[asterisk-users] Call Queues

Rob Schall rschall at callone.net
Thu Jul 5 15:44:46 CDT 2007


Noah Miller wrote:
> 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
>
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Possibly do a combination of things. Check if those zap chans are in 
use/busy. If they care, then create a call file using a script. I 
haven't played too much with it, so I don't know if those will queue 
until they can complete or if it will just error and delete itself. If 
you really are determined, you might even be able to route all requests 
to a script. Then have it check if there are any open lines... if so, 
create the call file... if not, then put it in a queue (in python, 
etc... not an asterisk queue), and try again in a min and see if a 
channel has opened up.

Disclaimer - I have no idea if this idea will work. :)
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