[asterisk-users] PRI failover to SIP trunk

[Steve Totaro]

On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin <jmartin at metrixmatrix.com>wrote:

> Hello,
>
> I've found a little documentation on voip-info and on the asterisk-
> users list, although I was hoping for an example of a tried-and-true
> failover setup between PRI and SIP.
>
> We are an outgoing call center that uses asterisk 1.4 connected to 2
> PRIs from the local telephone company in one group (g1) and a SIP
> trunk from bandwidth.com. The PRIs are the primary outgoing service,
> however we have been experiencing some issues where one or both of
> them can fail randomly. We are working with the telephone company to
> have this resolved.
>
> In the meantime, we want to have a good failover solution where if
> both PRIs fail, asterisk will dial out through the SIP trunk. I've
> found solutions as simple as two Dial commands one after the other,
> and others where the failover Dial is in a  jump to CONGESTION.
> Unfortunately we don't have a testing environment, so the solution
> really has to work.
>
> Does anyone else on the list have a PRI to VoIP failover setup that's
> worked for them in a high volume environment?
>
> Thanks!
>
> Jason Martin
> Metrix Matrix, Inc.
> 785 Elmgrove Rd, Bldg 1
> Rochester, NY 14624
> Office: 888-865-0065 x202
> Mobile: 585-705-1400
>
>
>
>
Simple enough,

exten => _.,1,Dial(Zap,g1,${EXTEN})
exten => _.,2,Dial(SIP,Provider,${EXTEN})

That is if Zap/DAHDI completely craps out.  If the dialplan/Asterisk thinks
it is working it will hang.

If totally out of commission, then the second priority gets called.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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