Monday, August 30, 2010

Google's VOIP Challenge to Skype: 10 Reasons It's a Serious Threat - IT Infrastructure from eWeek

Google's VOIP Challenge to Skype: 10 Reasons It's a Serious Threat - IT Infrastructure from eWeek

A NEWS analysis from eweek.com about google's new Voice-Phone calling services.It could pose a serious challenge to Skype. This challenge will take more effect in the US and Canada VoIP market, because google already offer free calling to this country for this year.

This analysis points out ten reasons, which could be serious threat forothers.

1. Its Readily available for Gmail user.

2. Its coming to more places (around the world).

3. Google voice service offer more option and services

4. Free calling in US and Canada

5. Google voice is becoming the One-Stop-Shop.

6. Skype is well known, but there are lots of Gmail user around the globe.

7. Future potential

8. Its already tested that the Voice Quality is outstanding.

9. Gmail already has voice chat (GTalk).

10. Simplicity regime supreme


Click here to read more about this analysis article.


Related Topic:

Googel Introduce Integrated Voice Service in Gmail

Saturday, August 28, 2010

Make Free Phone calls to Bangladesh, India using Best Mobile Dialer...


Do you want to make free phone calls to Bangladesh, India, Pakistan, Nepal or other countries around the world using our Best Mobile dialer and PC dialer ?
Most probably, the answer is YES !

If so, then start now; mail us today with your contact details using the following format:

Your Name:
Your Address:
Your Occupation:
Your Contact No:
Others (if any):


If you have really interests to know about VoIP (Voice over Internet Protocol), Route, Mobile Dialer, White, Grey, E1, Wholesale, Reseller etc…  just go for it. Mail us today; ask your question, tell us about your interests and we will provide our best answer to you.

This opportunity to make free phone calls using Mobile Dialer, PC dialer, Phone dialer through world wide, is mainly for new users who wants to know how to make VoIP phone calls using a Mobile Dialer or PC Dialer and for those also who wants to do some VoIP business with their own reseller system. If you are one of them mail us today.

After getting your mail, we will reply you within a short time. The replied mail will include:
  • Mobile Dialer and PC Dialer download link
  • Operator Code / Brand PIN (if applicable).
  • A Test PIN

This test PIN will include some balance using which you can make Free Phone Calls to your favorite destination. You can receive a test PIN by which you can make 20, 30, 50, or more free minutes to make free phone calls to Bangladesh, India, Nepal, Pakistan, Malaysia or other countries.

Please mention your choice or other related question in others section. Like White / Grey / Gray / E1 / Wholesale or others.

Currently we provide reseller or Card for the following Operator Services:
  • G Ring
  • Apollo
  • S1 Voice
  • Saymoom Telecom
  • DTL
  • BD Express
And more…………..


There are lots of other ways to make free phone calls worldwide. Actually you can make PC to PC calls which is also called Voice Chat for free using Gmail, Yahoo, Skype, Nimbuzz, and lots of others Services.

But if you want to make free phone calls online directly to other Cell phone or Land phone numbers then you should spend some money from your pocket. I can give you up to 20 minute, 30 minute, 50 minute, 100 minute…….

But it will be very much harder for me to give you more because it costs for me.

I will provide latest information always how to make free phone calls using best mobile dialer around the world, just keep reading my blog. If you have any questions, if you need any type of solution, if you need reseller and other VoIP related information, feel free to mail us at:


Related Topic:

Free call to IndiaFree call to Pakistan

Friday, August 27, 2010

Google introduce integrated Voice Call Service in Gmail



Google has introduced its new voice service in Gmail, its  about VoIP calling. Any Gmail user can make and receive VoIP call directly from his/her inbox. If you didn’t already get this service message in your inbox, don’t worry its coming.

To user Google VoIP Voice Service from your inbox just install the free Voice and Video Chat plugin  to your Gmail account. Follow the simple instructions graphically shown to you. You can start calling within a minute.

This new voice service is totally integrated with Gmail account. It provides free VoIP call to anywhere in US and Canada and at a flexible and low rate to other countries worldwide. Google offer this free service for US and Canada will be continue this year and other country rates will be low. Robin Schreibman (Software Engineer at Google) said, in future the rate to Germany, France, UK, China, Japan and other countries will be as little as 0.02 US $. This is a new and steps for google in VoIP.




To start dialing in any phone number from Gmail is very simple and easy. When you got enabled this service in your inbox just follow the simple steps:
 
Click – "Call Phone" (A dialing box will appear)

If you already have a chat list, just dial a number or enter a contacts name. If you have already Google Voice number, this number will display as the outbound caller ID. 




Select the country you want to make VoIP call and type the rest numbers of the Cell Phone or Land Phone number and make VoIP Phone call...

Gogole has taken on the rivals like Vonage, Skype, and others. Let see who can survive in the voice service.

See Google Voice Blog for more details.


Other VoIP related Topics:

Thursday, August 26, 2010

Make Free Phone Calls Online to more than 70 countries from OziCall



Make a Free Phone Calls Online to more than 70 countries using OziCall. The process is easy, just call an access number provided by OziCall, follow the instructions, dial your destination number and start making a free phone call from your mobile or computer. The access numbers provided by OziCall are Australian mobile numbers.

Using OziCall you can make free phone calls online directly to cell phone numbers and land phone numbers more than 70 countries worldwide:

Andorra
Argentina
Austria
Bahamas
Bahrain
Bangladesh
Belgium
Brazil
Brunei
Bulgaria
Canada
Chile
China
Colombia
Croatia
Cyprus
Czech Rep
Denmark
Dom. Rep
Estonia
Finland
France
Georgia
Germany
Greece
Guam
Hong Kong
Hungary
Iceland
India
Indonesia
Ireland
Israel
Italy
Japan
Latvia
Lithuania
Luxembourg
Macau
Macedonia
Malaysia
Mariana
Mexico
Netherlands
New Zealand
Norway
Panama
Peru
Poland
Portugal
Puerto Rico
Reunion Isl
Romania
Russia
San Marino
Jeddah
Singapore
South Africa
South Korea
Spain
Sweden
Switzerland
Taiwan
Thailand
Turkey
U.K.
U.S.V.I
USA
Vatican City
Venezuela



Now the question is it really free ?
 

While making international phone calls, the international leg of the call is free but when you will be billed only for calling the local access numbers provided by OziCall.
 

You can talk like normal calling, the voice quality is very good, when you make phone calls from 3 or Telstra network, and it include any MVNO operators that running under these providers.

If you want to make free phone calls from Vodafone network, than read this.

The overall process to make free phone calls from computer or mobile is easy. Just follow the following steps:

  1. Make sure that your country is included in the list of OziCall. For detail country list go to their Countries Page  
  2. Dial the access number provided by OziCall from your cell phone.
  3. Follow the instruction, when the server promote your destination number provide it.
  4. Wait until the you call is connected
  5. Enjoy talking your free international phone calls.

The process is really easy. Feel free to talk as long as you want. OziCall does not charge for making a phone call and it wouldn’t interrupt your call with any kind of ads, and you don’t need to worry about the tine limit.

To know more about how to make a free phone calls online using OziCall go to their site and start calling today: OziCall



Related Topic:

Wednesday, August 18, 2010

VOS - Carrier-class VoIP Operation Support System



VoIP Operation Support System – VOS 3000/2000 is a system designed for carrier class operations. It provides the following solutions:
  • Exchange Rate Management
  • Card Management
  • Package Management
  • Account Management
  • Gateway Management
  • Phone Management
  • Softswitch Management
  • Number Management
  • Data Query and Web self-service system
  • System Management
  • IVR and User Management

This system also includes an add-on modules like the Global Card Business System (support 10 million cards) and extreme media proxy. VOS3000 Softswitch supports H323 / SIP / SIP-H323, with a capacity of up to 10,000 concurrent calls. 


System Performance

  • Stand-alone performance
  • For Single Server total number of calls >= 10,000 (signaling forward)
  • Medial forwarding capacity of Server >= 2,000 (B/W limit 100 Mbps)
  • Call per second >= 1200 CPS (4,320,000 BHCA)
  • Supported protocol: HTTP, SIP, H323, TCP / UDP, etc


Important features of VOS 3000 System:

  • For VoIP (Voice over Internet Protocol) operators
  • Real-time control of soft-switching platform
  • Routing by (ASR) and Least Cost Routing (LCR)
  • Full support for Voice calls, Fax and Video call support.
  • Gateway group, preset profit rate
  • Overload protection, load balancing and redundancy backup mechanism.
  • Voice codec priority settings and Softswitch platform Control.
  • Dual Hot-backup mechanism
  • Multi-server distributed placement and centralized management
  • Connect analysis, interruption analysis and Excellent Compatibility
  • Multi-gateway encryption standard
  • Real time monitoring of Call performance
  • Area code automatically added by phone.
  • Global Card management business, IVR service helps to achieve one million global cards.
  • Various Package features, providing daily/weekly/monthly basis or annual rent and other available methods
  • Efficient media access transponder modules, forwarding up to 2000 concurrent calls.
  • Value added business expansion
  • A better after sales and technical support service
  • After rigorous testing, VOS system provides stable and reliable high-performance carrier-class services.
Increasing value-added service modules bring more rich business modes to the customers.
After thorough testing, the VOS3000 VoIP softswitch proves to be a constant and dependable high-performance carrier-class system.

For more details:
http://www.vos3000.com/



related posts:


Tuesday, August 17, 2010

Class 5 VoIP Softswitch - Total VoIP Business solution



For successful implementations of various VoIP related services, VoIPSwitch is a complete IP telephony platform that integrates all the elements into one comprehensive solution.
 
Functional specification of Hosted SoftSwitch:
A SoftSwitch is the main elements of the platform. It merges the functionality of the following elements:
  • SIP Proxy
  • SIP registrar
  • H323 gatekeeper
  • H323 switch
  • SMS gateway

 
Supported protocols for VoIPSwitch:
  • H323 v.2 (H225 v4 and H245 v7) with or without FAST START.
  • SIP (RFC 3261)
  • SMS through HTTP, SIP and SMPP.

 
Main characteristics of VoIPSwitch include:
  • Transparent and Simultaneous support for H323 and SIP protocols (sip <->h323 translator)
  • Support various types of proxy methods like, Signaling proxy, Full proxy (with RTP-proxy) and others, possibility of selecting a proxy method per destination, route or per client.
  • Full interoperability with industry standards compatible VoIP equipments (Switches, Gateways, Terminals and ATA etc).
  • Bi-directional NAT support both for H 323 and SIP equipments.
  • Routing based on Priorities per routes, Prefixes, depending on allowed voice codec per destination
  • Highly developed routing system (support for virtual prefixes allowing to create separate dialing plans to handle different groups of accounts)
  • Rerouting Support (failover), configurable end reasons initiating failover, support based on priorities.
  • Advanced algorithm to handle traffic being evenly distributed according to defined percentages for multiple routes, which is called Load Sharing Support.
  • Least calls routing and Quality routing
  • Internal numbering plans support.

 
VoIPSwitch supports various authentication methods:
  • By caller ID (ANI)
  • By IP address
  • By SIP credentials
  • By H323 ID
  • PIN


Call setup data modifications method for Clients, for destination in the dialing plan: 
  • Modifying dialed number, adding prefixes or suffixes, wild cards, maximum or minimum number length.
  • Modifying caller ID / SIP display.
  • Defining allowed and primary codec for clients and terminators
  • Codec auto negotiation
  • Import and export dialing plan and accounts from and to .excel or .txt file.
  • Strong database using MSSQL or MySQL.
  • Scalability supported cluster configuration with numerous VoipSwitch instances running connected in a cluster, sharing SQL database server, thus increasing performance by dividing the traffic among multiple servers while retaining central point of management with one, main IP address for clients (load balancing)
  • Redundancy support for seamless traffic handover in case of the main server failure, the service allows for controlling availability of particular ports (SIP, H323 listeners) real-time SQL data backup.

IP Telephony features:
  • Hold function
  • Music on hold
  • Do not disturb
  • Call transfer: blind and attended
  • Follow me / Find me (based on caller ID of incoming calls), sequential or ring to all
  • Voicemail boxes with personalized voice greetings for different caller IDs
  • Hunt / Ring groups


Unified messaging:
  • Voicemail to email with attachment (Mp3)
  • Voicemail notification via SMS or email
  • Voicemail transcription to SMS
  • SMS forwarding (internal SIP SMS forwarding to other GSM numbers)

Monday, August 16, 2010

DigiNet - Provide high quality VoIP related Consultancy



DigiNet is Hong Kong based VoIP (Voice over Internet Protocol) solution and service provider and a sister concern of Digital World Communications Limited. DigiNet (Digital Network Communications) provide VoIP softphone,  VoIP billing solutions, server monitoring system etc.

DigiNet is one of the fastest and leading VoIP solution providers in the market. It provides end to end solutions from wholesale to end users, small, medium and big VoIP providers. It provides equipment leasing and VoIP SoftSwitch applications to carriers and enterprises who are implementing VoIP networks and high quality VoIP business related constancy. 


DigiNet helps service providers minimize the risks of entering a new market by providing you security, technical training, support and a reliable billing platform.

DigiNet always try to keep its customers on the cutting edge of the rapidly growing VoIP technology. The company has business partners in USA, UK and Hong Kong and all hardware are placed at secured co-locations.


 

Digital Network Communications (DigiNet) has:
  • Direct connection to leading ISP backbones and IT hosted leaders.
  • Billing based upon aggregate bandwidth transfer.
  • Fully managed dedicated servers
  • 99.99 % uptime guarantee for network
  • Fastest setup.


DigiNet offers various business packages. Current Business Packages for large and medium VoIP service providers:

  • VoIP SoftSwitch rental solution
  • Hosted Linux Server Solution
  • Hosted Windows Server Solution
  • End user CDR solution (customized)
  • Customized Reseller Management
  • Relax to operate VoIP Switch
  • Customized PC to Phone Dialer
  • Call Back platform – SMS / SNI / PIN
  • Calling Card by interactive Voice Response (IP IVR)
  • Mobile VoIP – Symbian Softphone
  • VOS3000 – VoIP Operation Support System
  • Google Android SoftPhone – Mobile VoIP
  • Billing and Provisioning
  • SoftSwitch Class 5
  • VoIP SoftSwitch Overview


Please visit for more information about VoIP business and other related topic:
http://diginetcom.net

Friday, August 13, 2010

iTel Call Back Dialer – Experiences like regular call from your Phone



Reve System is one of the best VoIP solutions provider. It has various types of dialer like iTel Mobile dialer, Call through Dialer and Call Back Dialer. 

In this post I am going to discuss about various features of iTel Call Back Dialer. Using this user can experiences like a regular phone call. User can get hassle free call back service using this dialer. 


Some important key features of this call back dialer are:
  • Opportunity to create your own Brand as a service provider.
  • Direct integration and calling directly from phone book.
  • Details call log information on screen.
  • Great sound quality.
  • User can get fully automated and hassle free call back service.
  • Integrated with existing Call Back System
  • Support most common brand handset like Nokia, Samsung, LG, Sony Ericsson, iPhone, Blackberry, HTC, Motorola, BenQ, Siemens and many others.
  • Dialer can work on background or keep hidden.
  • Software can be distributed easily through WAP, Web and OTA.
  • Various routing choice can be implemented like all the local calls. These calls will go through the local telecom service provider and all the international calls will go through the VoIP Solutions Provider around the world.


Some advantages of iTel Call Back Dialer:
  • Takes shorter time to connect.
  • Easy user interface.
  • No need to enter long PIN.
  • Cheap call costs.
  • No need to memorize numbers because user can make call directly from their phone book entry.

 

Thursday, August 12, 2010

gPlex Wholesale Voice Carrier Route – TDM and SIP


Setting up a small / medium VoIP business is not so hard actually. Many VoIP services provider are available here in Bangladesh right now. The market here is very competitive and price sensitive.

After opening a brand with your own Softswitch and Dialer, you need good quality voice carrier route for several destinations. If you want to create a better brand of your own you need a better carrier route.

Genusys Inc. (www.genusys.us) offer a good quality voice carrier route to A - Z destinations. Their direct route for international voice traffic has a good ASR (average success ratio) and ACD (average call duration). Genusys Inc. provide both TDM and IP telephony routes.

Some important features and advantages of gPlex (genusys) Voice Carrier Services:
  • Provide both TDM and IP network
  • Interconnection with regional and major partner around the world, provide wide network coverage.
  • More than 30 destinations are directly connected including many countries in Asia, Europe, Middle-East and USA.
  • Voice termination is uncompressed for some TDM destination.
  • Our IP telephony and TDM are integrated which support, C7 signaling with TDM-based and SIP/H 323 with IP-based voice transport.
  • Scalable and reliable infrastructure provide unlimited no of voice interconnect through POPs in UK and USA.
  • G.711 and G.729 codec supported.
  • Open CLI premium quality wholesale routes for many destinations.
  • 24 x 7 hours NOC support.
  • Both IP and TDM carrier for A to Z call termination.

gPlex (Genusys inc.) is one of the best wholesale voice carrier route provider in Bangladesh. To setup a termination account and see various packages of Genusys wholesale Voice carrier services browse:

iTel Call Through Dialer provide hassle free Calling Card use

Use calling card service by iTel call through dialer is more easy and reliable.

If your current calling card services take longer time to connect, not user friendly interface, the PIN number is long. If you can’t access your contact number directly from your cell phone contact list and you need to write down the numbers or memorize it, mistakes for entry and paying for delay. Higher call cost.

To avoid this kind of hassles, iTel call through dialer is the man solution for you. It will give you an experience of regular calling like your cell phone. This dialer is pre configures with coupled PIN and user does not need to enter same PIN every time. 


iTel call through dialer is best in the market for some of its important feature:
  • You can make your own brand as a service operator.
  • User does not need to remember the PIN number. 
  • PIN numbers are pre-configured, so client does not need to enter long PIN. 
  • User can dial their phone number directly from cell phone contact list. 
  • User satisfaction 
  • Lost cost comparatively to others. 
  • Call log and balance display on screen. 
  • Automated system calling card usage. 
  • Web, OTA, WAP can be used to distribute the software. 

    Most common branded handset like Nokia, LG, Samsung, Siemens, Motorola, Sony Ericsson, BenQ, iMate, Pantech, T-mobile, Vodafone and some others are supported. 


    How to use the Call through Dialer: 
    • First download the dialer from provided link and install it on your Handset. 
    • After installation run it, and it will ask you to configure Access number, PIN number and own phone number. 
    • You will see balance on the display, now enter your number or select from the phonebook 
    • Press call and start talking.
     

    Click to know more about iTel Call Through Dialer.


    Read other topic related to this post:
    gPlex Mobile Dialer for Symbian S60 OS based habdsets
    Pioneer VoIP Mobile Dialer

    Sunday, August 8, 2010

    iTel Mobile Dialer – A pioneer VoIP solution in Bangladesh


    A Mobile dialer is a software that user can make VoIP phone calls directly to other Cell phone or PSTN numbers from his/her Cell phone using internet. iTel Mobile Dialer is one of the best Mobile Dialer in Bangladesh.

    The main specification of this mobile dialer is its best performance in low bandwidth and behind NAT. SIP protocol of this dialer handled by its SIP stack. For its NAT and firewall traversal using STUN supporting, clients can make VoIP phone calls behind the firewall using it.

    It is very much compatible with almost SIP (Session Initiation Protocol) switch and Gateways. This mobile dialer can enhance the quality of sound using appropriate codec because it has competent implementation of Jitter Buffer which helps to the voice running smoothly. 


    Some important features of iTel Mobile Dialer:
    • User friendly interface designed that user can operate easily.
    • SIP (Session Initiation Protocol) signaling support.
    • It supports G711, G729 and GSM codec system to sending audio data.
    • This mobile dialer works on private IP or behind NAT.
    • Using IVR customer can hear credit balance anytime.
    • For internet connectivity iTel Mobile Dialer supports WiFi, 3G, GPRS and Bluetooth.
    • This software runs on windows 5 and 6 or Symbian OS based mobile Handset.
    • This dialer is very much easy to install and use. Client can download the software directly form the given IP. After installation he does not need to configure it. He just has to input PIN.
    • Any kind of hosted Softswitch that support SIP, iTel mobile dialer can use efficiently with those.
    • This dialer has been designed specially keeping in mind the needs of operators.

    iTel Mobile Dialer is not just a software that can help user to make VoIP phone calls, operator can make a brand of their own service as an operator.


    Due to availability of GPRS, 3G and WiFi service in many countries, calling through IP (VoIP – Voice over Internet Protocol) is going to be more popular day by day. For those who want to start their own VoIP business today than iTel Mobile Dialer can be the easy and efficient way to do that.
     

    Friday, August 6, 2010

    gPlex Hosted SoftSwitch - for Small or Medium VoIP Business



    gPlex SoftSwitch is a comprehensive and scalable SIP SoftSwitch which enables establishment of secure and reliable Voice over Internet Protocol (VoIP) networks. gPlex offers flexible routing, centralized authentication and reliable billing. It is comprised of RTP Proxy, SIP Signaling, CDR, NAT Traversal, Billing, IVR, Voice Logging, and High Availability (HA).
     
    A standard gPlex hosted SoftSwitch system can handle up to 2,000 simultaneous calls and it is easily scalable to 40,000 simultaneous calls with carrier edition. The High Availability (HA) feature provides robust redundancy to ensure that calls never get interrupted in the event of power failure or hardware. Using gPlex SoftSwitch, service providers can benefit from improved call completion rates, less revenue leakage, and improved network security and availability.

    Features of gPlex Hosted SoftSwitch:

    • SIP & H.323 protocol
    • User-friendly web interface
    • Web panel for Master user, User & Sub-user
    • 99.9% uptime
    • Redundant IP network in hot-standby mode
    • Redundant power supply with duel power drop
    • Intelligent routing
    • SIP / H.323 protocol conversion
    • 24x7 support and free upgrades of new releases
    • Free training program for engineers, assistance with initial system configuration


    High Availability (HA)
    gPlex - HA module, offers virtually 100% uptime which ensure high network availability. It can be configured with gPlex - HA module on a backup server which provides true reflect of master gPlex SoftSwitch. HA server will take over all running calls within fraction of a second in the event of hardware or power failure. Thus user of gPlex-HA will never misplace billing.

    High Scalability
    gPlex SoftSwitch is offered in standard and carrier editions to meet the capacity needs of rising and established telecoms and service providers respectively. The standard edition of gPlex SoftSwitch supports up to 2,000 concurrent calls and carrier edition can handle up to 40,000 concurrent calls in full RTP proxy mode.

    Multiple Routing
    gPlex SoftSwitch has central and local routing capability. Central routing offers Least Cost Routing (LCR), ASR Routing, Priority Routing, Preferred Routing, and Route Fail-Over features. gPlex enables providers to select the most money-making and best quality routes for individual calls and helps increase call completion rates that translates into high revenues and earnings.

    Enhanced RTP Proxy
    gPlex SoftSwitch performs RTP Proxy for all calls and monitor each RTP packet to detect one way calls, hung or dead  calls. On detection gPlex terminates those types of calls immediately and sends messages to billing server to ensure accurate billing.

    Topology Hiding
    gPlex SoftSwitch can securely separate their VoIP networks from the outside IP world. It can act as a traffic proxy between Termination network and Origination network, and it prevent outsiders from seeing the true topology of the protected network.

    Auto Route Switching
    gPlex SoftSwtich offers route switching capabilities to ensure higher call completion rates and reduced revenue loss. It monitors all endpoints in real-time and automatically pulls off routes in the event of quality parameters like ASR, ACD, and PDD falling below acceptable values preset by system administrators.


    For medium and small size VoIP telephony service providers and resellers who want to enjoy carrier grade VoIP SoftSwitch with limited budget. gPlex Hosted SoftSwitch started from 100 channel to 1000 channel for them. This is very user-friendly web-interface for flexible routing, reliable billing, and real-time monitoring. User can handle everything using cell phone data network.

    gPlex Hosted SoftSwitch offer various Packages. For more details visit the following sites:

    www.gplex.us
    www.gplexdialer.com
    www.gplexswitch.com
    www.genusys.us


    related posts:
    gPlex Mobile Dialer for Symbian S60 OS based habdsets
    iTel Mobile Dialer
    Call through Dialer - for Hassle free Calling
    iTel Call Back Dialer
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