[Freeswitch-users] sip trunking question

[Steven Ward]

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.
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