I think I was able to answer my own question. At first the Linksys / Cisco SP8000 did not have any trunking options so I had to go to cisco's website and follow the tutorial on updating the firmware on the unit.https://www.cisco.com/c/en/us/suppo...ttings-on-the-spa8000-phone-adapter.html
After I did that the trunking options showed up.
* I went to Sipura web interface by typing the IP of my unit into Chrome, clicked on Admin login and Advanced
Clicked on the SIP tab scrolled down to bottom to ensure Trunk Parameters section Hunt Policy was "onhook only"
* Then I went to T1 tab at the top (Trunk 1)
* It essentially gets setup like a single SIP DID Line (FXO ports) would be so it has all the pertinent parameters that would need to setup with your SIP provider like proxy, port, User ID, Password, etc. I mirrored my settings of my main DID
* It also had a section called Contact List: which was set to 1,2,3,4,5,6,7,8,hunt=re;*;1 (appearantly to hunt through all the lines)
* I submitted changes and it took a lot longer to refresh then the 5 seconds it claims it needs. Then when it got back up all my lines could dial out (over the main DID) and now when I call in it starts on line 1 if its busy line 2 if its busy line 3 and so on.
I verified that my main DID can handle up to 25 concurrent calls. This is something that I assume could potentially throw off this functionality. My provider VoIP.ms has an unlimited subscription DID which only supports 3 concurrent calls which I am sure would not work if you wanted all 8 lines in the Trunk (Hunt Group) as I do. Perhaps you could have 3 in the Trunk in that case.
I imagine the Grandstream interface is much less combersom and more user friendly but I think its working. I also had to go into the Panasonic software (KX-TD Programmator) and configure the TRG #1 to include all of the lines and then I went into each extension for the office and set lines 1-8 to ring immediately on Day and Night. I have not been able to get my Dial Plan working that was setup on the individual Lines/FXO ports so I just left that as at default ([*#0-9A-D][*#0-9A-D].) which means I have to dial the whole number with the 1 on the front.
Anyway thanks for all your feedback, it is nice to know that they have added this functionality now to my ATA and that I don't have to rely on my SIP provider to have the option. I really appreciate this forum. Every time I use it you guys do an amazing job of pointing me in the right direction. Now on to that KX-TVS50...