Hi,
A friend recently give me a old Cisco 7970 IP phone. I am really new on Cisco phones is the first I get and I am trying to understand how to connect this phone to my local FreePBX server. So I am reading some articles about TFTP server and some xml configurations. Is any tutorial/video or notes with steps I should do to join this phone to FreePBX?
Thank you!
The first thing you have to figure out is if the phone is even running SIP firmware. Those phones when originally manufactured were shipped with SCCP firmware.
The easiest way to do this is setup your TFTP server to hand out the Cisco-phone specific info and then look in your TFTP server logs to see if the phone is attempting to load a firmware file that starts with P00 or P0S
If P0S then it’s running SIP if P00 then it’s got SCCP firmware on it. You will have to update it to SIP.
This isn’t trivial. If you are really really REALLY interested in doing this I can post a lot of info on how to do it but I don’t want to spend the time posting it if all your going to do is read it, and think “that’s too much work” and then not do it.
The reality is that there are many other Cisco models that are newer such as the 88xx series that are very cheap and have a lot more cooler features (such as video) Getting your 7970 running can be done but it’s in the camp of “I can call myself a FreePBX God if I can do this” instead of “I just want to play with a cheap phone” So what do you want?
My load file is SIP70.8-3-4SR1S, so I guess is SIP firmware.
I have an TFTP server and I am thinking about content of cnf.xml, the file should be SEP(MAC).cnf.xml and I am trying to make the xml file for the SIP extension what I have on FreePBX.
If any example code ready for 7970 exists please share it to test it.
My friend will give me another 2 7970 phones so I am trying to use these phones and not buy some newer modes, I want just to works as phones with basic functionallity, all others functions do the FreePBX.
I test a lot of SEP(MAC).cnf.xml and SIP(MAC)cnf.xml but nothing works, an on FreePBX nothing on logs. Is any working minimal xml and instructions?
There is a reason why used Cisco phones are so cheap. Good hardware…awful SIP-software.
The 7970 is more than 20 years old…I think.
Go to the nz-website and read…it takes weeks for a newbie to configure a Cisco phone with freePBX. Yet, they work pretty well once you spent the required time. I still use 10 Cisco 8961 phones with a freePBX16 server. The phone hardware is very robust…not like current phones.
No. Just stop. I don’t even think that’s a 7960 load file in the first place.
You can’t just stick any old file into your TFTP directory for these phones.
Let’s start at the basics here. Setup your TFTP server so that it logs every request. For example on Debian 12 you would do an apt-get install tftpd-hpa and the config file would be something like:
root@phony:/etc/default# cat /etc/default/tftpd-hpa
/etc/default/tftpd-hpa
TFTP_USERNAME=“tftp”
TFTP_DIRECTORY=“/srv/tftp”
TFTP_DIRECTORY=“/tftpboot”
TFTP_ADDRESS=“0.0.0.0:69”
TFTP_OPTIONS=“–secure -vvv”
root@phony:/etc/default#
then you would see the transfer requests logged in /var/log/syslog or some such
For your DHCP server (isc-dhcp) you need this option:
dhcp-option=150, 172.16.1.3,172.16.1.16 # Cisco phones TFTP Server
that would be if your tftp server was at 172.16.1.3 and a second tftp server at 172.16.1.16
Now, plug in the phone and make sure it’s powering up. It will check for option 150 in the DHCP ip address handed to it. Then it will attempt to load a basic config file from the TFTP server. (it actually may attempt to load a lot of config files)
Look at the logs and post the list of configuration files the phone is attempting to load, and I’ll explain the next step you need to do.
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I flashed the 9.2 sip firmware successfully using a tftp server. I fix the SEP(MAC).cnf.xml file with the basic using the .nz dicumentation and other sources on net.
All good, but stucks on “Registering”. I use a FreePBX 16 and asterisk 18. I make a chan_sip extension using 8 digits passwords and I try later 10 digits.
On sip debug I have the error 401 Unauthorized. I have disable nat on sip settings and set static ip the ip of freepbx. On .xml file I have the nat to false. Any ideas?
Try enabling sip on both TCP and UDP. Your config file may have it registering on TCP and you only have UDP enabled or vis-versa
Post your current xml-config-file…
Yes, after hours and hours I find it, now all works with TCP.
Thank you all!