On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: >What could the reason be audio in 1 direction is dropping? (Normally from >the Asterisk server to the mentioned SIP clients.) No clear information is >in the logs (it is like the call ended normally) and not all calls are >having problem (most not, but it happens to often for us to start using VoIP >more at the moment).
While the most usual problem is packet filtering / NAT, this generally manifests as no audio at all in one direction, not a drop in mid-call. But it's possible that one of the intermediate transit providers is doing something "clever". (Disabling ping, as you mention in your later email, is often a good indicator of a company with insufficient Clue.) Are you in a position to tunnel the traffic over a VPN or similarly flat and unfilterable network link? (This might be a good idea anyway.) Roger -- _____________________________________________________________________ -- 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
