[Freeswitch-users] Outbound Dial Configuration

[Joseph L. Casale]

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