Get a trace using Ethereal when the phone boots up and look in the warning field of the sip message, if it lists your firewall type as symetric theres a good chance your out of luck using that firewall. I'm a bit confused regarding your port selection, as 3478 is cleared stated as the broadcast port in the RFC. I have a stun server running at sp01.to-talk.com or 140.186.104.157 if you want to make a quick test. At the end of the day I'd need to see a trace of what's going on between the client and server. Let me know if you need any additional help.
Mike Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 508-295-2826 [EMAIL PROTECTED] ----- Original Message ----- From: "Matteo Brancaleoni" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 01, 2003 6:32 PM Subject: Re: [Asterisk-Users] A solution for SIP and NAT > Could you give some details about setting up a stun server? > I'm doing some tests, and were successful using snom + stund > from vovida . But I got a no-go with budgetones > (that needs stund on a standard port that's 3478). > When my snom contacts the stund server, I get a lot > of info about the connection type, the ip, blah blah > When the budgetone contacts it, I get only "Receive something len[20]" > 3 times. Nothing more. > > Matteo. > > Scrive Michael Kane <[EMAIL PROTECTED]>: > > > Hello, NAT/Firewall is truely a problem in the ITSP arena. There is one > > solution I know of that works well as an integrated DHCP/NAT/Firewall into a > > SIP aware firewall. Check out www.intertex.se and look at the IXX66 > > products. They even have a device that integrates DSL/NAT/Firewall. Or, one > > can purchase a SIP device that supports STUN(Grandstream and SNOM are the > > only vendors I know of that do) and install a STUN server. If anyone is > > interested I have a STUN server running to test with. Hope this helped.... > > > > Mike > > > > > > > > > > Michael Kane > > To-Talk Communications LLC. > > 37 Sandusky Dr. > > Wareham, Ma. 02571 > > 508-295-2826 > > ----- Original Message ----- > > From: "John Todd" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, July 01, 2003 3:47 PM > > Subject: Re: [Asterisk-Users] A solution for SIP and NAT > > > > > > > I'm uncertain why you're not able to get SIP working for your user > > > agents (SIP clients.) With Cisco equipment, as an example, it works > > > quite well and almost every 79xx or ATA-186 I have is behind a NAT, > > > and this configuration is duplicated across a dozen or more systems > > > now running behind almost every conceivable NAT/PAT situation* > > > > > > Known working config: > > > > > > UA -> (NAT) -> Internet -> Asterisk > > > > > > Can you be more specific about your problems with SIP? Perhaps you > > > have done so in the past, but re-state and maybe someone can see what > > > the problem is. > > > > > > JT > > > > > > > > > *Note: the Cisco PIX, while supposedly SIP-friendly, has been the one > > > box that has not worked with NAT/PAT SIP sessions. I have not been > > > the admin on that system, but a fairly clueful Cisco wrangler has > > > been unable to make it work for originating calls in both directions > > > - only one-way origination works.) > > > > > > > > > >Hi all. > > > > > > > >I have come to the conclusion that there just isn't anything out there > > > >for allowing SIP and NAT to work together nicely. This is rather amazing > > > >considering that as far back as March 2000 there are documents > > > >describing how to do it. > > > > > > > >So I've started a really simple SIP and RTP proxy project, SaRP, on > > > >sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. > > > >This is the first general release and should work for most people. We > > > >are using it quite successfully for standard calls between all sorts of > > > >NATed clients. All you need to do is forward UDP/5060 from your > > > >firewall/router to the box running SaRP if you want incoming calls to > > > >work and also allow UDP traffic from the ports listed in the config file > > > >out. > > > > > > > >The project can be found at http://sarp.sourceforge.net/ > > > > > > > >I would be very interested in any feedback you may have. > > > > > > > >Regards > > > > > > > >Andrew Radke. > > > >_______________________________________________ > > > >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 > > > > > > -- > > Matteo Brancaleoni > Espia System Administrator > http://www.espia.it > > ------------------------------------------------- > This mail sent through IMP: http://horde.org/imp/ > _______________________________________________ > 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