<html>
<head>
<style>
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Verdana
}
</style>
</head>
<body class='hmmessage'>
Can you do a 'sip show peer bt201' and show us what the output is? I don't see anywhere that indicates that your phones are even registered or trying to talk to Asterisk.<br><br>Also do a 'sip show peers' and show us what that gives you.<br><br><br><br><hr id="stopSpelling">From: jonas.kellens@telenet.be<br>To: asterisk-users@lists.digium.com<br>Date: Mon, 13 Apr 2009 23:10:13 +0200<br>CC: asterisk.org@sedwards.com<br>Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)<br><br>
<br>
[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>
<pre>On Mon, 13 Apr 2009, jonas kellens wrote:<br><br>> 1) IP-phones get there IP from a DHCP<br><br>The source of the address is not the issue.<br><br>> I still see no register-message on the CLI. This really should happen<br>> now, as they are defined host=dynamic !<br><br>I suspect you have not [correctly] configured the phones to register to the Asterisk server.<br><br>> I will now hang my portable on the switch and monitor the network with<br>> wireshark to see if the phones send a SIP-register to the<br>> Asterisk-server...<br><br>"sudo netstat -a -n -p | grep 5060" will show you if Asterisk is actually <br>listening. It should look something like:<br><br>udp 0 0 0.0.0.0:5060 0.0.0.0:* 3283/asterisk<br><br>"sudo tcpdump port 5060" will show you if the phones are talking to the <br>box. It should look something like:<br><br>13:11:30.432163 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, length 431<br>13:11:30.432443 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 398<br>13:11:30.432520 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 460<br>13:11:30.451350 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, length 578<br>13:11:30.451525 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 398<br>13:11:30.460889 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 481<br>13:11:30.461231 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 476<br>13:11:30.461541 IP asterisk.example.com.sip > spa841.example.com.sip: UDP, length 540<br>13:11:30.474515 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, length 383<br>13:11:30.497854 IP spa841.example.com.sip > asterisk.example.com.sip: UDP, length 319<br><br>"sip debug" at the Asterisk console will show the messages as the are received and responded to by Asterisk. It should look something like:<br><br><-- SIP read from 192.168.0.19:5060:<br>SIP/2.0 200 OK<br>To: <<a>sip:spa841@192.168.0.19:5060</a>>;tag=d732d5ba46660f68i0<br>From: "asterisk" <<a>sip:asterisk@192.168.0.1</a>>;tag=as51d58666<br>Call-ID: <a href="mailto:7e81b5850a48114430b5bd505bfd31dd@192.168.0.1">7e81b5850a48114430b5bd505bfd31dd@192.168.0.1</a><br>CSeq: 102 NOTIFY<br>Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK25449e4a<br>Server: Sipura/SPA841-3.1.4(a)<br>Content-Length: 0<br><br>--- (8 headers 0 lines) ---<br>Destroying call '<a href="mailto:7e81b5850a48114430b5bd505bfd31dd@192.168.0.1">7e81b5850a48114430b5bd505bfd31dd@192.168.0.1</a>'<br>12 headers, 0 lines<br>Reliably Transmitting (no NAT) to 192.168.0.19:5060:<br>OPTIONS <a>sip:spa841@192.168.0.19:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5<br>From: "asterisk" <<a>sip:asterisk@192.168.0.1</a>>;tag=as079a9a44<br>To: <<a>sip:spa841@192.168.0.19:5060</a>><br>Contact: <<a>sip:asterisk@192.168.0.1</a>><br>Call-ID: <a href="mailto:16bb21000690e22e53bff2f90b43d6e2@192.168.0.1">16bb21000690e22e53bff2f90b43d6e2@192.168.0.1</a><br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 13 Apr 2009 20:18:30 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Length: 0<br><br>Thanks in advance,<br>------------------------------------------------------------------------<br>Steve Edwards <a href="mailto:sedwards@sedwards.com">sedwards@sedwards.com</a> Voice: +1-760-468-3867 PST<br>Newline Fax: +1-760-731-3000<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br></pre>
</blockquote><br /><hr />Quick access to your favorite MSN content and Windows Live with Internet Explorer 8. <a href='http://ie8.msn.com/microsoft/internet-explorer-8/en-us/ie8.aspx?ocid=B037MSN55C0701A' target='_new'>Download FREE now!</a></body>
</html>