I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops.
I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. Kind Regards, Chris -- _____________________________________________________________________ -- 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
