We have some Sip Clients registering on the asterisk box and if they dial a number the call will be forwarded to another Switch where the call will be sent to a carrier.
What we are searching for:
If the users dials a number which is registered as cli from another user on the asterisk box the call should go directly only via the asterisk box to the other sip client.
If the calls will be connected only using the asterisk box there should be voicefile telling it to the calling party.
-) There should be a cdr list including only calls which have been connected.
-) No call rating is required.
-) What happens if one or both user are behind NAT?
-) What happens if the called party is busy?