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<DIV>------ Original Message ------</DIV>
<DIV>From: "Ernie Dunbar" <<A href="mailto:maillist@lightspeed.ca">maillist@lightspeed.ca</A>></DIV>
<DIV>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <<A 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>
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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></BLOCKQUOTE>
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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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<DIV><BR>The OpenVPN client is an Asus RT-N66U router, which if I'm not mistaken, runs a somewhat modified version of Tomato. <BR><BR>I've got the VPN tunnel working well enough. I can do practically anything from a computer hooked up to the client router as if I were in the main office where the server is. But any SIP client I use - whether it's a hardware SIP phone or a soft phone like Zoiper, can connect to the Asterisk server without issue. Making calls can work, accepting calls works, but I only get 1 way voice traffic. I can hear voice data coming in FROM the Asterisk PBX, but I cannot send any. <BR><BR>In my experience with SIP, this usually means a firewall is breaking the connection from the client phone to the Asterisk server. I just can't for the life of me find what could be wrong. None of the other traffic is being blocked. The ipfw firewall on the Asterisk PBX is extremely open (see below). The firewall on the client router is turned off, and as far as I can tell, most NAT routers don't even block outbound traffic in the first place.<BR><BR>I can't see how traffic from the TUN interface on the OpenVPN server even can be blocked going to another IP address on the same box, but here are the IPFW rules:<BR><BR>root@ldinfo:/etc/asterisk# iptables -L -n<BR>Chain INPUT (policy ACCEPT)<BR>target prot opt source destination<BR>ACCEPT all -- 192.168.0.0/24 192.168.0.3<BR>ACCEPT all -- 192.168.1.0/24 192.168.0.3<BR>ACCEPT all -- 10.8.0.0/24 192.168.0.3<BR>ACCEPT all -- X.X.X.X 192.168.0.3<BR>ACCEPT all -- 192.168.0.3 X.X.X.X<BR>ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194<BR>REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with icmp-port-unreachable<BR><BR>Chain FORWARD (policy ACCEPT)<BR>target prot opt source destination<BR><BR>Chain OUTPUT (policy ACCEPT)<BR>target prot opt source destination<BR><BR>Chain POSTROUTING (0 references)<BR>target prot opt source destination<BR><BR>192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server are on.<BR>192.168.1.0/24 is the network that the remote router is on.<BR>10.8.0.0/24 is the network that the TUN device creates.<BR>X.X.X.X is our datacenter.<BR>192.168.0.3 is the IP address of our PBX.<BR><BR>Any assistance would be greatly appreciated.<BR><BR></DIV>
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