Need help with asterisk router!

hi everyone,
i am trying to configure an * server to route calls from the PSTN to our internal PBX

This is the IDEA
Currently we have a PANASONIC KT1232 PBX that provides intercom calls and facility for call out lines on it (8 call out lines are pressent on it)

we also currently have 4 remote sites with which we communicate wil over VOIP using quintum boxes (the quintom box at the HQ has two lines connected to the PBX as a CO lines).

Now, i would like to implement * to take the place of the quintum, implement IVR and voicemail

My plan is to install a TDM2424E (FXO=16ports, FXS=8ports)card on the * server, connect 8 of the FXO ports to the PSTN analog lines we use, connect the rest (8 FXO prots) to the PBX and configure them as extensions (intercome lines) (this would be easier that trying to convince the KT1232 to see an E1 line to * as a TIE connection, and also eliminate the cost of purchasing a TIE/E1 card). The FXS ports will be connected to the CO ports on the KT1232, thereby replaing all the call out lines (since the call out lines have been connected to the * FXO ports).
For VOIP, the quintum boxes(AS series) at the remote sites will have to register to * as peers (using the SIP UA on the quintum), the last collection to this is that a GSM gateway (Voiceblue) wold be added to the whole scheme to terminate and originate GSM calls. Voiceblue would communicate with * as a peer using SIP.

now, when calls come in on the analog PSTN, * awnsers the call and plays a file for the dialing party to enter the extention of the person he/she would like to talk to, and if the recipient is unavailable or busy, the message could be delivered into the voicemalbox (IVR + voicemail). Then, * would dail the respective user through the intercom lines attached to the PBX

Dial (Zap/16/${EXTEN})

in this way, the call is completed to the PBX and the call rings at the recipient’s desk.
The quintum SIP lines would be configured as extensions (intercom) withing the dial plan so that anyone could also call in from the PSTN to the remote locations. Also, remote users using VOIP (quintum) would be able to dial any PSTN numbers across *.
So also calls comming in from the V/blue could be routed either through the PSTN lines or VOIP lines or directly to a called extension.

Now, for users within the office using extensions (intercom), they would be able to dial out (oringnate calls) through the 8 connected FXS ports (the KT1232 could be configured to route all outbound calls directly to the CO group assigned). GSM, PSTN and VOIP calls would be routed accross the * server to their various destination.

Hope you got the picture

Now, my question is this

  1. What PC spec would be able to handle this. (at least, the system should be able to handle nothing less that 25 simultaneous calls at a go)
  2. Is this possible to execute??
  3. How do you configure * to try multiple extensions, i.e, if one line is busy, try the next…

All useful advices are welcomed

Maybe these will help … ntegration … mensioning