Week 12

This is my last week in Superceed, after that I am free ! Starting of this week, my supervisor Mr. Jeffrey continue discuss with us about the call flow. Basically, it is the same routing like last week.  After some discussion, I have to correct the error and upload the wave file in the website.

 

Sample of call flow

Week 11

After finish doing the 30 Q & A and the call flow, Mr Jeffrey continue discuss with us about the Beta Testing. About the last week call flow, my colleagues and I suppose to edit to more details. For example, under sales department I need to break to subcategories like features, packages, rate and reseller info. Besides that, I need to convert the call flow and Q & A into mandarin. Then transfer into sound using text to speech software.

Mr Jeffrey is explaining the call flow.

By using wave pad I can edit the sound.

Tasks:

1) Anticipate Q & As for Phone Explanation and create a call flow option branching out from Press 1 for Sales and Press x for Customer Service that will provide automated explanation users.e.g. Press 1 for Sales; Press 4 for frequently asked questions

Press 1 on how to signup

Press 2 on how to make a payment

Press 3 on activation process

2) Create agent profile

Week 10

This week I had a meeting with my supervisor, Mr. Jeffrey and my collegues. This meeting is to brief us about the upcoming task we going to do which is Beta Testing. Firstly, he explain to us about Call Routing Strategies.

This is our meeting room

My collegue and I taking notes on what my supervisor said.

Call Routing Strategies

1. Lowest total talk time
2. Last agent
3. Fewest received calls
4. Round Robins

Round Robins

It is one of the oldest, simplest, fairest and most widely used scheduling algorithms, designed especially for time-sharing systems. A small unit of time, called timeslice or quantum, is defined. All runnable processes are kept in a circular queue. The CPU scheduler goes around this queue, allocating the CPU to each process for a time interval of one quantum. New processes are added to the tail of the queue.

The CPU scheduler picks the first process from the queue, sets a timer to interrupt after one quantum, and dispatches the process.

If the process is still running at the end of the quantum, the CPU is preempted and the process is added to the tail of the queue. If the process finishes before the end of the quantum, the process itself releases the CPU voluntarily. In either case, the CPU scheduler assigns the CPU to the next process in the ready queue. Every time a process is granted the CPU, a context switch occurs, which adds overhead to the process execution time.

Last Agent

Last agent optimization is a technique that reduces the number of sync point flows in a sync point tree. The initiator picks one adjacent agent as a last agent. Then the initiator sends prepare to the agents that were not selected as a last agent. When these agents all respond with a request commit back to the initiator, the initiator sends the last agent a request commit rather than the usual prepare. This last agent is free to select one of its cascaded initiators to be the last agent and so on. The ultimate last agent is given the commit decision for the entire sync point tree.

During the meeting, Mr Jeffrey assigned us some task. We have to design a call flow, for example when customer call our customer services center what they will heard. Besides that, we also need to create 30 Q & A for preparation.

sample of call flow

Week 9

This is the last month of my industrial training, I am going to make prepareration for beta testing. We will cover these things as below:

1) Canned Responses

     Industries

     a) Government

     b) Telcos

     c) Finance

2) Customer Service

3) Supervisory Skills

This week I will tackle the call script. A call script consists of Q & A designed anticipate questions from the caller, and what answers we could provide to each of the anticipated question. I was assigned to Dell, I need to prepare 10 Q & A touching on Sales, Customer Service, and Billing.

 

Week 8

This week research is about Erlang. Erlang is a general-purpose concurrent programming language and runtime system. The sequential subset of Erlang is a functional language, with strict evaluation, single assignment, and dynamic typing. While threads are considered a complicated and error-prone topic in most languages, Erlang provides language-level features for creating and managing processes with the aim of simplifying concurrent programming. Though all concurrency is explicit in Erlang, processes communicate using message passing instead of shared variables, which removes the need for locks.

An Erlang is a unit of telecommunications traffic measurement. Strictly speaking, an Erlang represents the continuous use of one voice path. In practice, it is used to describe the total traffic volume of one hour.

Erlang B is a modeling formula that is widely used in call center scheduling. The formula can be used to calculate any one of the following three factors if you know or predict the other two:

• Busy Hour Traffic (BHT): the number of hours of call traffic during the busiest hour of operation

• Blocking: the percentage of calls that are blocked because not enough lines are available

• Lines: the number of lines in a trunk group.

Erlang B can determine the number of trunks, or lines, needed to handle a calling load during a one-hour period. However, the formula assumes that lost calls are cleared; i.e., if callers get a busy signal, they will never retry. This assumption means that Erlang B can underestimate the number of trunks needed. For this reason, it is best used in situations with few busy signals. The Erlang B Extended formula takes into account the callers who will immediately retry if their calls do not go through.

Formula for Erlang B

Erlang C is a traffic modeling formula used in call center scheduling to calculate delays or predict waiting times for callers. Erlang C bases its formula on three factors: the number of reps providing service; the number of callers waiting; and the average amount of time it takes to serve each caller. Erlang C can also calculate the resources that will be needed to keep wait times within the call center's target limits. This method assumes that there are no lost calls or busy signals, and therefore may overestimate the staff that is required.

Because Erlang B is so simple to use (insert two numbers, it calculates the third), many managers assume that Erlang C will be similarly easy. That’s a mistake — even basic Erlang C calculations are difficult, and more complex ones can be daunting indeed. Erlang C is most commonly used to calculate how long callers will have to wait before being connected to a human in a call centre or similar situation. This adds complexity in at least four areas.

Formula for Erlang C

Week 7

This week my task is to create a summary to summarize supplier for text-to-speech and speech-to-text. I need to divide in six subtopics which is supplier name, best virtual person, URL, Integration with Vendor Products, Integration with custom 3rd party application, and Product Summary.

Suppliers that I manage to find are as listed below:

• 
  
  Nuance
• 
  
  IBM
• 
  
  LumenVox
• 
  
  Amcom software
• 
  
  Fonix
• 
  
  CMU Sphinx
• 
  
  MacSpeech
• 
  
  SRC
• 
  
  Sensory
• 
  
  Philips

Other than that, I also need to do some research about Premise-based and Hosted Contact Center vendors. Premise-based are the traditional equipments and software that we need to install in the office. Hosted are those which you do not need to buy.  

Below are the contact center vendors (premise-based):

• 
  
  eON Communication
• 
  
  Stratasoft
• 
  
  Vertical Networks
• 
  
  APEX Voice Communications
• 
  
  Avaya
• 
  
  Cantata (Dialogic website)
• 
  
  ClickFox
• 
  
  Spoken
• 
  
  Prosodie
• 
  
  Convergys

Below are the contact center vendors (hosted/on-demand):

• 
  
  Eagle ACD
• 
  
  Angel.com
• 
  
  Five9
• 
  
  Cincom
• 
  
  UCN
• 
  
  Envox Worldwide
• 
  
  Kunnect
• 
  
  Packet8
• 
  
  TeleTech
• 
  
  Virtual PBX

Week 6

This week I continued my research on text-to-speech (tts), speech-to-text, and Predictive Dialing.

Text-to-Speech

Text-to-speech (TTS) is a type of speech synthesis application that is used to create a spoken sound version of the text in a computer document, such as a help file or a Web page. TTS can enable the reading of computer display information for the visually challenged person, or may simply be used to augment the reading of a text message. Current TTS applications include voice-enabled e-mail and spoken prompts in voice response systems. Below is one of the example of tts:

Here is the link : http://www2.research.att.com/~ttsweb/tts/demo.php

Speech Recognition/ Speech-to-text

Speech recognition (also known as automatic speech recognition or computer speech recognition) converts spoken words to text. The term "voice recognition" is sometimes used to refer to recognition systems that must be trained to a particular speaker—as is the case for most desktop recognition software. Recognizing the speaker can simplify the task of translating speech. Speech recognition applications include voice dialing (e.g., "Call home"), call routing (e.g., "I would like to make a collect call"), dogmatic appliance control, search (e.g., find a podcast where particular words were spoken), simple data entry (e.g., entering a credit card number), preparation of structured documents (e.g., a radiology report), speech-to-text processing (e.g., word processors or emails), and aircraft (usually termed Direct Voice Input).

How it works?

Speech recognition enables the operating system to convert spoken words to written text. An internal driver, called a speech recognition engine, recognizes words and converts them to text. The speech recognition engine may be installed with the operating system or at a later time with other software. During the installation process, speech-enabled packages such as word processors and web browsers, may install their own engines or they may use existing engines. Additional engines are also available through third-party manufacturers. These engines often use a certain jargon or vocabulary; for example, they may use a vocabulary that specializes in medical or legal terminology. They can also use different voices allowing for regional accents such as British English, or use a different language altogether such as German, French, or Russian. We need a microphone or some other sound input device to receive the sound. In general, the microphone should be a high quality device with noise filters built in. The speech recognition rate is directly related to the quality of the input. The recognition rate is significantly lower or may be unacceptable if you use a poor microphone. The Microsoft Speech Recognition Training Wizard (Voice Training Wizard) guides you through the process, recommends the best position to place the microphone, and allows you to test it for optimal results.

Predictive Dialing

Predictive dialing uses a computer-based system that automatically dials groups of telephone numbers, and then passes calls to available operators or agents in a calling center once the calls are connected. The most common use of predictive dialing is in call centers which make large amounts of calls, such as those run by telemarketing companies. Predictive dialing was introduced for the purpose of increasing efficiency within calling centers. Prior to its development, most call centers used devices known as auto dialers, which were merely computers equipped with telephony boards that could dial a number without a caller having to manually enter it on a keypad. Predictive dialing is far more advanced than using an auto dialer because it monitors calls made to see how they are answered. If the call goes unanswered, is met with a busy signal or answering machine, or reaches a fax machine, the predictive dialer immediately ends the call. Only calls that are answered by a live person are put through to an operator. Therefore, productivity is increased because callers do not have to listen to unanswered calls or wait for someone to pick up.

Week 5

This week I did some research about Call Queues system and Automatic Call Distribution System.

Call Queuing
Call Queuing is a sophisticated queuing system that allows to accept more calls into the telephone system than have extensions or employees capable of answering them.It allows dealing efficiently with calling peaks without losing valued customer’s calls and projects a professional image of business. With Call Queuing, instead of getting an engaged tone the customers are answered automatically and held in a queue. While they are waiting for a representative they receive personal messages about how many calls are in front of them followed by music while they are waiting

ACD
In telephony, an Automatic Call Distributor (ACD), also known as Automated Call Distribution, is 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. Routing incoming calls is the task of the ACD system. ACD systems are often found in offices that handle large volumes of incoming phone calls from callers who have no need to talk to a specific person but who require assistance from any of multiple persons (e.g., customer service representatives) at the earliest opportunity.

After doing all these research, I did some practical testing about call queuing. Here are the steps:

Step 1: Agent login
Example: An agent in the telemarketing queue picks up the phone and dials #9000. The agent hears an invalid login message and is asked for his/her name and password. The auditing queue follows the same procedure.

Step 2: Queue
Once in the queue, the agent will hear MOH if defined. When a call comes in the telemarketing queue, the agent will hear a “beep” and will be connected to that call.


Step 3: Call ending
When the agent finishes the call, he/she can:
• Press ‘*’ to disconnect and stay in the queue.
• Disconnect the phone, disconnecting to the queue.
• Press #8000 to transfer the call for auditing.

Week 4

This week we did some practical task regarding last week research which is find me/ follow me test call and conferencing.

Here are the steps for Find Me/ Follow Me test call:

1. Create subscriber account in asterisk
2. Register the subscriber account in Zoiper
3. Edit ring group in asterisk. (If want to use find me choose ring all if is follow me choose ring in order.)
4. Testing Find Me/ follow me

These are the steps for conferencing:

1. Registered the extension that wanted to put in 3-party conference.
2. Create a conference bridge in asterisk
3. Call the conference bridge
4. Enter the conference

Week 3

This week we are asked to do some research regarding 3 topics which is

1. IVR Customization

2. Find Me, Follow Me

3. 3-party voice conferencing

IVR Customization

IVR stand for Interactive Voice Response. It allows customers to access a company’s database via a telephone keypad or by speech recognition, after which they can service their own inquiries by following the instructions. IVR systems can respond with pre-recorded or dynamically generated audio to further direct users on how to proceed. Other than that, IVR system also can used to control any function where the interface can be broken down into a series of simple menu choices.IVR systems are to be deployed to be used by normal people for availing services, solutions from it. IVR system interacts with its user (caller to the IVR) through voice prompts. For example, if the voice prompts announces, “To know read your voice mail please press 1, or press 0 to end the call.”

Find Me, Follow Me

Find me and follow me are two call forwarding services that are commonly used in conjunction with each other. Find me service allows the user to receive calls at any location; follow me service allows the user to be reached at any of several phone numbers. Find Me, Follow Me gives users a choice of ways to be found or followed. They can have an incoming call to their business number ring any landline or mobile phone, including their desk phone. Users can have the phones ring simultaneously or in sequence, and they can have a message played that asks the caller to wait while their party is being located. Users can order the various actions in different sequences and combinations, and they can choose to answer a call with the press of a key. Finally, they can have the call go to voice mail only if all the other options fail to reach them. More important than what the system can do is how it decides what to do. Users can have Find Me locate them according to time of day and the day of the week — such as between 9am and 9pm, Monday through Friday — and go to voice mail at other times. They can also specify answering options for calendar days, a rarity in Asterisk systems. Additionally, users can set up a VIP list, which some call a boss-and-spouse list, of people who can reach them at any time.

 

3-party Conferencing

3-Party Conferencing allows multi-line systems to have a 3-way conversation between you and persons on 2 other lines. The 2 other lines can be inside lines, outside lines or a combination. (Under certain circumstances, the far end parties on a conference call may not hear one another clearly.) If we start either a call-in or call-back teleconference, we can control audio for attendees during a meeting - that is, we can mute and un-mute their microphones to allow only certain attendees to speak. It is also possible to host a meeting that includes a hybrid voice conference-that is, a voice conference that includes both a teleconference and an Internet phone conference.

 

Week 2

12 April 2010:

Today I did research on Session Initiation Protocol (SIP). It is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams. I am suppose to focus on codes g.729, g.726 and g.711

SIP Proxy Server

13 April 2010:

Today I was asking to create a SIP account on asterisk and setting up X-Lite. First I need to log in my company asterisk server to create a user account. Then I need to download soft phone X-Lite from internet. My extension number is 1234. At last, I can call other user. For mail box message, can dialed 8888. Other than that, I also did some research on Host Media Processing (HMP). It is a software-based technology which replaces hardware telephony boards. These boards did all the media processing work and had special interfaces for connection with PSTN. Gridborg HMP, since it is software-based, requires no special hardware and runs on PC. Connection with PSTN is established using VoIP technology.

Soft phone X-Lite

HMP was introduced, telephony boards can be replaced with a software running on PC and using VoIP for PSTN connectivity.

14 April 2010:

Today I learnt about WellRec 5600 VoIP Recorder and WellSIP 6500 Telephony SIP Proxy Server. WellRec 5600 is a dedicated VoIP recorder for Welltech SIP network architecture. WellRec 5600 will do the recording on demand no matter incoming call or outgoing call. Through the web interface, administrators can easily search and play the recorded VOIP call anywhere. It provides a very cost/effective solution to meet your VOIP logging service. WellRec 5600 can also be a lawful interception or voice monitor system. Administrator can monitor the conversation real time either by recording channel or selected subscriber without any delay.

The Welltech WellSIP 6500 series SIP Telephony Server is the best choice to your convergence VOIP network which covert the requirements from enterprise to service provider. With built-in rich telephony services, WellSIP 6500 enables traditional PABX features to your VOIP convergence platform. Also you can easily upgrade the license or provide the high available service according to the growth of your business without any hardware changes.

15 April 2010:

Today i tested the IP phone and call the extension number. Instead of using X-Lite, I download Zoiper soft phone and configure the setting.

IP Softphone

Testing the IP phone

Week 1

6/4/2010 :

Today we learnt how to use wave pad and virtual PBX. Firstly, we use AT & T Labs Text-to-Speech or IBM to generate voices by typing the text in the message box. Next we can download the voice and browse into wave pad. IN wave pad we can modify the signal and editing accordingly.

7/4/2010:

Today we learnt about asterisk and asterisknow. Asterisk is most popular open source telephony project. It also a software that turns an ordinary computer into a feature rich voice communication server. Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution.

AsteriskNOW makes it easy to create custom telephony solutions by automatically installing the "plumbing". Much of the complexity of Asterisk and Linux is handled by the installer and the administrative GUI. Application developers and intergrators can concentrate on building their solution. AsteriskNOW was built for application developers, systems integrators, students, hackers and others who want to create custom solutions with Asterisk.

8/4/2010:

Today Mr. Jeffery wants us to do user manual on sms, fax, contact list, and uc setting for uniceed.com. It is to guide our customer to be more familiar with the software they using.

9/4/2010:

Today Mr. Jeffery wants us to do user manual on sms, fax, contact list, and uc setting for vos.com. It is to guide our customer to be more familiar with the software they using .

1st day of my industrial training

Day : 5 April 2010
Time : 10 am - 7 pm
Venue : DARE BPO SDN BHD
Plug and Play Technology Garden
Suite 7.01, level 7, The Gardens South Tower, Mid valley City
Lingkaran Syed Putra 59200 Kuala Lumpur

Today is my first day of industrial training, early morning around 8.30am I went to take ktm to go to my office which is mid valley..

The Gardens lobby

At 9.45am I already reach Gardens lobby to meet my friends and my supervisor Mr Jeffrey. After that, he bring us go to the office.

This is the place we going to spent for 3 months

Mr Jeffrey introduce us the product of the company and what our job scope for this 3 months. Basically our company is more on IT and Telecommunication. We created the program easier for the customer to receive their message, fax and email. For example, the person can set his/her cell phone, office, and house number to our program and if someone call he/she this 3 phone will ring at the same time and once he/she picks up one of it the others two will automatic turn off.

This is our company product

We are trying to use the product

This is my first day of work, just everything fine.