Compliance Update: FCC Solicits Comments to Redefine “Expressed Consent”

The FCC has released a Public Notice soliciting comments on the most recent petition by Craig Cunningham and Craig Moskowitz seeking to redfine the “express consent” standard for non-marketing automated calls and texts to wireless phones.  This would primarily apply to automated collections calls, informational calls and the like.  In the past, calls like this have been allowed without written consent so long as the recipient had previously provided their cell number to the company, for the purpose of “non-marketing” activities.

The FCC petition argues that “express consent” should be in writing, list a specific phone number and specify what dialer technology will be used to make the call.

The petition, if granted, will seriously affect the way collections companies do business and other companies who send non-marketing messages to people who provided their numbers voluntarily and who have not opted out or had their numbers reassigned.

Contact us at for more information.

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


Choosing a Proper “Dialer Friendly” SIP Provider

When people that are new to setting up a outbound call center choose a SIP provider for a Vicidial (or similar) based call center, they just choose a SIP provider based purely on the lowest cost per outbound minute.  But there are MANY more factors to consider.  With many of our customers, we have consulted with them after they have signed a multi-year contract only to find that the provider they have selected won’t fit their needs.

In the following article, we will outline a few of the factors you need to consider when choosing a SIP provider for dialtone for your outbound call center.

True Cost Per Minute

I realize that in the first paragraph it may seem like I said this wasn’t the prime factor, but in fact it IS an important factor.  HOWEVER, there’s more to it than just raw cost/minute.  An average cost for an outbound call in the contiguous United States (i.e. not Alaska and Hawaii) is $0.015/minute.  If you are calculating all your outbound dialing costs on that figure, it would be simple, however you can get better rates for dialing with what is known as “rate deck” plans.  That is to say, calling a certain state might cost you $0.008/minute, but calling another state would cost you $0.017/minute.  Depending on your target regions – you might be able to save money on your outbound costs by doing a little research.

Additionally, the billing method is important.  Some carriers will bill you for the first 30 seconds of a call, then bill you in increments after those first 30 seconds.  Others will bill you purely in 6 second increments.  If your dialing patterns give you a lot of short calls interspersed with a few longer calls (as most dialing traffic typically is), this is most likely the best option.

Short Call Duration Penalty

One little billing fee most customers are not aware of that exists with many SIP carriers, is what is known as a “Short Call Duration Penalty”.  Dialer SIP traffic is unusual.  Phone companies don’t make a lot of money from it because the bulk of the calls don’t connect and thus can’t be billed directly per minute (unless its written that way in the contract).  For that reason, many non-dialer-friendly carriers impost a the short call duration penalty when the percentage of calls that don’t connect versus those that do goes over a certain threshold.  This can add up quickly for larger call centers.

Calls Per Second Capacity

Here’s an area that people who don’t do their research can get stuck in very quickly.  An autodialer is designed to send out lots of calls at the same time, and with some dialers, there can be hundreds of calls going on at once, with more being added each time a call gets disconnected.  So the maximum calls-per-second, or CPS, rate is important for how quickly a dialer can recover from dropped or disconnected calls thus keeping agents on the phone and a call center productive.

Maximum CPS starts around 1-2 CPS and goes up from there.  In call centers with simultaneous calls in the hundreds, this number should be greater than 10 CPS.  This is where the smaller carriers fall short.  Most smaller SIP carriers can only handle a MAXIMUM of 5-10 CPS (although of course this varies from carrier to carrier).

Call Setup Time

One of the most crucial metrics for a outbound dialing is the time it takes from when the dialer initiates the call to when the call starts ringing on the other end.  This may not seem important, but it affects the outbound call center metric known as “drop rate”.  If the call takes too long to set up, the dialer may treat it as a failed call, and mark it as “dropped” and move on to the next call.  The problem is that this also affects a parameter known as “contact rate” that affects how fast the dialer has to run in order to keep agents on the phone.  What does all that mean?  Essentially, if the SIP carrier you choose has too high of a call setup time, it will take longer to make sales or cost more to make sales because you have to make more calls than with a carrier that has a lower call setup time.

Often, this occurs with carriers who claim they can give you the “best possible rate” and come back with an outrageously cheap price (I’ve seen quotes of $0.004/minute – that’s less than a half a penny per minute).  These carriers can do some funky stuff with the traffic (like send it offshore, and route it back through different carriers, or use funky routing, etc) to get these rates, but consequently they will have much higher call setup times.

The best way to get the best call setup time, is to go with a recommendation from a currently operating call center.  Or contact us so we can give you the benefit of our expertise with various SIP carriers friendly to dialer traffic and help you select one.

Inbound Number Availability

Surprising as it may sound, there are in fact a few “pro-dialer” carriers that meet the above criteria swimmingly, but do not have the ability to provide numbers that can be called back and routed back into your call center.  This is a major problem because the large majority of outbound calls go unanswered and if a valid number is on the caller ID, the person who was called could call back in and a possible sale could be made.

Focusing on the quality and cost of the outbound calls is important, it is just as important that the people you call can call you back, and that your SIP provider will facilitate that.

Number of Supporting “Backbone” Carriers

Let’s face it, there are literally hundreds, if not thousands, of SIP providers available today.  Each are competing for a share of the dialer market.  Some smaller carriers though are “mom-and-pop” operations.  While there is nothing inherently wrong with that, few have the capacity to hand the traffic that dialers generate.  Most often a small SIP provider will have a single connection to a major carrier.  The upside is that they can usually get pretty good rates to pass on, the downside is that they can only handle so much traffic.  Plus, if that one connection to their provider goes down, that’s it.  All their clients loose their ability to make calls.

Single Carrier SIP Provider

SIP Provider With a Single Carrier

Some of the better carriers have multiple connections to other providers to avoid this problem.  The problem is that many of the carriers they connect to, aren’t what are known as “backbone” carriers (i.e. the “500 pound gorillas of telecom – Level 3, Verizon, AT&T), but instead are just other more regional carriers.  They can handle more volume, and are more redundant, but they do have a higher cost factor, and they typically have a much higher call setup time than a smaller carrier.

The best carriers have multiple, direct connections to the backbone carriers (as shown in the image below).  They can maintain a high call volume, keep rates relatively low (although admittedly will be higher than the “mom-and-pop” operations), have extremely low call setup times, and have multiple routes for redundancy.

multi homed SIP provider

Multi-Homed SIP Provider

Local Caller ID Presence Numbers

These days, when sales calls are made, the person on the ringing end of the phone can usually see who is calling by use of caller ID.  But, as most of us who have received telemarketing calls know, if its a toll free number (800, 844, 888, etc), generally speaking it is a telemarketer or someone you don’t want to talk to.  But sales call centers need to reach people to make sales.  So what they often do is buy “local” DID numbers that point back to their call center from anywhere in the country.  Then, the call center will setup their dialer so that when they call a number, the dialer will send out a caller ID that is “local” to the caller.  For example, if the customer’s number is 909-111-1111, when the dialer dials that number, it will recognize the 909 area code, look up in its database of DID’s, and send out the caller ID of a phone number that points back to the dialer (say 909-222-2222 for example).  The caller is more likely to pick up, or if they miss the call, call them back. Using this tactic greatly increases contact rates.

NOTE: Vicidial is one dialer that has this capability

The issue is that most carriers require that you purchase INDIVIDUAL phone numbers in EACH area code.  With over 290 area codes in the US, that’s a lot of phone numbers to buy one at a time. Not only that, but people eventually will recognize these local numbers as telemarketing numbers and start blocking them, so you would have to buy new numbers every few months.  So while it has benefits, if not done properly this approach can be costly.

A good dialer SIP provider will have a “local presence” DID package (name of the package varies by carrier) that will provide local phone numbers in each area code.  Some providers even include FREE minutes with each number.  Additionally, some providers will allow you to “refresh” this package a few times a year to get new numbers to keep them working.


When choosing a SIP provider for your dialer call center, remember that cost is only one factor.  Multiple factors contribute to the proper interoperability of a SIP provider and a call centers and all these factors should be considered when making your choice.

For more information on SIP providers, Vicidial, call center design and implementation or help choosing a SIP provider, please contact us at

Vicidial Scaling and Fault Tolerance Notes

One of the most impressive features of Vicidial is its ability to scale, while at the same time enabling fault tolerance.  It has the ability to scale from a few users, to hundreds of users and thousands of simultaneous calls.  This is accomplished through the multi-server architecture ability of Vicidial and the ability to separate the various components of Vicidial onto separate servers and provide redundancy.

Single Server Setup

In the figure below you can see a basic Vicidial installation.  In the most basic configuration, you would typically have less than 50 agents which would only require one server.  Since the installation is so small (and most likely the call volume is low as well) you would most likely only have a single dial tone provider.  All of the services would run on the single server and all of the agents would connect to the same server.

Vicidial Single Server Installation

A Vicidial Single Server Installation

This configuration is the simplest, easiest to configure, and the cheapest, but subsequently the least fault tolerant.  The performance limits are determined by the hardware of the single server.  It has been our experience that while the recommended configuration is a maximum of 50 agents for this configuration, with a low volume call center, it is possible to increase that number to as much as 100 with some parameter tweaking.

Multi-Server Setup

As the call volume and number of call center agents increases, the number of servers required can be increased to handle the increased load.  We have built call centers with 300+ agents that do outbound dialing at a ration of 5:1 – meaning 1500 simultaneous calls.  And the Vicidial group has documented cases of even larger installations.


Vicidial Multi Server

A Vicidial Multi-Server Installation

The multi-server installation requires several components:

  • Primary Database Server – This server contains the master operational database and is the “key” server for the proper operation of the system.  It should be the fastest server with the most RAM in the entire group of servers
  • Secondary Database Server – This server serves as a backup for the primary database server and is used for redundancy, as well as to offload report generation from the Primary server to increase performance
  • Web Server – Because Vicidial is a web-based call center system, a high performance web server is key to proper operation.  In smaller setups, this can be handled by the same server as the database server, but as web traffic increases, it is best to offload the web traffic to a separate server.  The added benefit is that a separate web server is more secure if the Vicidial system has to be exposed to the Internet.
  • Network Attached Storage (NAS) Server – Vicidial has the ability to record EVERY call.  This will take up a very large amount of disk space.  A NAS device can provide a large amount of storage at a cheap price where you can store all your recordings in a central location and access them through the Vicidial interface.  Additionally, you can use the NAS device to store backups of each server’s configuration.  Finally, as an added benefit, many NAS device manufacturers have the ability to sync any storage in the device to cloud services (such as Amazon), thus making all the information available anywhere.
  • Telephony Servers – This is where the “heavy lifting” is done for Vicidial.  These servers are responsible for making and receiving the phone calls within Vicidial.  Some of these servers can also be dedicated to hosting specific agents to balance agent load across multiple servers.  Here’s where the magic truly happens.  Need more agents or more calls?  Simply add more telephony servers (assuming you have the database and web server infrastructure already in place).  The other great part is that these servers require the least amount of hardware and disk space since all they are doing is making phone calls, essentially.  Plus, if one of the telephony servers dies, the system as a whole is only mildly affected.  At the very most, a few agents will not be able to log in and dialing capacity will be reduced.  But the rest of the system will continue to operate

In order to get this level of performance and fault tolerance in a proprietary system, it would cost hundreds of thousands of dollars.  This type of setup can accomplish that in a fraction of the cost.


As you can see, Vicidial has the ability to scale from small call centers to very very large systems when designed properly.  Contact Verterion today at for more information about designing or expanding your Vicidial system.

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

Verterion Designs New Rapid Deployment Network and Telephony Platform

Verterion has completed design and implementation work on a rapid deployment IP phone system and network infrastructure for Phone Guys of Huntington Beach, CA.  The system is designed to be implemented in areas and buildings that have no cabled infrastructure but need networking and telephony immediately.

The system is rack mounted in a aircraft-shippable case, and can be shipped anywhere in the country and deployed immediately.  It contains a complete wired and wireless network, server, PBX, and Internet and phone connectivity appropriate to the location and conditions.

More information, contact 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.


Why Use Vicidial for Your Call Center?


Vicidial is an excellent choice for both small and large businesses to use for their call centers.  In this article we will examine the reasons why.

Vicidial has been around for many years.  It originally started out as a GUI for Asterisk to be used in call centers (hence occasional references to ASTGuiClient in docs and file nameses).  But because of its ease of use, reliability, and extremely low cost of operation it has become one of largest installed base of call center software.

Let’s take a brief look at why Vicidial is such a popular choice for call centers.

1. Vicidial is VERY inexpensive

From a management perspective, the software is FREE.  Those that want to delve into the finer points of the Open Source licensing Vicidial uses will understand that “Free” comes with many caveats (e.g. you can’t sell it, you must maintain original credit in the code if you modify it, etc etc etc), but from a business owner’s perspective, they don’t pay anything for the software.  Download it.  Install it.  Done.

Furthermore, since Vicidial is Linux-based software, the hardware that Vicidial requires to run is not only off-the-shelf PC hardware, but only has minimal specifications (see hardware specifications link here).   Rather than spending tens of thousands on a proprietary dialer or inbound call center system, a single server class machine could handle 50+ agents or more and cost less than $3000 (even less if refurbished equipment is used).

2. Vicidial is VERY reliable

As previously mentioned, Vicidial runs on Linux.  More specifically, it runs on Linux with a collection of supporting software packages, all well known, documented, and robust: Apache (webserver), MySQL (SQL Database), PHP (Programming Language), and Asterisk (Telephony Engine).  Working together, these pieces of software provide a system that is VERY reliable.  While the overall system reliability is heavily dependent on the hardware it is operating on, because redundant systems can be built to spread the dialing and call load over multiple redundant servers, the system can be made to be up to 99.999% reliable

3. No Proprietary Hardware or Software Components

No proprietary hardware or software is required at ALL to use Vicidial.  Everything from the hardware it runs on, to the agent computers, to the headsets are non-proprietary, and therefore, inexpensive.

4. The System Can Be Rapidly and Cheaply Scaled

Most proprietary call center systems require that you purchase additional licenses and possibly additional proprietary hardware modules to expand as your call center grows.  Vicidial requires only that you purchase additional, relatively cheap, standard servers to handle the growth in call and agent volume.

5. Easy Development and Integration

Because the architecture of Vicidial is completely Open Source and based on standards, it is extremely easy to integrate it with other software without learning complex API’s or additional development languages.

6. Multiple Contact Points between Customer and Agent

With the latest version of Vicidial, agent sessions are not limited to phone calls.  It is also possible to hand chat and email communications in the same distribution patterns as phone calls.  This would allow a staff of agents to handle all inbound communications more efficiently than a separate group of individuals not functioning as a “contact center”.

7. Constant New Feature Rollout and Updates

Because Vicidial is Open Source, there is a very large development community with a vested interest in maintaining it.  New features are always being tested and released.


Those are some of the largest reasons you should take a long look at Vicidial.  For more information on Vicidial, or how to get started with your implementation, contact us or email us at


Verterion Announces Vicidial Agent Performance Dashboards 

Verterion has developed a custom package that will allow call center managers to track individual daily performance INTEGRATED with various CRM software packages (both hosted on premise-based)

Vicidial Custom Dashboard

Sample Dashboard

Managers and agents can now see on a “wallboard” how their agents are doing on a daily and weekly basis with respect to ACTUAL sales performance, not just Vicidial disposition statuses.

If necessary the dashboard can even integrate with other operational functions such as live “floor bonuses” and commissions displays.

Contact us at for more information

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