<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div><br><br>Sent from my iPhone</div><div><br>On 19/04/2017, at 11:43 AM, Ernie Dunbar <<a href="mailto:maillist@lightspeed.ca">maillist@lightspeed.ca</a>> wrote:<br><br></div><blockquote type="cite"><div>
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On 2017-04-18 03:38 PM, Duncan Turnbull wrote:<br>
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<div>------ Original Message ------</div>
<div>From: "Ernie Dunbar" <<a moz-do-not-send="true" href="mailto:maillist@lightspeed.ca">maillist@lightspeed.ca</a>></div>
<div>To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'" <<a moz-do-not-send="true" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>></div>
<div>Sent: 19-Apr-17 10:25:59 AM</div>
<div>Subject: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.</div>
<div> </div>
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<blockquote class="cite2" cite="ff7e561a-bc8b-097d-5b3f-6657ea162b4f@lightspeed.ca" type="cite">Hi everyone. I'm having some trouble with an
OpenVPN tunnel that isn't working *quite* as well as we'd
hoped.<br>
<br>
First, here's our technical details:<br>
<br>
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind
a NAT router. The router has UDP port 1194 forwarded to our
server. This server also runs our office Asterisk PBX, so
there isn't any networking hardware or firewall between the
VPN tunnel and the Asterisk PBX.<br>
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<div>Asterisk maybe replying from the TUN address which may
confuse your sip client - if you set the TUN address as a
proxy that seems to solve it. If asterisk is bound to every
address then implicitly it shouldn't matter where it replies
from, but in the openvpn case it seems to reply from a
different address to the one it was called on and that can
definitely fool clients. tcpdump on the tunnel can help you
see whats happening</div>
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</blockquote>
<br>
I think I'll need a bit more detail about how to set the TUN address
as a proxy. Is this done on the OpenVPN server, or at the client
end? I'm also going to tell Asterisk to bind to all IPs and then
restart it when there's no calls in progress, perhaps that's all I
need to do?<br>
</div></blockquote><br><div>Set it as a proxy server in your sip phone client, we found using the tun ip on the vpn server works, we keep the actual asterisk address as the sip server and use the tun ip as the proxy server</div><div><br></div><div>Asterisk is probably already bound to all the addresses netstat -nupl should show you the addresses it's listening on for udp, if it says 0.0.0.0 it means all addresses</div><div><br></div><div>sudo tcpdump -i tun0 -s0 -A udp port 5060</div><div><br></div><div>Should show you the sip messages going through the tunnel and you can check the reply addresses </div></body></html>