Greetings, This week I have been trying to find a solution how to make my Asterisk IP PBX box to work behind cisco router with NAT.
I currently have problems with incomming calls after I have registered it with my sip provider. The sip registration works fine. When the call enters the linux (from internet) my ip pbx does pick up the call but no sound was passing thru. It seems like that the rtp ports arent passing thru.
Could this problem be solved with "ip inspect sip" ? Or what else should I try?
Thanks, Alex