[asterisk-users] Asterisk not receiving call from VPN address

[Duncan Turnbull]

On 21/01/2014, at 10:24 am, David Cunningham <dcunningham at voisonics.com> wrote:

> Hi Paul,
> 
> The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine?
> 
> 

Have you got a static route on asterisk or your default gateway showing how to get back to the 172. addresses i.e. pointing to the vpn box for 172 addresses?

Cheers Duncan
> 
> On 21 January 2014 05:30, Paul Belanger <paul.belanger at polybeacon.com> wrote:
> On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham
> <dcunningham at voisonics.com> wrote:
> > Hi,
> >
> > We have a Kamailio and Asterisk cluster, both machines being on a real 103.x
> > IP address and also on a 172.x OpenVPN address.
> >
> > The problem is that when Kamailo receives a call from the VPN and forwards
> > it to the Asterisk server on it's 103.x address, Asterisk never sees the
> > call.
> >
> > If Kamailio receives a call from the VPN and forwards the call to the
> > Asterisk server on it's 172.x address then it works. However, if the call
> > isn't from the VPN then forwarding it to the 172.x address doesn't work. So
> > basically the problem is going between the real network and the VPN.
> >
> > The question is, how can we make this work when calls are received on either
> > network on the Kamailio server and are forwarded to Asterisk?
> >
> > Using ngrep on the Asterisk server we see that it does receive the INVITE,
> > but Asterisk's logging shows no sign it at all. We guess it's a Linux
> > networking issue rather than Asterisk's fault, but don't know where to fix
> > it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk
> > servers.
> >
> > Thanks in advance for any help.
> >
> > The ngrep on the Asterisk server:
> >
> > U 2014/01/17 13:15:15.599557 172.x.x.x:5060 -> 103.y.y.y:5060
> > INVITE sip:9067268 at 103.y.y.y:5060;transport=udp SIP/2.0.
> > Record-Route: <sip:172.x.x.x;lr=on>.
> > Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0.
> > Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997.
> > From: "9067271" <sip:9067271 at 172.x.x.x>;tag=198791249.
> > To: <sip:9067268 at 172.x.x.x>.
> > Call-ID: 1905625787 at 192.z.z.z.
> > ...
> >
> > 172.x.x.x is the Kamailio server's VPN address
> > 103.y.y.y is the Asterisk server's real address
> > 192.z.z.z is the calling phone's LAN address
> >
> Sounds like a routing problem opposed to an application issue. You'll
> have to fire up tcpdump on Kamailio and see what happens to the
> packet. The look at the local routing tables to see where it is
> getting routed.  If Asterisk is not receiving the patch, then Kamailio
> is not routing it properly.
> 
> You'll be able to see everything once you have a pcap of the call.
> 
> --
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
> 
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> 
> -- 
> David Cunningham, Voisonics
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