On 2017-04-18 05:21 PM, Duncan Turnbull wrote:
Sent from my iPhone
On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
------ Original Message ------
Sent: 19-Apr-17 10:25:59 AM
Subject: [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.
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
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?
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
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
sudo tcpdump -i tun0 -s0 -A udp port 5060
Should show you the sip messages going through the tunnel and
you can check the reply addresses
Hmm. I also can't ping the phone's IP address on the 192.168.1.0/24
network. Perhaps that's the real problem there. This VPN should work
both ways, shouldn't it?
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users