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Problem with Cisco CP7911G registration
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techtechnique
Brekeke Newbie


Joined: 17 Feb 2011
Posts: 2
Location: UK

PostPosted: Thu Feb 17, 2011 8:33 am    Post subject: Problem with Cisco CP7911G registration Reply with quote

1. Brekeke Product Name and version: SIP Server 2.4.7.3/286.1

2. Java version: 1.6.0_23

3. OS type and the version: Windows Server 2003

4. UA (phone), gateway or other hardware/software involved: Cisco CP7911G (8-5-4S SIP firmware), BT HomeHub 2.0 router (Rooted)

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 4

6. Your problem: I have several Cisco handsets that I have flashed with SIP firmware and am trying to get them to register to my SIP service, which is hosted in an external data-centre.

After much fiddling about, I have managed to get the handset to appear to be registered on the Ondo server, but it seems that it is ignoring the 200 OK message sent from the server.

I know that Cisco handsets work on an asymmetric NAT basis, which is RFC compliant, but I have been told that the Ondo will only work in a symmetric NAT configuration.

I have successful registrations and full operation from X-Lite softphones and Sipura SPA942 hardphones, but my remit for the company I work for is to get these Cisco handsets to work, as we have over 800 units to deploy.

My internet connection does not have a static IP address, but I can use a Dynamic DNS host, although this does not seem to make any difference.

I have disabled SIP ALG in my router to no effect.

Not knowing too much about the configuration of the Ondo, am I missing something that could be causing the problem server-side?

Any help would be very gratefully received!
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Haddas
Brekeke Guru


Joined: 17 Jan 2008
Posts: 170

PostPosted: Thu Feb 17, 2011 12:50 pm    Post subject: Reply with quote

Is the SIP server connected to the Internet directly?
I means.. does the SIP Server's PC have a global IP address?

Are you using a STUN at the CISCO phone?
If yes, disable it.

Are you sure that the SIP Server's 200 OK message reaches the CISCO phone?
Is it the correct IP address?
Is it the correct UDP port??
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techtechnique
Brekeke Newbie


Joined: 17 Feb 2011
Posts: 2
Location: UK

PostPosted: Fri Feb 18, 2011 2:45 am    Post subject: Reply with quote

Thanks for your response.

The SIP server is hosted in a Data Centre in London somewhere (the details I am not too clear of as it is managed by another member of my team and he is on vacation at the moment), and I believe it to be directly connected to the internet, albeit firewalled with all the relevant ports open.
The Brekeke SIP server listens on port 5065, and the actual SIP PBX (Advoco) is on port 5060. (Again, I've no idea why it was set up this way...)
Our "live" system has the Brekeke server on a different machine, naturally with a different public IP address, and the Cisco has a problem with sending the FQDN and the port of 5065 to a different IP address, so for the purposes of this exercise I have installed a 60-day trial version of the Brekeke SIP server on the same box as the SIP PBX, which seems to get round this problem.

There seems to be no option for STUN on the Cisco handset, and yes, I'm pretty sure that the 200 OK messages are getting through. Cisco are deliberately unhelpful in providing zero support for SIP connectivity and configuration to non-Cisco VSPs, but I have learned a great deal from other forums (in particular the Asterisk ones...) about how the configuration should be.

I've got a couple of Wireshark traces taken this morning, but I'm not sure how to paste them to this thread, being a bit of a noob to this forum!

*edit*
Files below...
http://www.4shared.com/file/TOGm1G3f/cisco-005.html
http://www.4shared.com/file/k31pfw4z/ondo-005.html
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Haddas
Brekeke Guru


Joined: 17 Jan 2008
Posts: 170

PostPosted: Sat Feb 19, 2011 1:54 am    Post subject: Reply with quote

From packet traces, it seems you set the "Add 'rport' parameter (Receive) = on" because 200's Via header has "rport" parameter.

The CISCO phone may not accept it..
so try the "Add 'rport' parameter (Receive) = off".
You can tune it at the [Configuration] > [SIP] page.
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