How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible
----- Original Message ----- From: "Uriel Carrasquilla" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration > Very valuable help. It is now working like a champ. > > This is a solution with SIP--NAT---Internet---Asterisk. No problems here. > > What I would like to do next is to move Asterisk behind a NAT as follows > SIP---NAT---Internet---NAT---Asterisk > do I need a STUN server? is there a chance this could work? > The Google results seems to indicate that I will get an ulcer attempting > this step. is that true? > > Regards, > Uriel > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . > Sent: Wednesday, September 24, 2003 9:05 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration > > > Try adding nat=yes to your config.. > > Also if you want to make SIP to SIP extension calls and don't want to fight > with the NAT set canreinvite=yes to canreinvite=no.. > > Finally set dtmfmode=info for the GS phones.. > > Later.. > > > 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 > > > > -- > ______________________________________________ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
