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Finances and General



General and Finance Questions

Do you support 911/e911

We use e911 (Enhanced 911) and are 100% compliant with FCC and CRTC and cover 100% of USA/Canada.

Is there a volume commitment?

No. There’s no volume commitment and you can spend your credits at the pace you want. Your balance never expires.

What payment types are accepted?

We accept PayPal, Wire Transfers and Credit Cards.

What is the minimum deposit amount

Minimum deposit is $25USD

Do you have DIDs

At the moment, we have DID numbers available for USA in 49 states and 4 provinces of Canada. We also have USA toll free numbers. We have international DIDs available in 30 countries.

What’s the difference between your value and premium route for USA/Canada.

Value is the greatest price we could find for Canada. This permit us to offer Canada starting as low as 1/2 cent per minute, depending on location. Our value rate is of the best quality we could find and targeted at end users and resellers who are looking for the best wholesale prices without any volume commitment.

Premium has a flat rate of 0.01 for both USA and Canada and is routed through established and renowned tier-1 carriers always delivering the same level of quality, at a price that is a little higher than our value option. This price is intended to end-users with critical business calls or resellers who are willing to pay a little more for assured quality, but still less than other US48 tier-1 providers.

Which one to use? Best to do is to try both (value and premium) and settle for the one that best suit your needs.

Can I check call details online?

Yes, you can access your Call Details report online. It is updated every 60 seconds.

Do you offer downloadable call detail reports?

Yes, from the customer area, you can download CDR in CSV, Excel, XML and SQL format. We also have a printable version of the cdr.

How do I know if a rate has changed?

If we need to change rates on some destination, including raising or lowering a price, these changes are made on the 2nd day of each month at 7PM CST. When you download our rates, there is a column indicating old rate and the date of the change. You can also use our XML API to check rates.

What are the billing increments?

USA and Canada: 6 seconds initial, 6 seconds increment
Mexico: 60 seconds initial, 60 seconds increment
World: 6 seconds initial, 6 seconds increment

What does billing increment mean?

It’s the way we calculate our rates in order to bill your calls. For example, if you call USA for 10 seconds, you will be charged for 12 seconds (2 x 6 seconds since this is a 6 seconds increment call) of a minute, not the whole minute.

Where are you located?

The main office is located in Montreal, Canada and we also have an office in Merida, Mexico for our South America market. Credit Card charges are made in US Dollars (USD) Currency by Swiftvox Inc via Internet secure.

Do you accept predictive dialer call center traffic? (Dialer/Telemarketing)

We do not accept dialer traffic at this time.

Do you accept inbound call center traffic?



Do you pass callerid?

On our premium route, All US/Canada destinations will receive proper callerid. On the value route, we can not guarantee callerid will pass 100% of the time but it should in most cases.

How should I dial calls

For your convenience, we support 3 standards. 011 Prefix, 00 Prefix and direct country code. For example, to call to UK you can use 01144+number, 0044+number or 44+number. To call to USA you can use 1+Area Code+Number or 001+Area code+number.

Do you offer termination in every country?

Yes we do provide termination in every country. We are an A-Z VoIP termination provider. We always do our best to find and keep quality working routes. If you were to experience some problems with a particular destination, let us know and we’ll make everything possible to fix the problem.

What are the supported codecs?

G.711 (μ-law / pcmu) , G.729 and GSM.

What codec should I use?

For best sound quality, if you have the bandwidth available, we recommend G.711u. However, you can still maintain an excellent voice quality and lower bandwidth usage with codecs like G.729.

Codec Bandwidth Specifications

Bit Rate
Nominal Ethernet Bandwidth (Kilobits)
64 Kbps
87.2 Kbps
8 Kbps
31.2 Kbps
13 kbps
29.2 kb/s

What are the supported protocols?

We support SIP and IAX2.

Can you give me some details about these different protocols?

“The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.” (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 from the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is widely used as a signaling protocol forVoice over IP, along with H.323 and others.

Source and more information: Wikipedia

IAX is the Inter-Asterisk eXchange protocol used by Asterisk, a dual licensed open source and commercial PBX server from Digium and other softswitches and PBXs. It is used to enable VoIP connections between servers, and between servers and clients that also use the IAX protocol.

IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favor of IAX2

Source and more information: Wikipedia

Source and more information: Wikipedia

What protocol should I use

We recommend SIP. However we fully support IAX2

What type of equipment is required?

Any type of software or device (unlocked) which support SIP or IAX2. Including free softphones such as X-lite, free open source PBX Asterisk, Voxalot, Trixbox distribution, VoIP ata’s and VoIP phones, Hardware CISCO VoIP switches etc. Basically, any hardware or software that support one of the 2 protocols we offer.

Do you have an API for rates?

We do have an XML API. You can read more about it here.

Where are your servers located?

We have VoIP servers located in: Atlanta, Chicago, Dallas, Denver, Houston, Los Angeles, New York, Seattle, Tampa, Montreal, Toronto and London. As a customer of FringeTek, you may use the server of your choice at any time.

What kind of equipment do you use?

Our VoIP servers consist mainly of OpenSER and Asterisk running on Redhat Enterprise 64 bits OS.


In what language can you support me?

English, French and Spanish by phone, email and live chat.

What are the support hours?

Our live support hours by phone are from 9am to 5pm EST (Eastern standard time). However, we do have staff attending your email requests beyond these support hours. Check at the bottom of the page to check current EST time.

Do you have a phone number?

Yes we do. USA and Canada toll free at 888-850-2447. Worldwide, you can use our local Falls Church, Virginia, USA number: (1) 703-542-9960.

I’ve a problem configuring my device/server, can you help?

We always do our best to have you up and running. Customer support is an important part of our philosophy. We’ll do our best to help you no matter the type of equipment you are using. In the case of Asterisk, we have detailed configuration guides with screenshots in our Wiki, along with samples for more types of PBX.

Do you have configuration examples?

We do have configuration samples for Asterisk, FreePBX/Trixbox, Voxalot, Generic LinkSys/Sipura ata/phone. We’ll add more configuration examples in the future. Configuration to use our service is very straightforward with most softwares/devices. If your equipment is not included in the configuration examples and you have difficulties setting up VoIP, we’ll be glad to assist you.