On Thu, 2009-06-18 at 14:55 +0200, Giorgio Incantalupo wrote: > Hi John, > > I already have the ccd dir with the iroute (mandatory for routing to > pc/phone connected to vpn client). During the last test I could register > and make a call but voice disappears after 1, 2 seconds. I'm trying to > understand if it is a bandwidth problem. At the moment I have my phone > connected to the openvpn client (which is its gateway) but I have to use > the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip > (192.168.1.12) is not working. I suppose it is a sip protocol problem: > I had to change the sip.conf setting nat=yes to make the phone dial and > domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). > I keep on working on the vpn since it seems so little is missing to have > a clear conversation. Let me know if your tests are successfull. > > Thank you. > > Giorgio <snip> Hi, Giorgio. So far so good. I have twinkle running on my laptop (the VPN client), a Snom 320 and a Snom 360 on the internal network routing through my laptop. I haven't done much more than register and execute a very basic dialplan but it is all working so far.
I hit a couple of small bumps but nothing to do with *. I had forgotten to tell my DNS to accept requests from the test network. One of the phones somehow decided the data center firewall was an outbound SIP proxy. Once I removed that setting, it all worked just fine. I am using native addresses across the VPN; there is no NAT. I've not yet had sustained conversations. I'll be doing that in a while hopefully - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society _______________________________________________ -- 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