[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)

Mon Apr 13 16:10:13 CDT 2009

[root at asterisk asterisk]# netstat -a -n -p | grep 5060
udp        0      0 0.0.0.0:5060                0.0.0.0:*
3047/asterisk 

[root at 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:spa841 at 192.168.0.19:5060>;tag=d732d5ba46660f68i0
> From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as51d58666
> Call-ID: 7e81b5850a48114430b5bd505bfd31dd at 192.168.0.1
> 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 '7e81b5850a48114430b5bd505bfd31dd at 192.168.0.1'
> 12 headers, 0 lines
> Reliably Transmitting (no NAT) to 192.168.0.19:5060:
> OPTIONS sip:spa841 at 192.168.0.19:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5
> From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as079a9a44
> To: <sip:spa841 at 192.168.0.19:5060>
> Contact: <sip:asterisk at 192.168.0.1>
> Call-ID: 16bb21000690e22e53bff2f90b43d6e2 at 192.168.0.1
> 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      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                             Fax: +1-760-731-3000
> 
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