<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">Cool<div><br></div><div>That looks like it is arriving at Asterisk - are you sure asterisk is not getting it? If you turn on sip debug in asterisk can you see the SIP packets? It maybe asterisk is ignoring them or replying to them but its going out an interface you hadn’t thought of, I have had that a few times.</div><div><br></div><div>I should have mentioned to print out your route table and ifconfig. Asterisk can reply on a different address to the original destination especially if it came through a tunnel. Often it will be the tunnel interface address. Usually then we set the secondary address as the outbound proxy on the phone so the phone will also respond to it. </div><div><br></div><div>Cheers Duncan</div><div><br></div><div>On 21/01/2014, at 7:18 pm, David Cunningham <<a href="mailto:dcunningham@voisonics.com">dcunningham@voisonics.com</a>> wrote:</div><div><div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><div>Hi Duncan,<br><br></div>Thank you for your reply. Here's the netstat:<br><div><br>[root]# netstat -udpln | grep asterisk<br>udp 0 0 <a href="http://0.0.0.0:5000/" target="_blank">0.0.0.0:5000</a> 0.0.0.0:* 6672/asterisk <br>
udp 0 0 <a href="http://0.0.0.0:4520/" target="_blank">0.0.0.0:4520</a> 0.0.0.0:* 6672/asterisk <br>udp 0 0 <a href="http://0.0.0.0:5060/" target="_blank">0.0.0.0:5060</a> 0.0.0.0:* 6672/asterisk <br>
udp 0 0 <a href="http://0.0.0.0:4569/" target="_blank">0.0.0.0:4569</a> 0.0.0.0:* 6672/asterisk <br><br></div><div>Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server:<br>
<br>17:13:17.103771 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228<br>E.......@.>/g.v.............INVITE <a href="sip:*1@172.y.y.y:5060;transport=udp">sip:*1@172.y.y.y:5060;transport=udp</a> SIP/2.0<br>Record-Route: <<a href="sip:103.x.x.x;lr=on">sip:103.x.x.x;lr=on</a>><br>Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0<br>
Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850<br>From: <<a href="sip:9067273@103.x.x.x">sip:9067273@103.x.x.x</a>>;tag=1880695235<br>To: <<a href="sip:*1@103.x.x.x">sip:*1@103.x.x.x</a>><br>Call-ID: 1898224288<br><br><br>Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server:<br>
<br>17:13:17.093676 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228<br>E.......?.?/g.v.............INVITE <a href="sip:*1@172.y.y.y:5060;transport=udp">sip:*1@172.y.y.y:5060;transport=udp</a> SIP/2.0<br>Record-Route: <<a href="sip:103.x.x.x;lr=on">sip:103.x.x.x;lr=on</a>><br>Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0<br>
Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850<br>From: <<a href="sip:9067273@103.x.x.x">sip:9067273@103.x.x.x</a>>;tag=1880695235<br>To: <<a href="sip:*1@103.x.x.x">sip:*1@103.x.x.x</a>><br>Call-ID: 1898224288<br><br></div><div><br>
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