The server is at a remote datacenter - no nat, no firewall, pure public IP.
The phones are at home offices (i.e. DSL or Cable with Linksys-type firewall/routers). My initial testing was with a single SIP phone at the home office - and everything worked fine. But when I have two SIP phones at the home office, things start behaving badly. I understand the issue of phone-to-phone, where both phones are behind a nat at the home office - but that is not the issue I am having. My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. ---------------------------------------------------- >From : Robert Norton - SophTelecom.com <[EMAIL PROTECTED]> To : Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 18:25:16 -0700 > 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 <[email protected]> > 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 _______________________________________________ -- 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
