On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien Cramatte<[email protected]> wrote: > Hi, > > This sounds udp RTP problem. > Might be you have some firewall rules that block this kind of traffic ? > As soon I remember, Asterisk by default use random port between 10000 > and 20000 for rtp traffic (you can adjust this in rtp.conf).
In theory, there should be no firewalls between my asterisk server and the remote phones. I've opened a ticket with ATT with that exact question, as well as a question of rather any NATing is going on, though, I doubt this is the case, and this is the first time this type of problem has happened in over 4 years. The idea of RTP being to blame would make sense though. I can still transfer and such, and watching the console, I see when I press various keys on the phone, so it seems that the SIP traffic is working out fine. (I do understand that right? SIP == control RTP == voice in a very generic sense?) I plan to take a packet trace in the morning on the asterisk server and see what is going on at that level. Hints as to what I should be looking for? -jonathan _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
