motty cruz wrote: > Hello Jonathan, > I thank you for prompt reply to my post. > > I'm using SIP trunks with Polycom sp450 devices. > > > Also, I was wrong to mention meetme, my conference does not involve > using meetme feature on Asterisk. > > > It does not happen often, it happens random.
So if you aren't using MeetMe, what are you using? I'm guessing it's not confbridge (which isn't really well supported in 1.8 anyway) since you probably would have just mentioned it if it were. Is this some sort of service external to Asterisk? If so, you problems are probably more related to the external service than to Asterisk. You could probably check SIP debug to see if they were sending BYEs to your individual phones. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users