I can’t see why they have to hunt through the 3 lines before voicemail. Especially if it is sip, even going through ata’s

With the 1232 you don’t have sip option, you are still working with analogue lines at the system side. Normally old lines would only roll over if the other one is busy. If the line is ringing on line 1, what use is it going to line 2 and act as a second call on the system

I would upgrade to NS and connect via sip directly and not use the Ata adapter or get a voicemail unit for the 1232 and don’t use the providers mailbox

“I have not failed. I've just found 10,000 ways that won't work.”