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...NAT or not, but here it is.. My problem is when I call out, my asterisk system routes the call to my SIP provider, whoever, as soon as the other party answers, asterisk tries to make a native bridge for the call, and then the call drops instantly. However, if I keep asterisk in the middle (by anyable transfers), no bridge is made and the call stays just fine. My setup is so: Sipura-2000 -> NAT (Netgear router) -> cable/internet -> colocated asterisk server -> SIP provider The native bride I assume is asterisk trying to connect the RTP stream directly from the Sipura to my SIP...