On 21/01/2014, at 10:24 am, David Cunningham <[email protected]> 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 <[email protected]> wrote: > On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham > <[email protected]> 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:[email protected]: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:[email protected]>;tag=198791249. > > To: <sip:[email protected]>. > > Call-ID: [email protected]. > > ... > > > > 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: [email protected] | IRC: pabelanger (Freenode) > Github: https://github.com/pabelanger | Twitter: > https://twitter.com/pabelanger > > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > David Cunningham, Voisonics > http://voisonics.com/ > USA: +1 213 221 1092 > UK: +44 (0) 20 3298 1642 > Australia: +61 (0) 2 8063 9019 > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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