<br><br><div class="gmail_quote">On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@first-notification.com">stotaro@first-notification.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br><br><div class="gmail_quote"><div><div></div><div class="h5">On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin <span dir="ltr"><<a href="mailto:jmartin@metrixmatrix.com" target="_blank">jmartin@metrixmatrix.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br>
<br>
I've found a little documentation on voip-info and on the asterisk-<br>
users list, although I was hoping for an example of a tried-and-true<br>
failover setup between PRI and SIP.<br>
<br>
We are an outgoing call center that uses asterisk 1.4 connected to 2<br>
PRIs from the local telephone company in one group (g1) and a SIP<br>
trunk from <a href="http://bandwidth.com" target="_blank">bandwidth.com</a>. The PRIs are the primary outgoing service,<br>
however we have been experiencing some issues where one or both of<br>
them can fail randomly. We are working with the telephone company to<br>
have this resolved.<br>
<br>
In the meantime, we want to have a good failover solution where if<br>
both PRIs fail, asterisk will dial out through the SIP trunk. I've<br>
found solutions as simple as two Dial commands one after the other,<br>
and others where the failover Dial is in a jump to CONGESTION.<br>
Unfortunately we don't have a testing environment, so the solution<br>
really has to work.<br>
<br>
Does anyone else on the list have a PRI to VoIP failover setup that's<br>
worked for them in a high volume environment?<br>
<br>
Thanks!<br>
<br>
Jason Martin<br>
Metrix Matrix, Inc.<br>
785 Elmgrove Rd, Bldg 1<br>
Rochester, NY 14624<br>
Office: 888-865-0065 x202<br>
Mobile: 585-705-1400<br>
<br>
<br>
<br>
</blockquote></div></div><div><br>Simple enough,<br><br>exten => _.,1,Dial(Zap,g1,${EXTEN})<br>exten => _.,2,Dial(SIP,Provider,${EXTEN})<br><br>That is if Zap/DAHDI completely craps out. If the dialplan/Asterisk thinks it is working it will hang.<br>
<br>If totally out of commission, then the second priority gets called. <br></div></div><font color="#888888"><br></font></blockquote><div><br>Let me clarify that I think that is how it works. Been a long time.<br><br>Maybe it was the old N+101 trick? Not sure why that was ever deprecated. <br>
</div></div><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>