When I send an inbound call to a PSTN extension (my mobile phone, part of a ring group) and pick up the PSTN, the call connects but I get no audio. I'm about 98% sure this is a double NAT issue, as the PBX is behind a firewall. The firewall does have siproxd, which works as a sip proxy at the firewall - so no NAT at all. I'm wondering what I have to change in the PBX to make this work. As best as I can figure asterisk SIP settings: NAT set to "never" In my vitelity trunks (vitelity has you set up separate outbound and inbound trunks) nat=no outboundproxy=[firewall LAN address] What am I missing? Thanks!