search for: 081da978

...ave been made to believe that it can work... eventually. However, currently SIP call set up seems to go fine, but no media is transferred in either direction. For example, the following is output on the asterisk CLI despite no voice being heard. -- Executing [101 at john:1] Playback('SIP/john-081da978', 'hello-world') in new stack I have used tcpdump, ethereal and RTP debug to trace the problem. I have the following data: 1. My SIP client is correctly receiving and processing SIP/SD packets. Ethereal indicates that the client is told to send RTP traffic to server port 50296 and al...