[Freeswitch-users] Outbound Dial Configuration

[Rupa Schomaker]

Yes.

On Thu, Dec 31, 2009 at 2:17 AM, Joseph L. Casale
<jcasale at activenetwerx.com> wrote:
> 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
>
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-- 
-Rupa