On Fri, 2003-07-11 at 06:40, Simon Woodhead wrote: > >Our problem was that we all of a sudden would get dropped audio, and I > >had one user complain of extreme lag occasionally. I didn't have anyone > >else experience the lag, but the dropped audio would come and go. It > >sometimes would drop out for a second or so. Sound quality when there > >was still just perfect. > > >For your link to the Pace Vega Stream, what codec are you using? I would > >assume it would be more of a problem in codec shifting bits or > >something, but then again this is a wild guess. > > Thanks for that. We're using G.729 over H.323, incoming and outgoing. > Outgoing works perfectly but on incoming we get the underwater sound > periodically. It clicks in randomly but once there the only way to clear it > is to end the call and try again. > > One thing we have thought of is co-loing an * box directly at the Telco and > plugging in to their switch directly. We'd then be in control of the VoIP > part and know that over IAX it would work fine. Can anyone enlighten me as > to how we'd connect to them physically on-site? Would it be a PRI or would > there be a different method as the PSTN wouldn't be between us and their > switch?
You would still use PRI if you need bulk lines. You could use channelized T1, but you get a lot more options with PRI. Currently our phone server is in our colo rack and our phone lines are sent down to us via our data T1 line. -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
