Cisco IP Phone CP-7821 Configuration

I’ve been trying to configure the Cisco CP-7821 IP phone through the web interface. I go to the phone’s local IP address and there is no web configuration portion of it. Just an overview of everything that is set on the phone. I have tried looking everywhere to see how I configure the phones with their secret password as well as telling the phone which extension it has. Please help, thanks!

I believe the 7821 have a similar configuration file as the 9951 and 9970 as I see here: https://supportforums.cisco.com/discussion/12589941/configuring-cisco-cp-7821-sip

If i’m right then this configuration template and guide should work: http://labs.wrprojects.com/configuring-a-cisco-9951-phone-for-asterisk/

Regarding the web interface, there is no web interface to configure sip parameters but this: https://github.com/xpheres/sipManagerSuite/

This configuration tool is unfinished and for the moment just allow you to see the key parameters of the config file in a form. It will allow you to change and save those parameters in the future. Even making easier to configure the config file, you still have to send it to the phone yourself with a TFTP server.

I hope it helps

ah yes, this helps me with the config file as I have nothing to go off of. I need to convert the 7821 from skinny to sip, but I need a little bit more knowledge on the tftp server within freepbx or what to use outside of it for things like this. I appreciate all of the help you are doing for me

I saved for myself this great guide from a website that is not available anymore: http://lingoworld.eu/websites/zenbixsavedsite/DHCPCisco.pdf

Please if the website is available again let me know, I just want to keep the guide available as long as the original site is down.

The guide is for the case that the phone is bricked but it documents very well the process from the begining even if your telephone is not bricked and you install sip firmware for the first time.

Let me resume the process by steps:

  1. Install sip firmware. The first time you do this is tricky, because you can not tell the phone the IP where the TFTP server is installed, so the workaround is to change the IP address of the computer that host the TFTP server to 192.168.1.108 and start the TFTP server with all files (I’m not sure if this problem applies to your model too, otherwise just tell the phone the ip of your TFTP server). Please follow the guide I linked to complete this process if you are with windows (software used can be found here http://tftpd32.jounin.net). For linux install any tftp server and move files to tftp folder (in my case /srv/tftp/)

  2. Copy a configuration template for your telephone model and change parameters for your asterisk server.
    As I said, I believe you can follow the guide for 9951 I posted above, please give it a try. Upload it with your TFTP server and let me know if it works.

If everything is allright you are done.

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I appreciate how helpful you are being. I am curious though on where can I get a configuration template from? I tried doing a google search but nothing had came up for it. and if I am understanding this correctly I just place all of this in the root of the tftp server folder and label the configuration file like this SEP[MAC_ADDRESS].cnf.xml with the other files for the firmware?

yes, try the configuration file of the 9951 and 9970 from the site I gave you above. Key parameters are: < processNodeName >IP OF YOUR PBX< /processNodeName >
< sipPort >PORT USED BY YOUR EXTENSION< /sipPort >
< startMediaPort >START PORT USED FOR MEDIA< /startMediaPort >
< stopMediaPort >END PORT OF THE RANGE OF PORTS USED FOR MEDIA< /stopMediaPort >
< line button=“1” >
< featureID>9< /featureID >
< featureLabel >NAME OF THE LINE< /featureLabel >
< proxy >USECALLMANAGER< /proxy >
< port >PORT OF THE EXTENSION IN PBX< /port >
< authName >EXTENSION AUTH NAME< /authName >
< authPassword >EXTENSION PASSWORD< /authPassword >
< /line >
< loadInformation >NAME OF THE FIRWARE. EXAMPLE: SIP41.8-0-4SR3AS< /loadInformation >

In model 9951 USECALLMANAGER keyword should be used inside the labels < proxy > anything else won’t work