>>> my dial plan is like callers dial into asterisk(span1) , hear an IVR >>> option and they are connected to the agents via the legacy pbx (which is in >>> sync with asterisk on span2)....This works perfectly fine until about 200 >>> calls or so...After that time when asterisk tries to dial to the legacy pbx >>> - the call drops with error "All are busy congested at this time" .the same >>> is indicated on asterisk -rvvvvvvvvvv , but the spans are up and active at >>> that time... can anyone throw some light on this ?
Sounds like you have some kind of call limit on your legacy pbx side. You'll need to get a consultant for your legacy pbx to tell you why 200 is a magic number. I've run into all kinds of problems with getting asterisk to talk to Inter-tel. Found all kinds of buried limits, which weren't documented, but we discovered when the Inter-Tel would act up when a certain number of calls would go into that part of the system. The worst limits would crash large parts of the Inter-Tel and drop all calls. Not a very nice way to discover bugs in a proprietary PBX. Be glad you at least get a busy, which you can check for with Asterisk, and route around. Our solution has been to add hardware to the Inter-Tel only when absolutely necessary, and do everything possible to keep calls in Asterisk and only send those calls into Inter-Tel after screening with IVRs, making them queue, etc. I think a similar strategy is what you should consider. And of course, our overall strategy is to dump the Inter-Tel as soon as possible, while we're migrating out various pieces as we run into scaling limits. Everybody I've talked to says "name your proprietary PBX" has issues as well, making the name brand in your situation largely irrelevant. _______________________________________________ -- 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
