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.......@.>/g.v.............INVITE sip:*1@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@103.x.x.x>;tag=1880695235 To: <sip:*1@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@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@103.x.x.x>;tag=1880695235 To: <sip:*1@103.x.x.x> Call-ID: 1898224288 On 21 January 2014 16:56, Duncan Turnbull <dun...@e-simple.co.nz> wrote: > > On 21/01/2014, at 6:40 pm, David Cunningham <dcunning...@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@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..@.....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@192.168.51.7>;tag=as32fe455a > To: <sip:sip.2talk.co.nz> > Contact: <sip:0412345678@192.168.51.7:5060> > Call-ID: 10c0242d16529fff78572ef91ef47237@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@192.168.51.7:5060>;tag=as32fe455a > To: <sip:sip.2talk.co.nz>;tag=as7b633145 > Call-ID: 10c0242d16529fff78572ef91ef47237@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.belan...@polybeacon.com>wrote: > >> On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham >> <dcunning...@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.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> Github: https://github.com/pabelanger | Twitter: >> https://twitter.com/pabelanger >> >> -- >> _____________________________________________________________________ >> -- 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 > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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