FAQ

Welcome to our Frequently Asked Questions list!

We have tried to organize the many questions you may have and our answers in subheadings, which you can access below. On this page you can find answers to the most general questions.

General Questions

How do you compare to Skype?

Our service is similar to Skype in that it travels over the Internet, but differs in that it is an open standard. This means that any equipment you might invest in with Skype cannot be used elsewhere once you discontinue the service. Any equipment to be used with our service (called SIP) can be used with a multitude of providers. Competition in the SIP space is likely to bring prices down further, whereas Skype can increase prices at whim, as their customers are unaware of competing offers and cannot switch without losing their hardware investment (if any).

Do you pass Caller ID?

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

Can I fake my Caller ID?

We prefer to let our customers decide for themselves what Caller ID information to display on their outgoing calls. For instance, if you prefer to receive all incoming calls on another line, you might want to set your Caller ID number for your outgoing calls to display the phone number of your incoming line. However, your caller ID must be the same as your line if you ordered e911. By default, we set the Caller ID number to match your actual phone number, but if you would prefer to control it from your side, please let us know and we will remove it. The Caller ID name is entirely controlled by the your SIP (VoIP) client. Please see the Setup pages for details. Note that it is generally a bad idea to set it to show your Username.

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.

Technical

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 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. Note that GSM may sometimes be disabled as it provides inferior voice quality. If you specifically require it, please let us know.

Codec Bandwidth Specifications

What are the supported protocols?

We support SIP and IAX2. We recommend SIP.

What type of equipment is required?

Any type of software or device (unlocked) which support SIP or IAX2. This includes 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. You can use any hardware or software that supports at least one of the 2 protocols we offer.

Where are your servers located?

We have VoIP servers located in Houston, Dallas, Los Angeles, New York, Montreal, Toronto and London, UK. As our customer, you may use the server of your choice at any time.

Support

In what language can you support me?

English and French by phone, email and live chat.

What are the support hours?

Our live support hours by phone are from 9 am to 5 pm EST. However, we try to accommodate your request via email beyond this time interval.

I have a problem with my ATA can you help?

We always do our best to have you up and running in no time. Your satisfaction is our number one priority. We'll do the best to help you no matter the type of equipment you are using. In the case of Asterisk, we can have a technician enter your server and configure the basics for you (trunk and extension).

If your phone is not working, you might want to follow the following list:

1. Is your Internet connection working? It probably is, otherwise you would not be able to read this. (Thus you might want to save this page on your computer.)

2. Are you able to use the service from another device? For instance, if you primarily use an ATA, can you also make phone calls from your computer softphone or from your smartphone / iPod (call the echo service at 4443 so that you don't incur charges)? If yes, then the problem is with your device.

3. Sometimes the problem has a simple and obvious solution - such as a phone of the hook or a malfunctioning phone.

4. If the problem is with your ATA, consider rebooting the device. The easiest way to accomplish this is to disconnect power to the device by unplugging the power adapter, wait at least 30s, then reconnect it.

To minimize the chances of problems, consider investing in a UPS (uninterruptible power supply) and unplugging your line from the demarcation point. It is also a good idea to protect your ATA configuration with a password.

What is "Demarcation Point" and why is it important?

If you are using your ATA as a replacement for your regular phone line, you may want to disconnect your home wiring from the Bell lines. This is much easier to accomplish if you have a "demarcation point jack" installed, such as the one above (disregard the thick white cable, it is unrelated). Disconnecting your line is as easy as unplugging the jack. This ensures that you will not broadcast your conversation on the Bell (i.e., incumbent) trunk; also, you will not be receiving any interference from possibly malfunctioning Bell equipment.

Do you offer Asterisk consulting and programming for custom solutions?

Yes we do. We've gone as far as programming call shop interfaces and setting up multiple load-balanced asterisk servers for call centers. Contact us.