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

[David Cunningham]

Hi Duncan,

The Asterisk machine also has a VPN IP address, so it has a route for 172.x
addresses to go to tun0 VPN interface.



On 21 January 2014 08:30, Duncan Turnbull <duncan at e-simple.co.nz> wrote:

> 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
> http://voisonics.com/
> USA: +1 213 221 1092
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>  --
> _____________________________________________________________________
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>
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>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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