[Asterisk-Users] Inbound call center - reliability \ scalability with queues

Tue May 24 11:42:52 MST 2005

The asterisk machines will not have anything to do with the T1's, when they
receive the call it will be SIP VOIP.  There will be media gateways (i.e.
cisco media gateways) to change all T1 signals to VOIP before it reaches the
PBX.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mattf
Sent: Tuesday, May 24, 2005 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

OK, If you are going to be recording all calls you will need to rethink
things a bit. Recording calls limits you to 50-60 consecutive conversations
per server before audio distortion starts to occur.  You will probably want
to think about limiting yourself to 3 T1s per machine. There are many ways
to set this up and I think you will probably have to go through some
trial-and-error before you find the perfect system layout for your
operations.

I would first try setting up machines that would just have the T1s on them
and take the calls in(or out) and record them. Then have those
connect(through IAX or T1 crossover) to the servers that have your queues
and phones set up on them. You will also need some really big archiving
mechanism if you want to keep those recordings around to reference in the
future. audio recordings can take up a lot of space if you need to keep them
for 3 years like we do.

SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they
can hit the PCI-bus bottleneck and have issues. You may see the
"ast_channel_walk_locked" warning in Asterisk when this happens.

MATT---

-----Original Message-----
From: Warren Smith [mailto:warren at serverplus.com]
Sent: Tuesday, May 24, 2005 12:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

I would say that we would need to be able to scale to the 200+ consecutive
call range in the near future (6 months), and hopefully to the 500+ within
the next two years.

We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K
rpm SCSI).  We also are planning on recording all queue calls that are
answered, and possibly outbound calls made by support agents since it has
proven to be extremely helpful when it comes to training.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability \
scalability with queues

For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1 card
because they are about 30% more efficient than Digium T1 cards. 

When you say that you need to scale to 100s of consecutive calls, is that
closer to 200 or 900? and what timeframe is that planned for?

We have a distributed in/outbound call center environment across 4
geographic locations with over 20 T1s connected so it is possible for
Asterisk to handle over 1000 consecutive calls across the system if you
design it right. One of the reasons we don't use Asterisk queues, other than
the difficulty in customizing the code to work with the ManagerAPI and
client apps, is that it was hard to scale across multiple servers. That's
why we use the astGUIclient suite which is more customizable and scalable
across multiple servers, although (and it pains me to say this because we
developed it) it is not as easy to install and setup than just creating an
Asterisk queue.

We use a combination of SIP, IAX and Zap client phones depending on the
system and the user and yes 711 is always best to use when you can. And if
you have many remote phones using a codec like GSM it may actually be better
to have a dedicated machine doing nothing but the transcoding from GSM->711
and then just using IAX or a crossover Zap T1 to the inbound server to
reduce processor load.

In any case it is always advisable to have a backup server that is fully
ready to jump in production with a minimum of reconfiguration.

A couple more questions, will you do much recording? and what kind of disks
do you plan on using?

Hope this helps,

MATT---

-----Original Message-----
From: Warren Smith [mailto:warren at serverplus.com]
Sent: Monday, May 23, 2005 8:12 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Inbound call center - reliability \ scalability
with queues

We are wanting to move off of our legacy inter-tel phoneswitch and move to
VoIP and asterisk.  We are looking for a new PBX because the inter-tel
switch is too difficult to integrate our existing (and new) software into.  

We are a technical support center.  All our calls currently come in on toll
free numbers via T1's, and there are 3 of them.  I want to use a media
gateway to convert the T1's into SIP VOIP (I want reliable hardware for the
gateway), and use asterisk as the PBX having all incoming and outgoing
channels as SIP.  Almost all dialplans will be using Queues, and there will
likely be no more than 10 queues, with (currently) about 80 incoming
toll-free numbers.  There are approximatley 30 agents, but as of right now
there are no more than 15 agents logged in at a time.  We need to be able to
support 60-70 simultaneous calls initially and we have to be able to do this
reliably.  We also need to be able to scale into the 100's of simultanous
calls range.

What would be the best option, to have 2 powerful machines (dual
powersupply, ) with one as a hotswappable backup or have multiple machines
with a sort of load balancer setup?  Having multiple machines could possibly
cost less, but I'm not sure how the queues and agents would be managed
across multiple machines.  I.e. how would the agent 'login' to each asterisk
machine so that the calls could be handed to it, and how would the calls get
handed to an available agent by 4 seperate asterisk machines?  I've read
through the wiki, but I'm not sure how much overhead queues would put into
the system.  I want to have all the codecs the same, so the asterisk
machines doesn't do any transcoding, and have all channels as SIP.  There
will be music on hold.  Would a dual 2.8 ghz xeon in this config be able to
handle 80 simultaneous incoming calls?  Would using the 711 codec make a
difference in available processing power?

I'm sorry if this has been answered a million times already, I just didn't
see many configs close to what we're trying to do to compare to.  Thanks for
any input you may have.
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