Matt,
I am sure this is a RTFM and I am pretty sure you are using meetme
rooms. Just not too sure how you do the magic.
28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
665 lines. My client's call volume has shot from 5,000 to about 10,000
calls a day. Due to recent product offerings/advertising, I expect to
be eating up 6 T1 (peak) by the end of October. They will eventually
have every channel in use during peaks, whether that is in November or
December, I am not sure. I just know it can't break at that point due
the the sheer expense of revenue lost for downtime.
Thanks,
Steve
Matt Florell wrote:
How many lines and agents are you looking at?
What kind of call volume?
Average expected hold time?
VICIDIAL could be an option for you since it does not use Asterisk
Queues and can already easily scale across many servers.
MATT---
On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
I have been tossing around some ideas about scaling a call center with
load balancing and redundancy and would like the comunities input,
thoughts, criticism and anything anyone wants to toss in.
The most evident thing is to start with beefy servers and only run procs
that are required. All of the TDM boxes run stripped down versions of
Linux and Asterisk, they just take the call from the PRIs and convert
them to SIP, everything stays ulaw end to end.
*Shared queues across multiple servers would be ideal*. I don't think
it is possible in asterisk, as is. Maybe DUNDI could be useful but I am
not up to speed on it enough to really know.
I was toying with a concept of a DB server tracking the number of calls
to queue(s), number of agents logged into the queue(s). Some agents
will be logged into multiple queues and providing the logic to a series
of Asterisk servers. Calls could be made to the db to determine which
queue/server to route the call to. In this situation, duplicate queues
would exist on several servers, so balancing would work somewhat if the
DB made the selection on which box to route the call to and which box an
agent should log into. FastAGI and the manager interface will provide
the routing and DB updates.
Another thought was to have one central server with all of the queues
and agents, then somehow the central server would cause a "recording/CDR
server" to send re-invites to the two SIP endpoints so that the call/RTP
stream is moved to another asterisk server which would record the call
and keep the CDR info. Again, this would be done with a DB to decide
which asterisk (recording/CDR) box has the lightest load. It would take
the burden of maintaining the call from the "Queue" server. I/O is the
first bottleneck in scaling when you record each and every call.
Would it be difficult to have asterisk send two SIP endpoints re-invites
and then bridge the call? Then it is just a matter of the "Queue"
server checking the DB which recording/CDR server the call should go to
and send it a message to re-invite and bridge the endpoints. A transfer
to a meetme is another possiblility but I want the "Queue" server out of
the stream.
Has anybody else thought through the best way to scale something like
this. I have a DS3 and will be using all of the channels in the
semi-near future. I need to come up with a workable plan before then.
Thanks,
Steve
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
