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
