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

Tue Jan 21 05:23:14 CST 2014

Hi Duncan,

We have "sip set debug on" and nothing is shown, even though tcpdump/ngrep
on the same server does. It's very strange.

The output of "ip address list" is:

[root]# ip address list
1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue state UNKNOWN
    link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
    inet 127.0.0.1/8 scope host lo
    inet6 ::1/128 scope host
       valid_lft forever preferred_lft forever
2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc mq state UP qlen
1000
    link/ether 44:1e:a1:4e:2f:b8 brd ff:ff:ff:ff:ff:ff
    inet 103.y.y.19/24 brd 103.y.y.255 scope global eth0
    inet6 fe80::461e:a1ff:fe4e:2fb8/64 scope link
       valid_lft forever preferred_lft forever
3: eth1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
    link/ether 44:1e:a1:4e:2f:ba brd ff:ff:ff:ff:ff:ff
4: eth2: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
    link/ether 44:1e:a1:4f:30:a4 brd ff:ff:ff:ff:ff:ff
5: eth3: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
    link/ether 44:1e:a1:4f:30:a6 brd ff:ff:ff:ff:ff:ff
6: tun0: <POINTOPOINT,MULTICAST,NOARP,UP,LOWER_UP> mtu 1500 qdisc
pfifo_fast state UNKNOWN qlen 100
    link/[65534]
    inet 172.x.x.14 peer 172.x.x.13/32 scope global tun0

The output of "netstat -rn" is:

[root]# netstat -rn
Kernel IP routing table
Destination     Gateway         Genmask         Flags   MSS Window  irtt
Iface
172.x.x.10     172.x.x.13     255.255.255.255 UGH       0 0          0 tun0
172.x.x.13     0.0.0.0         255.255.255.255 UH        0 0          0 tun0
172.x.x.1      172.x.x.13     255.255.255.255 UGH       0 0          0 tun0
172.x.x.18     172.x.x.13     255.255.255.255 UGH       0 0          0 tun0
192.168.234.0   172.x.x.13     255.255.255.0   UG        0 0          0 tun0
192.168.235.0   172.x.x.13     255.255.255.0   UG        0 0          0 tun0
103.y.y.0   0.0.0.0         255.255.255.0   U         0 0          0 eth0
169.z.z.0     0.0.0.0         255.255.0.0     U         0 0          0 eth0
172.21.0.0      172.x.x.13     255.255.0.0     UG        0 0          0 tun0
10.0.0.0        172.x.x.13     255.0.0.0       UG        0 0          0 tun0
0.0.0.0         103.y.y.1   0.0.0.0         UG        0 0          0 eth0

On 21 January 2014 17:44, Duncan Turnbull <duncan at e-simple.co.nz> wrote:

> 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=udpSIP/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=udpSIP/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
>
>
>
>
>
> --
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-- 
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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