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anable
2005 Jan 29
2
SIP native bridge problem
...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 p...