By the editors and CT Labs technicians
03/28/2005 11:42 PM EST
URL:
http://www.callcentermagazine.com/shared/article/showArticle.jhtml?articleId=159907479
Digium sent us their Asterisk open source software PBX to CT Labs. It was installed on a dual-processor Linux server. With help from the Digium representative, we were able to connect three SIP phones as internal extensions and configure a number of features on the Asterisk system.
We found the Asterisk to be highly configurable; but since configuration is done via Linux command line instructions and text-based configuration files, it is not a product that would be easy for some to install and configure and maintain.
The Asterisk documentation is also a work-in-progress, with only a few completed documents -- the rest are pending completion by the Asterisk open source community. While the Asterisk system has no official graphical user interfaces, we evaluated a downloadable open source administrative GUI for Asterisk, and a client voicemail GUI that comes with the default voicemail system, Comedian Mail.
The GUIs were limited in functionality, but they worked well to accomplish specific tasks. The Comedian Mail system (the default voicemail system included with the Asterisk system) was easy to navigate and use through the telephone user interface, but one limitation we noted was a lack of message prioritization (such as marking a message "urgent").
We found that all of the Asterisk features that were configured worked well and that we were able to place calls successfully between the internal extensions and also between the extensions and an "external" phone.
Overall, we Labs found that the Digium Asterisk system to be best suited for users with a Linux background who are unafraid of tinkering with configuration files. With those requirements met, it's certainly hard to beat the price.
Download the Full CT Labs Report.
Test Setup
Digium supplied us with the Asterisk open source PBX software pre-installed on a Linux server that included a 4-port T1/E1/PRI card providing trunk access (their own Digium Wildcard TE410P board).
A Linksys 8-port workgroup switch was used to connect the Asterisk with three SIP-based phones: a Cisco 7960 provided by Digium, a Cisco 7040 phone and a Siemens optiPoint 600 phone that CT Labs had. In order to simulate "external" phone calls into and out of the Asterisk system, a Hammer IT call generator was connected and used to generate test calls into the ISDN-PRI port of the PBX using simple calling and answering scripts.
The Digium Asterisk program is open-source. Of course, nothing in life is free; there are therefore some costs for hardware and software in order to install and run Asterisk, including a nice PC running distribution scale Linux. The pricing for the 4-port T1 card is only $1,495. Their Pre-paid Configuration Packages include: Small Office Support Basic (either analog or SIP) $750; Small Office Support Premium (either analog or SIP) $1,200. Visit their website at http://www.digium.com/index.php?menu=config_packages to view details of each package. You'll need to buy phones too.
Installation
Digium provided us with their Asterisk open source PBX already installed and partially configured on a dual-processor server running Linux. They also sent along one Cisco 7060 SIP phone that was pre-configured to use in our testing. The basic configuration of the Asterisk system consisted mainly of having the phone accounts pre-configured for SIP and voicemail, and some basic dialing plans (e.g. SIP to SIP, internal voicemail access, and dialing out of the system).
For this test, we enlisted the help of a Digium support representative by phone. This person provided configuration assistance, including his logging into our system several different times in order to configure particular features, and to help us get the system started correctly after it was shut down over the Christmas holiday. The fact that everything in this system seemed to be configurable was found to be both a plus and a minus. On the upside, if you know both Linux and Apache and are familiar with using text-based configuration files, you will love the complete flexibility of this system.
However, if you are not "Linux-friendly," this will have you pulling your hair out within a few hours. As delivered, our system had forty-one different configuration files.
For example, the SIP phones are listed in one file, the phone extensions are listed in another, and the voicemail accounts are in a third file. We found it hard to figure out which file(s) to open and change in order to edit a particular feature. In fact, there were only a few configuration changes that we were able to make on our own, following the documentation available with the Asterisk system.
This is notable considering our extensive experience with a wide variety of IP PBX products. For all the remaining configuration changes we needed, we had to call Asterisk support and get their help.
For some of the simple changes, he walked us through making the changes ourselves; for more complicated ones, we set up remote access for the system so he could go in and change the configuration himself. We checked with Digium and found that they do offer service packages to help with installing and configuring the Asterisk PBX.
Overall, we rated the Asterisk installation and configuration a "6.0."
Features
See the Full CT Labs Report for a complete feature grid. There wasn't anything spectacular that jumped out.
Overall, we found that the system supports a full complement of features that would be needed by a small-to-medium sized business. The only caveat is that they require a varying amount of configuration through the Linux command line interface.
The only feature that we specifically noted as missing was the ability to set message priorities in the voicemail system, which we're sure more than few companies can do without. Overall, we rated the Asterisk features an "8.5."
Documentation
The documentation for the Asterisk PBX is available online. As this is an open-source product, the documentation is slowly being put together for the current version as people have time to contribute their efforts to write up information about various aspects of the product.
At the time of this review, the most comprehensive user guide we found on the Digium site was the Asterisk Handbook -- project version 1. It's a 48-page Adobe Acrobat .pdf file that you can read online or download. There is no Table of Contents, Index, or Adobe Bookmarks, so the document is hard to navigate -- the easiest method we found was to use the "search" function within Acrobat to find key words within the document.
This document presents an introduction to Linux Telephony, Voice over IP, an overview of the Asterisk system and the telephony interfaces, and configuration samples for Asterisk and associated modules (like the voicemail system).
We also evaluated the Digium Quick Install Guide, which is a 1-pager of text mainly explaining how to prep the Linux server and perform the installation of the Asterisk software itself, but nothing on configuring the system.
They got a "6" here.
GUIs
While Digium does not have any management GUIs that are included in the Asterisk package, there are GUIs that have been created by the open source community that are available to use with Asterisk.
Digium provided us with an example of one management GUI for our evaluation: the free download of Gastman GTK Asterisk Manager. This program required us to first download and install a prerequisite "runtime" program, and then download and install the Gastman program itself.
The Digium representative set up a login for us within the Asterisk configuration files, so once we installed the program, it was very simple for us to start the program and log into it. The Gastman management GUI allows the user to view the existing calls in graphical view (like the 4-party conference call shown in below).
One of the nice things about the graphical view is that you can choose a unique icon for each user from a gallery of people icons like engineer, secretary, big boss, etc. By hovering the mouse over an icon, you can see the information about the user (as shown, and if you right-click an icon, you can choose to redirect or hang up on a call.
We found that the graphical view is very nice for the small number of phones we used in this test, but once you get more than about 10 calls going at once, the screen would fill up and be very hard to view and manage.
The Gastman GUI also allows the user to access the command line for the system (as shown below) and enter commands as desired.
The default voicemail component of our Asterisk server (Comedian Mail) provided a client web GUI for listening to and managing voicemail messages. When a user logs in, the program opens the Inbox, showing the user's messages (as shown below in slightly modified view).
When the user presses the "play" button, the screen displays details about the call. A built-in media player allows the user to control the playback and volume of the message. The user can also choose to forward the message to another user, or save the message to a different folder.
We found that both of these GUIs were somewhat limited in their range of functionality, but they were easy to navigate and use.
Overall, we rated the GUIs for the Asterisk system an "8.0."
The Asterisk has two telephone user interfaces: the auto-attendant, and the voicemail system. There is no default auto-attendant, so the Digium representative made us a simple menu where we could choose to direct our call to one of three different departments, or go to a second-level menu for several other options that controlled how to direct a call.
Our system did not have a user directory configured, so we did not test any directory functionality within the auto-attendant. The voicemail that is included in the Asterisk system is Comedian mail. We found that the Comedian mail menus were easy to navigate and the prompts were descriptive while not being overly lengthy.
Overall, we rate the Asterisk Telephone User Interfaces a "8.0."
Performance
CT Labs manually exercised the Digium Asterisk media platform with a variety of phone and call types. During these calls, CT Labs testers noted perceived call quality, call connection capability, and any call handling problems.
All calls placed in this manual functionality testing evaluation were completed successfully with high perceived call quality. In order to complete the calls to and from the "external" phone, we used simple Empirix Hammer scripts to make the Hammer IT call generator place calls into the system voicemail and directly to the SIP phones. SIP phones were also used to dial out from the system to the Hammer IT, verifying this functionality.
No problems were noted with call connectivity or call quality. We rate the Asterisk functionality a "9.5."
Technical Support
Throughout the evaluation process, each tester was instructed to keep track of when questions were posed to the technical support department, and when a callback or email response was received. The status of each reported bug or product defect, if any, was maintained in a log. The relative responsiveness of the vendor as well as the quality of the answers received was also noted, if any.
For this test, CT Labs was assigned a particular technical support representative, but they reached him through the normal technical support phone number. They found that they had to contact Digium technical support a number of times for help with configuring various components/features of the system.
CT Labs rated their Digium technical support experience an "8.8."