[Asterisk-Users] ASTERISK CALL ROUTER

[Dumpolid Exeplish]

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

CONNECTION
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) 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.

DIAL PLAN
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

i.e
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
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