[asterisk-users] callers in queue passed to agents who accept only one call at a time

Vieri rentorbuy at yahoo.com
Fri Mar 28 08:33:42 CDT 2008


--- Atis Lezdins <atis at iq-labs.net> wrote:

> On Thu, Mar 27, 2008 at 6:32 PM, Vieri
> <rentorbuy at yahoo.com> wrote:
> > I have a queue I configured as "strict" and a cron
> >  script I use to QueueAdd and QueueRemove agents
> >  according to my company's requirements. Usually I
> have
> >  2 or 3 agents at a time and the ring strategy is
> >  ringall.
> >
> >  These agents use non-open-source Windows
> softphones
> >  that do not let you configure it so that if
> they're on
> >  the phone, a second call will be rejected (agent
> >  busy). Instead, it's as if they had call waiting
> and
> >  incoming calls keep popping up while they're
> >  conversating with the first caller and they would
> like
> >  to avoid this.
> >
> >  I guess the easiest solution would be to find an
> >  open-source or free softphone that can be
> configured
> >  to accept only one call at a time (currently
> using
> >  SJphone).
> >
> >  Another solution would be if I could tell the
> Queue()
> >  application that if an agent is InUse then don't
> pass
> >  the call.
> >
> >  Still another yet more delicate solution would be
> to
> >  have a custom script "receive" manager events
> related
> >  to the queue which in turn replies with an agi
> >  command. For example, whenever an agent answers a
> call
> >  I think that an event such as QueueMemberStatus
> can be
> >  triggered (although I don't know how). If the
> custom
> >  script could receive this event in realtime then
> it
> >  would run an agi command such as
> >  QueueRemove(busyagent...). When the agent is free
> >  again I suppose the same event is triggered and
> the
> >  custom script can QueueAdd(freeagent...).
> >
> >  Could anyone please give me some pointers on
> this?
> 
> In queues.conf set ringinuse=no
> Also make sure that you don't use realtime sip peers
> (or use
> rtcachefriends with that). Probably you also need
> call-limit set to
> any value in sip.conf

Thanks Atis and Rodrigo.

However, I can't use ringinuse=no in queues.conf
because I'm running 1.2.27 (or is there a
backport/patch?).

If I use call-limit=1 then I get all sorts of problems
(see
http://lists.digium.com/pipermail/asterisk-users/2008-March/208558.html
)

Besides, call-limit=1 would not allow the agent to do
attended transfers.

I guess I'm forced to upgrade to 1.4 although there
have been several instability issues lately, even on
this mailing list.

Vieri



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