FREQUENTLY ASKED QUESTIONS 1:
are the basic components of a CallHandler system?
CallHandler Audio Conference System consists of four basic elements.
you want to add more lines, or change from analogue to digital as your business
grows, simply change the telephony card and the incoming telephone line type.
PC Chassis Running Microsoft Windows 2000
PCI Card(s) that plug into the Computer Slots. Analogue, ISDN Basic Rate, or
ISDN Primary Rate.
the case of the American T1 Trunk which has 24 incoming lines x 4 per card. (4
max. telephone sockets per card) The trunks are supplied by your Telco.
operator in the in the form of a socket on your wall - probably next to your
company telephone exchange. A single trunk connection looks a bit like a normal
RJ45 computer network socket. You will also need a LAN/WAN socket next to the
system so that you can control it - if needed.
connect Primary Rate ISDN cables into this board (Think of it as a very, very
a 24 line system you need one card with one socket - In the UK these trunks
(E1) have 30 lines capacity.
need to talk to your telecom. provider for and incoming trunk(s) that connects
CallHandler Operating Software that manages the whole system and provides other
Application Software (e.g.CallHandler Audio Conference) Consisting of: a)
Software that runs on the CallHandler box b) Software that runs on PC on a
LAN/WAN to set-up and control the conferences
||What is Power/Preview Dialling in Call Center
are its dialling principles?
agent script passes the power/preview dialler a telephone number to dial, a
message is sent to the CallHandler, the CallHandler dials the number, then
connects the agent to the ringing trunk line. When the call is answered an
event is sent from the CallHandler to the agent which can trigger a screen pop
on the agent's terminal.
is its design criteria i.e. what are its benefits & attributes?
benefit is that the dialling process is automatic, so less prone to error and
can be linked to an Agents script. It is generally used for smaller Agent
systems where Predictive Dialling is not effective
does it sit relative to the agent desktop and PABX?
The CallHandler can replace a PABX,
or b) The CallHandler can interface to
a PABX via a link e.g. DPNSS or QSIG
Does it interface to any common "Contact Management Systems" e.g. Sales
No direct interface, interfacing should be very easy using the
PowerDial/PreviewDial COM object.
components are used in its construction?
Windows 2000 OS Platform, Aculab Telco Spec Telephony
Cards, Pika Telco Spec Headset Connectivity Cards, Industrial specification
Chassis, and PC Components Modular COM/ActiveX software components, Hand
Optimised C/C++ Software, VB Scripting
C++ Software COM Component Communicates via TCP/IP to
is SMS messaging dealt with?
version 2.0 has integrated SMS messaging. This means that you can send an SMS
command via a LAN/WAN to the CallHandler and this will send an SMS message to a
We can support several delivery
via GSM modem / Mobile phone attached to
ISDN/Modem dial-up to an SMSC (SMS Message
via Internet to an SMSC
has developed the SMS messaging software from the ground up to be very reliable
No or very little set-up fee Anything between 3p to 7p per
message for 1-10000 messages
1. Web Servers -
hundreds try these UK based ones:-
or type: "bulk sms uk" into
There are several of ways to do this, depending what the service provider
a. Forward the
email to the SMS service provider using their SMTP mail interface
b. Retrieve the email,
convert it to an HTTP packet, queue it then post it to the SMS service
above, but using TCP/IP sockets connection.
low level logging info is available from E1/T1
is the actual low level logging information
This is a sample log trace from
the Traffic log.
^-- Date/Time Trace Was Recorded ^-- CallHandler
Unique Resource Identifier 01/07/2002 04:40:16 | 005 |
^-- Start Date/Time Of Call ^-- Call
Duration In Seconds
^- DDI Number (If Set) ^- CLI Number
(If Set) ^- Real Destination Number ^-Real Originating Number
DDI: CLI: DestAddr:96789354112 OrigAddr:
Reason The Call Ended ^-- The Raw Clearing Cause Code Taken From The Protocol
signalling protocol over that E1? Is it MFC-R2? Is it "Digital E and M" with
DTMF?. Maybe "Primary-Rate" (PRI) with signalling over the D-Channel
can support just about any signalling protocol by loading firmware drivers to
support the protocol. For example, we support PRI Q931, T1 PRI, MFC-R2
(Standard in South Africa, Egypt and Israel) also Basic Rate Euro ISDN CTR3 for
connection to COs. We support both 'User End' and 'Network End' connections, so
the CallHandler system can also be a CO. Also, we support Q.SIG and DPNSS for
direct connection to PBXs. Your chassis should behave as if it is a PBX towards
the "Central-Office" exchange. This is the 'User End' configuration, this is no
problem. We fully implement all features of the protocols we support, this
includes Call Clearing reporting, Direct Dial Inward (DDI) and Caller Line
Identification (CLI) signalling
CallHandler support more than a single E1?
can support up to 80 E1 connections, or 2400 lines from a single chassis
depending on the application. Multiple PC Chassis can be clustered to gether to
form very large systems.
*** Compact PCI (cPCI) Solutions ***
in a Compact PCI System Dual Processor
(Limited by H110 Bus Capacity)
*** Passive Backplane PCI Solutions ***
2040 Lines Call Switching Only
(Limited by Chassis Size, 17 slots)
1080 Lines Audio Record/Playback on
Dual 1GHz PC
(Limited by Host Processor / PCI Bandwidth)
are a number of things that limit the number of mailboxes you can have:
Disk Space Every message left uses up disk space at the rate of 3.7MB per
minute (uncompressed). So, 5000 mailboxes with say, 5 messages each with an
average 2 minute length will use up 190GB, for example. The maximum size we can
support in a single chassis is 109GB, or 2900 mailboxes. We can make better use
of the storage space available by using audio compression 'on-the-fly' as the
messages are recorded. We can support several rates of compression, 4:1 or 2:1,
although audio quality degrades with compression. To support a larger number of
mailboxes we would use an external RAID array. One possible RAID system would
take disk capacity up to 3.4TB, or 91000 mailboxes.
Number of DDI Numbers Available If you have 5000 mailboxes and you want to give
every user a separate telephone number, you will need 5000 DDI numbers from
your telecoms provider. You need to ask your telecoms provider whether they can
supply this many numbers. Assuming they can, whenever someone wishes to leave a
message, they will dial a phone number say: '01189016812', the telecoms company
will switch the call through to the CallHandler and pass either the whole
number or the last few digits of the number to the CallHandler, in the example
'6812', before the call is answered. The CallHandler will use this number to
locate the correct mailbox. We can support either a part of the number or the
whole number being forwarded as a DDI, so can support any number of DDI
numbers, and hence mailboxes.
||Aculab Text to Speech - what do I need?
uses a separate PC to run the TTS Engine, and links to the CallHandler via
100BaseT LAN link. This configuration can support 120 concurrent TTS channels.
& Hauspie Text to Speech - what do I need?
uses the DSP resources (opposed to a separate host PC). We can support up to 26
concurrent channels per card - 8 Cards per PC
SAPI Support - what can it do?
will allow any SAPI compatible TTS engine to work with the CallHandler e.g.
Lucent TTS, Microsoft (L&H) TTS, L&H, Philips etc. This will support
any number of TTS channels, distributed over one or more TTS servers, and
linked to the CallHandler via a high speed LAN.
custom controls does CallHandler have?
are several custom controls. Each Device and Device Manager in the CallHandler
has two controls: Configuration Control and Status Control. The Configuration
Control is used to configure the device prior to it being "Run" by the
CallHandler. The Status Control is used to monitor the status of the device and
to alter device settings while it is being "Run". The Status and Configuration
Controls run from within the IE5 browser and form part of the CallHandler
does the Developer Kit include regarding Text to Speech
Hooks for Aculab and Lernout & Hauspie Text to Speech Systems
I need special equipment to run the Conference Demo Disc?
the demo software is Install Shield based and contains all the necessary
modules and simulators.
we use the existing E1 Trunk which has 15 lines currently used for the office
providing the E1 is looped through the CallHandler. The CallHandler would strip
out its 15 lines, and then pass the remaining ones to the local exchange. We
would supply a system with additional physical ports to facilitate
do you still use NT when there are later versions of MS Windows
uses Windows 2000 because this is a tried a tested platform. We tend to
software and hardware that has a track record, and has been fully tested in our
laboratory before making a general release.
version of SQL do you support?
is the format of the audio data?
format of the audio data conforms to the ITU-T Recommendation G.711 Pulse Code
Modulation (PCM) of Voice Frequencies. This is a companded audio format for
improving noise performance of audio channels. The G.711 format covers two
incompatible companding formats: A-Law and mu-Law.
format apply to specific countries depending on the telecommunications system
the country supports. Generally, Europe operates using the A-Law format, and
North America operates using the mu-Law format. This only applies when the
CallHandler system is installed in a particular country, it must support the
companding format for that country. But if a call is made from a CallHandler in
Europe to North America, the CallHandler would use A-Law companding, and the
international telecom provider will convert A-Law to mu-Law and visa-versa
you have any references on Audio Formats?
Microsoft Multimedia Standards Update - New Multimedia Data Types and Data
Techniques Rev3.0 April 15, 1994 - Available as part of the Multimedia
Registration Kit (MRK) from Microsoft, the MRK is distributed as an MSDOS
self-extracting archive, which is available from:
ITU-T Recommendation G.711 [11/88] - Pulse code modulation (PCM) of voice
frequencies. Available from the International Telecommunications Union (ITU) at
MSDN Library - Resource Interchange File Format Services - Multimedia: Platform
your Telephony Cards work in Egypt?
Digital Telecom cards have been tested in Egypt. We understand that Egypt uses
a standard MFC-R2 but there can be certain variations depending on the Egyptian
exchange the card is run against, and this may vary between the Egyptian
you give me more information on video over Voice Over IP (VoIP), this is a real
benefit in the call centre or strategic customer service environment. i.e.
picture and voice point to point.
Both the H323 and SIP standards support the streaming of video. H323 and SIP
are basically both protocols for establishing streaming connections between 2
(or more parties). Both H323 and SIP use the RTP (Realtime Transport Protocol)
and RTCP (Realtime Transport Control Protocol) for streaming. RTP can stream
just about any format of audio and video. So the protocol is pretty much
in-place to support any sort of multimedia. Comparing audio and video
streaming, a typically compressed audio stream uses 8kbits/sec, whereas a
typically compressed video stream uses typically 64-128kbits/sec. Compressing
and streaming video around and enterprise is a much larger problem than audio,
you need much bigger network bandwidth, and much bigger codec cards, and hence
is a lot more expensive.
What about security and Voice Over IP (VoIP), will or could the connectivity of
telephone networks to IP present some sort of security breach to the corporate
are several issues with security:
i) At present with most VOIP systems it
is not possible to encrypt the audio channels, so anyone with a VOIP network
monitor can listen in on a VOIP conversation.
ii) H323 is not very good at
connecting through firewalls. The H323 handshake is done via one IP port, then
the actual stream connection is made on another port, which is agreed upon
dynamically via the handshake process. To make connections through a firewall
it is essential to know which port connections to allow and which to deny, and
H323's dynamic port allocation makes this pretty much impossible. There are a
couple of products that get around this problem, but they're not very mature
yet. This is not an issue with SIP as the handshake and stream connections are
made on fixed pre-defined ports.
Can you support for D-channel for E1 line?
support D-channel protocol on E1. There are 2 D-channels on E1, one for framing
and the other for protocol signalling.
example of a D1 protocol is Q931
||PLEASE SUBMIT YOUR OWN