Heh, should have guessed it would be you that replied Gareth ;) Sorry yes, this box is on public IP with no NAT as is the upstream providers box (or so they say).
So we have had audio cease outbound towards the provider. We have a couple of volunteer customers who are being routed via this new test upstream. It's very difficult (basically impossible!) to replicate the failure it's so infrequent. Looking at PCAPs between us and the upstream we stop sending them audio for example and then a little while later the call drops. Without PCAPs between us and the customer at the same time I can't say why we stopped sending audio (where we receiving any from the customer, did their connection drop for example). We have also had the reverse where we stop receiving audio then a short period later, SIP BYE from us to them! I have read up on rtpkeepalive and rtptimeout. I will put this to one side for now until we have a direct connect to the new test provider there are to many variables in the equation. Thanks for your input though Gareth!. Kind regards, James. -- _____________________________________________________________________ -- 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