<div dir="ltr">Hi Eric,<br><br>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.<br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On 21 January 2014 09:18, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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.<br>
<br>
The *bindaddr options are seldom useful.<br>
<div><div class="h5"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of David Cunningham<br>
Sent: Monday, January 20, 2014 5:15 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Asterisk not receiving call from VPN address<br>
<br>
Hi Duncan,<br>
<br>
<br>
The Asterisk machine also has a VPN IP address, so it has a route for 172.x addresses to go to tun0 VPN interface.<br>
<br>
<br>
<br>
<br>
On 21 January 2014 08:30, Duncan Turnbull <<a href="mailto:duncan@e-simple.co.nz">duncan@e-simple.co.nz</a>> wrote:<br>
<br>
<br>
On 21/01/2014, at 10:24 am, David Cunningham <<a href="mailto:dcunningham@voisonics.com">dcunningham@voisonics.com</a>> wrote:<br>
<br>
<br>
Hi Paul,<br>
<br>
<br>
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?<br>
<br>
<br>
<br>
<br>
<br>
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?<br>
<br>
Cheers Duncan<br>
<br>
<br>
<br>
On 21 January 2014 05:30, Paul Belanger <<a href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a>> wrote:<br>
<br>
<br>
On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham<br>
<<a href="mailto:dcunningham@voisonics.com">dcunningham@voisonics.com</a>> wrote:<br>
> Hi,<br>
><br>
> We have a Kamailio and Asterisk cluster, both machines being on a real 103.x<br>
> IP address and also on a 172.x OpenVPN address.<br>
><br>
> The problem is that when Kamailo receives a call from the VPN and forwards<br>
> it to the Asterisk server on it's 103.x address, Asterisk never sees the<br>
> call.<br>
><br>
> If Kamailio receives a call from the VPN and forwards the call to the<br>
> Asterisk server on it's 172.x address then it works. However, if the call<br>
> isn't from the VPN then forwarding it to the 172.x address doesn't work. So<br>
> basically the problem is going between the real network and the VPN.<br>
><br>
> The question is, how can we make this work when calls are received on either<br>
> network on the Kamailio server and are forwarded to Asterisk?<br>
><br>
> Using ngrep on the Asterisk server we see that it does receive the INVITE,<br>
> but Asterisk's logging shows no sign it at all. We guess it's a Linux<br>
> networking issue rather than Asterisk's fault, but don't know where to fix<br>
> it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk<br>
> servers.<br>
><br>
> Thanks in advance for any help.<br>
><br>
> The ngrep on the Asterisk server:<br>
><br>
</div></div> > U 2014/01/17 13:15:<a href="tel:15.599557%20172" value="+15599557172">15.599557 172</a> <tel:15.599557%20172> .x.x.x:5060 -> 103.y.y.y:5060<br>
<div class="im"> > INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0.<br>
> Record-Route: <sip:172.x.x.x;lr=on>.<br>
> Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0.<br>
> Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997.<br>
> From: "9067271" <sip:9067271@172.x.x.x>;tag=198791249.<br>
> To: <sip:9067268@172.x.x.x>.<br>
> Call-ID: 1905625787@192.z.z.z.<br>
> ...<br>
><br>
> 172.x.x.x is the Kamailio server's VPN address<br>
> 103.y.y.y is the Asterisk server's real address<br>
> 192.z.z.z is the calling phone's LAN address<br>
><br>
<br>
Sounds like a routing problem opposed to an application issue. You'll<br>
have to fire up tcpdump on Kamailio and see what happens to the<br>
packet. The look at the local routing tables to see where it is<br>
getting routed. If Asterisk is not receiving the patch, then Kamailio<br>
is not routing it properly.<br>
<br>
You'll be able to see everything once you have a pcap of the call.<br>
<br>
--<br>
Paul Belanger | PolyBeacon, Inc.<br>
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