How to be a Good VoIP Provider for Asterisk Users

There is more to being a good VoIP telephony provider for Asterisk users than simply offering the best balance of price and reliability. Usability is also very important. Here are some things which make for a service more convenient for Asterisk users.

Provide a Simple Dialplan

It is all right to allow your users to dial numbers in a variety of ways, perhaps to conform to local practice in your target market. But make sure to allow your customers to dial numbers in full international format, starting with the country code, if they so choose. I suggest you follow the European Union practice and designate 00 as the prefix to indicate that the number will be dialed in international format.

This is important because otherwise your users will have to program Asterisk to divide numbers into classes and select the appropriate prefixes. Lets take Vitelity’s dial plan as an example. Vitelity expects calls to the USA and Canada to be dialed as 10 digits, starting with the area code. (You can dial 1 first if you like.) But calls to other countries, including countries in the North American Number Plan must be dialed with a prefix of 011. So, a call to Jamaica, which would normally be dialed from the USA as 1-876-555-1212 must be dialed 011-1-876-555-1212. This means that we must give Asterisk a list of about two dozen NANP area codes which require special treatment.

Most European providers does this. Voip.ms does this too.

Allow us to Disable Recorded Fault Messages

Asterisk users frequently have several providers for outgoing calls so that if one fails or your account with it is out of money Asterisk can roll over to the next. They do this to improve reliability and to get better rates.

A common cause of failure is running out of funds. If you play an “you are out of funds” message, the caller will most likely hang up even though, had he waited, Asterisk might move on to another provider. You should simply reject the call with an appropriate error code.

Of course, such messages are helpful to those customers who are using your service directly from an ATA, but there should be a clearly explained way to turn them off, perhaps by indicating whether one is using an ATA or a PBX.

Whether or not you play a message, never, ever answer a call which you do not intend to put through. If you must play a message, play it as early audio.

The same goes for other problems which would provoke a recorded fault message. Don’t play messages for all-routes-busy, cannot-be-completed-as-dialed or any other condition where another provider may be able to put the call through. True, you will make no money all the call, but you reduce the chance that the Asterisk administrator will cut you out of the route.

Provide Machine Readable Rate Tables

Many Asterisk users want to know exactly how much their calls will cost so that they can do least-cost routing. While web pages which give rates for such things as “Peru Landlines” or “UK Mobiles” are fine as an advertisement, a computer cannot take a phone number and automatically find the corresponding rate.

You should provide a rate table which lists your rates by prefix. By convention the longest match wins, so you can list a rate for a whole country or other region and then list longer prefixes begining with the same digits for those blocks which have other rates. For example:

1 1.05 ; United States
1808 1.46 ; United States - Hawaii
1907 1.25 ; United States Alaska

But, be sure your switch rejects the call gracefully, without playing any messages, for any ‘holes’ in your coverage.

Finally, make sure that your rate table can be downloaded by fetching a single URL. Customers hate having to write programs to scrape your website.

Let DID’s Work Without Registration

For most customers using SIP phones directly with your service, periodic registration is the best way for them to inform you of their current IP address. However, Asterisk switches frequently have fixed IP addresses and DNS names. So, let your customers specify a SIP or IAX2 address to which to send the calls. This eliminates the constant chatter of registrations. When there are no calls there is no load on your network and servers and no load on ours.

For an example of a provider that does this well, take a look at how voip-user.org has users configure their second and subsequent incoming numbers. Call With Us allows this to, though they do not explain how to do it.

Make it Easy to Check One’s Balance

Your Asterisk-using customers are quite likely to want to check their balances automatically and gather all of their provider balances into one place. Make it easy for them to do so. Provide a way for them to do this with a single URL fetch.

Voicepulse for Asterisk and Call With Us do this well. Most other providers make you scrape their websites in order to determine your balance automatically

Treat Customer Suggestions as Free Labour

If your service is weak in some of these areas, knowledgeable customers may point it out to you. Train your staff to recognize these customers and forward their suggestions to your system administrators for consideration. While it may cost more to fix the problem than you will ever make from that customer, a more usable service will help you to attract additional customers.

asterisk, telephone, voip

Leave a Reply

Your email address will not be published. Required fields are marked *