Steve,  

I take it this also works for SIP?

Regards
David

My appologies to the list, I did not realise that the first attempt
earlier today hit the list.

On Apr 5, 2005 7:38 PM, Steve Mann <[EMAIL PROTECTED]> wrote:
> 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: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of David John
> Walsh
> Sent: Tuesday, April 05, 2005 1:18 PM
> To: [email protected]
> 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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