Sunday, May 30, 2010

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:




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.

Monday, May 17, 2010

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.

 

Sunday, April 18, 2010

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


Saturday, April 10, 2010

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 .

Wednesday, April 7, 2010

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.