[Freeswitch-users] Outbound Dial Configuration

I am starting to migrate an Asterisk box over with a tdm card w/ 1 fxo port this office uses for redundancy when either their sip provider or net connection drops. In Asterisk, I had a long macro for attempting the sip providers first, then finally getting to the dahdi line. Can I simply do the following: <action application="bridge" data="sofia/gateway/sip.example.com/$1|openzap/1/1/$1"/> to attempt my sip provider first always, then hit span 1/port 1 of my tdm card if it's the provider is not available? Is this an elegant enough way to do this for an office of < 10 phones? Thanks, jlc