VOIP Network Help

A couple of months ago I began looking into how to design, implement and manage VOIP networks. I am very new to this in both the data and telco side, needless to say it has been confusing and need some help in filling holes.

I am trying to design a converged network that supports eveyday IP functions (web, e-mail, ftp) as well as Phone services with Digital and Analog phones. Whereas, an individual can have a “regular” phone , not having to buy a IP hard phone or use soft one and be able plug into a rj-11 jack and place phone call. However, I wish the voice traffic to travel of IP transparent to the user.

So here it goes,The network would consist of 50 ppl all living in seperate apartments. They would have a RJ-45 jack and RJ-11 in there room. One for computer and one for phone.

Here are some the elements of the local newtork. For the most part the data side I understand. But I have still have a few questions as to fill in the holes.

1.) T-1 Line ( 3 channelled out for voice for 911 calls etc.)
2.)Gateway-3600 Cisco Router QOS configured with the CSU/DSU card.
3.) Gatekeeper-SIP soft switch server running on top of Windows 2000 for Authentication,management etc.

Here is where I get lost.

What device is used to house CODECS understand and convert DTMF signaling to IP traffic?

What sort of device do I plug all the phones lines into?

What do DSP’s do and do I need them?

In addition, on the receiving end of the T-1 I want to co-locate a Media gateway with a phone or cable company that will allow access to the traditinol PSTN network with a trunk line. Also, this where the IP traffic is converted back Digital Voice and SS7 and routed over the PSTN and allow for intereoperability with regular phone users and long distance?

Can someone help with a suggestion on a proven quality device thay suports SIP and/or H323 v2 or above
that could do this.

I am sure I have missed many components and any help is greatly appreciated.

Still learning

Harris C