I was wrong. While setting the bindaddr=outsideipaddress does fix the format of the SIP message (the VIA and Contact IP addresses are now correct) , it forces the * process to run on that address , which is useless when the * server is behind a NAT.
Lee Goodman ----- Original Message ----- From: "Lee Goodman" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 11, 2003 10:04 AM Subject: Re: [Asterisk-Users] running * on a VPN gateway > I just tried this, I set the bindaddr=outside NAT address and my sip > registration messages now have the correct ip address in the VIA field!!!!!! > > I tried the fromdomain=outside NAT address , but it didn't change anything > in the sip message. > > And setting the bindaddr=outside NAT address didn't break the SIP thread. As > the registration messages work fine. > > Lee Goodman > > > > ----- Original Message ----- > From: "Ian Blenke" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, September 10, 2003 3:23 PM > Subject: Re: [Asterisk-Users] running * on a VPN gateway > > > > Lee Goodman wrote: > > > Could the bindaddr=x.x.x.x be a way to make * work through a NAT? > > > > > > I have * and a few 7960 phones behind a NAT. I am trying to register > with a > > > proxy on the outside of the NAT. Registration is ok, but the VIA field > has > > > my inside NAT ip address (192. 168.0.7). So the proxy doesn't know how > to > > > send a call to *. Would adding my outside NAT ip address to the bindaddr > > > statement cause the * to put the outside address in the VIA field??? > > > > Use "fromdomain=" in your sip.conf entry for your external proxy to > > override this. > > > > >>If like me you run * on a VPN (or multihomed) gateway and want to serve > > >>remote SIP clients, make sure you have > > >> > > >>bindaddr = 192.168.0.1 ; or whatever is your box's private IP > > >> > > >>otherwise * might bind to its public IP and send it as return address in > > >>the SIP call setup, which will (should) be rejected by your firewall. > > >> > > >>To * experts: might this setting interfer with NATed SIP clients? > > > > There appear to be real issues with multi-homed Asterisk installs in > > more than simple call appearances in the SIP messages. > > > > At one point in testing with a recent CVS build (while bound to > > 0.0.0.0), I was getting SIP messages from the public IP interface and > > RTP streams *from* the private IP interface, resulting in one-way audio > > (the called party could hear me, I could not hear them). Very confusing, > > to say the least. > > > > -- > > - Ian C. Blenke <[EMAIL PROTECTED]> > > (This message bound by the following: > > http://www.nks.net/email_disclaimer.html) > > > > > > _______________________________________________ > > 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
