have you tried to put nat=yes in the user definition in sip.conf ? Also, the * server is on a public IP?
Matteo Il mer, 2003-09-24 alle 15:35, Uriel Carrasquilla ha scritto: > Hi there! > I installed the BudgetTone (GrandStream) on my LAN without any > problems. Then, I moved it to another location using a D-Link NAT. > I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP > address of the BudgetTone. > When I receive a call on my Asterisk, it would ring my FXS as before. > However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 > in the log). > The configuration I have in * is the following: > sip.conf > ----------- > [general] > port=5060 > context=sip > maxexpirey=3600 > defaultexpirey=60 > disallow=all > allow=ulaw > allow=gsm > [1000] > contet=sip > type=friend > username=1000 > secret=????? (not the real one) > host=dynamic > mailbox=1000 > canreinvite=yes > dtmfmode=rfc2833 > > I did not change the above configuration when I moved the budgetTone > from the LAN to the Internet (Wan). > I am not using a "register" statement in the sip.conf and I am > wondering if I need to. > I did change the sip server IP address in the Grandstream > configuration. > > I suspect my problem is with the router (NAT). I don't quite > understand the symetric discussions but I downloaded a paper to learn > more. Right now, all my public and private ports are the same. > > Regards, > Uriel > -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
