[r...@asterisk asterisk]# netstat -a -n -p | grep 5060 udp 0 0 0.0.0.0:5060 0.0.0.0:* 3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes 23:04:59.522498 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:01.233460 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:23.521076 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:24.520486 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:25.232068 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:26.231229 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:26.520308 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:28.231050 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:30.519957 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:32.230693 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:34.521843 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925 23:05:34.530587 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530 23:05:35.519255 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925 23:05:36.230336 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540 23:05:37.519077 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925 23:05:41.518720 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925 Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3047) Verbosity is at least 3 asterisk*CLI> sip debug SIP Debugging re-enabled asterisk*CLI> and it stays that way... Greetingz, Jonas. On Mon, 2009-04-13 at 13:21 -0700, Steve Edwards wrote: > On Mon, 13 Apr 2009, jonas kellens wrote: > > > 1) IP-phones get there IP from a DHCP > > The source of the address is not the issue. > > > I still see no register-message on the CLI. This really should happen > > now, as they are defined host=dynamic ! > > I suspect you have not [correctly] configured the phones to register to the > Asterisk server. > > > I will now hang my portable on the switch and monitor the network with > > wireshark to see if the phones send a SIP-register to the > > Asterisk-server... > > "sudo netstat -a -n -p | grep 5060" will show you if Asterisk is actually > listening. It should look something like: > > udp 0 0 0.0.0.0:5060 0.0.0.0:* 3283/asterisk > > "sudo tcpdump port 5060" will show you if the phones are talking to the > box. It should look something like: > > 13:11:30.432163 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, > length 431 > 13:11:30.432443 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 398 > 13:11:30.432520 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 460 > 13:11:30.451350 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, > length 578 > 13:11:30.451525 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 398 > 13:11:30.460889 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 481 > 13:11:30.461231 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 476 > 13:11:30.461541 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, > length 540 > 13:11:30.474515 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, > length 383 > 13:11:30.497854 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, > length 319 > > "sip debug" at the Asterisk console will show the messages as the are > received and responded to by Asterisk. It should look something like: > > <-- SIP read from 192.168.0.19:5060: > SIP/2.0 200 OK > To: <sip:[email protected]:5060>;tag=d732d5ba46660f68i0 > From: "asterisk" <sip:[email protected]>;tag=as51d58666 > Call-ID: [email protected] > CSeq: 102 NOTIFY > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK25449e4a > Server: Sipura/SPA841-3.1.4(a) > Content-Length: 0 > > --- (8 headers 0 lines) --- > Destroying call '[email protected]' > 12 headers, 0 lines > Reliably Transmitting (no NAT) to 192.168.0.19:5060: > OPTIONS sip:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5 > From: "asterisk" <sip:[email protected]>;tag=as079a9a44 > To: <sip:[email protected]:5060> > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 13 Apr 2009 20:18:30 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > Thanks in advance, > ------------------------------------------------------------------------ > Steve Edwards [email protected] Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
