[Freeswitch-users] sip trunking question

[Michael Collins]

2009/3/16 Steven Ward <steve.d.ward at gmail.com>:
> 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]

What is your network setup? The gateway you created is using the
external profile and trying to do a STUN lookup. Is that what you are
trying to do? Just confirming.

-MC