Are the phones on the same subnet as your server's inside NIC? If not, you will need a manual route added or your server will try sending the audio to the internet, when it should be for an inside phone.
Also, verify your DNS. The SIP proxy address on the phone should be pointing to the internal server address. -- -- Steven http://www.glimasoutheast.org "William Piper" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] That did the trick. Thanks for the tip. Interesting though. Although technically it is behind a NAT, it is also connecting with the server who is also behind the NAT, I figured that in the eyes of the server... it would need NAT=no because neither device is connecting to it *through* the NAT. Whatever... thanks a million. bp On 8/15/06, Earl Terwilliger <[EMAIL PROTECTED]> wrote: how about nat=yes qualify=yes canreinvite=no according to: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
