It seems that you are using the wrong XML config file for the Cisco 7975.
5:46:10p TFTP Error : softKey9971.xml
It looks like its attempting to register, it shows the phone lines sometimes then the phone restarts, I’ll see what I can do on some traffic captures
That part I was able to fix, the main config had it labeled wrong
I’m not too sure what to do for this, I got a xml that was already preconfigured for SIP and I have been putting the edits in it
A XML config file for e.g. the Cisco 8961 does not work with the Cisco 7975. Your main config file seems to be for the Cisco 9971, because it points to a 9971keys file.
You are wasting your time with a 20 year old phone. If I were you, I would go with the new Sangoma P-series phones. They are supposed to be plug & play.
Every Cisco 79xx phone thread is 100 posts long and full of trial and error, but mostly error. We are only at 46 posts so far. Hang in there; it’ll be another week until ultimate defeat.
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I feel like its worth trying at least, a lot of progress has been made and I think the only issue I am having left is that it cant pull locale information, which may fix the issue with it not fully connecting to the pbx
No, the locale info is not important. @Charles_Darwin is correct that you cannot use a 9971 config for a 7975.
Ok, do you know where I can get the correct file?
No, I don’t. I’m running SCCP on my 7965 phone and got a config template from the chan-sccp-b project.
I posted a correct file above, but you would have to adapt it based on the nz-website…because otherwise you would end up with a german Interface and ringtone and time 
EDIT: sorry my fault…my config needs the patched Asterisk…
So I dont understand how to do the patch, is there any info on that?
To patch Asterisk in freePBX you have to be a more advanced user. What do you want to achieve? The Cisco 7975 has only rudimentary functionality with a non-Cisco-phone system (without adaptation). It has not been designed for the SIP-protocol. The SIP-firmware is full of bugs.
You can buy the freePBX Endpoint Manager module (EPM), but it gives you only basic functionality (no phonebook, no BLF, etc). I am not sure, if there exists a (old) EPM template for the Cisco 7975.
10 years ago, I spent many, many, many hours to adapt Cisco 7975 and 8961 phones for use with Asterisk/freePBX. I still use the Cisco 8961 phones at one location, because they have superior audio (g722) and they can handle a live image of a door camera…but at all other locations I use Digium/Sangoma phones…and will do so in the future!
A deskphone without a phonebook and without BLF (busy lamp field)…what is it good for??? Think again…
I wouldnt need to worry about a phone book since this is for personal use, not using within a company, BLF would be nice but its not super important since the phone will only be used to contact one other phone. Would a CP-8851 work better for freepbx?
Here is a description of the status quo…but you would have to buy EPM (freePBX)
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=216182415#content/view/216182415
Please don’t recommend this to anyone. It uses chan_sip and there’s been no effort to make this work with the supported SIP driver in Asterisk. Unless the maintainer gets off their butt and does something to make this work in chan_pjsip this has a finite lifespan (4 more years) unless you decided to never go beyond Asterisk v20 to support your $10, 15yo phones.
I’d not encourage it! Probably better for this hacky Cisco stuff to go away with chan_sip than for it to perpetuate.
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I 100% agree. The maintainer keeps it updated for current versions of Asterisk. It supports the current LTS versions of Asterisk and keeps getting updated at least every 6-8 weeks to follow the Asterisk release cycle.
So really at this point, if they are doing active maintenance for this but haven’t bothered with a chan_pjsip offering I don’t think it’s going to happen. It very well could be they can’t do certain things in chan_pjsip like they could with chan_sip. I’m guessing the fact that chan_pjsip isn’t just one huge bloated module and is a collection of like 30+ modules made it too complex.
So what would I be able to do to get the 7975 and/or the 8851 to work on the free side of freepbx? I know that the 7975 is at least talking to the server for tftp data but still cant get the phone to pull good locale info and register and the 8851 is just stuck on detecting network and in status it says tftp timeout
I still don’t understand, what you are doing and why. Why do you need a freePBX system, when you just want to call another phone?
Register for an online SIP-number and use any other IP-phone with a web-user-interface, e.g. an old Snom.