[Serusers] Call routing Problems ser->asterisk->ser with NATed UAs

[Kai Militzer]

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