I have little issue here that My provider telling me to connect SIP and media on different Ips. that is why they shared me the 2 different ips for signalling and media. So I am confused here how to connect on both ways. If I sending calls only to signalling ip media going blank.
thats probably for your firewall so that you can allow those ips.
the scenario you describe is standard and you should not have to make any changes.
asterisk and you providers exchange information and agree on ips and ports.
that is the part od SDP. you should only configure trunk with the ip where trunk recieves SIP traffic.
Thanks for response. I tried and also disabled the firewall on the server. Even this is 2nd server which I am trying…when I tried free switch I put rtp ip. there and voice worked