Asterisk VOIP Telephone System

The Asterisk voice-over-IP (VOIP) telephone system, combined with the FreePBX web-based management portal, provides excellent service for businesses of all sizes. We install the software on a dedicated Linux server.

Warnings

First and foremost, you need to be aware of three limitations of VOIP:

No emergency 911 service.
No operator services (411, collect calling, etc.)
Poor fax support. We recommend that you use a fax-thru-email service.
In a small number of cases, it is necessary to replace the router or cable/DSL modem.

Three Tiered Features

This is a partial feature list.

Asterisk	FreePBX	HTN Support
Automated attendant Blind transfer Call forward on busy Call forward on no-answer Call monitoring Call parking Call queuing Call recording Call routing (DID & ANI) Call transfer Call waiting Caller ID Conference bridging Dial by name Direct inward system access Distinctive ring Do not disturb Interactive voice response (IVR menus) Music on hold Music on transfer Remote call pick-up Remote office support Roaming extensions Route by caller ID SMS messaging Three-way calling Voicemail Visual indicator for message waiting Stutter dial tone for message waiting Voicemail to email Voicemail groups	Full GUI for all functions Reports & Graphs Call Detail Records (CDR) Voice Mail Operator Control Panel	Initial set-up and configuration of all software and hardware Reconfiguration as-needed Backups of all configuration files Documentation Telephone and email support as-needed

"Telephones"

We put "telephones" in quotation marks intentionally because you have three options.

You can retain your old, analog phones by purchasing phone adaptors. These have one or two telephone jacks on one side and an ethernet port (or two) on the other.
You can purchase special VOIP telephones which plug directly into the ethernet.
You can use "soft phones" which run on your laptop or desktop computer. We strongly recommend use of a headset with soft phones.

In all cases, HTN will pre-configure the device so that it is ready to use with your system.

Making Calls

Your Asterisk system uses a VOIP provider to place and receive calls to/from the public switched telephone network (PSTN). HTN recommends VoicePulse Connect. As of September 6, 2006, their rates are:

Outgoing calls to US numbers vary from 0.5¢ to 1.9¢ per minute
Incoming calls to US phone numbers are free
$11 per US phone number per month

It is also possible to configure multiple VOIP providers. For instance, you might use VoicePulse for calls within the United States but you might find another provider which offers lower rates to a particular foreign country which you call frequently. We can configure Asterisk so that calls are automatically routed through the least-cost provider.

Remember, also, that there are no charges whatsoever for calling between extensions on your Asterisk system, even if the callers are in different countries. Likewise, there are no costs associated with using the conference calling features of the Asterisk system (beyond the normal rates for the individual phone calls).

Pricing

There are two components to the pricing: initial set-up and monthly charges. Every system is unique and we will provide quotations for the initial set-up and the monthly fee.

Initial Set-up	Monthly
Linux Server	Linux Server
Linux Configuration	Support Fee (per extension)
Asterisk & FreePBX Installation & Configuration
Telephone Devices (Purchase & Configuration)

In addition to these fees, you will establish your own account with VoicePulse Connect and may have additional charges with a phone company for hosting your toll free number(s). We will help you establish these accounts and get everything "hooked up" and working properly.