On 2017-04-18 03:39 PM, Sebastian Nielsen wrote:
You need to ensure that traffic to the SIP box is sent to the
correct IP. Also if you use split-tunnel (eg: not redirect-gateway
def1) you must make sure NAT and traffic redirection works as is
so the Asus router knows it should send the traffic through tunnel
and not via WAN.
I'm not that well versed in OpenVPN, but it's worth noting that we
have the `push "redirect-gateway def1 bypass-dhcp"` directive set on
the server. I have two independent DHCP servers on either side of
the VPN, so that the clients are getting their IP addresses for
their appropriate networks - 192.168.0.0/24 on the server side, and
192.168.1.0/24 on the client side.
IMPORTANT: Then you must, in the ASUS RT-N66U make a port
forward inwards from TUN to the phone client.
I'll give that a shot, but it will have to wait until tomorrow. :)
I would suggest wiresharking on the client side and see which
IP Asterisk suggest the client should connect back to. This
should be the internal IP of the asterisk server as seen from
the openvpn server's point of view.
Another important thing: The local network in the Openvpns
machine locatiin, may NOT have same subnet as the network behind
the asus.
All these must be separate, like:
server network: 192.168.1.0/24
Openvpn tunnel network: 192.168.2.0/24
Asus network: 192.168.3.0/24
I'm pretty sure that I've got this subnet separation in place. If I
didn't cover it in my original post, the network looks like this:
Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting through the VPN (Asterisk output is
truncated):
ToHost :
Addr->IP : 10.8.0.6:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: FrontDesk1
SIP Options : (none)
Codecs : (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing : No
Status : Unmonitored
Useragent : Aastra 6731i/3.2.2.1136
Reg. Contact : sip:[email protected]:5060;transport=udp
Else you get bizarre routing problems when states appear in
the state table.
-------- Originalmeddelande --------
Datum: 2017-04-19 00:25 (GMT+01:00)
Rubrik: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.
First, here's our technical details:
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.
The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato.
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.
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.
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:
root@ldinfo:/etc/asterisk# iptables -L -n
Chain INPUT (policy ACCEPT)
target prot opt source destination
ACCEPT all -- 192.168.0.0/24 192.168.0.3
ACCEPT all -- 192.168.1.0/24 192.168.0.3
ACCEPT all -- 10.8.0.0/24 192.168.0.3
ACCEPT all -- X.X.X.X 192.168.0.3
ACCEPT all -- 192.168.0.3 X.X.X.X
ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194
REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
Chain POSTROUTING (0 references)
target prot opt source destination
192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.
192.168.1.0/24 is the network that the remote router is on.
10.8.0.0/24 is the network that the TUN device creates.
X.X.X.X is our datacenter.
192.168.0.3 is the IP address of our PBX.
Any assistance would be greatly appreciated.
|
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