I'm trying to set up a sip trunk between a FS and * box, and right now I'm having trouble getting things set up so I make a call from a sip phone registered with my FS box to a sip phone registered w/ my Asterisk box. I have a gateway defined as in directory/default/example.com.xml and in my dialplan I'm trying to do a bridge w/ something like "sofia/gateway/${default_gateway}/12345." When I try to make the call I see from the console: ... New Channel sofia/external/12345 ... ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] The Originate fails. I tried sticking to what the instructions laid out for this in the Connecting FS and Asterisk wiki page, so I'd appreciate some help in figuring out what's going on. Thanks.