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 <dun...@e-simple.co.nz> wrote: > On 21/01/2014, at 10:24 am, David Cunningham <dcunning...@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.belan...@polybeacon.com>wrote: > >> 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 >> > > > > -- > 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 > > > > -- > _____________________________________________________________________ > -- 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
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