I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening.
C. Chad Wallace wrote: > At 5:22 PM on 08 May 2008, Forrest Beck wrote: > > >> I have a client that is using the Sangoma A200DE with two phone >> lines attached. >> >> The problem is: >> >> They use their phone (Grandstream GXP2020) to dial out of the system. >> Instead of getting ringing, there is someone on the other end of the >> line that happened to dial in at the exact same moment. >> >> So now they are stuck talking with this person, instead of the one >> the originally called. >> >> The ZAP channels are in a dial plan context that instructs it to >> just dial the office phones. >> >> [zap1] >> exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003) >> exten => s,n,Voicemail([EMAIL PROTECTED]) >> >> Anyone know how to get around this? >> > > This is known in the telephony world as "glare", and there's not much > you can do about it, especially if you only have one line. > > If you have multiple lines on an over-ring (or hunt group or whatever > you call it), the best thing to do is find out which way the telco > assigns calls to those lines wrt how they are assigned to the Asterisk > box. And then allocate outgoing calls in the other direction. > > On our installation, the calls are allocated from the first FXO port > (Zap/25) up. So we set Asterisk to dial out starting from the last FXO > port in the group by calling Dial(Zap/G2) (capital G means dial down > from last, lowercase g means dial up from first). That minimizes glare. > > But, as I said before, if you only have one line, you can't do that... > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users