On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro <stot...@first-notification.com > wrote:
> > > On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin <jmar...@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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