The problem is * not supporting or handling early media. I have looked through the sniffer traces and I see the RTP stream being setup between * and the gateway during the invite and or 183 message, but * does not setup a corresponding stream to the client until it sees an OK (200) message. The result is the end user never hears ringing, although the call is completed. I have looked trough the messages, we, wiki and there seems to only be a few messages on this subject. I have tried to "fake" the ringing by adding "r" to the dial command, but it does not seem like a good solution. It works great in the case where there is little chance of getting a busy signal or when the called party has an answering machine or voice mail. Any ideas, work around? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041222/889d1878/attachment.htm