[asterisk-users] asterisk and openvpn and sip

[John A. Sullivan III]

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
jsullivan at opensourcedevel.com

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