[Asterisk-Users] SIP ACD system for station to station calls

[Colin Stefani]

[A bit long...sorry...]

I have a unique application where we're handling *only* internal calls
between extensions in a call center format. We are exploring moving this
to a SIP based solution and I'm looking for
recommendations/ideas/guidance on how to approach this since I'm new to
VoIP in general.

Before I get too far, here's the scenario:

This system is used for handling truck traffic and processing truckers
and their loads in and out of a container facility. We have N number of
kiosks the truckers use, average is 20-30 extensions that are ringing in
at once on the other side the call center is approximately 10-12 agents
receiving these calls. Right now we have an Inter-Tel PBX with analog
phones on one side (set up as house phones) and the call center uses
digital phones as you'd see with any office phone system. The truckers
pull up to a kiosk and press the call button, since these extensions are
programmed as house phones to ring an ACD queue they then enter the
queue they are designated to go to. There are several queues defined to
handle different types of truck traffic. The call center is on the
digital end and they login to the ACD queues and take calls from them,
the agents may login to one or more queues at once. Thus this creates a
closed system by which all calls are routed internally to the system and
there are no outside lines. We have a software system that uses OAI to
detect events from the PBX and thus provide call control through our
client software (answer, hangup, DND, acd login/logout).

Enter VoIP:

We'd like to have all the end-points (agents and kiosks) be SIP clients
(either pure software, a hybrid or simple ATA based). Then we'd like to
put a highly available SIP server in the middle. Asterisk may fit this
bill, but first off it's not really highly available and my dilemma is
this; Asterisk is really good at acting as a proxy and there are also
some good modules for advanced ACD/ICD functionality
(http://www.voip-info.org/wiki-ICD). However, since we're attempting to
design a pure SIP based system end to end, does it really make sense to
put something like Asterisk in the middle (or any soft-pbx for that
matter)? The goal would be to have the SIP clients setup the calls and
then communicate directly with each other using whatever codec we find
we like (as you'd expect in an end-to-end SIP system (think interoffice
calls, ext to ext). We'll have total control over the entire system and
can define and control the SIP clients, the router, the codec on down.
So I don't have to worry about Joe User with some crazy codec or SIP
client that's not compliant.

My sense is no, it doesn't necessarily make sense (and might be
overkill) to put a full blown soft PBX in the middle. It seems better to
write our own SIP router or use one that's already out there. This would
be lighter weight (gotta keep administration in mind too). Further there
seems to be very little attention paid to high availability in the PBX
world and I can't have the central server fail and lose all the calls at
once, there has to be some transparency in failover or at least recovery
of state. We have the server infrastructure in Java (www.jboss.org) to
support JAIN and SIP servlets should we create our own. But I'd like to
find an already existing solution as I hate reinventing the wheel.

That said, I'm not sure what to do about the call routing ACD/ICD
functionality. Does anyone know of a SIP router that does that or add-on
software to perform that function with SIP clients? Where do I start in
terms of looking for this?

I'd love to hear your ideas or thoughts on this, any approaches that we
should consider. I'm in research mode and have my eyes and ears open. We
maybe going some place no one else has, but I would find that hard to
believe as call centers are so common and it would surprise me if no one
has attacked this from a pure VoIP point of view.


Colin Stefani
Tideworks Technology


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