Archives for vicidial

Open Source Call Center Platform Comparison

Let’s face it, its expensive to start and run a call center.  Open Source-based call center software is very popular because of its low cost of implementation when compared to commercial options.  In this article we compare the top Open Source call center platforms that are available.  To be absolutely clear, this means a complete platform.  NOT just the ability to make and take calls.  The system must have an agent interface to manage the call and data, an admin interface to design call flows, system setup, and to report reports,  as well a live dashboard of some sort to view call center activity in real time.

We’ll briefly cover the primary benefits and drawbacks of each and show brief examples of the “look” of each of the primary functions just listed.


Vicidial is arguable has the largest installed user base of any Open Source-based call center platform.  In fact, it is so well designed that many of the other of those discussed in this article are actually BASED on the code written in Vicidial.  This will be very apparent when you look at the graphics if any call center is based on Vicidial.  Typically the agent interface is the main component that is changed and the admin interface remains basically the same with only a few minor changes.

Vicidial is the leader in open source call center software for many reasons.  First of all, it just WORKS.  Once set up, it is not uncommon for Vicidial-based systems to work for years without intervention or service.  Additionally, it contains all the most popular features necessary to run an effective call center, such as auto-dialing, inbound routing, caller ID management, live dashboards, agent statistics, call recording and so on.  Also, it can scale to hundreds (in some cases up to 1000+) users with multiple, redundant servers providing 99.99% uptime (assuming similar redundant networking and power hardware).

The biggest complaint people have (and really the only complain of significance) of Vicidial is the “look” of its agent interface.  It could use some “prettying up”.

Below you can see examples of the agent interface, the administration interface, and the live “Wallboard” screens that come with Vicidial.

Agent Inferface Admin Interface Live Dashboard
vicidial agent interface Vicidial admin interface vicidial live dashboard



OSDial is one of the “forks” of the Vicidial code, is OSdial (  It was designed to make the Vicidial interface simpler and a bit easier to use and a bit less “clunky”.  It maintains the same basic code base and most of the same functions as Vicidial (inbound, outbound automatic, and manual call handling, reporting, etc).  Therefore, if you are familiar with the operation of Vicidial, you can administrate and operate an OSDial implementation.

However, because of the “improvements” made, there are some drawbacks.  First of all, the code base, while based on Vicidial, is maintained separately.  Therefore, updates to the Vicidial code don’t make it to the OSDial system until quite a bit later (often times up to a year or more – if at all).  Additionally, some functions may be left out and left as the Vicidial “defaults” in hopes that you will fall into the 90% of customers who don’t need to change those settings.  If you do, you would have to manually modify config files, or database settings, or other items (a difficult proposition at best).

Below you can see examples of the agent interface, the administration interface, and the live “Wallboard” screens that come with OSdial.  Note how they compare with Vicidial’s (shown earlier in this article.)

Agent Interface Admin Interface Live Dashboard
osdial agent interface osdial admin interface osdial live screen



GOAutodial (  is yet another attempt to put a “pretty face” on Vicidial.  However, it actually includes a few additional utilities for server management that make it a more rounded platform (like database management tools, and other server management tools).  Additionally, the “live wallboard” is much more polished than other flavors.  It also offers wizard-based configuration for many of the more common tasks, like configuring campaigns, which makes administration easier for call center managers.

It still suffers from the same basic problem as OSDial.  That is, it is based on the Vicidial code base, and as such, it can’t make updates until the Vicidial code base does.  And once Vicidial releases code, GOAutodial must test it and integrate its code into its own system.  But, again, it still doesn’t have as many features or the quick release of features, or broad support based that Vicidial does.


Agent Interface Admin Interface Live Dashboard
goautodial agent interface goautodial admin interface goautodial live report

Asterisk With QueueMetrics

Queuemetrics ( is a unique piece of software in this category.  It is designed primarily to work with Asterisk queue based systems and as such, mainly handles INBOUND call distribution and reporting.  However, there are add-ons to enable it to function as an outbound call center as well.  Another difference is that it is one of the few call center platforms that is not based on the Vicidial core engine.  It uses Asterisk’s own queue management and tracking, then expands on it with a database connection and agent tools.

Agent Interface Admin Interface Live Dashboard
Queuemetrics Agent Interface Queuemetrics Admin Interface Queuemetrics Realtime Wallboard


Vicidial Avatar Soundboard Feature

The Vicidial open source call center platform in its latest release (Version: 2.12-11 Build: 161111-1646) has included a feature long awaited by off shore outsourced call centers: An Avatar Soundboard.

For those not familiar with Avatar call center software, the concept is simple.  Essentially, instead of a live operator speaking directly to a customer on the phone, instead the operator will play pre-recorded sound files back to the customer on the phone.  This is useful in many cases, including agents that may understand english speakers, but have thick accents  that are difficult to understand by customers over the phone.  Additionally, by using Avatar agents, recorded responses are universal in voice and tone and thus can be planned in advance.  Often, Avatar campaigns are used as “opener” campaigns to dial through bulk leads to qualify them, before transferring them to closers.

Vicidial’s Avatar Soundboard (shown below) can be configured with pre-recorded responses, organized on a single page that allows the agent to simply click on the appropriate response to play the associated sound back to a customer.

Vicidial Soundboard

A Sample Vicidial Soundboard in the Agent Interface


If you’d like assistance setting up a Vicidial Avatar Soundboard system, or any other Vicidial related item, please contact us or email us at

Verterion Announces Vicidial Integration For Hubspot CRM

Verterion has just completed design and implementation of its first integration of Vicidial with Hubspot CRM.  This integration connector automatically:

  • Logs agent calls in Hubspot
  • Records call dispositions in Hubspot
  • Post individual call notes to Hubspot
  • Play Vicidial recordings from within Hubspot
  • Posts call events in contact timeline
hubspot call

An example of a call sent to Hubspot CRM from Vicidial

Additionally, if you are using the Chrome browser, there are plugins available that we can use to implement click-to-call functionality.  If you currently use Hubspot as your CRM and Vicidial for your call center, please contact us today 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

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

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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