On Thu, 2005-01-27 at 16:06, Kim Lux wrote: > I've got Grandstreams (SIP devices) working behind double NATs, none the > less. > > I recommend turning STUN off and make sure that your SIP devices are > generating random port numbers. If they generate static port numbers, > you'll get port collisions. > > The other parameter to watch is the "keep alive" interval. I'm not an > expert, but I think this has to be long enough so that the device > doesn't disconnect from the router while the various signalling is > getting set up. (I've got it set to 20 seconds.) > > Maybe I'm missing something, but I thought it works quite well without > STUN. They've never ever dropped a call. > > > > On Fri, 2005-01-28 at 00:18 +0400, Jean-Michel Hiver wrote: > > Hi Guys, > > > > After days of fiddling, I can't really get my SIP device to work > > communicate with Asterisk behind NAT. Sometimes the STUN server is > > flaky, sometimes the device isn't reachable if the connection is dropped > > and then put back on, sometimes it registers OK, sometimes it doesn't, etc. > > > > I've come to the same conclusion as the wiki: it's probably better to > > avoid this horrible mess by either using IAX or doing VPN. Letting the > > IAX option aside, are you aware of any SIP devices that support some > > simple, easy to use VPN protocol? > > > > Cheers, > > Jean-Michel. > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Will you Please share your configuration, I was ready to give up, thinking no one had been successful. TIA db _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
