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

[Steve Mann]

In the zapata.conf where you define your channels, you would also define
them as part of a group.
Then in your dial plan, when you execute the dial command, you would pass it
the ZAP/group_name

This will tell the dial command to use the first available channel within
the group you have defined.

see: http://www.voip-info.org/tiki-index.php?page=Channels%20and%20Groups
for more info.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of David John
Walsh
Sent: Tuesday, April 05, 2005 1:18 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] "Multiplexing" (or what ever the term is)
FXOports into a "Trunk"


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
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