> 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
