And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall?
In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -----Original Message----- From: Greg Oliver <[EMAIL PROTECTED]> Sent: Saturday, February 02, 2008 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall On Feb 2, 2008, at 2:11 PM, John Von Essen <[EMAIL PROTECTED]> wrote: > I posted an email a few days regarding a problem with hearing the > voicemail greeting on my sip phones. > > It turns out to be a phone/stun/linksys issue - not an asterisk issue. > Which brings up a couple of questions.... > > I always assumed that you can have multiple SIP phones behind a > Linksys > firewall/router (WRT54G) all using the same STUN server/port. > > But apparently thats not the case. Is it a Linksys bug, a > Grandstream bug > in the BudgeTone-100 phone, or am I off base and just doing something > wrong? > > I cleary have problems as soon as I try to use a second phone behind > the > Linksys - registration issues, cant hear voicemail greeting, etc.,. > > My next test was to run multiple STUN servers on the same machine with > different ports. Then, for my multiple SIP phones behind the > Linksys, have > each phone use a different stun port. > > Any thoughts? > > John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users