[asterisk-users] Call Queues

[Noah Miller]

Hi Eve -

> The thing is that i have a tdm422p with the two fxo
> ports connected to the pstn. I want my sip users to be
> able to call other numbers(any number) in the pstn
> through my zap fxo channels. I have a big number of
> sip users so as you can imagine there will be
> congestion when some of them(more than two!!) want to
> call outside, that is why i want to be able to put
> those outgoing calls in a queue. For example if i want
> to call someone in the pstn and the fxo port is
> already in use, i want to be placed in a queue and
> when the channel is free my call is routed to the
> aproppiated destination. As far as i have read the
> queues are not for this kind of stuffs,  there are
> just agents or extensions that attend the calls in the
> queue and nothing more. am i wrong???

I think your suspicions may be correct.  You could add your ZAP
channels as members in queues.conf, maybe something like this: members
=> ZAP/1, and then use queue() on your outbound extensions.  The
problem is how will your agents, in this case your ZAP trunks, know to
"pick up the line" when they are not busy.  You'd have to get these
lines to somehow go offhook if they're not already busy.  Maybe you
can do this with an AGI script.  I don't know, I've never tried to
artificially control hook status.

Personally, I'd probably just skip the whole queue idea and get some
cheap SIP or IAX trunks and fall back to them when the ZAP lines are
busy.


- Noah