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<title>SIP ACD system for station to station calls</title>
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Colin,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Asterisk does exactly what you are after,
as for your comment about high availability – it can be.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Either research or hire in the expertise –
your requirements are already being met in a number of very similar
installations.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoPlainText><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Cheers,<o:p></o:p></span></font></p>
<p class=MsoPlainText><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Dean <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Colin Stefani<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, April 13, 2005
12:11 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] SIP ACD
system for station to station calls</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>[A
bit long…sorry…]</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>I</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>have a
unique application where we’re handling *<b><span style='font-weight:
bold'>only</span></b>* internal calls between</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>extensions in a
call center format.</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>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.</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Before
I get too far, here’s the scenario:</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
system is used for handling truck traffic and processing truckers and their
loads in and out of a container facility. We</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Right now we have an Inter-Tel</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>PBX with analog
phones on one side (set up as house phones) and</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>the call center
uses digital phones as you’d see with any office phone system. The
truckers pull up to a kiosk and press</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>the call</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>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</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>this creates a closed system by
which all calls are routed internally to</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>the system and there are no outside
lines. We have a software system that uses OAI to detect events from</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>the
PBX and</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>thus provide call control through our</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>client
software (answer, hangup, DND, acd login/logout).</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Enter
VoIP:</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>We’d
like to</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>have all the end-points (agents and kiosks) be SIP clients
(either</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>pure</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>software, a hybrid or simple ATA
based). Then we’d like to put a highly available SIP server in</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>the
middle. Asterisk may fit this bill, but</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>first off it’s not really
highly available and</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>my dilemma is this; Asterisk is
really good at acting as a proxy and there are also some good modules for</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>advanced</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>ACD/ICD
functionality (</span></font><a href="http://www.voip-info.org/wiki-ICD"><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>http://www.voip-info.org/wiki-ICD</span></font></a><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>). 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</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>have the SIP</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>clients
setup the calls and then</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>communicate directly</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>with each
other</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>have
total control over the entire system and can define and control</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>the</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>My
sense is no, it doesn’t necessarily make sense (and might be overkill) to
put a full blown soft PBX in</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>the middle.</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>It seems
better to write our own SIP router or use one that’s already out there.
This would</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>be</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>lighter weight (gotta keep
administration in mind too). Further there seems to be very little attention
paid to high availability in</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>the PBX world and I can’t have
the</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>central server fail and lose all the calls at once, there
has to be some</span></font> <font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>transparency in failover or at least recovery of
state. We</span></font> <font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>have the server infrastructure in Java (</span></font><a
href="http://www.jboss.org"><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>www.jboss.org</span></font></a><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>)</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>to support
JAIN and SIP servlets</span></font> <font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>should we create our own.</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>But
I’d like to find an already existing solution as I hate</span></font> <font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>reinventing
the wheel.</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>That
said, I’m not sure what to do about</span></font> <font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>the call routing
ACD/ICD functionality. Does anyone know of a SIP router that does that or</span></font>
<font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>add-on
software to perform that function with SIP clients? Where do I start in terms
of looking for this?</span></font><o:p></o:p></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.</span></font><o:p></o:p></p>
<p><a name=""></a><i><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial;font-style:italic'>Colin Stefani</span></font></i><o:p></o:p></p>
<p><i><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial;
font-style:italic'>Tideworks Technology</span></font></i><o:p></o:p></p>
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