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June 25, 2010 / MyDivert

Complete VoIP Service was established almost 2 years ago as a Next Generation long distance call provider using Voice over Internet Protocol (VOIP) technology.

The availability of DID (Direct Inward Dialing) numbers for VoIP has allowed us to offer communication solutions worldwide that a few years ago could have only been dreamt about. today, as an international communications provider, strives to maintain the highest quality service for all our business and residential customers. We pride ourself in providing high quality VoIP services both domestic and international.

Complete VoIP

Complete VoIP

Among the many features of are:

Premuim call routes & voice quality
Cheapest VoIP calling rates
Premium customer support
DID numbers from over 50 countries
DID numbers for your own pbx or asterisk server
Direct DID to IP SIP peering
24 Hour trial period on every DID number
ATA and Asterisk compatible
Unlimited Free Incoming Calls
Low per monthly rates
Instant Number activation

Services for Call Centers provide DID numbers and VoIP services to run and setup a call centre at the lowest rate possible in the market. Providing geaographical VoIP numbers and 24 hour In-bound and Out bound traffic to A-Z routes.

IP Phone Users offers a comprehensive service for IP phone users. You just need to connect your Ethernet cable with the IP Phone, configure it and you are ready to call any where in the world. Receive inbound calls with your purchased DID number.

PC2Phone Users
Pc2phone technology is a technology that allows PC users with a softphone dialer to make long distance and international calls to anywhere in the world. Customers can recharge their accounts and purchase call credit from their account area.

Softphone software is available free (Google: SJPhone or X-Lite) to download and install on their computers. Users can then make calls to anywhere in the world – all they need is a sound card, speakers and a microphone on their PC. They can also log in anytime to to check their account balance and call history.

International Call Services for Business
Many of our customers use a virtual phone number as a great way to create a local presence even without a physical office in the country or city where the phone number is located. Whether you need to forward your callers to the U.S. or any country you are visiting, a virtual phone number can be your solution.

Call Forwarding Services
At you can buy a Local Telephone Number and have your calls forwarded to any other phone number, worldwide. You can buy a private phone number in any country where numbers are available and direct callers who dial that number to ring your mobile phone, business line, home, or any other phone internationally.

For further information email sales[at]

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June 25, 2010 / MyDivert

Making Calls With a SIP VoIP Device Or Softphone

Making Calls with a SIP VoIP device

You can also place outgoing calls with MyDivert, but you will need a MyDivert username and password in order to call via our network. So first sign up for your account. Calls can be made from a softphone on your computer, or from a normal telephone with an adaptor. The SIP settings you will require can be accessed from your user account area.

Receiving Calls on your SIP Phone or PC
If you wish to receive your incoming phone calls on your softphone or SIP device directly you can simply disable the call forwarding feature in your user account. Calls placed to your incoming VoIP number will then ring to your SIP phone or PC without diversion.



Using a Softphone on your PC

A softphone is a small application that works with VoIP technology. It enables you to make regular phone calls from your computer. The interface normally looks and behaves like a traditional phone’s dialling pad. A softphone can be used with your MyDivert account in a similar way to skype, except our call charges are lower. Many softphone applications are available as free downloads.

Examples include: X-lite, ExpreeTalk, and SJ Softphone

Using a Telephone Adaptor

An analogue terminal adaptor (ATA) is a piece of equipment which allows you to use a normal telephone with a VoIP service. The device acts as a bridge between your existing phone and your broadband router/modem. This allows you to make calls as you would on a normal phone line, using a standard telephone plugged into a small adaptor. Examples include: Linksys PAP2

Using Custom SIP/IAX Settings for you Incoming Phone Numbers

The user account area also allows you to direct purchased DID numbers to any sip provider of your choice. Simply choose the ‘custom sip’ or ‘custom iax’ option from the phone number management area. Please check with your sip providor that incoming SIP calls are allowed. Please contact us if you require our IP addresses to authenticate channels.

June 25, 2010 / MyDivert

Virtual Telephone Numbers From Poland – All Area Codes Now Available, one of the web’s most popular call forwarding services today announced the introduction of complete geographical coverage for Polish VoIP virtual numbers.

Now you can have a local phone number from any of the 50 local area codes in Poland, and have your family, friends, or business contacts call you – dialing in at local rates.A virtual telephone number allows you to receive calls on a landline, mobile phone, VoIP telephone, softphone on a PC, or send calls to your own VoIP provider or pbx where ever you are in the world!

The announcement has been described as a ‘breakthrough in communication for Polish migrant workers and their families’.There are estimated to be over 2 million Polish nationals working abroad in the EU alone, with 700,000 living in Britain and an estimated 200,000 in the Republic of Ireland.

Joel Driver, CEO of, said today, “Our partnership with the leading national provider is great news for all Polish people working or living abroad. The new partnership enables us to pass on the services to our users at a fantastic rate. Anyone can have a local number in Poland now for just €2.95 per month. This is going to help thousands of families to keep in contact. That has to be a good thing.”

“We’re very excited to be the first service to partner with a Polish voip provider and provide this level of coverage.”, said Josh Stephens, technical director at “Integrating the Polish DID into our platform opens great communication possabilites for the individual as well as for business use. The ability to receive local calls without the need for a physical presence in the area is at the core of our business. Our server grid is located in Germany and the Polish DID are mapped directly to us. This enables us to provide the highest quality calls for customers with at a competitive price.”

The service at has no set-up charges and allows for unlimited incoming calls. If calls are forwarded to a landline or mobile phone then a small call forwarding charge applies. With a local telephone number on your phone, or mobile, your callers pay Local Call Rates when they call you. – Polish VoIP Numbers

June 15, 2010 / MyDivert

VoIP Equipment and Providers – Things to Consider

Today, VoIP communication is commonly used to make calls online. The use of VoIP has become so popular because it allows users to make long distance and international telephone calls over their internet connection at a fraction of the cost associated with PSTN and mobile network providers.

With virtual telephone numbers now available from websites, such as, the VoIP telephone can now be used to make and receive calls just as you would with any other traditional telephone. In order to do this the VoIP user can use software on their PC, which will work in a similar manner to skype, with a headset and microphone, or they can use a hardware device such as a VoIP telephone or ATA adaptor.

For VoIP to replace the traditional landline telephone, clear 2-way audio, reliable transmission of DTMF tones, and connection availability are important issues.
The most significant factor that affects the call quality and functioning of the VoIP client is the speed of the internet connection used for making and receiving phone calls. The overall quality and reliability of your VOIP communications are completely reliant upon the quality, reliability and speed of the Internet connection that it uses. The internet speed (or bandwidth) in both UP and DOWN directions is important. A minimum bandwidth of 128 kpbs is recommended for VoIP calls.

Another consideration is the means used to make and receive calls. In general a VoIP hardware device has several advantages, and produces far better results than a software softphone or dialer. Some of the advantages of the hardware based solution include:

# A VoIP hardware device remains available and connected when your PC is turned off.
# A hardware VoIP telephone is not impaired when your PC is under heavy load (running multiple applications, or games etc.)
# Audio for your hardware device is not interrupted by other applications and PC sounds (email arriving etc.)
# SIP Registration is more reliable with a hardware based device.
# Hardware VoIP telephones generally have far more configuration options available.

There are many brands, and types, of VoIP telephone available. These range from single line VoIP telephones to fully functioning pbx systems. Another good solution is the ATA telephone adaptor. These small adaptors, about the size of a cigarette packet, allow you to use a standard telephone for VoIP calls. If you are planning to use just one or two VoIP lines then a simple ATA adaptor will suffice. More information about ATA devices can be found by doing a web search for ‘PAP2 ATA’.

In order to make the most from a VoIP line you will need a Direct Inward Dialing (DID) telephone number. This will enable callers to reach you on your VoIP telephone by dialing a standard telephone number. DID for VoIP use are still not available for every country, and all area codes, in the world although the coverage is increasing every day. DID numbers for most of the major business cities of the world are available, and an internet search will soon through up a list of providers.

Direct Inward Dialing (DID) virtual numbers and VoIP work very well together when everything is configured correctly. As most DID/VoIP calls rely on more than one server or carrier working together discrepancies in configurations can result in unstable communication. Common problems include, one way audio, loss of DTMF tones, echo distortion, and dropped calls. These problems generally arise from incompatible codec translations, or other restrictions affecting one or more of the services being used.

If you are planning to use a VoIP carrier it is important to check that an incoming call from your DID is going to be received. You may need to determine the originating IP address for your incoming DID calls and check that calls from this IP are accepted by your carrier. If you are using an Asterisk server, or some other pbx, you may need to add this IP address to the configuration (as an IP authenticated peer).

Check that your DID provider is supplying free, or very low cost, incoming call minutes. A DID with unlimited free incoming minutes, or at least 5000 minutes per month, would be considered reasonable in todays market. Do your research and check the per minute charges if you do exceed the monthly limit.
Check that your DID, VoIP carrier, and PBX are using compatible codecs.
The G711 codec is the most basic, and widely used, telecommunication audio codec. The G711 codec commonly ensures the most reliable transmission of DTMF tones (required for ‘ring tones’ IVR answering systems), but does not use audio compression (causing distortion where internet bandwidth is restricted).
The G729 codec is also very widely used and produces excellent results for both DTMF and audio quality. However, with the G729 codec a licence is required by the PBX administrator if any transcoding is required.
It has been known for service providers to have an inadequate licence for the G729, restricted to a fixed number of simultaneous transcoded calls. In the circumstance users could experience loss of audio at times of high server usage. If all parties involved in the call route have G729 available then no transcoding is required and the calls will pass straight through.

The use of codecs will be negotiated between the servers involved in relaying your incoming call. The exact codec used will be determined by the codecs available on each server, and the preferred codec order list. Sometimes it may be necessary to experiment with calls to find the optimum configuration. – VoIP DID Virtual Numbers