[asterisk-users] Asterisk not receiving call from VPN address
David Cunningham
dcunningham at voisonics.com
Tue Jan 21 00:18:47 CST 2014
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
On 21 January 2014 16:56, Duncan Turnbull <duncan at e-simple.co.nz> wrote:
>
> On 21/01/2014, at 6:40 pm, David Cunningham <dcunningham at voisonics.com>
> wrote:
>
> Hi Paul,
>
> Using ngrep/tcpdump shows the packet clearly going from the Kamailio
> server and arriving at the Asterisk server. This is why it's a mystery that
> Asterisk doesn't see the call coming in. We tried removing the firewall (so
> iptables -L shows no rules at all) but that didn't help unfortunately.
>
> Can you show a packet dump of the SIP invites arriving at the asterisk PBX
> , mostly just confirming the ip address that the server is receiving
> packets on
>
> *root at zespri*:*~*# tcpdump udp port 5060 -A -n
> tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
> listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
> 18:52:23.063862 IP 192.168.51.7.5060 > 27.111.14.65.5060: SIP, length: 534
> E`.2.L.. at .....3..o.A......u.OPTIONS sip:sip.2talk.co.nz SIP/2.0
> Via: SIP/2.0/UDP 192.168.51.7:5060;branch=z9hG4bK45a08b58;rport
> Max-Forwards: 70
> From: "Unknown" <sip:049343953 at 192.168.51.7>;tag=as32fe455a
> To: <sip:sip.2talk.co.nz>
> Contact: <sip:0412345678 at 192.168.51.7:5060>
> Call-ID: 10c0242d16529fff78572ef91ef47237 at 192.168.51.7:5060
> CSeq: 102 OPTIONS
> User-Agent: FPBX-2.10.1(10.6.1)
> Date: Tue, 21 Jan 2014 05:52:23 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
>
>
> 18:52:23.084330 IP 27.111.14.65.5060 > 192.168.51.7.5060: SIP, length: 472
> E.......9....o.A..3.......r.SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 192.168.51.7:5060
> ;branch=z9hG4bK45a08b58;received=192.168.51.7;rport=5060
> From: "Unknown" <sip:049343953 at 192.168.51.7:5060>;tag=as32fe455a
> To: <sip:sip.2talk.co.nz>;tag=as7b633145
> Call-ID: 10c0242d16529fff78572ef91ef47237 at 192.168.51.7:5060
> CSeq: 102 OPTIONS
> Server: 2talk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> Accept: application/sdp
> Content-Length: 0
>
> Also the udp ports asterisk is listening on
>
> e.g
> netstat -udpl
> Active Internet connections (only servers)
> Proto Recv-Q Send-Q Local Address Foreign Address State
> PID/Program name
> udp 0 0 0.0.0.0:4520 0.0.0.0:*
> 1413/asterisk
> udp 0 0 0.0.0.0:4569 0.0.0.0:*
> 1413/asterisk
> udp 0 0 0.0.0.0:5000 0.0.0.0:*
> 1413/asterisk
> udp 0 0 0.0.0.0:5060 0.0.0.0:*
> 1413/asterisk
>
>
>
>
>
> On 21 January 2014 15:29, Paul Belanger <paul.belanger at polybeacon.com>wrote:
>
>> On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
>> <dcunningham at voisonics.com> 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?
>> >
>> Well, you need to use tcpdump on each hop across your network. If are
>> Asterisk is not getting anything, either it is not receiving anything
>> (check transmit side) or the firewall is dropping it.
>>
>> --
>> Paul Belanger | PolyBeacon, Inc.
>> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
>> Github: https://github.com/pabelanger | Twitter:
>> https://twitter.com/pabelanger
>>
>> --
>> _____________________________________________________________________
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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
> --
> _____________________________________________________________________
> -- 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
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