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[root@asterisk asterisk]# netstat -a -n -p | grep 5060<BR>
udp 0 0 0.0.0.0:5060 0.0.0.0:* 3047/asterisk <BR>
<BR>
<BR>
[root@asterisk asterisk]# /usr/sbin/tcpdump port 5060<BR>
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode<BR>
listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes<BR>
23:04:59.522498 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:01.233460 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:23.521076 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:24.520486 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:25.232068 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:26.231229 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:26.520308 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:28.231050 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:30.519957 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:32.230693 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:34.521843 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925<BR>
23:05:34.530587 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 530<BR>
23:05:35.519255 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925<BR>
23:05:36.230336 IP 192.168.4.112.sip > 192.168.4.248.sip: SIP, length: 540<BR>
23:05:37.519077 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925<BR>
23:05:41.518720 IP 192.168.4.114.sip > 192.168.4.248.sip: SIP, length: 925<BR>
<BR>
Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3047)<BR>
Verbosity is at least 3<BR>
asterisk*CLI> sip debug<BR>
SIP Debugging re-enabled<BR>
asterisk*CLI> <BR>
<BR>
and it stays that way...<BR>
<BR>
Greetingz,<BR>
Jonas.<BR>
<BR>
<BR>
<BR>
On Mon, 2009-04-13 at 13:21 -0700, Steve Edwards wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
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: <<A HREF="sip:spa841@192.168.0.19:5060">sip:spa841@192.168.0.19:5060</A>>;tag=d732d5ba46660f68i0
From: "asterisk" <<A HREF="sip:asterisk@192.168.0.1">sip:asterisk@192.168.0.1</A>>;tag=as51d58666
Call-ID: <A HREF="mailto:7e81b5850a48114430b5bd505bfd31dd@192.168.0.1">7e81b5850a48114430b5bd505bfd31dd@192.168.0.1</A>
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 '<A HREF="mailto:7e81b5850a48114430b5bd505bfd31dd@192.168.0.1">7e81b5850a48114430b5bd505bfd31dd@192.168.0.1</A>'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.19:5060:
OPTIONS <A HREF="sip:spa841@192.168.0.19:5060">sip:spa841@192.168.0.19:5060</A> SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5
From: "asterisk" <<A HREF="sip:asterisk@192.168.0.1">sip:asterisk@192.168.0.1</A>>;tag=as079a9a44
To: <<A HREF="sip:spa841@192.168.0.19:5060">sip:spa841@192.168.0.19:5060</A>>
Contact: <<A HREF="sip:asterisk@192.168.0.1">sip:asterisk@192.168.0.1</A>>
Call-ID: <A HREF="mailto:16bb21000690e22e53bff2f90b43d6e2@192.168.0.1">16bb21000690e22e53bff2f90b43d6e2@192.168.0.1</A>
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 <A HREF="mailto:sedwards@sedwards.com">sedwards@sedwards.com</A> Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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</PRE>
</BLOCKQUOTE>
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