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

[mattf]

We have several different setups, but on a couple servers we are doing upto
50 concurrent conversations of recording. We ran into the 50-60 recording
ceiling about a year ago and it's mostly the hard drive that limits it to
that number, really it's a lot if you think about it, Asterisk is having the
hard drive write 100-120 audio files(-in and -out for each conversation)
several times a second. It is also important to note that we mix them with
sox after hours to reduce load on the system and load on the drives.
Although this does mean that the recordings are not available until the next
day. We also have setup 2 systems to copy the in and out files off to
another machine to be mixed more closely to realtime so that is an
alternative.

MATT---


-----Original Message-----
From: Ilan Rabinovitch [mailto:irabinovitch at gmail.com]
Sent: Tuesday, May 24, 2005 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound call center - reliability \
scalabil ity with queues


Matt,

Are you doing any call recording / monitoring?  What percentage?  

Ilan

On 5/23/05, mattf <mattf at vicimarketing.com> wrote:
> 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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