[asterisk-users] PRI failover to SIP trunk

[Steve Totaro]

On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro <stotaro at first-notification.com
> wrote:

>
>
> 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.
>
>
Let me clarify that I think that is how it works.  Been a long time.

Maybe it was the old N+101 trick?  Not sure why that was ever deprecated.

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