contact center technology platform for Asterisk

December 27, 2009

Basic network topology for Call Center setup

Efficient network setup can enhance the quality of your VoIP experience.  Even if you are averse to VoIP terminations for routing long distance traffic, within your call center, you can take advantage of VoIP. Here is a simple but efficient topology for Asterisk based call center setup.

Asterisk based contact center network layout
Asterisk based contact center network layout

Here all the agents, supervisor and admin phones are within the LAN and have assured bandwidth. Good quality IP phones from Polycom, SNOM, Linksys and other vendors can serve as the admin phones. Agents can use softphones like X-Lite or Eyebeam. An enterprise grade call center software like Indosoft Q-Suite will provide all the necessary features to setup and manage an advanced contact center. I recommend DELL 2950 category of servers which has served as very well in the past. The Dell web managed switches are inexpensive and work well within the contact centers. We implemented our first VoIP based contact center using Asterisk in 2003 with the long distance traffic terminated on PRI using Digium TDM board.

December 25, 2009

Considerations while setting up contact centers using Asterisk

Asterisk opens up a lot of options while setting up contact centers. Last ten years have heralded a revolution in the contact center technology platform. During this period VoIP technology has dramatically altered the landscape of of telecommunications.  Asterisk being an hybrid PBX with inbuilt support for VoIP and TDM, is the force behind the change in the contact center technology platform. Let us consider some of the options that are available for Asterisk based call centers.

You can have all your call center agents on VoIP  using IP phone or soft-phone within your internal LAN.  This is a relatively safe option where you can ensure required bandwidth using inexpensive web managed switches. There should be no measurable delay within your LAN to ensure very good quality voice. Any sane LAN design should achieve this.

You can also get a VoIP provider for terminating your calls to and from the outside world. Here the bandwidth and delay are critical. An uncompressed voice will require 100 Kbps and using G729 CODEC you can get away with 50 Kbps per voice conversation approximately. Generally SIP termination is a great option if you are maintaining your infrastructure in a co-location. The back-end bandwidth and latency to your SIP termination provider will not be a big issue. This is a great option for setting up at-home agents.

Another option is the Skype for Asterisk where you can take advantage of the Skype connectivity and its calling rates to make calls. Skype interconnectivity to Asterisk allows Asterisk users to make use of Skype’s superior codec and voice quality for VoIP.

When VoIP is not good and not an option for your site, the tradional TDM (PRI E1 or T1) is available through add-on telephony boards from Digium or Sangoma. What is more is if PRI is not available, you can use SS7 (ISUP) to connect to the next level.

You can decide to record selectively or all calls in Asterisk. One hour of voice recording at 64Kbps will consume 29 MB of memory. This should give you a rough idea on the storage requirement for uncompressed voice. You can also listen to live conversations.

You can also setup a fully functional PBX for the administrative section of your contact center. At Indosoft, we offer the feature rich contact center software Q-Suite which takes the mystery out of setting up enterprise grade call centers. Q-Suite offers all the features required for a contact center, right out of the box.

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