Hi, I noticed today, that there was a stale SIP call on my 1.2.24 A*k server. One call (X-lite client) started yesterday into a meetme conference. For some reason the call stayed established.
>From network stats, I see transmit data but no receive (as obviously the client went offline). Luckily this won't of cost the company anything, as it was all soft/IP. However if it'd been PSTN based there would've been a cost which concerns me. A SIP SHOW PEER shows that the peer is offline. Is there a flag/setting somewhere to somehow control stale sessions like this? Should A*k not of closed the call down, say X minutes after the peer went offline? A. _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
