[Freeswitch-users] SIP dialout problems

[Nicolas Brenner]

Chris,

Thanks for your help. Actually I first configured my Gizmo account
like you say, following the instructions from here:
http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing

But that doesn't seem right to me, because what I want to do is have a
'SIP trunk', not a SIP user, and for what I understand, the directory
configuration is for setting up user accounts. That's why I set up the
sip_profile instead of the user in the directory.

Anyway, I'm going to try out your suggestion and see how it goes. How
should I bridge/route the call? I mean, what should I replace <action
application="bridge" data="sofia/gateway/gizmo/0056$1"/> for?

Thanks!

Nicolas