Nat=yes with the phone behind a nat box and asterisk on a registered IP works just fine with Cisco, Snom, Xlite and others (I haven't tried many of the others, however).
------------------------ > I don't think you can use NAT = yes unless there is a STUN server > involved. See my post yesterday for my Grandstream settings. > > > On Fri, 2005-01-28 at 10:28 +0100, Radovan.Mihalik wrote: > > Hello, > > > > I try to connect VoIP phones to Asterisk on private network, > > And use Asterisk as outbound proxy via his public IP. > > But the localnet and externip with nat=yes, just is not working, > > I believe it might only rewrite SIP headers but does not touch > > The rtp stream. Am I right ? > > > > R. > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux > > Sent: Friday, January 28, 2005 1:29 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess > > > > Comments below. > > > > On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote: > > > > > > Kim Lux wrote: > > > > > > >I was expecting to have to port forward too and yet our setup doesn't > > > >require it, not on the laptop nor on the wireless router. > > > > > > > >I think as long as the SIP clients open a port on the NATing device > > and > > > >keep them open so the SIP provider can connect to it, all is well, > > even > > > >if STUN isn't used. > > > > > > > >I was surprised by how easy it was to NAT the Grandstreams. I had > > > >visions of having every device being assigned a static IP and having > > a > > > >fistful of port forwards assigned to them on the router. > > > > > > > > > > > You're connecting to a SIP provider or just Asterisk? > > > > Just a provider right now. I'll tackle asterisk in a few days. > > > > > Most SIP provider > > > use a far-end NAT traversal device like Jasomi, Acmepacket or Kagoo. > > The > > > NAT traversal device has the intelligence to figure out the UDP port > > > mapping used by the NAT. SER + nathelper has the effect. > > > > I guess ignorance is bliss in this case. > > > > > For my SER > > > setup, most of the time we can just plug the SIP phone into a router > > and > > > it will work without any special config. Unfortunately, there're > > certain > > > firewalls like PIX and MS ISA that will fail. In those cases, your > > best > > > bet is to do port forwarding or use an outbound proxy. IIRC, Vonage > > also > > > has the same problem. > > > > Thanks for sharing this. It may help some poor soul trying to get his > > SIP device working in these situations. > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Kim Lux, Diesel Research Inc. > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
