I spent the better part of the day trying to figure out why my SIP calls going through * were just going dead after 20 seconds. I was sure it was a nat issue but now I'm not so sure anymore.
I have * on a public ip and clients behind a nat. I was using simpletelecom to terminate my calls. I could connect fine if I went direct from client -> simpletelecom. If I used * as a proxy the audio just went dead after 20 seconds. A few minutes ago I tested the same setup using FWD instead of simpletelecom. Everything works, no dead audio. And of course iax didn't have any problems at all. Where should I be looking for the cause? Chris _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
