Hi,
I have the following setup:
Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints
With directmedia=no I can make a call between the two SIP endpoints; the RTP
stream being passed through the Asterisk box.
Obviously, this is sub-optimal. I attempted to enable bridging of the call
betwe
Hi,
I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone
behind a different NAT network.
Asterisk -> Nat -> Internet -> Nat -> Softphone.
I can register my softphone to the asterisk box ok via SIP but the RTP
stream from the asterisk box is addressed to the private non-route