[Asterisk-Users] SIP native bridge problem

[Nathan Goodwin]

I'm having a problem, I'm not sure if it has todo with the fact that my 
phone is behind a 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 provider (thus asterisk keeping it's 
self out of the media stream), and this is exactly what I would like to 
have.

But I can't for the life of be figure ot why it's just hanging up once 
the bridge is made.

Does anyone have any ideas how I could fix this, this is sort of 
important, if it's just me because of my NAT causing it, would doing so 
part forwarding and disable NAT support on asterisk and the Sipura fix 
this problem?

I'll welcome any input,
Nathan Goodwin
Diamonleaf Communications LLC