Configure Asterisk Settings and SIP Trunking, and integrate with vTiger

I have 2 VPS. One has the Digium Asterisk GUI installed the other has the AsteriskNOW FreePBX GUI. ***I only plan on keeping one, whatever is easiest to use, most convenient, and cheaper to work with. So I will let you select which one is most convenient and I will only need work done on that one. Please bid accordingly and give me some of the info on this through message. I need a very basic setup. I have a VOIP phone in my office that I want it to ring to and I need to have the voicemail setup. I also want to have the ability to to make it ring simultaneously to my cell phone sometimes through call forwarding. I have SIP Trunking through [login to view URL] Which I plugged in the information but I do not know if I did this right. Probobly not so I would need your help with that. You also have to be able to explain to me what and how you are doing everything just in case I need to go back and make simple changes. The most important thing I need done is I need it to be integrated into my vTiger. Looks relatively simple but as I do not have experience with this I do not know how to do it. I am relatively new to asterisk, however I have some basic knowledge with coding making websites, CRMs, etc, so I have a good idea as to the prices and how long things should take so please be fair and everything will go smoothly. *** Now I do not know if this is possible, and its not really necessary, but I am also curious to know if there is a way to log in to the asterisk GUI without using putty. I don't understand the point of having a hostname if I cant even use it.