WEEK 13 ( 28 JUNE- 2 JULY)

This week I did some testing on Microsoft Outlook. I learned how to fax using Microsoft Outlook. This was done through UNICEED account. Then, the testing was recorded step by step in manual form. 

Microsoft Outlook - Fax

I edited the FAQ for UNICEED. Besides that, I also did some settings in Superceed website for the call flow. Since there was error in the website, I was asked to upload the voice file again.

Finally, it was time for Beta testing. My colleagues and I pretended to be agents that answers the calls. Then , we answered the calls for different paths.That was my last task for Industrial Training.

WEEK 12 (21-25 JUNE)

My task for this week was fully involving the superceed/vcc/inbound website. I had to create path for each options and level in the website. I also had to upload the voice file that I created last week. Before doing this, I had to create certain categories in the website. 

Superceed website

One of the part of the call flow

Next, I had to create agents/supervisor to attend the calls for each path. I have created about 20 agents for all the paths in the call flow.  

Agents/supervisor list

Besides that, I also created 30 frequently asked question by customers and prepared the answers for each questions. I type out the Q&A and convert it to PDF file. This was then uploaded in the website under knowledge base section. This was done for the agents so that it will be easier for them to answer the customer's questions. 

 

FAQ in knowledge base

Finally was the attributes for agents. I had to divide it into categories and sub-categories.

For example:

Cat                                Subcat

Languages                    English

Department                   Sales & Marketing

Sub                               Features

Specialized on               Voice

WEEK 11 (14-18 JUNE)

This week I was asked to design a call flow for the company's product UNICEED. A call flow is an interactive voice response application. The call flow describes how the caller enters the application, the options and inputs (key presses) that are provided to the caller, and the application's response to these inputs. Before designing a call flow, certain criteria need to be planned. For example, how many languages are available for this customer service or what departments do I route the calls to. 

Call Flow Diagram

The call flow is divided into 4 levels and 3 options. When a customer call UNICEED customer service, firstly they have to choose their preferred language. Then, departments that they wanna deal with. Next would be the subcategory of the department and finally is what they wanna specialized on. 

There 3 options available are English, Bahasa Melayu and Mandarin (level 1). Each language are divided into sales & marketing, technical assistance, billing, general enquiries and automated guide. These are called the departments (level 2). 

Under sales & marketing department, there are features, packages, rates and reseller information. Whereas under the technical assistance, the choices available are voice, fax, email and sms. Payment, renewal and reload options are available under billing department. Other than that, if the customer choose general enquiries, it will directly connect them to the telephone operator. Under the automated guide are sales, technical and billing. All these are sub-categories (level 3).

The final level is level 4 (specialized on). This level is only available for sales & marketing and automated guide. Under sales-features, there are voice, fax, email and sms. Whereas under rates, available options are domestic and international. For reseller information, there are uniceed seller program and private label program. Finally, the under each sub-categories of automated guide, there are few frequently asked questions (Q&A). This option will assist the customer when it's after office hours. 

All these levels are then translated to Bahasa Melayu and Mandarin in order to cater customer's needs in other languages. Besides that, I had to convert those text to speech. This was done by using Text-To-Speech software. After recording the voices, I edited the voice file in wavepad editor by changing the sample frequencies, bit rate and many more. 

WEEK 10 (7-11 JUNE)

Beginning of this week, my colleagues and I had a meeting with our boss. We discussed about all our task that have been done for the past 2 months. In the meeting, I have been briefed about call routing strategies and virtual contact center. Besides that, I was also given new tasks for the week.

 In the meeting room

Nowadays, someone in England can make a phone call to Australia, or that someone in United States can read web pages that are on a computer in Canada. A large network is affected by many factors which are often hard to predict. There can be busy and quiet periods through the day. For example, if a television program has a phone-in vote, there can be a sudden overload at one point in the network. This is when routing calls comes in handy. These routing schemes work by searching out the spare capacity in the network so as to route calls away from parts of the network that are broken or full and into parts that are underloaded.

Call routing strategies is divided into:-

• last agent
• round robin
• lowest total talk time
• fewest received calls

Last agent

Route new calls to the general queue and repeat calls to the last agent contacted with overflow to the general queue. In this routing strategy, calls that queue for a specific agent are overflowed to the general queue after a pre-set period of time (i.e, wait, 1, 10, 20 and 30 seconds for a specific agent). Repeat callers waiting for the last agent have a higher priority than general callers.

 last agent

Round robin

If more than one agent is available, the system will try the agent who goes after the last one who took a call in a round robin fashion. Imagine 3 agents are in a circle. A call comes in, and the first agent takes it. Next time a call comes in, agent 2 takes it. Next time it will be agent 3, and then back to agent 1, as long as all of them are available. This routing method is appropriate for situations when most calls are of comparable length and all agents need to take a comparable number of calls. In other words, it balances the load of incoming calls based on number of calls.

I also learned about contact center.

Virtual Call Centers

Skills Based Routing can be implemented as part of a much more elaborate call center scenario, the virtual call center. Several vendors are now providing fresh looks at how a virtual call center would operate within a network environment. The application sits on top of Centrex, central office or PBX environments, allowing agents from multiple geographic locations to log into a single virtual ACD environment. Think of it as an ACD without the equipment, but totally dependent on the public switched network. In this scenario, routing calls based on agent skills is a network requirement and easily implemented. Calls are labeled as a primary skill or a non-primary skill. Matching calls to agents becomes easier and less costly within the network and easier for the customer and less costly for the call center.

Vendor Approaches

There are a variety of companies providing products and services with special features focused on Skills Based Routing.

Blended Call Centers

Blended environments are call centers where inbound calls and outbound calls are handled by the same agents. Inbound call activities are assigned as one skill and outbound call activity is assigned to a second skill. Vendors supplying call routing systems for this environment may include enhanced ACD and IVR functions with CTI features.

Premise-Based Contact Center

Call center have been built on PBX equipment that is owned and hosted by the call center operator. The PBX might provide functions such as Automatic Call Distribution, Interactive Voice Response, and skills-based routing. The call center operator would be responsible for the maintenance of the equipment and necessary software upgrades as released by the vendor.

Lastly, I learned call queue and call distribution strategy.

• Call Queue is to offer customer the best waiting time possible.
• Call Distribution Strategy : supervise talk time (average 3 min) and supervise the etiquette of agents.

WEEK 9 (31 MAY- 4 JUNE)

Finally, it's the last month of Industrial Training. Yahoo!!! Last month of the internship I'm gonna be doing beta testing. So, this week was preparation for the beta testing in order to participate in real contact center environment.

Beginning of the week, I was asked to write a call script. A call script consists of Q&A designed anticipate questions from the caller, and what answers you could provide to each of the anticipated question. I was assigned to Syarikat Bekalan Air Selangor (SYABAS). This call script contains of 30 Q&A's in General Enquiries, Customer Service and Billing.

End of the week, I compiled a list of 50 contacts in Mid Valley. The contacts are from companies in South Tower, North Tower and Boulevard. After that, I did an additional 30 contacts for KL Sentral companies. This was done in order to test the inbound and outbound calls. 

WEEK 8 (24-28 MAY)

Erlang algorithm is a common algorithm used in Workforce prediction in a contact center environment. This week I learnt about Erlang. Basically, erlang is a unit of traffic density in a telecommunications system. One erlang is the equivalent of one call (including call attempts and holding time) in a specific channel for 3600 seconds in an hour. The 3600 seconds need not be, and generally are not, in a contiguous block. Erlang is divided into 3 which are Erlang A, Erlang B and Erlang C.

In call centers, Erlang A is used, to support solutions of the staffing problem, such as how many agents should be answering calls during a specified time period. In order to apply Erlang A, it is necessary to input values for its four parameters: λ, μ, θ and n. Parameters λ and μ are calculated for every hourly interval. We also calculate each hour’s average number of agents n. Because the resulting n’s need not be integral, we apply a continuous extrapolation of the Erlang-A formulae, obtained from relationships developed in. Finally, for θ we use the formula P{Ab} = _ • E[W].

A traffic engineering model that assumes that an offered call is cleared immediately, with no queuing. In other words, Erlang B assumes that a call encountering blockage will not appear again. Either the caller will hang up and not attempt to place the call again, or the call will automatically be routed over another circuit if one exists, even if the use of that circuit is more expensive. 

Erlang B is the formula to use when a blocked call is really blocked. For example, when somebody calls your phone number and gets a busy signal or tries to access a tie trunk and finds it in use. It is built around three variables which are Servers, Traffic, and Grade of Service.

The most common traffic engineering problem involves sizing a trunk group. For example, how many trunks are needed to carry your toll-free calls, how many tie trunks between two offices, how many ports into your voice mail system, or some similar question. Erlang B handles that relatively easily, in four steps:

1. Collect traffic data
2. Determine the Average Busy Hour
3. Choose a target Grade of Service
4. Use Erlang B

Example of Erlang B

Erlang B Formula 

Erlang C formula is used when a blocked call is delayed. For example, when someone calls your call centre and must wait for an agent to take the call. It uses the same three variables, plus the average length of each call, to calculate the probability of being delayed and how long the delay is likely to be.

Because Erlang B is so simple to use (insert two numbers, it calculates the third), many people assume that Erlang C will be similarly easy. But it is not true, 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.

Erlang C Formula

After learning about Erlangs, I searched for erlang sample excel sheet calculations and formula to test it out. Firstly, I searched Erlang B.

Example of Excel Function for Erlang B:

ErlbBlockage(nsrv,trafficInErlangs) returns the Erlang B blockage for a specified number of servers and specified offered traffic. The nsrv is the number of servers (can be any non-negative number) and traffic in Erlangs is the offered traffic in erlangs (can be any non-negative number). For example, =ErlbBlockage(3,10.4) returns Erlang B blockage for 3 servers and 10.4 erlangs of offered traffic. This then returns the value 0.7411421.

Besides that, there are more functions for Erlang B in excel such as ErlbNsrvFromBlockage and ErlbTrafFromBlockage.

Example of Excel Function for Erlang C:

ErlcFractionDelayed(nsrv, trafficInErlangs) returns the probability that a customer arriving at the queuing facility will experience a delay before beginning service. For example, =ErlcFractionDelayed(11,10.1) returns the probability that a customer will experience a wait, when there are 11 servers and 10.1 erlangs of offered traffic. This then returns the value 0.71095477.

Besides that, there are more functions for Erlang C in excel such as ErlcFractionOk, ErlcNsrvFromFractionOk,  ErlcWait and ErlcNsrvFromWait.

WEEK 7 (17-21 MAY)

This week task looked simple but it's actually very hard. I had to find out Text-to-Speech plugin asterisk suppliers. I also had break it into 4 sub-topic such as best virtual person, integration with vendor products, integration with custom 3rd party application and product summary.

Suppliers that I manage to find are as listed below:

• The Festvox Project
• Cepstral
• Acapela
• Power Text to Speech Reader 2.31
• Sayvoice Text to Speech Reader
• Word Talk
• ACE-HIGH
• Sayz Me
• NextUp Talker
• Loquendo

Besides that, I also had to find out contact center vendors (premise-based and hosted).  Many contact centers are heavy users of communications technology. Premise-based are the traditional equipments and software that you need to install in the office. Hosted are those which you do not need to buy, just need to subscribe. 

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

• Nobel Systems
• Aspect
• Cisco
• Computer Talk.Tech
• Upstream Works
• Zeacom
• Interactive Intelligents
• LumenVox
• TouchStar
• Siebel

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

• Cincom
• Contactual
• Envox Worldwide
• LivePerson
• Smoothstone
• TeleTech
• UCN
• Five9
• Packet8

WEEK 6 (10-14 MAY)

This is my 6th week in Dare BPO. This week I continued my research on Text-to-Speech (TTS), Speech-to-Text and predictive dialing.

Firstly, I shall discuss about Text-to-Speech. TTS is the artificial production of human speech. TTS transforms any text into speech in real time. It literally reads out loud any written information with a smooth and natural sounding voice.The automatic intonation reflects the meaning of the text, with respect to pauses, breath groups, punctuation and context. The most important qualities of a speech synthesis system are naturalness and intelligibility. The computer system used to achieve this is called a speech engine. You can try using the TTS application at http://www2.research.att.com/~ttsweb/tts/demo.php. 

Example of TTS application

In order to reproduce the natural sound of each language, a narrator records a series of texts which contain every possible sound in the chosen language. These recordings are then sliced and organized into an acoustic database. During database creation, all recorded speech is segmented into some or all of the following: diphones, syllables, morphemes, words, phrases, and sentences. To reproduce words from a text, the TTS system begins by carrying out a sophisticated linguistic analysis that transposes written text into phonetic text. A grammatical and syntactic analysis then enables the system to define how to pronounce each word in order to reconstruct the sense. We call this the prosody. It gives the rhythm and intonation of a sentence. Finally, the system produces information associating the phonetic writing with the tone and required length of the pronunciation. The chain of analysis ends here and sound is generated by selecting the best units stocked in the acoustic database. 

A TTS capability for a computer refers to the ability to play back text in a spoken voice. The Text-to-Speech tab located in the Speech Control Panel presents the options for each TTS engine. Below are the steps for configuration: 

Step 1: Set Up Speakers.
Step 2: Select an Audio Output Device.

Step 3: Set Audio Output Device Options.
Step 4: Configure Text-to-Speech Options.

Next topic to discuss would be Speech-to-Text. Speech-to-Text (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.

Speech recognition applications include voice dialing (e.g., "Call home"), call routing (e.g., "I would like to make a collect call"), domotic 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). 

Speech recognition fundamentally functions as a pipeline that converts PCM (Pulse Code Modulation) digital audio from a sound card into recognized speech. The elements of the pipeline are:

1. Transform the PCM digital audio into a better acoustic representation.
2. Apply a "grammar" so the speech recognizer knows what phonemes to expect. A grammar could be anything from a context-free grammar to full-blown Language.
3. Figure out which phonemes are spoken.
4.  Convert the phonemes into words.

When a person speaks, vibrations are created. The speech recognition technology converts these vibrations, for example analog signals into a digital form by means of an analog-to-digital converter (ADC). Digitization of sound takes place by its measurement at regular intervals. The sound is filtered into different frequency bands and normalized, so that it attains a constant volume level. It is checked whether the sound matches with the already stored sound templates. The next step in the speech recognition procedure, is dividing the analog signals into segments that range from a few hundredths to thousands of a second. These segments are matched with phonemes that are already stored in the system. Phonemes are specific sounds that are understood by people speaking a particular language.

Final topic to discuss is 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 is far more advanced than using an autodialer 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. Predictive dialing is so named because it predicts when callers will become available to take a new call, and dials calls in advance. When a person answers the phone, predictive dialing puts the call through to an agent, although there is sometimes a brief delay as the predictive dialer attempts to determine whether the person's voice is a recording, in which case the call is ended.

In call centers and other applications where predictive dialer is employed, information pertaining to telephone numbers of people and businesses to be called is stored in a network server. All agents are linked to the server. The network is also linked to the predictive dialer, which can be either a hard dialer or a soft dialer. With agents at work, the server and or the dialer start dialing the numbers. The calls are then managed by the dialer. In case of silence at other end, the dialer will hang up. From the other calls, the dialer will screen out busy, unanswered, and answering machine calls. Only the live calls are put through to the agents. The instant agent gets connected to a call, all information pertaining to the call gets displayed on the agent's screen. 

WEEK 5 (3-7 MAY)

This week task was based on R&D. The topic given to me was Call Queuing. Call Queuing is a method of handling calls until they can be answered by an agent. Call Queuing is a sophisticated queuing system that allows you to accept more calls into your telephone system than you have extensions or employees capable of answering them.

A Call Queuing system handles incoming callers and prevents an engaged tone when the intended recipient, or group of recipients, are already on the phone. For example, if a member of the Sales Team is on the Phone, or even the entire Sales Team, a caller will hear an engaged tone unless Call Queuing is in place. The caller can be held in a queue and played music, or pre-recorded announcements, or given the option to leave a message. Call queuing significantly reduces the likelihood of missed calls and inquiries. 

The queue works retaining call while finding an unoccupied agent to answer the call. Usually, a call queue works as below:

• Calls are queued.

• Agents answer the queue (logged in agents).

• A queuing strategy to distribute the calls is used.

• Music on hold is played while the caller waits.

• Announcements can be made to callers, warning about waiting time.

The next task was Automatic Call Distribution (ACD). An ACD is a system that automatically distributes phone calls to a specific group of agent work stations.

A department generally asks for one to be set up when it wants incoming calls to go to a single or main number and be distributed among a number of staff because the volume is too large to be handled effectively by one person. This single or main number is the one you will see published in the phone book. It is not assigned to a particular telephone. When someone calls this single or main number, the ACD routes the call to one of a group of working telephone extensions that have been assigned to the ACD group. During busy times when an agent is not immediately available, an ACD can queue calls, provide a recorded announcement, forward the call to someone designated to handle call overflow, offer a voice mail option, or give the caller a busy signal. A department determines what options a caller will be offered when the ACD is initially set up.

An ACD is really one component of an automated call processing system. It works in conjunction with the Phone Mail system to route incoming calls to staff whose job it is to answer the calls. If you have ever called a Medical Center Clinic, you probably heard, "If you are calling for an appointment, press 1, if you are calling for a prescription refill, press 2, etc." These options are provided by the Phone Mail system. When you press 1 for an appointment, the call is forwarded to an ACD. When you press 2, it is forwarded to a voice mailbox, etc.

The number of people in an ACD group is determined by the department when the ACD is set up. At some future time, a department may decide that there are too many or too few people in the group and adjust the number. An ACD forwards incoming calls to the staff member who has been waiting (idle) the longest. When all the extensions are busy, the ACD handles the call according to instructions in a "routing table" specified by the department. For example, the department might decide that if all agents are busy, it wants the caller to hear a recorded message and placed in a queue ("All agents are busy. Calls are answered in the order they are received."), or it may decide that a caller should be given the option of being put in a queue or leaving a voice mail message. There's a lot more to routine tables than this, but now you have an idea of how they fit in.

Lastly, I did a little research in Avaya, Genesys and Cisco. This products also uses call queueing and acd. Genesys supports the integration of IP telephony services (phone-based services) with a broad range of customer contact channels, such as other phones, e-mail or chat. ACD is a very popular feature within Cisco Unified CME for locations that don’t have the luxury of either having a Cisco Unity Express, Unity or IPCC Auto Attendant. Its purpose is to provide basic call handling service for incoming calls to a CME.

Here are several steps needed to configure CME B-ACD: 

• Download TCL Scripts and Audio Prompts

• Set up incoming Dial Peers for AA Pilot Numbers

• Set up Ephone Hunt Groups

• Set up Call-Queue and AA service 

Whereas, Integrating SpeechAttendant in an Avaya ACD environment involves the following steps:

Step 1: Creating a hunt group

Step 2: Programming agents

Step 3: Programming stations

Step 4: Programming a call vector

Step 5: Programming a vector directory number

Step 6: Configuring the settings related to automatic call distribution

WEEK 4 (26-30 APRIL)

After doing some research in the last week's 3 topics, this week I did practical in these 3 topics. The first practical task was Find Me/Follow Me test call which the subscriber 1 calls in, rings simultaneously or sequentially on subscriber 2, 3 and 4's device (IP phone/softphone).

A customized find me / follow me schedule created right in the Web account. The days and times the customers will be available with the phone numbers where they can be reached is entered in the system. If the user uses follow me when the user receives a call, the system dials the numbers in the order they specify. If there is no answer at the first number, follow me will call the next number and so on. The numbers are dialed sequentially. If the user uses find me when the user receives a call, the system dials all the numbers simultaneously. When the user answers any one of the number, the calling will stop for the other numbers.

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

1. Create an user extension in file editor. (Created user extension will appear in ring groups. The user extension is created by typing coding into the config files.)


2. Edit Ring Groups.


3. Select strategy either Ring all or Ring in order. (Ring all is Find me which rings simultaneously. Ring in order is Follow me which rings sequentially.)


4. User extensions in available channels are sent to ring group members.


5. Test Calls ( 3 different subscribers receive calls simultaneously and sequentially from one caller)

The next practical task was 3-party voice conferencing which test call from subscriber 1 to subscriber 2 and subscriber 3 able to listento the conversation.

The basic concept of audio conferencing uses two or more speakerphones and several dial up numbers. With most services, the user would set up a meeting for a specific time and there will be either local or toll free numbers available for each person to call in on. They would in turn enter the required information to gain access to the meeting that you have been invited to attend. Then everyone is either on a single handset or at a table surrounding a speakerphone of some type. This is accomplished by the use of a pin number entry system and then the only people that will have access to the meeting are those that have provided with a pin number. The user can change the pin number for each meeting, providing greater levels of security.

Below are the steps for 3-party voice conferencing:

1. Create an user extension in file editor. (Created user extension will appear in ring groups. The user extension is created by typing coding into the config files.)


2. Configure Conference Bridge Extensions. (New extension is created by clicking new in conference bridges)


3. Edit conference room options.


4. Test Calls. (3 different subscribers able to do audio conferencing)

Audio Conferencing

After carrying on all the practical task, I documented all the testing and findings. The test plan document was done to be sent to the main branch.

WEEK 3 (19-23 APRIL)

Beginning of the week, I continued last week pratical task which was testing the IP phone and softphone. I had to configure the IP phone in order to make and receive calls. The configuration was done by excessing the phone's asterisk server. The phone that I used was Linksys SIPURA SPA 921. After configuring it, I was able to make calls to the softphone which I configured last week.

How an IP phone works

Configuration in Asterisk server

End of the week, I was asked to do some research. I was given 3 research topics. First topic was IVR customiztion. IVR stands for Interactive Voice Response. It is the technology that we use everyday when we call a bank or a hospital or any large business. It is the voice activated prompts that brings us to the correct person or gives us the answer we want. IVR automates inbound call routing and fosters self-service.The whole process of placing an order by the caller, and checking the status with the help of the order number can be automated with the customized IVR flow. During wait time or hold time, messages about other products and promotional offers can also be played allowing you to capitalize on cross-selling and up-selling opportunities. For example a customer may ask for a "Meal for two" instead of two individual pizzas after listening to the offer while he was in the queue.

IVR application

The second topic was Find Me/Follow Me. Find me and follow me are two call forwading services that are commonly used in conjunction with each other. Find me service allows the user to receive calls at any location. Whereas follow me service allows the user to be reached at any of several phone numbers. Find me / follow me numbers use VoIP (Voice over Internet Protocol) technology to route all of the incoming calls to one single telephone number. These numbers let the users consolidate their communication devices.

Find Me/Follow Me

The last topic for the week was 3-party voice conferencing. Voice conferencing is one of the most critical elements of group communication. Excellent voice quality is essential for both voice-only calls and visual communication environments, ensuring that everyone can clearly hear and be heard to improve productivity and deliver a seamless, natural communications experience.

Below are the steps to setup a third-party conferencing service to be the main audio conference solution:

1. As the host, enter the meeting room.
2. Select the Meeting menu.
3. Select Set phone conference options. The Phone Conference Options appears.
4. Select Other.
5. Enter the conference number.
6. Enter the participant code.
7. Enter the Moderator code.
8. Click OK.

Audio Conferencing

WEEK 2 (12-16 APRIL)

This week was kinda exciting. Beginning of the week, I did some research on Session Initiation Protocol (SIP). SIP is a signaling protocol used for establishing sessions in an IP network. The protocol can be used for creating, modifying and terminating two-party (unicast) or multi-party (multicast) sessions consisting of one or several media streams. SIP responses are the codecs used by SIP for communication. The codecs that I focused was g729, g726 and g711.

SIP configuration

After doing some research on SIP, the next day task was involving application on SIP. Firstly, I logged in into AsteriskNow server of this company and created an user extension. My extension number is 1115. Next, I downloaded free softphone XLite from the internet and configured the settings of the softphone. Finally, I tested the softphone by making calls to networked users. To hear the mail box messages, extension number 8888 can be dialed.

Softphone XLite

Besides that, I also learned about Host Media Processing (HMP). Dialogic HMP software performs media processing tasks on general-purpose servers without requiring the use of specialized hardware.


The next day, I was exposed to Welltech 6500 and Wellrec 5600. Welltech is the IP PBX system to operate on a variety of VoIP applications. The Welltech WellCIP 6500 series SIP Telephony server is the best choice to convergence VoIP network which convert the requirements from enterprise to service provider. Wellrec 5600 is a VoIP recoder for Welltech SIP network architecture.

Welltech 6500 and Wellrec 5600

My last task for the week was building a custom Interactive Voice Response (IVR). IVR is a technology that automates interactions with telephone calls. I downloaded Zoiper softphone from internet and configured the settings with my user extension. Then, I connected the IP phone and made phone calls from the IP phone to the softphone and vice versa.

IP phone



Testing the softphone and IP phone

WEEK 1 (6 - 9 APRIL)

Since this was just my first week, I learned the basics of telecommunication. Firstly, I learned how to use wavepad. Wavepad is an audio editing software that allow us to make and edit music, voice and other audio recordings. This software was kinda cool. It had many features like amplify, trim, reverb, echo, pitch change, equalizer and many more. We can change a voice to a totally different voice by doing some editing.

Wavepade editor

After that, I was assigned with my first task. My first task was to create an operator service (Virtual PBX). PBX used to calls answered with a custom business greeting, offer an audio scroll-through of your employee directory, provide a connection directly to a specific person or department, play music while the system is in hold, and even take voice mail for emloyees. Firstly, I google AT&T TTS and IBM. TTS is the creation of audible speech from computer readable text. Then, I typed out a speech text and clicked download. Next, I opened the downloaded wavefile in wavepad and edited the voice. Finally, the edited voice was uploaded in uniceed demo account under UC settings in multilevel Virtual PBX.

Besides that, I also learned about Web ticketing and online chat system this week. This was done by creating an account in supportuniceed/staffs and browsing the system. Other than that, I learned about Asterisk and AsteriskNow. Asterisk is an open source telephony project. It turns an ordinary computer into a feature-rich voice communications server. Asterisk includes all the building blocks needed to create a PBX system, an IVR system. AsteriskNow creates custom telephony solutions by automatically installing the "plumbing".

ASTERISK configuration

End of the week, I was required to create an user manual for uniceed. My friends and I divided the task to parts. I wrote the manual for UC settings which includes Multi-level PBX, call diverts to voice mail and Find me/Follow me. I was also required to edit the manual that I did previously to suit the VOS manual (Virtual Office Suite). The editing was done by replacing the relevant screenshots from UNICEED to VOS.

FIRST DAY OF INDUSTRIAL TRAINING (5 APRIL)

Today is my first day of my internship. I'm gonna do my 3-months internship at DARE BPO Sdn.Bhd. This company is situated at Plug & Play Technology Garden, Level 7 of The Gardens South Tower Mid Valley City. This company is based on Telecommunication. The product that DARE BPO offers is Unified Communication Service (UNICEED).


Services that UNICEED offers are:-

-Webmail
-Voice-to-Email/ Fax-to-Email
-SMS Messaging
-Divert calls
-Virtual PBX
-Interactive Voice Response
-Find Me/ Follow Me
-Multi-party Video Conferencing
-SMS Notifications
-Fax Forwading

My working hours is from 10am to 7pm. Since it's my first day, I left my house early and arrived at the Gardens South Tower lobby at 9.30am. At the lobby, I met 3 of my friends who also got the same placement. After reaching the office, Mr. Jeffrey, CEO, gave us a brief introduction about the company and the services. Since it's a new branch, the office is very small and not many staffs. The main branch is at Cyberjaya.

Our Workstation for 3 months

After the briefing, we were allowed to log in into a demo account to try out the service from customer's perspective. I typed a message from the account and send it to my phone. I also tried other applications such as Find me/Follow me. If "Find me" is selected, incoming calls can ring simultaneously on three user-defined numbers. Whereas, if "Follow me" is selected, incoming calls will ring sequentially through three user-defined numbers. Besides that, I also tried the Virtual PBX. For example:
Welcome to ABC company!
For sales, press 1
For customer service, press 2
For technical, press 3

Trying out the service

After trying out the service, my friends and I went out for lunch at 12.30pm. After lunch time, Mr. Jeffrey went to the other branch in Cyberjaya. Before he left, he gave us the office key to lock the office at 7pm.