Hello !
Your port 5060 or the newer 4849 is just related to SIP messages… You can start testing if your extensions can call each other… answer calls… If these actions are working fine it means that your port 4849 is good. Now you are good to check the RTP side, that cares about voice. First try to check your /asterisk/rtp.conf there find rtpstart and rtpend these parameters show you your RTP range that must be configured in your NAT… If you check and sounds to be good, try to capture tcpdump and check SDP negotiation port… It is a start …