CP-6941 Sip/Register with Asterisk server

[shift24]

I am running into the issue of getting my Cisco 6941 operational on my network for the Sip firmware with the FreePBX installation (Asterisk). I know that Cisco has made it obsolete, but there is only the 6901 available on their website, and I would like to know if anyone has those files. If anybody could also help me configure the phone with the Sip firmware without the Cisco management software? Please let me know; thank you.

I have tried researching it myself, but I have been unable to get the phone to reset using the hold pound key method that I saw other models able to use. I found some links on previous related posts about the 6941 setup/installation, but they were dead links.

Cisco Unified IP Phone 6961 - Retirement Notification - Cisco

[Leo Laohoo]

Text files?  What text files? 

What is the syntax of the configuration file of the phone?  Is the filename extension "txt" or "cnf.xml"?

[shift24]

No, the text file was only the phone log via the IP address; they were copied and pasted from the console log. The files are being read by the phone since the NTP server and the admin password is correctly applied to the phone. The file extension uses the ".cnf.xml". I included the XML file in a previous message.

[shift24]

At the moment, I am trying different settings with the FreePBX server using the legacy SIP protocol.

[shift24]

I have spent about a day or so trying different XML files to see if maybe the issue is related to them with no luck, but maybe it could be because of my network. At the moment, I am using a Cisco-only network with a router and two redundant switches. My server and phone are on completely opposite sides of the network but are completely able to ping and communicate with the tftp server. The XML file is being loaded as the settings password updates to the new one that I change in every iteration of the file. I also used the legacy SIP (chan_sip) protocol and chan_pjsip with the deactivated option for NAT (no, force) and fiddled with the advanced settings like transport for TCP only / etc. I tried using this config Cisco-IP-Phone-Provisioning-Files/6921.cnf.xml at master · NamoDev/Cisco-IP-Phone-Provisioning-Files · GitHub. Is there anything else that I could try?

[Leo Laohoo]

<transportLayerProtocol>1</transportLayerProtocol>

Try that.

Upgrade the firmware of the phone to, say, 9.4(1)SR3.

[shift24]

Thank you, unfortunately, the version I had is the only one I can find online for SIP, as the Cisco software website only has 9.3 for 6901. I got the install files from a pretty sketchy website, but it seems to be legitimate (6921, 6941 & 6961 IP Phone (SCCP & SIP) (firewall.cx)).

EDIT:

I found this link on Softpedia for 9.4.1.3 claimed for 6921. Would you know if this firmware is legitimate and compatible? I'm not a huge fan of trusting websites that I just heard of today.

[Leo Laohoo]

"9.4.1.3" is only 9.4(1)SR and not 9.4.(1)SR3.  

Try that nonetheless.  

And I have no idea if the file is legit or not. 

[shift24]

Thank you for all of your advice, I finally got it to register with SIP. I ended up upgrading the version from 9.2.1 to 9.4.1.3 version from Softpedia. I ended up getting it to work using the pjsip driver with TCP transport in the driver settings set to enabled. My final config is attached below. 

My only question left would be that my 6941 phones are unable to place a call on hold and just end the call instead. Is there a way to fix this or strictly a restriction for SIP?

[Leo Laohoo]

The config file is going to help a lot of people.  

Thanks for taking the time to upload it. 

In regards to putting a call on hold, check the settings of the extension on Asterisk.  Run an Asterisk debug is another thing I would recommend.

[shift24]

Thank you, I'm still looking into the settings, but I got the debug out. I added a question to the freepbx community here: Configuring Cisco Phone Hold Button - FreePBX / Configuration - FreePBX Community Forums. The debug is attached below, as far as I can tell it may be something I can put in the XML, but im still looking into it.

[shift24]

I appreciate the template, but I noticed some things might need to be added. You mentioned that the username needs to be changed, but I don't see the <name>PEERNAME</name> operation, which I thought was required for registration.

[Leo Laohoo]

Same respond with the another thread, "Start by searching in this forum for "SEPmacaddress.cnf.xml" and Asterisk."

[Leo Laohoo]

To get this to work, you will need the following: 

1. A known working SEPmacaddress.cnf.xml template.  I have a known working SEPmacaddress.cnf.xml for a 9971 (LINK) and it will have no problem working with a 6941.
2. TFTPd64
3. DHCP Option 150
4. Dial plan
5. Phone firmware