[asterisk-users] asterisk and openvpn and sip
John A. Sullivan III
jsullivan at opensourcedevel.com
Fri Jun 19 07:37:21 CDT 2009
On Thu, 2009-06-18 at 19:18 -0400, John A. Sullivan III wrote:
> 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
Sustained conversations are working fine with reinvite=yes. Take care -
John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com
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