[Freeswitch-users] SIP dialout problems

[Chris Danielson]

Nicolas,
What you have for your bridging should work fine.  Here is how I 
performed a bridge on my system.
    <action application="bridge" 
data="sofia/gateway/sip.vonics.net/$1 at sip.vonics.net" />
-Chris


Nicolas Brenner wrote:
> 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
>
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