I'm still testing to make sure I have all the permutations looked at, but from what I can tell, what is happening is that in situations where stations behind the NAT call out, no audio is passed until after the party on the PUBLIC side generates some audio.
So that means if I call from the public side to one of the NAT boxes, I can't hear them answer. But when (while watching the console) I can see that the call has been bridged, I quickly hail them with a "Hello," then the RTP stream starts going and everyone is happy.
I have the exact same problem using iconnecthere when I call out (to the PSTN) from stations behind NAT: I see the call bridge on the console; my party answers but I don't hear it, nor do they hear me until I say something, and at that point the RTP stream starts up.
This must be evidence of something wrong with the way the initial RTP stream is commenced when SIP stations are behind NAT.
Does anyone know what's going on, or of course better, what I can do to rectify this?
Thanks.
B.
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
