Call Centre Solution

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Crystaline Call Centre

COPYRIGHT 2016 Crystaline Technologies Limited. All rights reserved.
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All company and product names are trademarks of the respective companies with which they are associated

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Overview
Crystaline Technologies is happy to present its Customised Call Centre and IVR Solution
which is based on the Asterisk Platform.
The Call Centre solution is built to handle a large volume of phone calls. Call Centres
typically handle customer service, support, telemarketing, telesales and collections
functions. The employees who staff call Centres are referred to as agents or customer
service representatives (frequently abbreviated as CSRs). Call Centres range from very
small informal operations to massive, highly optimized sites with hundreds or even
thousands of agents.
Call Centres use specialized telephone equipment to maximize productivity. Specialized
telephony switching systems called Automatic Call Distributors or ACDs are used to queue
and route inbound calls to agents based on a wide variety of criteria. Outbound calls are
frequently generated by an automated system called a Predictive Dialer that monitors the
status of agents and places calls on their behalf. Other common call Centre tools include
desktop integration (frequently referred to as screen pop), Interactive Voice Response
(IVR) applications, call recording solutions, productivity monitoring utilities, workforce
planning systems and various methods of historical and near real-time reporting.
QUEUE STRATEGIES
A simple ACD system consists of a source of calls (a pool of lines, trunks or virtual trunks), a
FIFO (first-in, first-out) queue and a pool of agents who are selected using a ring-all
strategy. In this case, when a call arrives the system rings the phones of all agents who are
not already on a call. The first agent to answer the call is connected with the calling party.
All the other phones stop ringing.
A more complex (and likely more useful) configuration would have the call offered to the
agent who had been in the idle state longest. This most idle strategy is frequently used
when all agents are considered equally qualified to handle a task. Other common strategies
include round robin, linear hunt, least-recently-called, fewest calls and random. In some
cases, the ACD can weight its selection based on the callers need (generally collected using
an IVR application) and a list of skills associated with each agent. This is generally referred to
as skills-based routing.

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CALLER EXPERIENCE
While waiting in queue, callers generally hear a combination of marketing messages, queue
status messages and music. Marketing messages are simply audio recordings that are piped
into the queue on a periodic basis. Status messages provide the caller with specific
information about their status the number of callers ahead of them in the queue, the
estimated wait time and sometimes alternatives to waiting in queue. Some more advanced
call queueing systems support virtual queueing. A virtual queueing system allows callers to
provide a callback number, then disconnect. Their position in the queue is preserved and
when an agent becomes available the system places an outbound call to the caller.
AUTOMATED ATTENDANT
Automated attendant systems have long been paired with ACDs, allowing callers to route
themselves into the appropriate call queue. Automated attendants are simply menu systems that
prompt callers to indicate their preference using the keys on their phone or, in some cases, by
speaking keywords. Callers are generally willing to accept up to two levels of menu before reaching a
live agent. More than two levels tends to annoy most callers and can result in an increase in
abandon

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System Customisable features List

Base Features

Additional Features

ADSI On-Screen Menu System

Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing

Call Features
SMS Messaging
Spell / Say
Streaming Hold Music
Supervised Transfer
Talk Detection
Text-to-Speech (via Festival)
Three-way Calling
Time and Date
Transcoding
Trunking
VoIP Gateways
Voicemail:
Visual Indicator for Message Waiting
Stutter Dialtone for Message
Waiting
Voicemail to email
Voicemail Groups
Web Voicemail Interface
Music On Transfer:
Remote Call Pickup
Roaming Extensions
Route by Caller ID

Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number Discovery (DUNDi)
Do Not Disturb
E911
ENUM
Fax Transmit and Receive
Flexible Extension Logic
Interactive Directory Listing

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Interactive Voice Response (IVR)

Local and Remote Call Agents
Macros
Music On Hold
Flexible Mp3-based System
Random or Linear Play
Volume Control
Privacy
Open Settlement Protocol (OSP)
Overhead Paging
Protocol Conversion
Remote Office Support

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System Requirements
Priority

Item

Mandatory Work Station/PC

Purpose

Minimum Specifications
Intel Core i5 processor or Quad Core
Procesor, 8 GB RAM, DVD/RW,
Network Card, PCI Expansion Slots,
500 GB HDD (HP 280 G1 i5 or
Equivalent)

Qty

Mandatory FXO/FXS Card

Software Host
Interface with
PSTN (FXO)
Network as well
as analog Phone
lines (FXS) within
the office

Optional

E1/T1 Line

Multiple lines
from Operator to
use with 3-Digit
Shortcode

Optional

E1/T1 PCI Card

PC Module for
plugging in the E1
Line(s)

Optional

CC Agent Equipment
Office Staff
Equipment

Optional

Optional

UPS

Up to 8 ports each through a

combination of FXS400 and FXO400
modules, Full-length analog card,
Optional Failover function with
A810EF11 only., RJ45 Connector

Headsets (with microphones)

connected to Desktops

Backup Power in
case of failure or
primary

TBA

IP Phones

1500 VA minimum

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Glossary
ACD (Automatic Call Distributor) - A device or system that distributes incoming calls to a
specific group of terminals that agents use. It is often part of a computer telephony
integration (CTI) system.
CODEC (Coder/Decoder) - A software library that contains the algorithms necessary to
convert an analog signal to and from a digital one. Examples: G.711 G.729 GSM
Context - The dialplan is composed of one or more extension contexts. Each extension
context is itself simply a collection of extensions. Each extension context in a dialplan has a
unique name associated with it. The use of contexts can be used to implement a number of
important features, such as security, routing, autoattendant, multilevel menus,
authentication, callback, privacy, macros, etc...
DAHDI (Digium Asterisk Hardware Device Interface) - A high density kernel telephony
interface for PSTN hardware.
Dialplan - A dial plan establishes the expected number and pattern of digits for a telephone
number. This includes country codes, access codes, area codes and all combinations of digits
dialed. For instance, the North American public switched telephone network (PSTN) uses a
10-digit dial plan that includes a 3-digit area code and a 7-digit telephone number. Most
PBXs support variable-length dial plans that use 3 to 11 digits. Dial plans must comply with
the telephone networks to which they connect.
E&M (Ear & Mouth) A type of signaling commonly used over T1 and E1 interfaces.
Encode - The process of converting an analog signal into a digital signal that can be
manipulated easily by a computer.
FXO (Foreign Exchange Office) - A device usually found on the customer end that is powered
by the channel and can interface into the telephone company's network. Digium makes FXO
modules that interface with PSTN lines using FXS signalling in either Loopstart(fxs_ls) or the
more common Kewlstart(fxs_ks) modes.
FXS (Foreign Exchange Station) - A device usually located on the telephony company's
property, a FXS device send power through a channel to a phone on the other end. Digium
makes FXS modules that interface with PSTN phones using FXO signalling in either
Loopstart(fxo_ls) or the more common Kewlstart(fxo_ks) modes.
G.711 - An uncompressed codec that samples a 64kbps channel at 8 bits per sample using
pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw.

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G.729 - The G.729 codec is an industry standard which allows for stuffing more calls in
limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes
64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to
this number). Many people are using Asterisk with G.729 to replace expensive gateways.
GSM - A compressed speech codec that uses a rate of 13 kbps.
H.323 - A VOIP protocol that was deployed early and is widely adopted.
IAX (Inter-Asterisk eXchange) - A VOIP protocol designed to be much more NAT friendly. IAX
currently only transports audio.
IVR (Interactive Voice Response) - An automated voice system that allows callers to navigate
a phone system and be directed to the correct extension by pressing a series of numbers on
a tuch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)
MGCP (Media Gateway Control Protocol) - A VOIP Protocol that has both signaling and
control and was designed to reduce complexity between media gateways.
Open source - An approach to the design, development, and distribution of software,
offering practical accessibility to a software's source code.
PBX (Private Branch Exchange) - A telephone exchange that serves a particular business or
office, as opposed to one that a common carrier or telephone company operates for many
businesses or for the general public.
PRI (Primary Rate Interface) - A PRI is a truly digital circuit, containing 24 ISDN channels. One
of these channels is a D channel and used for signaling. The rest are B channels and used to
transport audio.
PSTN (Public Switched Telephone Network) - Originally a network of fixed-line analog
telephone systems, the PSTN is now almost entirely digital and includes mobile as well as
fixed telephones. The network works in much the same way that the Internet is the network
of the world's public IP-based packet-switched networks.
REN (Ringer Equivalency Number) - A number which represents the ringer loading effect on
a line. A ringer equivalency number of 1 represents the loading effect of a single traditional
telephone set ringing circuit. Most modern telephones probably will have a REN lower than
1. The total REN expresses the total loading effect of the equipment on the ringing current
generator (FXS). The REN of a Digium FXS board is 5 (representing "extension," i.e., parallelconnected telephones). The actual number of devices on the line may be greater than the
REN limit, if their respective individual RENs are less than 1.
SIP (Session Initiation Protocol) - A signaling protocol, widely used for controlling multimedia
communication sessions such as voice and video calls over Internet Protocol (IP). SIP
adoption amongst hardware and software vendors continues to expand.
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TDM (Time Division Multiplexing) - A processes of splitting one medium into two or more
channels by using timed segments to transmit information.
Transcode - The process of converting a channel with one type of encoding to a different
type of encoding in real time.
VoIP (Voice Over Internet Protocol) - A general method for transporting voice through the
internet.
Zaptel - The Zaptel project has been renamed 'DAHDI' as of May 2008. DAHDI is a series of
drivers for telephony hardware devices.

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