Vicidial

Vicidial vs PBX In Call Center Operations

Deciding which platform to use to run a call center is key.  The phones, after all, are what bring in the money in a call center.  The Vicidial open source call center software is SPECIFICALLY designed for call centers.  PBX phone systems, however, are more “general use”.  That is they can perform many of the same call routing functions, but in a more general way.  Many call center owners will try to use traditional PBX phone systems in their call centers, but often find the features limited for what they actually want to do.

In this article, we will discuss the various similarities and differences between Vicidial and PBX systems as they relate to the operation of a call center

Components and Call Flow

Arguably the largest difference between Vicidial and most PBX systems is the components required to make the call center, and the flow of the calls.  PBX-based systems are more complicated because they use separate systems are are difficult to integrate with other data systems.  Vicidial is designed to integrate both telephone calls and data systems together from the start.  Lets take a look at each of these separately.

Components

In a PBX-based call center (whether inbound or outbound), typically you will need:

  • The PBX server or appliance for call processing
  • Telephone handsets (most often with headsets)
  • Computers for data access and entry
  • A server that contains customer data about who might be calling in or who you are supposed to call

The key thing to note here is that there is a MINIMAL level of integration by design (as shown in the figure below).  Inbound calls will allow the PBX to ring the phones separately and, in most cases, requires the agent to manually look up a phone number.  Or on outbound calls, require the agent to look up a number to be dialed and dial the numbers on the phone manually (or possibly use a click-to-dial software) before connecting the call.  Suffice to say there is a lot of MANUAL work to be done by the agents with the separated components in a PBX-based call center.

Additionally, the primary interface to the phone system for the agent is the phone.

PBX Call Center

PBX Call Center Components

In a Vicidial-based call center,

Vicidial Call Center

Vicidial-Based Call Center

Call Flow

Another big difference between PBX-based call centers and Vicidial-based call centers is the way calls are routed to agents.  In PBX-based systems, often the logic is limited to simply calls coming in and ringing a group of phones in a specific pattern.  Most PBX systems can apply “Weight” to individual phones (note not “people”, but “phones”) to allow specific phones to answer calls with a higher priority than others.  But that’s just about where the logic stops.  It’s very difficult for an inbound call to be routed with advanced logic.  And with PBX systems, outbound calls are pretty much “pick up the phone and dial the number”.

However, with Vicidial, a VERY large number of customizable advanced call flow functions are available for both inbound and outbound calling logic.  For example, to have a call routed to a specific agent or group of agents, based on the geographic area they came from, who their “area rep” is, the call routing criteria looked up from an external source, filter out unwanted inbound calls and redirect them to another source, automatically NOT call someone based on the disposition of the last call (or route it somewhere specific).  These are just some of the default logic pieces.  Functions are also opened to allow custom call routing routines to be written that haven been thought of, that might be required for a specific call center.

Agent Performance

A key metric for the effectiveness and proper operation of any call center is agent performance.  This metric can mean a lot of different things depending on the call center, but in a general sense, it means how effective the agent is at doing their job, i.e. handling phone calls and the data operations that go with them.  While we will get to the discussion of measuring agent performance with reporting in the next section, we would like to mention just one primary fact here: time.  In a “best practices” installation, agents do not pick up a ringing “phone” – the call simply “pops” into an established web session and the agent sees a “live call” on their screen and hears a sound in their ear.  All telephony actions are routed at the same time as the data and handled by the SERVER and the agent has little choice of which calls to answer.  This primary benefit of this is that agents answer calls QUICKER, and operate as a TEAM to answer calls.  Additionally, customers will go from waiting 15-30 seconds for someone to answer a ringing phone to 5-10 seconds maximum.

Based on our implementation gathering statistics over the past few years, for those call centers that have switched from a PBX-based call center to Vicidial, some interesting agent performance statistics have been found (these are averages):

  • Average answer time reduced from 30 seconds to 12 seconds (60% reduction)
  • Average calls per day handled increased 25%
  • Volume of outbound manual calls made increased 15%
  • Volume of outbound auto dialed calls made increased 95% (most PBX systems don’t have an auto-dialer – those that do are low performance)

Reports

Reports are extremely important to a call center.  They allow call center managers to ensure the center is performing at its maximum efficiencies.  One of the LARGEST strengths of Vicidial over a PBX is the sheer amount of call center data that can be gathered from the system because of its feature set and its tight integration of data and telephony.  In a general sense (and depending on the make and model of PBX system) the majority of the reports gathered from PBX systems are call-based and extension-based.  For example, you can see how many calls came in to a certain phone number, and to which EXTENSION they went to, and how many calls were answered within a specific amount of time.

While Vicidial can also see this information, it also adds MUCH more information to this.  Because dispositions are recorded at the end of each call as to the nature or result of the call, you can track how many calls of what TYPE came to a specific phone number, thus tracking sales percentages by phone number, or break out calls in even more detail.

Additionally, there are reports included by default for things such as:

  • Inbound calls per date range (including queue destinations grouping phone numbers together and disposition summaries)
  • Outbound calls per date range (including grouping disposition results)
  • Agent reporting by agent and disposition
  • Agent sales reporting
  • Hourly volume breakdowns
  • Server usage performance
  • Agent fronter/closer reporting
  • Outbound list calling reports
  • Outbound list calling performance
  • Timeclock reporting
  • Agent activity detail (login/logout/pause time and type coding)
  • Individual carrier and server performance (in multi-carrier and multi-server systems)

And those are just the default report CATEGORIES – there are at least 20 default reports – each of which can be automatically run and emailed each day if necessary.

Additionally – because the reporting data comes from a SQL-based database, you can create custom reports or dashboards using any number of the dozens of desktop or web based reporting tools available.  It is also possible to integrate dashboards from the call center data and other SQL-based CRM and ERP systems to get integrated performance metrics.

Overall Expenses

This is where things get a bit more complex.  It comes down to what type of call center you run as well as the size of it.  Some call centers simply can’t be done with traditional PBX systems (I.E. automated dialing).  For the purposes of discussion, we’ll keep it fairly simple to the “up front” costs of hardware and installation, as well as the ongoing costs of administration and maintenance.  We’ll assume that fixed costs like computers and software, as well as variable costs like labor and telephony costs should be the same.

Hardware and Installation

In a traditional PBX systems, you would have a PBX server and individual telephone handsets for each agent (as shown in the diagram earlier in this article).  If new, IP telephones are typically around $100 (on the low end) and can run up to $250 for higher end phones.  But, the expense doesn’t stop there.  Most call centers for efficiency’s sake, will have each agent have a headset.  The unfortunate part of this is that headsets for IP telephones get expensive quickly.  They cost as much, if not more, than the phones (in the range of $75-$200, depending on features of the headset).  And remember, this is PER AGENT.  So each agents telephone setup could run (on average) in the area of $200.  In a small call center of 10 agents – that would be $2000!.  That is not including the PBX server.

A PBX server is typically scaled according to the number of calls and phones attached to it, but its monolithic (meaning that its ONE server).  You  must choose a server that will meet a call center’s needs not only now, but in foreseeable future as well, since upgrading the server means replacing it completely, which is very expensive.

Finally, with respect to installation, not only does the server need to be installed, but each phone needs to have cabling to the network, which can provide challenges.  In all fairness, many of the phones provide “passthrough” for the network connections for ease of installation, but its still an extra step.  Additionally, best practices dictate they have their own network cables and switches.  Either way, it adds extra complexity and costs.

Vicidial on the other hand, works much differently.  The only telephony requirement is a USB headset (quality ones can be found for <$35) and a software phone, which are freely downloadable from the Internet, be installed on the agents computer.  These components working together provide the audio connection to Vicidial.  Vicidial handles the call routing to the agents.  Its straightforward and simple to install.

The other unique thing about Vicidial’s architecture is that it can be easily scaled.  You can start with a single server, and as your need for more calls and more agents grows, you can simply add more servers to handle the demand without completely reinstalling the system.

Administration and Maintenance

Here is where the playing field between the two is more level.  Administration, or more to the point, what we call “moves, adds, and changes”, is not too difficult on either platform.  The major difference, however, is that in PBX systems, you administrate PHONES.  In Vicidial systems, you administrate AGENTS.  You must keep in mind that calls are routed to phones in PBX systems, and agents in Vicidial.   When a new person is hired, on a PBX system, you must figure out which desk/phone they are sitting at and they must stay there in order for call center statistics to be accurate.  In Vicidial, on the other hand, an agent can log in at any station, but the calls and statistics will follow the user.  And when new people are hired, you just keep creating users, and their statistics follow them.

The other area where the systems are similar is in backups.  Most IP PBX systems use FTP to backup configuration information as well as system recordings and voicemails.  Vicidial is no exception.  All that is required is to have an FTP server to send the backups to on a regular basis.

Summary

Overall, a PBX and Vicidial are used for vastly different purposes.  And while a PBX system CAN be used for a call center, Vicidial is a much more efficient and scalable choice for a call center.  PBX systems are best used for managing the day-to-day telephone traffic of an office.

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

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

OSDial is one of the “forks” of the Vicidial code, is OSdial (www.osdial.com).  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

GOAutodial (http://www.goautodial.org/)  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 (http://www.queuemetrics.com) 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 info@verterion.com

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 info@verterion.com

 

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: http://my.digium.com/en/products/ivr/audio-converter/ 

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 info@verterion.com

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 info@verterion.com.

 

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.

Summary

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 info@verterion.com

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.

Summary

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 info@verterion.com 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.

Jitter

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

Echo

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.

Summary

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 info@verterion.com

Why Use Vicidial for Your Call Center?

Vicidial

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 info@verterion.com

 

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