Source: Gerald Pickford (c)2005

 

In early 2004, I had the opportunity to assist a private Catholic elementary school in the Seattle area with a telecom expansion project using an Asterisk open source telephony server.  The school had a legacy Lucent Partner II switch that provided service to the administrative offices.  This system was at capacity, but the school had wanted to expand phone service to all classrooms and several staff offices so staff members could make and receive calls from their classes.  The ability to transfer calls from the main office to classrooms was also desired.  This would require 14 new extensions that their current system could not provide.

A full replacement of their existing switch was out of the question for budgetary reasons.  So integration with the Partner II was proposed.

We configured an Asterisk server with 3 - X100P analog line cards from Digium, to provide dial tone to an asterisk server.  One of these cards connected to a POTS phone line from Qwest.  The other two cards connected to two available ports on the Partner II.

The school had been wired with Ethernet many years back, so Ethernet IP phones were easily added to existing outlets in classrooms.  To keep costs down, Grandstream Budgetone 100 IP phones were used in the classrooms.  These phones provided the basics (VM Message waiting lamp, Caller ID, Transfer Button, Hold, etc.) for a great price.  However, we did experience some pushback with the overall 'feel' of the phones as they are fairly lightly built and cheap feeling.  We ended up switching one of them out for a Uniden UIP200 for one user who was very unhappy with her Grandstream.  The Uniden is a much better phone, but also costs about $50 more.  A Grandstream Handytone ATA adapter was used to connect an analog cordless phone to the Asterisk server.  In hindsight, I would have preferred to use a Sipura ATA in this application, but the Handytone worked fine.  It just has less options and features than a Sipura for about the same price.

In order to integrate the Asterisk server with the Partner II, we created a hunt-group on the Partner II, which included the two ports we were using to interconnect the systems.  On the Asterisk box, we created a 'trusted' IVR context to receive inbound calls from the Partner II.  This allows the receptionist to transfer calls from the Partner II to a classroom on the Asterisk box by using a two-stage dialing process.  She hits 'transfer' on her Partner II phone, then dials the number for the Asterisk hunt-group.  The Asterisk IVR answers this line and the receptionist then enters the classroom extension number and completes the transfer. 

Outbound calls are routed via the first available line.  In the case of the lines that are interconnected with the Partner II, this involved an automated two-stage dialing process.  When a classroom dialed an outside number, the Asterisk system would open a connection to the Partner II, dial 9 to reach an outside line on the partner, pause, then dial the requested number.  This worked fine, but it did create a lag in connecting the user's call that took a bit of getting use to for the staff.  A side benefit of this interconnection was the ability to easily dial Partner II extensions from the Asterisk box.  We created a dial '8' prefix that allowed Partner II extension to be dialed.  For example if a staff member wanted to call extension 41 on the Partner II, they would dial 841 from an IP phone.  This worked well, and connection times were fast compared to accessing an outside line through the Partner II.

Parents or spouses of staff members wishing to reach a teacher directly could call into the system using the Qwest line.  This line was answered by an IVR menu that routed calls to classrooms after class hours, but transferred calls to voicemail during class so as not to disturb the classes.  An automated staff directory was provided on this line to allow for easy call routing.

Overall the installation went as expected.  However several weeks after the installation 'red alarms' started to appear on one of the X100P cards connected to the Asterisk box.  These alarms would clear if one disconnected the line and then reattached it.  But the alarm would return within several minutes.  We puzzled over this behavior for several weeks.  We replaced the X100P cards, cables… Everything we could think of.  In the end, it was a setting on the Partner II related to voicemail MWI.  Someone had managed to leave a voicemail for the extension of one of the ports we were using to interconnect to the Asterisk box.  This led the Partner II to signal that a message was waiting for the extension by oscillating the voltage on the port, which confused the X100P and caused the red-alarm.  Once MWI was disabled on the Partner II port everything functioned perfectly.

Early on, one complaint that we encountered was echo on the line, when placing calls outside the school.  The problem seemed to appear only on calls going out over the dedicated Qwest line.  Adjusting the echo training and RX/TX gains on the effected X100P card mitigated the problem to a great extent.  Echo was never a problem on the ports interconnected with the Partner II switch.

In the end, the expansion using Asterisk was a resounding success.  The school achieved its aim of adding phone service to the classrooms with minimal cost and no rewiring.  I see great potential for Asterisk in schools as it provides a feature rich platform for delivering telephony services as a price schools can afford.  Its powerful scripting environment allows for very customized dial plans and provides flexibility when integrating with legacy systems.

Four Loop Technologies, LLC today unveiled its latest easy to use product, the Switchvox PBX. Switchvox shows small to medium-sized business owners that high-end phone system features like Interactive Voice Response, call queues, and Advanced Call Routing are not out of their reach. "Typically, to buy a traditional phone system with the features of Switchvox, you're looking at spending tens of thousands of dollars, and even then it's a nightmare to get up and running." said Joshua Stephens, CEO of Four Loop Technologies. "With Switchvox we have an easy to use, intuitive portal to control your own phone system. If you can check your mail on the web, you can set up your phone system- and do it in less time than it takes to call out a technician."

As the commoditization and open sourcing of operating systems and applications continue to disrupt the software companies, telephony vendors have so far enjoyed a relative calm in the closed and proprietary phone systems market with substantial profit margins. That could now all be turned on its head with the proliferation of open source VoIP and PBX software. There are now a handful of these open source telephony platforms such as OpenPBX and Pingtel, but one of the most interesting is Asterisk, which even has its own communication protocol IAX in place of SIP for unified signaling and data transport.

Asterisk's IAX has all the attractive characteristics of SIP yet it plays nice with NAT and firewalls due to the fact that it uses a single UDP port for signaling and voice transmission. SIP, on the other hand, uses one port for signaling and another for voice, which makes it difficult to handle in NAT and firewall devices. Another benefit of IAX is its trunking capability. Think of it as the commuter lane for VoIP traffic, since multiple voice channels can share a single IP datagram (think of this as the enclosure for voice packets).

 

This is no small feat, since the overhead of an IP datagram can be anywhere from two to five times bigger than the actual voice data itself while traversing Frame Relay or VPN tunnels! If the IP header can be shared, it can mean the difference between supporting 19 simultaneous G.729 voice channels using SIP or 55 simultaneous G.729 channels when using IAX where 512 kbps of a Frame Relay link is allocated for voice traffic. To make Asterisk universally appealing, it supports nearly all communication protocols, including SIP, H.323, MGCP, and even some limited proprietary Cisco SCCP (Skinny) support.

 

With such a robust feature set, it’s hard to believe Asterisk is free (GPL license). Note that there is a $10 licensing fee per session per server if you want to use G.729 (arguably the best CODEC for voice compression); and, of course, you do have to buy some telephony interface cards for your cheap commodity LINTEL (Linux on Intel) PC or Server, but it is still by far the cheapest hybrid telephony system that money can buy.

Although it’s easy to think of Asterisk as just another VoIP server, that couldn’t be further from the truth. Asterisk is an extremely flexible communications platform that can serve as a VoIP Signaling Server, a Media Gateway (allows IP telephony to interface with analog phones, fax machines, or PSTN lines), a traditional analog or TDM-based PBX phone system, voice mail, IVR, Unified Messaging, and too many other things to list!

 

For example, you can build a phone system that can support 72 analog telephones or fax machines, 100 IP hard or soft phones on site or remote, a T1 line to the public telco for 23 simultaneous external PSTN connections, multiple IP-based IAX trunks to multiple remote offices for seamless toll-bypass 4-digit dialing, IVR, and almost unlimited voice mail for everyone – for under $6,000 in a 1U chassis. Such a price point is easily 10 or more times cheaper than a commercial alternative. Here is a graphic illustration of such a system.

 

 

Of course, this all might sound too good to be true and you’re probably wondering "what’s the catch?" There is one catch (but diminishing), and it boils down to ease of use and having a person who is really good with Linux. However, you can always hire a consultant to implement the system for you on an hourly basis or go to one of the many companies selling turnkey systems that work out of the box like Xoasis or Coalescent. Coalescent for its part has contributed a free Opensource management package called AMP (Asterisk Management Portal).

 

Asterisk itself also offers very reasonable support contracts to assist the do-it-yourselfers. Then there are companies like Xorcom, which has done a fine job in simplifying the installation of Asterisk with a simple CD-ROM ISO that you can download and burn. This will allow you to install Debian Linux with Asterisk as a fully functional server. Xorcom is also working on implementing a GUI management tool into their Asterisk installer CD, and they may be looking at some of these GUI Management tools. With all these developments, Asterisk is on the verge of critical mass to explode on to the VoIP scene – which will revolutionize the IP telephony market.