On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote: > Hey Matt, thanks for the response. > > I know it sounds impossible. Hell, I sound like a user :) But it *is* > happening. And only on the cisco phones. We're trying to lab it up > right now. What should I be looking for in the sip debug ?
Just something happening when the call gets cut off. Is there any DTMF being transmitted, why was the call disconnected etc. Or just take a snippet and put it up on pastebin/post here -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _____________________________________________________________________ -- 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
