Digium phones, which (as far as I can tell with my experience) do not support VPN yet.
On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen < [email protected]> wrote: > Or if it's just a couple phones, you might be able to setup a vpn > connection directly on the phone itself - have it vpn into 'HQ' and get an > address on that network. I'm not sure which phones you're using though or > what phones support that setup. > > Justin Killen > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Justin Killen > Sent: Thursday, February 07, 2013 9:55 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk calls between 2 private networks > > I don't see how that would really solve anything - instead of the server > sending the 192.168.x.x packets onto the local network, it will send them > up toward the internet and get black-holed. What probably makes more sense > would be to switch the subnet on one of the networks, AND put up a vpn > between them, adding the routes for the private networks to cross thru the > tunnels. > > Justin Killen > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Frank > Sent: Thursday, February 07, 2013 9:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Eric Wieling > Subject: Re: [asterisk-users] Asterisk calls between 2 private networks > > I thought about that. > I will give it a shot tonight and will post back my results in here. > Thanks > > On 2/7/13 12:39 PM, Eric Wieling wrote: > > The easiest thing to is renumber one of the networks so they are not > using the same address block. > > > > -----Original Message----- > > From: [email protected] [mailto: > [email protected]] On Behalf Of Frank > > Sent: Thursday, February 07, 2013 12:27 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Asterisk calls between 2 private networks > > > > AJS, > > > > That is a solution that I am envisaging. > > But I would really love to try to work out with my issue first. It will > allow me to deploy more phones in separates buildlings in the future. If I > do the IAX solution, it means that for every building, I need a box.. > > Which I would like to prevent. > > > > > > > > On 2/7/13 10:46 AM, A J Stiles wrote: > >> On Thursday 07 February 2013, Frank wrote: > >>> My apologies if this topic was already discussed in the past. > >>> > >>> Here is my scenario: > >>> Network A - 192.168.1.0 > >>> 1 Asterisk > >>> 1 Digium phone > >>> Router does NAT from the public IP to asterisk, and forward ports > >>> 5060tcp/udp and 10k-20k udp > >>> > >>> Network B - 192.168.1.0 > >>> 1 Digium phone, registering to the public IP of network A > >>> > >>> > >>> My SIP.CONF has: > >>> nat=yes > >>> localnet=192.168.1.0/255.255.255.0 > >>> externaddr=public_ip_of_network_a > >>> directmedia=no > >> > >> My (lazy) solution to this problem was to throw hardware at it ..... > >> > >> Bearing in mind that Asterisk will run on just about any old scrapper > >> (or even a Raspberry Pi, if you feel so inclined), there's little > >> point even trying to send SIP over the Internet. Just have an > >> Asterisk box at each end, and then you only need a much > simpler-to-configure IAX trunk between the two. > >> The routers at each end then just need one port -- UDP 4569 -- > >> forwarded to the Asterisk box (if it isn't configured as the default > DMZ machine). > >> > >> > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > 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 -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
