If Asterisk registers with a SIP long distance provider and I make a call from an IP phone through Asterisk to that LD provider, does the RTP (audio) traffic flow between the two end points directly (normally the IP phone and the LD provider) or does it flow through Asterisk?
I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
