SIP Registration Failure

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DonGould

3892 posts

Uber Geek

#124584 12-Jul-2013 21:48

Jul 12 09:40:58 192.168.89.233 [0]SIP:RegFailed;Retry in 30s
Jul 12 09:40:58 192.168.89.233 [0]SIP:RegFailed;Retry in 30s
Jul 12 09:40:58 192.168.89.233 [1]SIP:RegFailed;Retry in 30s
Jul 12 09:40:58 192.168.89.233 [1]SIP:RegFailed;Retry in 30s

The set up:

Elastix/Asterisk
SPA122
SPA303

The devices are failing to register.

The server is at one location and the phone/ATA is at another and I've got them connected via a VPN. It has all been running well and I have no idea why it's just decided to die.

Another phone on the local network to the server does register ok.

I can't figure out how to find out why the registration is failing.

The obvious thing was to look for log files, but there are no obvious messages in /var/log/asterisk/full

I just have a bunch of these messages:

[Jul 12 09:35:01] VERBOSE[1716] asterisk.c:     -- Remote UNIX connection
[Jul 12 09:35:01] VERBOSE[7945] asterisk.c:     -- Remote UNIX connection disconnected
[Jul 12 09:40:01] VERBOSE[1716] asterisk.c:     -- Remote UNIX connection
[Jul 12 09:40:01] VERBOSE[9700] asterisk.c:     -- Remote UNIX connection disconnected

How do I set some sort of sensible logging level so I can see why it's failing to register?

D

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DestRoYeDnz

52 posts

Master Geek

#854649 12-Jul-2013 21:59

In asterisk Console you can set

"sip set debug on"

Then Restart the device to force it to Re-register and then watch asterisk -rvvvvvvvvvvv this should show a more verbose output of SIP registrations.

Hope this helps

DonGould

3892 posts

Uber Geek

#854655 12-Jul-2013 22:18

DestRoYeDnz: In asterisk Console you can set

"sip set debug on"

Then Restart the device to force it to Re-register and then watch asterisk -rvvvvvvvvvvv this should show a more verbose output of SIP registrations.

Hope this helps

ok I'm getting lots of traffic in full now...

Sorry I didn't understand this bit...

"watch asterisk -rvvvvvvvvvvv"

is that a file somewhere, you mean to use the watch command, sorry this seems a bit idiot.

I'm getting lots of messages in full:

<------------>
[Jul 12 10:12:24] VERBOSE[1899] chan_sip.c: Scheduling destruction of SIP dialog                                      'b86efad1-176995e7@192.168.89.231' in 32000 ms (Method: REGISTER)
[Jul 12 10:12:25] VERBOSE[1899] chan_sip.c:
<--- SIP read from UDP:111.69.241.4:1024 --->
NOTIFY sip:192.168.253.5 SIP/2.0
Via: SIP/2.0/UDP 111.69.241.4:1024;branch=z9hG4bK-d6ba649d
From: "202" <sip:202@192.168.253.5>;tag=82c84af1bdd385e1o1
To: <sip:192.168.253.5>
Call-ID: bacbd4f1-2d9583e1@192.168.89.232
CSeq: 203 NOTIFY
Max-Forwards: 70
Contact: "202" <sip:202@111.69.241.4:1024;ref=202>
Event: keep-alive
User-Agent: Cisco/SPA122-1.3.2(014)
Content-Length: 0

<------------->
[Jul 12 10:12:25] VERBOSE[1899] chan_sip.c: --- (11 headers 0 lines) ---
[Jul 12 10:12:25] VERBOSE[1899] chan_sip.c:
<--- Transmitting (NAT) to 111.69.241.4:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.69.241.4:1024;branch=z9hG4bK-d6ba649d;received=111.69.241.4                                     ;rport=1024
From: "202" <sip:202@192.168.253.5>;tag=82c84af1bdd385e1o1
To: <sip:192.168.253.5>;tag=as7577a112
Call-ID: bacbd4f1-2d9583e1@192.168.89.232
CSeq: 203 NOTIFY
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                     H
Supported: replaces, timer
Content-Length: 0

<------------>

Doesn't seem to me to indicate why the registration is failing.

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ubergeeknz

3344 posts

Uber Geek

Trusted

Vocus

#854656 12-Jul-2013 22:24

That which you have there is a NOTIFY and associated response.

You need to look for a REGISTER and the associated response.

TBH it's usually easier to run a wireshark or tcpdump capture on the Asterisk box and then have a look, or if you are happy to, send it to me and I'll analyse it if I have some spare time in the weekend :)

DonGould

3892 posts

Uber Geek

#854664 12-Jul-2013 22:34

ubergeeknz: That which you have there is a NOTIFY and associated response.

You need to look for a REGISTER and the associated response.

TBH it's usually easier to run a wireshark or tcpdump capture on the Asterisk box and then have a look, or if you are happy to, send it to me and I'll analyse it if I have some spare time in the weekend :)

Hi,

More than happy to accept the help. I'm keen to keep as much in the public space as reasonable possible. I tried doing some googling and didn't really find much that is very helpful, so I'm keen to leave a public record to help others.

Can you tell me what tcpdump command you'd like to run to catch the relevant data?

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DonGould

3892 posts

Uber Geek

#854666 12-Jul-2013 22:46

http://stackoverflow.com/questions/11685903/capture-rtp-and-sip-traffic-using-tcpdump

Gave me the file I've linked to you in a PM.

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