[Asterisk-Users] Outgoing SIP Call to unregistered Users

[Klaus Hueske]

Hi!

I'm using asterisk with a few kphone SIP-Clients. The registration process 
seems quite OK. But there are some problems:

Calling other registered users is possible, but the rtp-stream is not reaching 
the right port, so you can hear nothing. In ethereal you can see, that the 
SIP/SDP fields addresses different ports at each client, so client A sends to 
port 32000 but client B listens on port 32002. One solution for this problem 
ist to use the canreinvite=no statement in sip.conf, but in this case every 
rtp-packet is going through asterisk. I think, only the SIP/SDP packets 
should go through asterisk and the voicetraffic direct from client A to 
client B. May be, I'm wrong about that, please correct me in that case.

Another problem is calling SIP users that are not registered to asterisk. 
Giving kphone the address sip:name at anyhost causes asterisk to search 
for the extension name, but there is no such extension. Are there ways to say 
asterisk, that these calls should only be forwarded to the given host?

I hope, somebody could write something about that. Thanks

Klaus