That is exactly what I said in my post, the first line says that the static can be heard on standard analog phones plugged directly into the line. I will check /proc/interrupts, but as I already stated, the lines have very poor quality as tested with standard analog phones. However, the calls don't drop as they do with the fxo card. how would I go about getting the cards on different interupts if they are on the same one? Tom
Quoting Rich Adamson <[EMAIL PROTECTED]>: > > I just installed a new asterisk box with a wctdm with 4 FXO modules. The > lines > > in the office have terrible static (using standard analog phones) and this > > static can obviously be heard through the asterisk box on the sipura sip > phones > > we installed. This by itself would not be a problem as the office is used > to > > and doesn't mind (I don't know how) the static. > > > > However it appears that this really bad line quality is causing the fxo > ports to > > drop calls. We tested all of the FXO ports in our office before we took > the > > box to install it, and it worked just fine... Here are the problems we are > > seeing: > > > > 1) Incoming calls, although immediate=no is set in zapata.conf the caller > hears > > one ring, and then when asterisk starts the simple switch, the caller hears > > static and dead air, as if asterisk had done an "answer()". The caller > doesn't > > hear any more rings. It takes asterisk about 3 seconds before it even > rings > > the internal sip phone, and then while the sip phone is ringing, until it > is > > answered the caller hears static and dead air. It seems as if the call has > been > > disconnected, or at least it will be very confusing for the customers of > this > > business, at any rate its unacceptable. > > > > 2) Outgoing and incoming calls: call quality is bad because of the static, > but > > randomly the zap channel that the call is on will hang up even though > neither > > side has hung up. It seems like the poor line quality is somehow > simulating a > > "hangup" signal from the CO, and the fxo line is dropping the call. > > > > has anyone seen poor line quality cause the digium fxo modules to have > strange > > errors such as these? > > > > Thanks in advance for any replies/ideas/solutions (besides obviously > calling the > > phone company and telling them they suck) > > >From your description, it sounds more like a shared interrupt problem > (cat /proc/interrupts) then it does a pstn line problem. > > If it really is a pstn line problem, then plug the line into a ordinary > analog phone set and listen. If the pstn line is bad, you'll hear > the same noise on the analog set. > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
