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

[Larry Moore]

Is Kamalio running on the same system as Asterisk?

On 21/01/2014 2:41 PM, David Cunningham wrote:
> Hi Larry,
>
> Thanks for the reply. We have all of those settings left out of our
> sip.conf, so this should allow everything, right?
>
>
>
> On 21 January 2014 17:38, Larry Moore <lmoore at omninet.net.au
> <mailto:lmoore at omninet.net.au>> wrote:
>
>     Have you checked your localnet=, deny=, permit=, contactdeny= &
>     contactpermit= settings?
>
>     My 2c worth.
>
>
>     On 20/01/2014 10:51 AM, David Cunningham 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
>         <tel:15.599557%20172>.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
>
>         --
>         David Cunningham, Voisonics
>         http://voisonics.com/
>         USA: +1 213 221 1092 <tel:%2B1%20213%20221%201092>
>         UK: +44 (0) 20 3298 1642 <tel:%2B44%20%280%29%2020%203298%201642>
>         Australia: +61 (0) 2 8063 9019
>         <tel:%2B61%20%280%29%202%208063%209019>
>
>
>
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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
>
>