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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:
Here you can see the my
sip.conf.
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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