On Sat, 2005-11-19 at 19:13 -0800, snacktime wrote: > If we can't do this then we will probably just have a password > protected voice menu they can call into via an 800 number. That has > some advantages anyways, but I personally think it would look a lot > better if their voip phone acted like it was a local extension to our > own asterisk server. > > Any thoughts on how we might accomplish this? The simpler the better.
You could have everyone register with your asterisk box, then directly connect the media streams (reinvite=yes or notransfer=no sip or iax respectively) to the actual ITSP. That saves you the bandwidth, reduces customer latency (usually), and overall gives you what you want. If asterisk is the endpoint of the call (ie they are accessing a menu local to your system) then the media stream obviously would terminate with your box and not be redirected. If howevr they call the PSTN it would go direct where it has to go, your partner ITSP. To reduce your costs you can hook into a dundi cloud and/or try enum.org to directly terminate voip->voip where possible. Does this setup do exactly what you want it to do? Or did I miss anything? CAUTION: if you connect the two end points directly there is *no* guarantee that you will have proper CDR records. You may but there is no guarantee. This is because once you are out of the loop things can happen that prevent any 'end of call' messages from ever being sent to your asterisk box. To that end you may want to partner with a provider that will provide you with their CDR records for your calls and insert that into your database locally (either by a deadAGI callback to your box to report on that call specifically or better by fetching the CDR data at periodic intervals). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group
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