How to setup a call center using asterisk

Asterisk Asterisk Support
1

I want to setup a call center about 20 seat using asterisk. Could you please give some advice on it.

2
  1. read some
    forums.digium.com/viewtopic.php?t=4208
  2. install favored linux
  3. install asterisk
  4. configure asterisk
  5. run Your callcenter
    if You have some problems in points 3-5 post it here.
3

also keep in mind asterisk is not currently compatible with telemarketing. There was some discussion a while ago about this but I haven’t seen a patch to enable it yet.

That said, a few issues to consider-

  1. what type of phones will your employees be using? hardphone, softphone? For call centers softphones often make a lot of sense as the user never takes their headset off

  2. what kind of connection to the PSTN will you have? PRI? VoIP? If VoIP, what codec? If it’s VoIP, you WILL need a quality of service control on your router…

  3. presumably you will want to monitor or record your employees? Consider how these recordings will be saved (what format) and consider how you will store/archive them, or if you are going to rotate them.

  4. consider how your server(s) will be set up. 20 seats is not too big but are you going to expand? Are you going to have the * box doing anything that requires some CPU, such as g.729?

lastly, as fd said, read up. read voip-info.org there is a lot of info there. look up the Queue and Agent systems, they will be very useful for you.
Also look up the book Asterisk: The future of telephony. If you read it cover to cover you will be an asterisk expert and could answer this question yourself :smile:

4

I am double aout how to queue the incoming calls and asterisk how to distribute calls to opertor? Are there any configuration example for my reference? Thanks!

5

orderlyq.com/asteriskqueues.html
voip-info.org/wiki/view/asterisk+cmd+queue
voip-info.org/wiki/view/aste … ueues.conf

6

Thanks, i had setup my call center successfully. But i found that my call center agent can not transfer the reiceved call from outside to other extension by press #. For example:
I bind agent as follow:
agent A to extension 101.
agent B to extension 102.
agent C to extension 103.
When agent A(extension 101) recevie call from call queue, he can not transfer call to extension 103 by press #

But when extension 101 call to extension 102(not through call queue), extension B can press # to begin transferr call to extension 103.

7

You have to use the “t” option in the Queue Application like this:

exten => s,4,Queue(Team-1|tW|||120)

See this link for different options:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial

Also just read the links provided by the previous guys THE WIKI IS YOUR FRIEND!

8

Well, Ive done 2 systems for telemarketing with 50+ users using a basic 2.8ghz with 1gig ram pbx, only doing sip though. Using Linksys pap2 atas and cheap att phones that cost 12 bucks at walmart, i chose this setup over grandstream phones. And if you want to be sneaky you can “monitor the calls” to view the telemarketer’s performance.

Good luck.

9

Thanks, my call center agent can transfer call now. But when they press # and the third extension number the call would transfer to the third.
How to make a attend transfer? Below is part of my feature.conf

…

[featuremap]
blindxfer => #1 ; Blind transfer
;disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
…

It seem that i can press *2 to begin an attend transfer, but when i press *, the call will be hung up.
BTW, blindxfer => #1 mean that press #1 to begin transfer call, but I only need to press # to begin transfer call. Why?