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Cool Asterisk Add-ons

February 14, 2017 by admin in Asterisk, Blog | Comments Off on 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

FOP2 Web Interface

More information about FOP2, including a full feature list can be found at https://www.fop2.com/

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

iSymphony Web Interface

More information about iSymphony can be found at http://www.getisymphony.com

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

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 http://www.vicidial.org

QueueMetrics

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

Queuemetrics Call Center Monitoring

For more detailed information about QueueMetrics, visit https://www.queuemetrics.com/

QueXS

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 https://quexs.acspri.org.au/

 

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 http://www.asternic.net/

 OrderlyStats

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

OrderlyStats Web Interface

For more information, please visit http://www.orderlyq.com/solutions/orderlystats

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

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

MonAST Web Interface

You can find more information about the features of MonAST at https://dagmoller.github.io/monast/

AstChannelsLive

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.

astchannelslive

AstChannelsLive

More information about this software can be found at http://www.astchannelslive.com/index.html

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.

PHP-AGI

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 http://phpagi.sourceforge.net/  If you need information about AGI – check out the informational resources on Wikipedia at https://www.voip-info.org/wiki-Asterisk+AGI

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 https://www.zoho.com/crm/help/zoho-phonebridge/asterisk-avaya-elastix.html

Activa

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 http://activa.sourceforge.net/index.html

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) – http://zaf.github.io/asterisk-speech-recog/

MojoLingo RubySpeach – http://mojolingo.com/open-source/

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 http://www.asterisk2billing.org/

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. http://oreka.sourceforge.net/

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. http://www.asternic.net/cdrreports/

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

CDR Stats Dashboard

To view the complete feature list, as well as a live demo, visit http://www.cdr-stats.org/

Asterisk World February 8-10

February 7, 2017 by admin in Asterisk, Blog | Comments Off on Asterisk World February 8-10

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

 

VoIP Call Recording Notes

February 3, 2017 by admin in Asterisk, Blog, Vicidial | Comments Off on 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.

 

Asterisk Call Quality Troubleshooting

February 1, 2017 by admin in Asterisk, Blog, Vicidial | Comments Off on 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

Verterion Designs New Rapid Deployment Network and Telephony Platform

February 1, 2017 by admin in Blog | Comments Off on Verterion Designs New Rapid Deployment Network and Telephony Platform

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

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

More information, contact us at info@verterion.com

 

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