HELP! No SIP device (truck or phone) will register with Asterisk server

HELP HELP HELP!! I am so frustrated. I’ve been looking online for hours and hours and can’t seem to find any solution. I’m a newbie to the FreePBX/Asterisk world and I’m sure the solution is simple but I can’t find it. No matter what I do I can’t get any device to register with Asterisk.

The Asterisk box, soft phones, and hard phones (Polycom 650) are all on the same LAN. On the outside I have a trunk with Aretta that won’t register either. Asterisk continually reports 0 SIP registrations.

I’m suspecting it may be a port issue as when I check for port 5060 inside or outside of the LAN, the connection is always refused. Here’s the netstat from the box itself for asterisk:

tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 7657/asterisk
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 7657/asterisk
udp 0 0 0.0.0.0:5000 0.0.0.0:* 7657/asterisk
udp 0 0 0.0.0.0:2727 0.0.0.0:* 7657/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 7657/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 7657/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 7657/asterisk
unix 2 [ ACC ] STREAM LISTENING 23668 7657/asterisk /var/run/asterisk/asterisk.ctl

Inside of FreePBX I’m assuming that the extension numbers are the SIP account name and secret is the password? I know it’s not the router (because this is over my LAN) and I’ve disabled the box’s firewall which still gives me connection refused when I try to access 5060 from my own computer.

The simplest test I’ve been running is using the soft phone AdoreSoftphone which continually gets rejected from the server. I’ve even switched ports from 5060 to 5061 with no success.

Any ideas anyone? I’m hoping there’s a brain out there who can point me in the right direction. I’m at a dead end!

Here’s the lowdown:
FreePBX 2.7.0.2
Asterisk 1.6.2.6
CentOS 5.4

Thanks in advance!

After you’ve created the extensions, have you clicked on Apply Configuration Changes?

I have the Adore SIP softphone, the X-lite softphone, and the Polycom 650s. None of them can authenticate and register with the server. They’re all on the same LAN as the server.

Yes, all configuration changes have been applied, updated, restarted, etc. A lot of what I’m reading on the web seems to point to domain problems. I don’t have a domain assigned to the server, just an IP. Whether I use 192.168.0.100 or specify the port in that 192.168.0.100:5060, nothing seems to work. The server won’t talk to the devices and it’s driving me nuts!

If you turn on SIP debug from the Asterisk CLI do you see the device trying to register?

You can also run tcpdump.

Here’s what showing up in the log:

[Apr 27 18:38:48] NOTICE[7692] chan_sip.c: Registration from ‘"Lee Hamblin"sip:[email protected]’ failed for ‘192.168.0.101’ - No matching peer found

I have an extension that is 1250 with the correct password so I don’t know why it’s coming up with no matching peer found. Any ideas?

Although this is different since I’m at my home now, here’s the output (and don’t worry, I’ve changed the IP address as not to compromise the server):

<------------>
Scheduling destruction of SIP dialog ‘NmFkM2JkNDQwYTIyYjc3YjUxMzY0MzhiNmI4NjQ0ZmQ.’ in 32000 ms (Method: REGISTER)
ixyo*CLI>
<— SIP read from UDP:68.112.74.59:50317 —>
REGISTER sip:70.121.03.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:13935;branch=z9hG4bK-d8754z-d537a6689206a405-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:13935;rinstance=1f40b8702f69aaaa
To: "Lee Hamblin"sip:[email protected]
From: "Lee Hamblin"sip:[email protected];tag=ce2c2f3e
Call-ID: NmFkM2JkNDQwYTIyYjc3YjUxMzY0MzhiNmI4NjQ0ZmQ.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.47 : 13935 (no NAT)

<— Transmitting (no NAT) to 192.168.1.47:13935 —>
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.1.47:13935;branch=z9hG4bK-d8754z-d537a6689206a405-1—d8754z-;received=68.112.74.59;rport=50317
From: "Lee Hamblin"sip:[email protected];tag=ce2c2f3e
To: "Lee Hamblin"sip:[email protected];tag=as3a45825c
Call-ID: NmFkM2JkNDQwYTIyYjc3YjUxMzY0MzhiNmI4NjQ0ZmQ.
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

It’s trying to register with your name, you have to use the extension number.

I spent some time on the phone today and found that my entire Asterisk system is corrupt. We wiped everything, reinstalled, and now the SIP devices are registering. Don’t know what the underlying problem was though.

Thanks for those who helped!

The name field it’s passing is the “Display Name” in FreePBX which they say is for the caller ID name and not for the extension. I changed it anyway to the extension number and I’m stilling getting the same SIP/2.0 404 Not Found error.

<------------>
Scheduling destruction of SIP dialog ‘OWQ2ZjIwMTQ0ZDAxZDBjMTA3ZjM1ODBkNmZjMTExMGM.’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘OWQ2ZjIwMTQ0ZDAxZDBjMTA3ZjM1ODBkNmZjMTExMGM.’ Method: REGISTER
ixyo*CLI>
<— SIP read from UDP:192.168.0.101:37724 —>
REGISTER sip:192.168.0.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:37724;branch=z9hG4bK-d8754z-4141213512791a70-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:37724;rinstance=67b1c6cd3c8e8292
To: "1220"sip:[email protected]
From: "1220"sip:[email protected];tag=3d75c251
Call-ID: NzRiNTBlZDM5M2Q0MDM4MjU1YjE2NzEzM2Q4ZDM2ZGQ.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.0.101 : 37724 (no NAT)

<— Transmitting (no NAT) to 192.168.0.101:37724 —>
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.101:37724;branch=z9hG4bK-d8754z-4141213512791a70-1—d8754z-;received=192.168.0.101;rport=37724
From: "1220"sip:[email protected];tag=3d75c251
To: "1220"sip:[email protected];tag=as55aeaed0
Call-ID: NzRiNTBlZDM5M2Q0MDM4MjU1YjE2NzEzM2Q4ZDM2ZGQ.
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Try following this webpage on setting up a softphone. I’m pretty sure that you extension/secret on the phones are not matching the extension/secret on the asterisk box. It’s going to be something simple…