Verterion Announces New VoIP Service

Verterion has partnered with Vitelity to offer reliable, cost effective VoIP service to our customers.  Services include SIP trunking, DID numbers, eFax services and SMS services.

Vitelity is an industry leader in VoIP products and services, so they are a natural fit with Verterion’s consulting services.  We are proud of our new partnership and look forward to continued success with them!

Click here for information on pricing and services available.

Using Digium Phones with FreePBX Systems With the DPMA Module

There are a multitude of VoIP phones that can be used with FreePBX.  However, one of the best additions to the FreePBX family of support phones are the Digium brand of phones.  Digium, as you may already know, is the prime supporter of the Asterisk Telephony Engine that FreePBX runs on.  They have developed phones specifically to take advantage of all the features of their commercial version of Asterisk in their Switchvox PBX.  FreePBX now has the ability to use these Digium phones natively through the use of the Digium Phone Module for Asterisk (DPMA).

While it IS possible to connect Digium phones to any Asterisk-based system through simple web-based configuration of SIP parameters, this requires manual configuration of each phone.  Once enabled, DPMA allows automatic configuration of most phone parameters.   Historically, provisioning phones has been slightly difficult, because it required a degree of manual configuration of the phone (configuration server for example).  Digium phones can search for phone servers on the local subnet using multicast DNS (mDNS).   Additionally, the phone has NATIVE applications that can control how the phones interact with the PBX.

It requires Asterisk versions 1.8 or above along with a Free License (available from

Digium D65

An Example of a Digium Phone (Model D65 Shown)

More information on the DPMA module for FreePBX and Asterisk can be found at The Digium Asterisk Wiki.  If you need help installing the DPMA module or planning for a Digium Phone installation, please contact us or email us at

How To Convert Audio Files for Use With Asterisk

Although Asterisk has many built-in audio files, most often you will want to have custom audio files for prompts for your auto attendants.  You can either record them using the phone system itself (often times this can be somewhat confusing), or you can record them with conventional computer recording software and convert them into the proper format.

Most recording software has the ability to record audio, then export it into the proper format that the phone system will use.  For Asterisk-based systems, this depends on the CODEC you are using.  90% of the time, the g711 uLAW codec is used, which means the sound file must be an 8kHz, mono, 16-bit audio file.

The best way to convert your audio files for use with Asterisk is to use Digium’s online audio file converter, located here: 

Instructions are provided on the web page.  Once you download the file, you can upload it onto your Asterisk server and use it in your prompts.

If you have any questions about how to use audio files, or how to convert and install custom prompts or audio files, please contact us at

Cool Asterisk Add-ons

Out of the box, Asterisk based systems have most, if not all, of the functionality that you need in a PBX or call center system.  However, there are some features that people consider to be “not up to par”, don’t fit their needs, or features that just simply don’t exist that a particular company simply must have.  Here are some software packages you probably didn’t know were out there for Asterisk-based systems.  In this article, we’ll discuss these Add-ons by category.

Operator Panel Software

An Operator Panel is any software that can be used by a phone operator or receptionist (or even every user on the system) to manage calls.  It typically shows all the phones on the system, whether they are on the phone, who they might be talking to, and allows call transfer and hold operations.  This software typically takes the place of a “button module” attached to a phone that would perform the same activities.

Flash Operator Panel 2

Flash Operator Panel 2 (or FOP2 as it more commonly known) is one of the most popular add-ons because it is free for < 15 extensions and for unlimited extensions it has a maximum price with all the bells and whistles of $80 (as of the writing of this article)


FOP2 Web Interface

More information about FOP2, including a full feature list can be found at

iSymphony Operator Panel

Another very popular browser-based operator panel is iSymphony.  It has similar features to FOP2 (including a few extra) and has a free version as well (only allows one login and shows no queues), but has quite a few more features and a bit more “polished” look to it.  The commercial version is much more scalable but still cost effective.


iSymphony Web Interface

More information about iSymphony can be found at

Call Center Operations and Management

Because Asterisk can handle call traffic so well, there is a whole category of add-ons specifically designed for managing calls in a call center environment.  Typically the packages include call reporting, call flow management, agent management, agent login/logout interface, and a “live wallboard” to see call center performance in real time.  In some cases, they are primarily reporting-only tools, but still add a lot of value and insight into the operation of an Asterisk-based call center.  Here are some of the most popular call center add-ons.


Vicidial is the most often used and most popular call center software for Asterisk.  While it is a whole set of scripts and web pages (and really a product all its own), it is still at it’s core, an add-on for Asterisk.  We do specialized work in ViciDIAL as well as other software packages, but since it is so popular, we get a large amount of requests for support and integration work, as well as dedicate several articles to it on our website.

vicidial live dashboard

Vicidial Live Dashboard

For more information about ViciDIAL, see


QueueMetrics is a unique software in that it can be used not only for simple queue reporting, but also as a call center queue management system for inbound (and to a lesser degree outbound) call management.  It can be configured to see agent realtime data, connect to a CRM, collect IVR response data, and many other statistical information a standard reporting package does not gather.


Queuemetrics Call Center Monitoring

For more detailed information about QueueMetrics, visit


While the other products in this category are specific to running an entire call center, this software is specific to operating and managing a specific type of call center.  Specifically, it is designed to manage survey campaigns.  The entire system is set up to allow agents to call prospects, execute a survey of some sort, and gather the results.  Managers can then run reports on the results as well as specific agent performance.

quexs admin interface

Quexs Survey Web Admin Interface

More information about QueXS can be found at


Call Center Stats

Call Center Stats is another software package developed by the same people who developed FOP2 and CDR Stats (discussed later in this article).  It is comprehensive in showing the inbound queue statistics for inbound and outbound calling for agents.  Statistics such as drop rate, number of calls, service level, hourly performance, etc are all available in a nice graphical format.  There are both free and paid versions available.

asternic call center stats

Asternic Call Center Stats Web Interface

For more information about Call Center Stats, visit


OrderlyStats is another call center performance tool (very similar to “Call Center Stats” by Asternic) that provides all performance data, as well as live statistics about the operation of a call center.  It will provide the inbound performance metrics by queue, by agent, as well as a “live” wallboard that can be put on a TV display in the call center to show “at-a-glance” call center activity.


OrderlyStats Web Interface

For more information, please visit

PBX Management

Operator panels, in general, are used to control what happens within an Asterisk PBX.  Products in this category, however, are designed primarily to display what is actually happening with ALL the functions of the PBX.  For example, who is on the phone, what channels are in use, processor utilization, and so on.

Let’s take a quick look at two of the more popular.


MonAST is a web-based software package that can be installed right on your PBX that will give you a tab-based view of all the functions of your PBX in a “live” view.  Additionally it gives you a command line interface (CLI) that you can use to do limited control commands directly with Asterisk.


MonAST Web Interface

You can find more information about the features of MonAST at


AstChannelsLive is a Windows-based program that will show you primarily a LIVE version of all channels in use (SIP, Analog, PRI, etc).  In addition, the latest version will show you a wallboard of queue activity.  It provides basic functionality, but is pretty powerful at the same time.  There is also a mobile version of the product.



More information about this software can be found at

Software Integration

Of major interest to owners of phone systems is the integration of their phones to other software, typically Customer Relationship Management (CRM) programs that track their customer interactions.  There is a great deal of software that readily facilitates these types of integrations as well as other more advanced integrations.


The first and possibly most powerful integration piece is the PHP-AGI library.  What this allows you to do is write special code in PHP that will directly interface with an Asterisk server via the Asterisk Gateway Interface (AGI) programming language to manage call flow.  With it you can use PHP functions to initiate calls, transfer calls, monitor calls, and just about all functions of the Asterisk PBX.  You will have to be familiar with PHP programming as well as

More information about PHP-AGI can be found at  If you need information about AGI – check out the informational resources on Wikipedia at

Zoho Phonebridge for Asterisk

Zoho is a leading web-based Customer Relationship Management (CRM) software.  It allows businesses to track customers and every contact that is made between the business and the customers, including emails and phone calls.  Zoho Phonebridge is a plugin for Zoho that allows an Asterisk-based PBX to speak to the CRM and display customer information on the screen when a phone call comes in, as well as allow “click-do-dial” functions for Zoho and call logging of each call.

zoho phonebridge

Zoho With Phonebridge for Asterisk


More information about Zoho PhoneBirdge can be found at


Activa is a very important component for Windows software users.  It is a TAPI driver for Asterisk.  That means you can initiate phone calls through the TAPI software interface on your computer (various Windows CRM software programs use this – like Outlook) as well as do “screen pops” based on inbound calls.

activa settings

Activa Settings Window

More information about Activa and the downloads can be found at

Feature Add-Ons

The last category of add-ons are those that expand existing functions of Asterisk, but add more capabilities outside of those that already exist.

Speech Recognition for Asterisk

By default, Asterisk has the ability to do Text-to-Speech (TTS).  That is, convert any text in the dialplan into machine-spoken (or in some cases-prerecorded female sound file) language.  However, out of the box, the reverse cannot be done.  Asterisk does not have the capability, without additional help, to recognize the spoken word and recognize it (e.g. Answering “Yes” or “No” to an IVR prompt or saying the name of the person they want to transfer to).

There are two primary speech recognition engines that are often use – “Speech Recognition for Asterisk (SRA)” and “Mojolingo RubySpeach”.  Each has their benefits and downsides.  SRA uses the Google speech recognition engine and Mojolingo RubySpeak API has a company doing technical support for it.

The information for the related API’s can be found at the following URL’s:

Speach Recognition for Asterisk (SRA) –

MojoLingo RubySpeach –

A2Billing for Asterisk

A2Billing is a unique add-on for Asterisk that turns it Asterisk into a hosted VoIP billing solution.  Essentially, you could put an A2Billing server in place and re-selling SIP dial tone to other Asterisk servers and maintain separate clients, billing plans, etc.  There are even calling card functions that allow you to make pre-paid calling cards and bill for them.

A2Billing Portal Screenshot

A2Billing Main Portal

There are hundreds of features for A2billing.  For a complete list, visit

Oreka Call Recording

In large call centers there is often a need to record not just a few calls but hundreds (possibly thousands) of calls.  For this it is often necessary to have a separate call recording appliance for this purpose.  There is a popular open-source software that does just this, and it is called Oreka.  It sits right in the stream of all calls and records the stream of audio data to a file and makes a searchable database of call recordings.

Oreka Admin Interface

Oreka Call Recording Admin Interface

For detailed information about Oreka, please visit.

Asternic CDR Reports

In Asterisk versions with a web-based GUI (FreePBX, Elastix, etc), they often include a basic Call Detail Reporting (CDR) search function.  Unfortunately because of how Asterisk records call records, and the simplistic nature of the GUI’s, it is often difficult to obtain truly useful data from these CDR tools.

Asternic has developed a plugin for FreePBX (also works with Elastix) that reformats all the CDR data into truly useful reports, such as by person, time of date, inbound, outbound, etc in more summary fashion than individual call-by-call (although that is still possible).  Overall it is more useful than the built-in CDR Reports tool.

Asternic CDR Reports

Asternic CDR Reports

For a detailed feature list and more screenshots, please visit.

Areski CDR Stats

The final add-on we will cover is another CDR statistic program, CDR Stats by Areski.  This is a not a plug-in for a GUI, but an entirely separate web GUI.  It has the ability to work not only with Asterisk, but with Freeswitch as well.  It covers not just pure call record reports and statistics, but can also monitor call traffic and send alerts for unusual patterns as well.  It can also “rate” calls (i.e. give a total billed cost for each call by entering the price plan you pay for minutes.  Overall it is possibly one of the best CDR stats programs available.


CDR Stats Dashboard

To view the complete feature list, as well as a live demo, visit

Asterisk World February 8-10

If you are at all interested in Asterisk products and services, make plans to attend Asterisk World in Fort Lauderdale, FL on February 8th-10th.  For more information go to


VoIP Call Recording Notes

We’re asked all the time…”configure my phone system to record ALL my calls”.  Our very next question is “Are you sure you want to record ALL your calls?”.  Clients will typically answer “absolutely” but then not understand what the implications of that decision.

To begin with, exactly how many of these are calls are really necessary to keep?  In all reality, only a small portion are ever listened to.  They are typically kept for “training purposes” or more often for legal “documentation” purposes.  But, if someone is being recorded, there are laws in various states that typically state they must be notified that they are being recorded and for what purpose.  This can be a bit of a hassle to ensure this is taking place.

Not every type of call needs to be recorded either.  Calls between people within a company are never between a customer and employee, therefore it is rare that recording ALL conversations between employees is necessary.

Once you have decided which calls to record, you should then also decide how long to keep those calls you have recorded.  Some state and federal laws require that you keep recorded calls for a specific amount of time for compliance purposes.

Considering disk space limitations it’s not always practical to keep them on the PBX so if you have a high call volume you might want to consider setting up automatic call deletion or archiving. The best way to do this is to have a storage system (typically NAS or a large ftp server) that you can archive the older calls to so you can retrieve them if necessary but not necessarily keep them readily accessible through the PBX or call center software.

Everyone’s call recording requirements are unique, but before you choose to record ALL your calls…consider the factors we have discussed and make your call recording plan.  If you need help, please contact us at


Asterisk Call Quality Troubleshooting

Excellent call quality with any phone system is crucial.  While businesses these days do run on email and web, the telephone is STILL the lifeline of the business. Asterisk as a call processing engine for a PBX is well designed and very efficient, so it has many features that can be used to mitigate the most common problems.  In this article we’ll briefly discuss the most common call quality issues that occur with IP PBX’s and how Asterisk-based systems can deal with him.

NOTE: These tips primarily deal with Internet-based SIP dial tone, except where noted.


This is arguably the most common issue with any IP-based PBX.  It especially occurs at some point on systems with Internet-based SIP trunks for their dial tone.  Jitter sounds like one of two things: Either “interruptions” in the audio, or “underwater sounding voices”.  The issue is caused by inconsistencies in the delay between data packets between two endpoints.  Most people, incorrectly, assume that they don’t have enough Internet bandwidth.  This is generally not the source of the problem (unless the Internet is painfully underpowered for the amount of calls going through at the same time – i.e. dialer traffic).  It has more to do with the QUALITY of the Internet than the QUANTITY of bandwidth.  You can even have this problem on a LAN, where there is NO Internet connection and plenty of available bandwidth, but the LAN is congested or of poor performance.

With a poor quality Internet connection where there are inconsistent delays between packets, there isn’t a lot you can do with the provider.  Once it leaves your network for the Internet, its out of your control.  HOWEVER, Asterisk (and other IP PBX’s) have a function called a Jitter Buffer. A Jitter Buffer takes a portion of the audio and “buffers” it (stores it in memory briefly) before sending or while receiving.  This has the benefit of removing or eliminating the jitter.

Occasionally a possible cause of jitter is processor overload on the firewall or switches.  You can use your firewall or switch management tools to see if they are overloaded, then either remove some of the load, or upgrade them and see if the jitter is reduced and call quality improves.

One Way Audio

One way audio is a particular problem  that is usually seen when a PBX is first installed.   What it sounds like should be pretty obvious, based on the name.  Only one party will hear the audio portion of the call.  The call will ring, but when the destination party picks up, the receiver hears the call, but can’t talk back to the originator.

This problem is due to a misconfiguration of the Asterisk PBX, the firewall between it and the provider, or both.  When a firewall sits between an Asterisk PBX and an Internet SIP provider, all packets go through Network Address Translation (NAT).  In other words, the private IP address in the network packet is replaced with the public IP address of the firewall when the packet travels out, and the reverse happens on the return.

The issue specific to SIP is that a SIP packet is unique in that it has TWO places for the IP address, and a firewall typically isn’t smart enough to look for the second IP address (located in the SIP header of the packet).  So when a SIP provider tries to send its traffic back to the firewall that sent it – it is going to try to send it to the IP address listed in the SIP header, which is a private LAN address can’t be routed over the Internet.

Some firewalls have a function called an Application Layer Gateway (ALG) – and it goes by different names by different manufacturers (SIP ALG, SIP Fixup, SIP-NAT, and so on), but they all basically do the same thing – look for SIP packets and get involved to do the SECOND IP address NATing.

Unfortunately, most firewalls do this incorrectly with Asterisk-based systems.  Asterisk has a function built into its SIP configuration for automatically taking care of this issue.  The function uses 3 parameters in it’s sip.conf configuration file (the location and use of which depends on which version and implementation of Asterisk you are using).  These parameters are:

  • NAT=yes
  • localnet=<local network definition>
  • externip=<external NAT’ed IP address of the PBX>

When these three parameters are configured correctly, the PBX automatically takes care of the NAT substitution for packets destined for Internet hosts.

NOTE: Because SIP ALG’s generally aren’t smart enough or often don’t work properly with Asterisk – we highly recommend turning them OFF and properly configuring Asterisk as described


Troubleshooting echo issues isn’t typically an issue with IP PBX systems using SIP providers, it is more commonly seen with systems that use more traditional connections to the telephone company.  Connections such as PRI, or more commonly, analog lines.

Echo, as you may guess, is simply a repeat of some portion of the audio back to either speaker.  There are two typical causes.  The first is easiest to solve.  Often, someone will be calling on a speaker phone with the volume turned up significantly and a portion of the audio is “feeding back” into the microphone.  The person on the other end of the call will hear a portion of the audio delayed by a few milliseconds.  The solution for this issue is straightforward, simply turn down the speaker volume and see if the quality improves (or pickup the handset and see if the problem goes away completely).

The more complicated cause is what is known as a “hot” volume on a trunk.  That means that the incoming voltage is too high for the interface card to handle and it must be adjusted down to compensate.  If the incoming voltage is too high, the caller from the outside will receive a portion of the signal echoing back.  If the outgoing voltage is too high, the caller from the inside will receive a portion of the signal echoing back.  If both are too high, both parties may receive echos.

Asterisk has the ability to adjust “gain” settings for its interface cards that can reduce or eliminate this echo if this is truly the source.  The parameters are “txgain” and “rxgain” located in the DAHDI or ZAPTEL configuration settings (depending on the technology used).

Some interface cards also have built in hardware to automatically eliminate echo on problematic lines.  If you are using interface cards without automatic echo cancelling, consider replacing them with cards with automatic echo cancelling cards.  Be warned, however, they are more expensive than their non-echo cancelling counterparts.

It is important to note that the amount of echo is also dependent on the volume of the person speaking.  If the echo problem is borderline, and the person speaking starts talking louder, then the echo may begin at that point because it has passed the threshold.


This article should give you a few items to look at when troubleshooting call quality issues on your Asterisk-based system (as well as any IP-PBX for that matter). For assistance with eliminating any of your call quality issues, feel free to contact us or email us at

FreePBX IP Phone Comparison

FreePBX is very flexible when in comes to which phones it can use, especially with the Commerical Endpoint Manager.  There are currently more than 20+ brands of phones and devices that can be auto-provisioned with FreePBX.  HOWEVER – just about ANY standard SIP-based phone or device can be manually configured, including software-based phones that run on desktop computers or mobile devices.

In this article, we’d like to briefly cover our experiences using FreePBX with specific brands of phones.   This is not meant to be an absolute comparison, just what our experiences have shown over time with our customer base (primarily small-to-medium sized businesses).

Here are some of the top brands, listed in alphabetical order.


Aastra 57i

Aastra 57i SIP Phone

Right off the bat, you may not have necessarily heard of Aastra as a major phone manufacturer, but they manufactured phones for Nortel Meridian systems, so you have more than likely seen a couple of their products.

For the most part, Aastra phones are a good balance between quality, reliability, features, and price.  Their 53i,55i,and 57i (pictured above), are the workhorses of many FreePBX installations because they often offer the most “bang for the buck”.  However, many people feel that the Aastra phones are essentially compromises in most areas and thus the phrase “Jack of all trades, master of none” would seem to apply.

Incidentally, Mitel purchased them in January, 2014.  So new Aastra phones are now Mitel phones.

Aastra IP Phone Positives:

  • High “Bang for the Buck”
  • Clear backlit LCD display on most models
  • Unique XML call control language allows for many different API uses

Aastra IP Phone Drawbacks:

  • Low handset volume complaints
  • Low speakerphne audio quality
  • “rubberized” keypad buttons – often not liked by high volume phone users


If there is one “500 pound gorilla” of the IT world – it is Cisco Systems.  And yes, they do make IP Phones.  However, they do make two different types of IP phones, the Cisco Small Business series and the Cisco Unified IP series.  The primary difference is that Cisco Small Business phones are SIP ONLY, whereas the Cisco Unified IP series can either use SIP or SCCP firmware (SCCP firmware is for use with Cisco’s Unified Communications Manager (UCM).  But, either series of phone will work with FreePBX.

The Cisco Small Business series of phones pictured below are based on technologies purchased from LinkSys.  As a matter of fact, they used to be LinkSys phones.  They are fairly inexpensive new, and have all the basic features required in a phone system.  As a matter of fact, they are a pretty rugged and feature filled phone.


Cisco 504G

Cisco 504G Phone


Cisco SPA962

Cisco SPA962 Color IP Phone

The Cisco Unified IP series phones (7960 phone pictured below) are much more robust and are somewhat of a “status symbol” phone.  These phones are seen everyone in corporate america and if you pay attention, movies and television.  Their key benefit is they are essentially “bullet-proof”.  Once you get them working, they just keep working until they mechanically fail somehow.

Cisco 7960

Cisco 7960 IP Phone

Cisco IP Phone Positives:

  • VERY reliable
  • Nice design
  • Great sound

Cisco IP Phone Drawbacks:

  • Difficult to get to work properly with Non-Cisco equipment (such as FreePBX)
  • Expensive
  • Not supported by Cisco for use with Non-Cisco equipment


One manufacturer who has been working with FreePBX systems the longest is Grandstream.  For the longest time, the GXP-2000 was considered the workhorse in the FreePBX phone system stable.  It was cheap, it did its job well, and was relatively easy to configure.  The latest Grandstream models (like the GXP-2160) are no exception – although as with older model Grandstream phones, their features can be a bit “clunky”, but they still do the job and do it well.  The build quality has increased in the newer models as well.


Grandstream GXP-2000



Grandstream GXP-2160

Grandstream IP Phone Positives:

  • Inexpensive
  • Fairly rugged

Grandstream IP Phone Drawbacks:

  • Some features “Clunky” to use
  • “Rubberized” buttons on older models not friendly to power users


Polycom Conference Phone

Polycom IP6000 Conference Phone

Polycom as a telephone equipment manufacturer is known for making the best conferencing equipment available.  For the longest time, they were the market leader in both video and audio conferencing.  The technology used in their award winning conferencing solutions has carried over into their line of IP phones.  Polycom phones are considered to have the best quality speaker phones, bar none.

Polycom IP 330 Phone

Polycom IP330 Phone


Polycom carries a wide range of IP phone models for a wide variety of phone systems, but their SIP phones are all compatible with FreePBX and are probably one of the best brands to use with a FreePBX system.  Their entry level Soundpoint IP 330 was the “General Desktop Phone” used in companies for years.  The SoundPoint series of phones has been replaced by the VVX series which includes not only Gigabit Ethernet, but in some cases color touch screen as well.

Polycom VVX 500 Phone

Polycom VVX 500 Phone


Polycom IP Phone Positives:

  • EXCELLENT sound quality
  • EXCELLENT build quality
  • Large number of customizations configuration options available for phones
  • Nice range of phones available, from entry level to high end, color touch screen phones

Polycom IP Phone Drawbacks:

  • On average they are more expensive than most other brands, but because of their abundance in the market, refurbished models can be found
  • Fairly complex XML configuration language that can prevent proper configuration of the phone if the firmware and the FreePBX configuration setup do not match



Now that FreePBX has been acquired by telecom giant Sangoma – there are a lot more resources available, including a phone SPECIFICALLY designed for FreePBX.

Sangoma S700 Phone

Sangoma S700

Sangoma has a complete line of phones specifically integrated into FreePBX (although it requires the commercial Endpoint Manager module to provision them).  The phones have the ability to “zero-touch” provision as well as work natively with many of FreePBX’s features.

Sangoma IP Phone Positives:

  • DEEP integration with FreePBX – including REST Apps
  • Easy setup with FreePBX
  • Excellent sound quality
  • Beautiful interface
  • Easy provisioning with Sangoma portal

Sangoma IP Phone Drawbacks:

  • Expensive price point
  • Requires commercial Endpoint Manager module


Snom is a German company that specifically designs SIP phones.  SNOM phones are designed to work universally across all SIP systems.  They are popular with system designers because they not only have a lower price point than most other phones with similar features but also have a complex feature set accessible through XML configuration files or through a web administration portal.


Snom 710

Snom 710



SNOM IP Phone Positives:

  • VERY inexpensive
  • Great for developers because of XML features

SNOM IP Phone Drawbacks:

  • Marginal build quality
  • Ring tones are strange and most clients do not like


Entry level Yealink phones are almost on par with Grandstream in terms of their ruggedness, just a bit less well known.  However, their button quality is higher.  But the high-end Yealinks really do have a lot of nice features and truly stunning displays.  However, where they fall short is in provisioning.  They are truly abysmal to provision in bulk.

Yealink T21P

Yealink T21P

Yealink T48G

Yealink T48G

Yealink IP Phone Positives:

  • Inexpensive
  • Rugged
  • Nice buttons
  • Excellent audio quality

Yealink IP Phone Drawbacks:

  • Difficult provisioning


Based on the information provided here, you can choose your own brand of phone based on your particular needs, however, it is worth noting that by FAR the most popular phone used with FreePBX phone systems is Polycom.  Their great feature set, combined with relatively low cost and excellent secondary (used/refurbished) market make them an excellent choice.  Followed closely by Grandstream and Yealink.


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