At 01:16 PM 6/5/2010, you wrote: > Please can someone help me clear up this mess. I'm completely > frustrated and >don't know what else to do, where else to look.
I've always forwarded port 5060 and all the RTP ports, in my case 16000-16100, directly to my Asterisk box and I've never had problems. For SIP you really need to forward the RTP ports you expect to use and you might as well forward 5060 as that's the only place you want it to go. Or, that's my belief. Try it and see, but find out what RTP ports you need by looking in rtp.conf for the lines: rtpstart=16000 rtpend=16100 I picked those numbers because I thought using the default huge range made no sense and I've never had a problem with my no more than 3 calls at a time world. Ira -- _____________________________________________________________________ -- 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
