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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