Open Architecture Computer Telephony Servers and Tools  
{info@chelston.co.uk} 
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FREQUENTLY ASKED QUESTIONS 1:
What are the basic components of a CallHandler system?
  The CallHandler Audio Conference System consists of four basic elements.
  1. The PC Chassis Running Microsoft Windows 2000
  2. Telephony PCI Card(s) that plug into the Computer Slots. Analogue, ISDN Basic Rate, or ISDN Primary Rate.
    • In 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.
    • You connect Primary Rate ISDN cables into this board (Think of it as a very, very sophisticated modem)
    • For a 24 line system you need one card with one socket - In the UK these trunks (E1) have 30 lines capacity.
    • You need to talk to your telecom. provider for and incoming trunk(s) that connects to this
  3. The CallHandler Operating Software that manages the whole system and provides other capabilities
  4. The 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
If 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.
What is Power/Preview Dialling in Call Center Applications?
 
  1. What are its dialling principles?
    • The 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.
  2. What is its design criteria i.e. what are its benefits & attributes?
    • The 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
  3. Where does it sit relative to the agent desktop and PABX?
    • a) The CallHandler can replace a PABX,
      or b) The CallHandler can interface to a PABX via a link e.g. DPNSS or QSIG
  4. Does it interface to any common "Contact Management Systems" e.g. Sales Logix
    • No direct interface, interfacing should be very easy using the PowerDial/PreviewDial COM object.
  5. What components are used in its construction?
    • The CallHandler Server
      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
    • PowerDial COM Object
      C++ Software COM Component Communicates via TCP/IP to CallHandler Server
How is SMS messaging dealt with?
  CallHandler 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 mobile phone.
We can support several delivery methods:
    - via GSM modem / Mobile phone attached to CallHandler.
    - ISDN/Modem dial-up to an SMSC (SMS Message Centre)
    - via Internet to an SMSC
Chelston has developed the SMS messaging software from the ground up to be very reliable and efficient.
Costs roughly:
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:-
        www.clickatell.com
        www.bulksms.com
        www.csoft.co.uk
        or type: "bulk sms uk" into Google
    2. There are several of ways to do this, depending what the service provider supports:-
        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 provider's server
        c. As above, but using TCP/IP sockets connection.
What low level logging info is available from E1/T1
  Here 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 | ResHandle:0x86200800

^-- Start Date/Time Of Call ^-- Call Duration In Seconds
StartDate:01/07/2002 StartTime:00:01:10 Dur:43

^- DDI Number (If Set) ^- CLI Number (If Set) ^- Real Destination Number ^-Real Originating Number
DDI: CLI: DestAddr:96789354112 OrigAddr:

^-- The Reason The Call Ended ^-- The Raw Clearing Cause Code Taken From The Protocol
Cause:NORMAL:0x0 Raw:0x0
What 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 (Q.931). 
  We 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 
Can CallHandler support more than a single E1?
  We 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 ***
2400 Lines 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)
How many mailboxes?
  There are a number of things that limit the number of mailboxes you can have:
i) 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.
ii) 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?
  This 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.
Lernout & Hauspie Text to Speech - what do I need?
This uses the DSP resources (opposed to a separate host PC). We can support up to 26 concurrent channels per card - 8 Cards per PC 
Microsoft SAPI Support - what can it do? 
This 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.
What custom controls does CallHandler have?
  There 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 Maintenance Centre.
What does the Developer Kit include regarding Text to Speech
  Developers Kit including
a) API Documentation
b) Example Code
c) Hooks for Aculab and Lernout & Hauspie Text to Speech Systems
Do I need special equipment to run the Conference Demo Disc?
  No, the demo software is Install Shield based and contains all the necessary modules and simulators.
Can we use the existing E1 Trunk which has 15 lines currently used for the office exchange?
  Yes, 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 this.
Why do you still use NT when there are later versions of MS Windows available
  CallHandler 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.
What version of SQL do you support?
  Latest Version
What is the format of the audio data?
  The 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.
These 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 between countries.
Do you have any references on Audio Formats?
  [1] 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: ftp://ftp.microsoft.com/softlib/mslfiles/mdrk.exe filename: RIFFNEW.DOC
[2] ITU-T Recommendation G.711 [11/88] - Pulse code modulation (PCM) of voice frequencies. Available from the International Telecommunications Union (ITU) at http://www.itu.ch
[3] MSDN Library - Resource Interchange File Format Services - Multimedia: Platform SDK
Do your Telephony Cards work in Egypt?
  Our 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 telecom providers.
Can 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
  There 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?
  We support D-channel protocol on E1. There are 2 D-channels on E1, one for framing and the other for protocol signalling.
An example of a D1 protocol is Q931
PLEASE SUBMIT YOUR OWN QUESTIONS


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