In the open-source PBX world, one platform is king: Asterisk.  Asterisk is an open-source software PBX platform originally created by Mark Spencer of Digium. Because of the open-source nature of Asterisk, many companies have used the Asterisk code as a base for their product and modified the code to suit their needs. There are a host of products built on top of Asterisk, such as Fonality’s products PBXtra and Trixbox, IntuitiveVoice’s Evolution PBX, etc. This post applies to all of them to some degree. Also, there are other PC-based PBX solutions out there that run on Windows, such as 3CX or NCH’s Axon product. These share many of the same pitfalls as described below (but have the additional problems associated with Windows-based instability and security issues).

Open Source.
Asterisk is open-source software. It is controlled by a company named Digium, which issues periodic updates and bug fixes to the code. Because it is open-source software, this means that (1) Asterisk is free to anyone and can be downloaded without charge; (2) Anyone with the proper skills is able to view the original programming code and modify it for their own purposes. There are a number of companies that have grown up around this model and are selling their own “take” on Asterisk.

Hardware.
Most Asterisk builds use off-the-shelf PC hardware. Because there is so much to consider in regards to PC-based phone systems (whether tower-based or rack-based), I can‘t enumerate it all here.  Suffice it to say, PC-based phone systems suffer from a slew of drawbacks, including longevity, expandability, security, complexity, migratability, and recoverability issues, and they are more of a headache than they’re worth.

There are also a number of appliance-based Asterisk systems out there. Many of these are PCs in disguise. Others, that are truly appliance-based (you’ll often see the term “embedded” when seeing references to these systems) are generally for the smaller-end market. They feature hardware that has specifically been engineered for that specific product. Many times there is a single circuit board inside that replaces the functionality of multiple boards in other systems. The reliability of these products varies, but is generally good. If something needs to be replaced, however, the whole system usually needs to be swapped out.

Telephony cards.
In the case of PC-based implementations, and some appliance-based installations, the box will require certain telephony cards in most cases. This is to provide support for dialtone (analog lines or T1/PRI lines), as well as other analog devices such as cordless phones, credit card machines or faxes. These cards can be purchased from Digium, but the majority of these are purchased from third-party vendors, such as Sangoma, Rhino, OpenVox, ZapMicro, etc. With all of these choices comes different implementation issues, as each card has its own drivers, idiosyncrasies, implementation, support, reliability, and integration with Asterisk. Some cards are supported better than others. Some companies will only allow certain cards to be used with their implementation.

A note on echo. One of the pervasive challenges with Asterisk is that of echo. This generally happens when a digital signal (like IP) is being converted to an analog signal, or vice versa, and there is an impedance mismatch on the line (there usually is-it’s just the nature of the beast). This causes one or both parties to hear their own voice echo in their ear. While software-based echo cancellers are standard and tend to cut down on this, many users have reported serious echoing issues. The only proper way to minimize echo with Asterisk-based systems is to make sure the cards have HARDWARE based echo cancellation (the Echo Cancellation, or EC, version of any given card is more expensive). The advantages of hardware-based echo cancellation are (1) it can take care of some echoing that software sometimes cannot; (2) it is faster (although you may still have a couple of seconds of echo at the beginning of the call; (3) it frees up precious CPU resources from having to do the echo cancellation.

Telephones.
Digium does not make the telephones for its Asterisk software, nor do the other vendors whose PBX products are based on Asterisk. These are third-party phones from various manufacturers, such as Aastra, Grandstream, Cisco, Snom, and others. Each phone model from each manufacturer has a different look and operation than the others.

While “freedom of choice” is touted by Asterisk proponents, this disparity between phone models, and the integration of phones with Asterisk, is actually one of Asterisk’s greatest drawbacks. By not having proprietary phones, integration with the phone system is looser. Some of the phone manufacturers illustrate some “whiz-bang” applications that can be done on their phones, but the reality is that it’s difficult to incorporate into the Asterisk platform (more on this in a moment), and most IP phones used on Asterisk-based phone systems function little better than plain old analog telephones.

Getting the phones to work with Asterisk in the first place has traditionally been a challenge, although many strides have been made in this respect.

One thing that anyone who has been looking at Asterisk for any length of time will notice is that the phones in general are severely restricted in the number of buttons available for lines and features. This requires people to remember and dial feature codes manually for most anything they would like to do feature-wise, or work their way through on-screen menus.

Features.
Now for a paradox on Asterisk. Asterisk is, admittedly, a platform with perhaps the greatest potential for flexibility, and yet, in practice most people find that they are locked into the features provided out-of-the-box. It is true that because their code is open-source that in theory someone who knows how to program can make it do special things. It is also true that Asterisk has a scripting language that allows for special things to be accomplished. By going into special, text-based configuration files, one can, in theory, open up the flexibility potential. Here are some of the configuration files:

amportal.conf; asterisk.conf; autofs_ldap_auth.conf; capi.conf; cbmysql.conf; cdr_mysql.conf; conman.conf; dhcpd.conf; disa-1.conf; enum.conf; extensions.conf; extensions_additional.conf; extensions_custom.conf; extensions_hud.conf; extensions_override_freepbx.conf; ez-ipupdate.conf; features.conf; features_applicationmap_additional.conf; features_applicationmap_custom.conf; features_featuremap_additional.conf; features_featuremap_custom.conf; features_general_additional.conf; features_general_custom.conf; fxotune.conf; globals_custom.conf; gpm-root.conf; grub.conf; gssapi_mech.conf; host.conf; iax.conf; iax_additional.conf; iax_custom.conf; iax_custom_post.conf; iax_general_additional.conf; iax_general_custom.conf; iax_registrations.conf; iax_registrations_custom.conf; idmapd.conf; indications.conf; initlog.conf; jwhois.conf; krb5.conf; ld.so.conf; ldap.conf; lftp.conf; libaudit.conf; libuser.conf; localprefixes.conf; logger.conf; logrotate.conf; manager.conf; manager_additional.conf; manager_custom.conf; meetme.conf; meetme_additional.conf; mke2fs.conf; modem.conf; modprobe.conf; modules.conf; mtools.conf; multipath.conf; musiconhold.conf; musiconhold_additional.conf; musiconhold_custom.conf; my.cnf; nscd.conf; nsswitch.conf; ntp.conf; oddjobd.conf; op_astdb.cfg; op_buttons.cfg; op_buttons_additional.cfg; op_buttons_custom.cfg; op_lang_de.cfg; op_lang_en.cfg; op_lang_es.cfg; op_lang_it.cfg; op_server.cfg; op_style.cfg; pam_smb.conf; pear.conf; phone.conf; phpagi.conf; privacy.conf; queues.conf; queues_additional.conf; queues_custom.conf; queues_custom_general.conf; queues_general_additional.conf; queues_post_custom.conf; reader.conf; request-key.conf; resolv.conf; rtp.conf; sensors.conf; sestatus.conf; sip.conf; sip_additional.conf; sip_custom.conf; sip_custom_post.conf; sip_general_additional.conf; sip_general_custom.conf; sip_nat.conf; sip_registrations.conf; sip_registrations_custom.conf; smartd.conf; snom.cnf; sysctl.conf; syslog.conf; updatedb.conf; vm_email.inc; vm_general.inc; voicemail.conf; warnquota.conf; wvdial.conf; xinetd.conf; yp.conf; yum.conf; zapata.conf; zapata-auto.conf; zapata-channels.conf; zaptel.conf

Because of this maddening array of configuration files (which are not in one single place on the disk, by the way), several projects (including the makers of Asterisk itself) have developed web-based configuration screens so that you don’t have to delve into all of these. By doing this, however, they are necessarily limiting the number and type of special configurations that can be done.

Asterisk features a special scripting language to tweak the behavior of the phone system. Again, in theory it’s great, but only if you want to learn this:

exten => s,n,Set(FROMCONTEXT=exten-vm)
exten => s,n,Set(VMBOX=${ARG1})
exten => s,n,Set(EXTTOCALL=${ARG2})
exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})})
exten => s,n,Set(CFBEXT=${DB(CFB/${EXTTOCALL})})
exten => s,n,Set(RT=${IF($[$["${VMBOX}"!="novm"] | $["foo${CFUEXT}"!="foo"]]?${RINGTIMER}:”")})
exten => s,n,Macro(record-enable,${EXTTOCALL},IN)
exten => s,n,Macro(dial,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,Set(SV_DIALSTATUS=${DIALSTATUS})
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"="NOANSWER"] & $["foo${CFUEXT}"!="foo"]]?docfu,1) ; check for CFU in use on no answer
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"="BUSY"] & $["foo${CFBEXT}"!="foo"]]?docfb,1) ; check for CFB in use on busy
exten => s,n,Set(DIALSTATUS=${SV_DIALSTATUS})
exten => s,n,NoOp(Voicemail is ‘${VMBOX}’)
exten => s,n,GotoIf($["${VMBOX}" = "novm"]?s-${DIALSTATUS},1) ; no voicemail in use for this extension
exten => s,n,NoOp(Sending to Voicemail box ${EXTTOCALL})
exten => s,n,Macro(vm,${VMBOX},${DIALSTATUS})

If you do have something custom programmed, you’ll need to do your best to ensure that it doesn’t get overwritten with a software update, or knowledge of it lost if the person who programmed it moves away.

Here is a real-world example of how the practical reality of Asterisk differs from theory. Consider some features that Asterisk-based systems have only recently added in, or that in some cases STILL have not been implemented yet. These are things that traditional phone systems have had for over 20 years:

• Day/night mode button (Fonality just implemented in 2008?)
• BLF, or Busy Lamp Field (Fonality just implemented in 2008?). This is where, by looking at a button on your phone, you can see if another person in your office is on their phone
• Shared Line Appearances (ability to press a specific line button to make a call, and see visually whether that line is in use by looking at the line button). Asterisk just added this capability in the last year, and some Asterisk-based systems still do not have this feature (including Fonality, based on their website).

Support.
Most national Asterisk-based vendors do not hire local support. There is normally an 800 number you can call with the major vendors, some of whom have 24-hour support. Bear in mind, however, that if and when your system goes down, there is only so much that can be troubleshot remotely. If the remote connection is down, then what? Most of these vendors do not have a mechanism in place for same-day parts replacement, let alone getting a body out locally to troubleshoot.

Voice over IP, or VoIP, has come a long way since the early days.  Back then, VoIP phones in general weren’t at all tolerant of network issues, nor were networks as good at prioritizing VoIP traffic to ensure quality.  QoS, or Quality of Service, a term used to indicate network traffic prioritization, was nonexistent on service provider networks.  And most IP phones out there were difficult to set up, could not be easily moved from site to site, and had very few features.

What a difference a product generation or two makes.  Not only are IP phone makers getting smarter and more capable in their offerings, but service providers are offering more bandwidth and many are routing VoIP traffic more intelligently.  Sure, things still aren’t perfect, but for the first time, business VoIP phones are truly useful.

ESI (the manufacturer whose phone systems we sell) was an early entrant in the VoIP market (they were the first, in fact, to be able to network 100 phone systems together via IP), and their experience has shown.  Unlike others in the market, they have for a long time offered the same feature set on their IP phones as their regular digital key phones (and believe me, they have a rich feature set).  On ESI’s Remote Feature Phone II, which is compatible with any IP-enabled Communications Server phone system, you’ll truly be able to participate as if you were sitting in the office.

ESI IP Feature Phone II (backlit)

ESI’s Remote Feature Phone II, while retaining the same form factor as its predecessor (which we like), has improved some things under the hood.  For example, it has become much more tolerant of network issues.  Not that there was anything wrong with the prior iteration, but the newer version is a bit like a car having better shocks on a bumpy road.  Of course, you’ll want to make sure that your network connection is as good as it can be, but much of the time we don’t need to do any special router configuration at the remote site for the user to have a good experience.

One thing that the Remote Feature Phone II also integrated was the ability to “plug & play” at virtually any site where there is a broadband connection.  In the prior generation, you would set up the phone and not move it, because it would generally take about an hour to get the phone operating wherever it was.  Not here–wherever you plug the Feature Phone II in, it begins communicating with the main phone cabinet right away, and connects automatically.

Another change ESI made was switching its codec to G.726 (a codec is the compression algorithm used to save Internet bandwidth).  I think this was a brilliant move; most other IP phones seem to be using G.729 or G.729a, which has a higher compression rate, and the higher voice quality of G.726 is noticeable.  While G.726 does use more bandwidth than G.729, it’s still much less than G.711, which is uncompressed.  Besides, provider bandwidth is increasing more all the time, and the 80K or so the phone requires (which includes voice in both directions, plus overhead for lighting buttons, updating the phone display, etc.) is very, very small compared to what standard cable modem connections provide around here (generally about 2Mbps upstream and 10Mbps or so downstream).

One thing I appreciate about ESI’s IP phones is that they don’t skimp on buttons.  I can’t believe that most manufacturers are getting away with putting only a few buttons on their IP phones (presumably to save on manufacturing costs?).  Sure, many of them try to compensate by having a larger screen and/or navigation buttons, but I have always found navigating this way to be cumbersome.  There is truly no substitute for having all the buttons you need for line appearances, station buttons, and feature keys.  I also note that ESI’s phones have tri-colored buttons; most competitors’ lighted buttons are either a single color, or in some cases, two colors.

Regarding features, I haven’t really delved into that in this post.  Features and ease-of-use are ESI’s strong suit, and their phones are truly a pleasure to use.  You can check out our web site to find out more on ESI products in general, or poke around in the other posts on this blog.

Almost every phone system, universally, supports cordless phones when properly equipped with an analog station port.  These are the phones you can get practically anywhere and are primarily intended for home use.  Many phone systems, however, do not support cordless phones via a digital interface.  Why does this matter?  Features, and ease of use.

An analog interface to a phone system is a very simple one.  Cordless phones connected this way are generally able to make and receive calls, transfer, and that’s about the extent of it.  Sure, a lot of manufacturers allow you to do more by pressing the Flash button and dialing a code, but few people ever venture beyond the basic functionality because to try to do more is, well, a hassle.  Even putting a call on hold is usually a chore, because you can’t see by looking at the phone what line a call is on, to be able to pick it back up.

This is where digitally-interfaced cordless phones come in.  These phones can communicate digitally with the phone cabinet, allowing for much tighter integration, making for more features and a phone that’s much easier to use.  What features?  Well, let’s take an ESI Digital Cordless Handset, for example:

This baby is small and light, making it very portable.  Though you can’t quite make it out from the picture, this phone has dedicated Hold, Redial, and Voice Mail buttons, as well as a button used for Transfer, Conference, or station programming depending on when you press it.  What’s really great are the 4 buttons along the bottom that can be used for almost anything, according to your needs.  Here is a partial list:

• Line buttons
• Extension buttons (including extension status)
• Company day/night/holiday mode
• Do Not Disturb
• Call Forward
• Overhead Page
• ACD (Automatic Call Distribution) agent logon/logoff
• ACD administrator for viewing ACD queue status
• Personal greeting switch
• Virtual Answer
• Door Unlock (used with Presence Management)
• Account code entry

The display indicates when you have new voicemail messages.  Also on the phone is one-button access to change the volume (including ring volume), as well as mute.  There is a micro-mini headset port at the top, which supports generally the same headsets you would use for a cell phone.

In addition to the digital version of this phone, a local IP version is available (for inside the building), and a remote IP version is available for off-site.  All three versions operate in the same way, with full functionality.  The remote IP version will also allow you to connect a phone line (for example, if you’re using it at home), so that you have the choice of making local calls if you wish (otherwise, calls are routed via your office lines).  There is also a slightly larger version of this phone (in all 3 flavors) which offers longer battery life and an extended range.

There has been a real buzz over Voice over IP (VoIP) during the last couple of years.  We get asked about it often by businesses looking to purchase a new phone system, who want to make sure they’re not being “left out”.  How does it work?  Will it save money on long distance?  What other benefits does it provide?  In many people’s minds, the concept is a bit undefined; they’re not sure exactly what it will do for them, but they do know that their next system should support it.

First, a small explanation of what VoIP technology is, and the basic reason it’s so promising.  In traditional telephony, voice conversations take place over a dedicated line.  This is generally a pair of wires that carry your voice to the other person you are speaking to, and their voice back to you.  Granted, there is sophisticated equipment that routes your call from point A to point B, so there are usually a number of sets of wires that are actually being connected, but the point is that you are using a dedicated connection when you are on that call.  Since there are a limited total number of connections that can be made, even the phone company can run out of connections, and this is why you can sometimes try to make a call and hear the “all circuits are busy” recording.

VoIP technology is more sophisticated.  When you are on a conversation using VoIP technology, your voice is broken out into many small bits at lightning speed and sent to the other end very quickly, where it is reassembled and played to the person at the other end.  These small voice bits, or packets, have several pathways they can take to their destination and will take whatever path is the most efficient at that millisecond in time.  As long as they can be reassembled properly at the other end within a reasonable time frame (on the order of milliseconds), it does not matter what path they take to get there.  Because a dedicated path does not need to be established in this scenario, certain things can be done to make calls more efficient, such as cutting out the parts of the conversation where no talking is happening, like between words, and voice compression can also take place.  The net effect is that the same resources that could once carry say, 24 calls simultaneously, may now be able to carry triple that number or more.

It’s important to note that, while all VoIP technology is the same in a basic sense as has been described, there are some different applications  that take advantage of this technology.

VoIP Dialtone

Dialtone providers have learned early on the tremendous efficiency gains (and cost reductions) that could be had with packet-based calls, and most carriers have upgraded their equipment so that, at least behind the scenes, they are using this technology.  Most are also now offering VoIP-based phone lines to both business and consumer customers.  In some cases, the interface to your home or business may not even be any different, but they can offer more competitive dialtone rates to you because of this technology.

Some companies, such as Vonage, deliver dialtone over the Internet, and they provide you with a special box called an ATA (or Analog Terminal Adapter) that converts the IP technology into an analog signal that will work with your phone or business phone system.  Companies that provide dialtone via the Internet in this manner are called ITSPs, or Internet Telephony Service Providers, and they can usually offer very aggressive rates in the form of lower monthly rates or free long distance.  Some newer phone systems have circuitry that can understand IP dialtone natively, and in such a case an ATA box is not required.

Phone System VoIP

Phone system manufacturers have also realized that they can do some fantastic things with VoIP technology.  These applications offer tremendous benefits by taking advantage of two things: (1) the fact that most Internet connections are based on a monthly fee, and that you are not charged based on the amount of traffic you generate, and (2) the global nature of the Internet.

• Local IP Phones: these are used instead of traditional digital phones, and rather than requiring separate cabling can use the same cable that your computer uses on your office LAN (Local Area Network).  Their functionality is usually the same as a normal digital phone by the same manufacturer.
• Remote IP Phones: these can be used off-site, such as at a home residence for telecommuters or executives.  While remote IP phones from most manufacturers offer a more limited feature set than a local IP phone, remote phones from ESI (the manufacturer whose phone systems we sell) offer exactly the same experience that a local IP phone does, meaning that a remote user can see who is on their phone in the office at a glance, intercom, answer incoming calls to the business, act as a customer service agent as part of an ACD (Automatic Call Distribution) group, and more.
• Remote “Soft” Phones: this is software installed on a computer, such as a traveling sales rep’s laptop, that, when used in conjunction with a headset, allows the user to have similar functionality to a Remote phone, without the phone’s footprint.  This is extremely beneficial for “road warriors” and can be used anywhere in the world there is a decent broadband Internet connection (wired or wireless).
• Site-to-site VoIP: this is beneficial for businesses with multiple locations.  These locations can be virtually connected together to make it as if the users in all offices are together in one large office.  In addition to bringing everyone together without having to dial outside phone numbers from one site to another, this can eliminate long distance charges between offices.  ESI’s implementation of this is called ESI-Link, and supports tying up to 100 phone systems together.

ESI offers all of the above-mentioned types of phone system-based VoIP in their Communications Server phone systems, and does so with a very rich feature-set and high-quality voice codecs, making it an outstanding investment value.

The ESI 48-key and 24-key office phones, which have been around for several years (but still look as sexy as ever), have been fitted with a backlight on the display:

ESI 48-key backlit phone

ESI now offers a backlit and non-backlit version of each phone, with the backlit version costing slightly more than the non-backlit version.  The good news is, they will work on any ESI phone system that supports either 48-key or 24-key phones.  This includes not only the newest Communications Servers models, but the IVX Gen I and Gen II models (48-key Remote IP phones for the Communications Servers only).

Apparently, some people have been clamoring for this.  For my part, I never gave it a second thought until they came out.  Then, I remembered that I have a 48-key Remote IP phone at home, which gets a bit dark at times.  I think I may need to beg for one….