Hi Eric, Thanks for the suggestion. It was on bindaddr of 0.0.0.0, but we tried removing that too, and Asterisk still doesn't see anything.
On 21 January 2014 09:18, Eric Wieling <[email protected]> wrote: > Make sure you do NOT have any *bindaddr options set in your sip.conf. If > you do, you are telling Asterisk to not allow the OS to pick the source IP > and hence the routing. > > The *bindaddr options are seldom useful. > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of David Cunningham > Sent: Monday, January 20, 2014 5:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk not receiving call from VPN address > > 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 <[email protected]> wrote: > > > 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<tel:15.599557%20172> .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 <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 <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> > > -- > > _____________________________________________________________________ > -- 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 > > > -- > _____________________________________________________________________ > -- 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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