Hello,
i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP
registration from VoIP provider and route the calls to the asterisk boxes where an IVR
will take care to answer.

Here's my network topology:

                                      +---> [asterisk1]
[public_ip]                           |    10.50.10.131
 [router]  <---NAT---> [kamailio] <---+
10.50.10.1            10.50.10.120    |   
                                      +---> [asterisk2]
                                           10.50.10.132

In my setup i planned to use UAC and DISPATCHER modules. I started from the
"kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER.

All is working fine when i do a test call from a softphone inside network 10.50.10.0/24.

When a call is coming from the sip carrier, troubles occurs because asterisk boxes
are sending their internal ip in SDP.

I understand that i need to rewrite SDP in that case, but i actually don't know how/where.

I've attached kamailio configuration and a sip trace taken with sngrep where the problem
is visible.

For security reasons, i would like to force the RTP through RTPProxy.

I'm missing something, and need your help me to understand my errors.

Best Regards,
Bruno

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