Hello everybody! I've been trying for three days to acomplish the following scenario with ser, asterisk and SIP NATed UAs and somehow didn't get any further. What I want in the end is the following. A call from an UA (with the extension 8002) to let's say the extension 98001 comes into ser, from there it is routed to asterisk, which does something (read: record the message for the archives), rewrites the destination and sends it back to ser. With rewriting I mean stripping of the first digit, in this case the 9, so it calls the 8001 on ser. 8001 is a registered UA behind a NAT. The problem i now have is, that the calles extension 8002 rings, but if I answer the call, I have no sound. I'm sure this does something have to do with the NATed UAs and the rtp-Stream, but I can't figure out what exactely it is. I'm sure I have something to do with the nathelper module and rtpproxy on the ser machine, but I haven't found any documentation where it tells me how to exactelly do it. What is strange is the fact, that if I forward a call only to asterisk (for example to a voicemail), without routing it back to ser, I have sound in both directions, meaning I can hear the anouncements of the vm and record a message. If anybody can help me by pointing me in the right direction (RTFMs are fine for me, as long as I got told where to read) I would appreceate it very much. If you need some more information (e.g. ser configurations, etc.), I will happily supply them. Thanks in advance for any help. Best regards Kai _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers