[Asterisk-Users] "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"

[David John Walsh]

Hi all,

For an event we are doing, we have been donated several analogue PSTN
lines and an 8 port FXO bridge.

On the bridge, we have set up each of the ports to work on the SIP
protocol, and have referenced them, line1, line2, line3 etc for their
username / password.

I have placed the config in sip.conf, and they all work fine, inbound
and out - for testing anyway!

How do I get asterisk, to treat these 8 lines as one 8 call limit
trunk?  From a users perspective, all he/she needs to dial is
9xxxxxxxx (where x's the number) to get any of the 8 outside lines?

Sure I could "hardcode" somthing in each part of the extensions.conf,
but if this trial is sucsessful, the number of lines may increase, and
it would be nice to define the array once as it were.

(I am aware that most of my troubles would go away if I used a more
intelligent termination such as ISDN, but for several issues, its not
possible)

Thank you for your time on this matter.

David