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I have several ongoing issues that Avaya have acknoledged as software bugs. I have been playing the caputre dubug and send it off for about a month now, and they keep telling me they will send me the intermin release any day now. I was just wondering if I should just upgrade to 4.0, or will...

This is a very annoying problem. When you hit the dialpage, all phones go off hook, a dialtone is heard, and then you can make your announcment. How can I fix this?

Our IP 406 V2 rebooted today for no apparent reason. The error message in Monitor was: ALARM: 14/05/2007 16:20:17 IP 406 DS 3.2(55) <TLB Data Error> CRIT RAISED addr=000004c3 d=13 pc=ff6a2cb8 00000001 ff8043e8 ff7fc56c ff7fc6cc ff7c33f4 IT also happened almost exactly a month ago. Does...

We are having strange issues where a person makes a page, and another person who is already on a call hears the page on the call, and then the call is dropped. Has anyone ever experienced this before?

When I make a change in Manager to anything, it seems as if for example a user has there folow-me ferature turned on, it turns it off. Also some users are complaining about setting there phone display to show the time, and it eventually goes away. is there an option to prevent this sort of...

I read that the call status application is only able to be running on one PC. Has anyone tested having it running on several places at once? We have 5 receptionists, and they neew to know when someone is already on a call.

Our incoming calls which gace Called-ID setup, only shows the first 4 disits. I have an IPO 406V2 with 16 Port analog trunk module. I am running Ver 5.2(55) code. Calls in the same area code as the trunk lines seem to show up poroperly. Very bizzare.

I have an issue where the system periodically changed its time to 2.5 hours forward. A reboot of the system sets it back. How can I determine where this is coming from, or how can I change the system from accepting time from elsewhere?

All my users have 3 apparence lines. They want to have the ability to send back a busy signal when they are on a call, instead of the next available line ringing? Is this possible?

I have SCN enabled between two sites. One site has extensions in the 200 range the other in the 300 range. I want to be able to have someone who calls in, to be able to key in a users extension in any location. I am using embedded voice mail, and if I do a Normal transfer fron a menu key, like...

We just installed out IPO, and are experiencing dropped calls periodically. I am trying to debug using SysMonitor, but I cannot find documentation on what filters I need on to debug a dropped call. I suspect its on the same analog trunk, but cannot find a common thread. Any help is...