On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham <dcunning...@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@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@172.x.x.x>;tag=198791249. > To: <sip:9067268@172.x.x.x>. > Call-ID: 1905625787@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.belan...@polybeacon.com | 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
