do you mean by site-to-site exactly ? You can have sip/iax trunk between asterisk servers to send and receive calls .

Registering is just a way of letting the provider discover the address of the Asterisk server. If the address is configured into the provider’s system by other means, they don’t need to register. I doubt there are many providers tht will work that way, though.

Tie line connections are trivial. Although, in that case, you are more likely to have a static address, so not need to use register.

Thanks for the prompt reply David.

My provider didn’t provided any sip account to register. He has shared only the static ip from which he will send INVITES.

From my side, I have natted my local server to static IP on UDP port 5060 and RTP Ports opened.

Now when he start sending INVITE, i am unable to receive it at my end.

Please suggest.

You will need to run with “sip set debug on”. If you don’t see any requests there, you will need to trace back within your network to see where they are getting lost.

Also, make sure that you are configured for NAT use. That is not the nat= parameter, but rather externip, externhost or stunaddr.

i am yet to test the scenario with my provider.

In between i had another issue in playing a prompt in my xlite.

i am playing demo-congrats file in my extensions.conf but unable to hear it.

it happens only when i register my sip user with static ip.

Please suggest.

sip set debug on

Solved this issue.

Since i was using two ip’s on the same machine, we were not getting the invites from far end.

after adding route on my machine for destination, calls landed into our server.

Thanks for all the support.