Hello!

Please help to fix problem with sdp headers

UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2)

When i call from UAC to 9002 i received INVITE/SDP from kamailio

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 192.168.1.52:27080;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080

    Record-Route: <sip:192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**>

    Record-Route: <sip:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes>

    From: <sip:user4@X.X.X.X>;tag=0748d948

    To: <sip:9002@X.X.X.X>;tag=as3914e1d1

    Call-ID: ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU.

    CSeq: 2 INVITE

    Server: Virtel.net Node2

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

    Supported: replaces, timer

    Contact: <sip:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*>

    Content-Type: application/sdp

    Content-Length: 278

    

    v=0

    o=root 732368067 732368067 IN IP4 X.X.X.X

    s=Asterisk PBX 11.17.1

    c=IN IP4 X.X.X.X

    t=0 0

    m=audio 15768 RTP/AVP 0 8 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

    a=nortpproxy:yes

Why Record-Route and Contact fields contain private IP of asterisk ?