[asterisk-users] Asterisk not receiving call from VPN address

Duncan Turnbull duncan at e-simple.co.nz
Tue Jan 21 00:44:07 CST 2014


Cool

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.

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. 

Cheers Duncan

On 21/01/2014, at 7:18 pm, David Cunningham <dcunningham at voisonics.com> wrote:

> Hi Duncan,
> 
> Thank you for your reply. Here's the netstat:
> 
> [root]# netstat -udpln | grep asterisk
> udp        0      0 0.0.0.0:5000                0.0.0.0:*                               6672/asterisk       
> udp        0      0 0.0.0.0:4520                0.0.0.0:*                               6672/asterisk       
> udp        0      0 0.0.0.0:5060                0.0.0.0:*                               6672/asterisk       
> udp        0      0 0.0.0.0:4569                0.0.0.0:*                               6672/asterisk       
> 
> Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server:
> 
> 17:13:17.103771 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228
> E....... at .>/g.v.............INVITE sip:*1 at 172.y.y.y:5060;transport=udp SIP/2.0
> Record-Route: <sip:103.x.x.x;lr=on>
> Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0
> Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850
> From: <sip:9067273 at 103.x.x.x>;tag=1880695235
> To: <sip:*1 at 103.x.x.x>
> Call-ID: 1898224288
> 
> 
> Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server:
> 
> 17:13:17.093676 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228
> E.......?.?/g.v.............INVITE sip:*1 at 172.y.y.y:5060;transport=udp SIP/2.0
> Record-Route: <sip:103.x.x.x;lr=on>
> Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0
> Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850
> From: <sip:9067273 at 103.x.x.x>;tag=1880695235
> To: <sip:*1 at 103.x.x.x>
> Call-ID: 1898224288
> 
> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140121/fcbab166/attachment.html>


More information about the asterisk-users mailing list