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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
