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
What is your setup, i.e. where is the NAT? Between UAs and ser, or between ser and asterisk?
Have you tried to caputure the SIP signaling (take a look at the SDP bodies, which IP addresses are announced there?) and the RTP stream? -->use ethereal to get a glue where the packets are sent to and check out what is going wrong.
Where does the UAs send the packets to? To rtpproxy or to asterisk?
AFAIK asterisk is capable of NAT traversal (symmetric RTP). So I would suggest: - for normal calls (without asterisk) do the NAT traversal with nathelper+rtpproxy - for "special" calls (with asterisk) do the SIP NAT traversal in ser (nathelper: force rport, fix natted contact, record_route) and do the RTP NAT traversal in asterisk.
regards, klaus
Kai Militzer wrote:
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
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Kai Militzer -
Klaus Darilion