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

[David Cunningham]

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?



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
http://voisonics.com/
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