[Asterisk-Users] No correct IP in RTP media stream

[Xisco]

    Hi everybody,

    I'm trying to configure * for make SIP calls. Now I'm doing several test but I have some errors.

    Firstly I will describe my scenario.

Client Software  (Private IP 192.168.0.181, SJ Phone over Windows 2000) ---- Router Adsl (Public ip A.B.C.D, and NAPT on port 5060 to 192.168.0.181) ----- FW+Router ----- Asterisk (Public IP E.F.G.H + e400p)------ Spain ISDN

    I make a call to my * server and this began a SIP (SDP) signaling comunication with Client SJ Phone, during signaling comunication all looks good. But when the client answer the call, the RTP media session began, I don't hear anything in noone of the two sides.

    I have sniffed the traffic RTP and I see that asterisk send to my private IP, here you can see a log line of this trafic:

      From E.F.G.H:9056 To 192.168.0.181:16384
    Here you can see the my sip.conf.

  sip.conf

  [general]
  port = 5060                     ; Port to bind to
  bindaddr = 0.0.0.0              ; Address to bind to
  context = default               ; Default for incoming calls
  allow = alaw


  [user1]
  type=friend
  secret=pass1
  host=A.B.C.D
  nat=yes
        I dont know how can I told to * the public IP in the RTP stream, and how can I determin only one RTP port in order to make NAPT in adsl router.

        If somebody can help me I will be very pleasured. 

        Thks a lot.
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