5 Things to Know About VoIP

2010-09-10T07:09:02+00:00 September 10th, 2010|Uncategorized|

What is Voice Over IP (VoIP)?

VoIP (Voice Over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.  The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption for voice networking.  

5 Things to Know About VoIP!

1.  VoIP doesn’t “Sound Bad”

2.  VoIP is quick to setup

3.   VoIP isn’t always cheaper

4.  Peaceful Coexistence

5.  SIP Trunks don’t exist…

Watch the video below to learn more about the VoIP technology.

Coretek Services is a Michigan based Systems Integration and IT consulting company that works with telephone systems, virtualization infrastructure, and technology support. Please contact us today for any IT requirements you may have.

Source: Digium

SpeechBridge for Switchvox:

2010-06-30T10:25:51+00:00 June 30th, 2010|Uncategorized|

Adding SpeechBridge speech recognition and text-to-speech solutions to Switchvox empowers customers and employees to gain access to important information and company resources using simple spoken commands. Learn how SpeechBridge can benefit your customers and help your organization sell more Switchvox!


  • Improve customer service
  • Comply with “hands free” cell phone laws
  • Reduce operating costs

SpeechBridge Solutions:

  • Custom speech-enabled IVRs
  • Speech-enabled Auto-attendants and Directories
  • 100% safe hands free access to email and calendar during “windshield” time

Incendonet’s SpeechBridge is a SIP-based, enterprise-grade, complete speech solution that we believe will be of substantial interest to businesses using Asterisk. It’s fast to deploy, easy to use and gives cost-effective Asterisk deployments the feel of much more expensive systems.” Said Rod Montgomery, Director of Product Management at Digium.

For more information go to http://www.coretekservices.com/?page_id=54

Source: Switchvox

Differences between Asterisk and Switchvox Phone Systems

2017-07-27T00:01:11+00:00 June 9th, 2010|Uncategorized|

Asterisk or Switchvox?

Digium is the creator and primary sponsor of the Asterisk project.  Asterisk is an open source communications engine that transforms commodity computers into powerful communications servers.  Asterisk is free.

Digium also makes and sells Switchvox, a turnkey unified communications system (IP PBX) based on Asterisk.  Switchvox is far less expensive than competitive IP PBX and UC systems based on proprietary technologies, but it is not free.

Users and customers frequently asked why Digium offers both the free-and-open Asterisk engine and the commercial Switchvox solution.  The answer is simple: while both products fit into the larger universe of telecommunications technologies, they have very different purposes and are geared towards very different audiences.

Asterisk is built by and for communication systems developers.  The open source project began in 1999 when Mark Spencer released the original Asterisk source code and began accepting submissions from a growing community of users.  The resulting product is an engine that handles all of the low-level details of initiating, maintaining and manipulating real-time media streams (calls) between endpoints (phones).  Since the initial release it’s been tested and refined by a community of more than 65,000 developers and integrators in 170 countries around the world.

Asterisk is to telephony what the Apache server is to web applications: essentially the exquisitely complex plumbing on which other applications are built.  Just as a web server does very little without web applications, a telephony server does nothing without telephony applications.  Web applications can be as simple as single static HTML page or as complex as Facebook or Google.  Likewise telephony applications can be very simple scripts or hugely complex suites of application software.

Low-level engines like Asterisk and Apache are extremely powerful precisely because they have no fixed function or specific purpose set by their creators.  The functions to which they are ultimately applied are determined not by the creators (the developers of the Asterisk and Apache development teams) but by application developers.

Application developers take engine-level components like Asterisk and Apache and build on top of them.  These developers craft purpose-built solutions that implement a specific set of functions.  Asterisk application developers write programs that make Asterisk behave as a PBX or as VoIP gateway or as a dialer or virtually an other type of telecom apparatus.

Some Asterisk applications are simple and use little more than the core Asterisk engine, a few configuration files and some scripts written in Asterisk’s Dialplan language.  More advanced Asterisk applications connect Asterisk with databases, web services and other external resources.  Finally, there are application suites that interconnect Asterisk with many other applications in a complex web of interactions.  These complex aggregate solutions do far more than could be done by Asterisk alone.  Digium’s Switchvox phone system is a perfect example of this class of application.

Where Asterisk is an engine, Switchvox is a complete vehicle.  The Switchvox development team has spent the past six years creating a powerful unified communications system that anyone with a minimum of computer experience can manage.  Where Asterisk is built for telecom developers, Switchvox is built for small and mid-sized businesses that need a powerful, cost effective phone system.

The Case For Switchvox


Digium’s line of Switchvox IP PBX systems make unified communications capabilities available to small and medium businesses.  Switchvox is administered through an easy to use graphical user interface (GUI) rather than raw configuration files and custom scripts.  Switchvox includes all of the standard features of phone system plus unified communication capabilities like advanced voice messaging, instant messaging, desktop fax, drag/drop call control, multi-party conferencing and advanced IVR.  Features that would cost thousands to bolt onto a traditional phone system.

With raw Asterisk, the process of configuring phones is entirely manual.  Each phone must be independently set up by the system administrator.  Switchvox automatically detects and configures phones, making it easy to deploy and manage users.  Switchvox also detects and configures Digium interface cards, making it easy to connect to the PSTN.  Setting up SIP trunks and tie-lines to other VoIP systems is even easier.

So who should really pick Switchvox instead of Asterisk?  People who aren’t telecom gurus who need a powerful, easy to install, easy to maintain, reasonably priced phone system for up to 400 users.

Don’t get me wrong: it’s entirely possible to create a powerful PBX system using raw Asterisk.  The major drawbacks to running Asterisk as a PBX are the deployment time and maintainability.  Building an IP PBX out of raw Asterisk requires some fairly advanced technical skills, including a good working knowledge of IP networking, Linux/Unix system administration, traditional telephony and script programming.  Even those who are fully versed in all four of these disciplines will need to overcome something of a learning curve to create a working system.  Once the system is up and running you will need someone on staff (or at least on call) who knows how the system works and how to handle any moves, adds or changes.

Digium’s flagship Switchvox SMB system (with all the bells and whistles you can imagine) starts at only $3600.  Our basic SOHO package is only $1600.  If you’re still tempted to use Asterisk, that’s cool but first do this:  Divide $1600 by what you think an hour of your time is worth.  Let’s use $50/hour as an example.  $1600 / $50 = 32 hours.  If you can learn enough Asterisk to build your own solution in 32 hours or less, go for it.  If not, take a good look at Switchvox.

The Case For Asterisk

Let’s go back to the engine/vehicle metaphor.  Asterisk is an engine.  It’s powerful.  It’s flexible.  It has enormous potential.  What it requires is a skilled engineer (or even a skilled shade-tree mechanic) who can take the engine and build it into a vehicle.  If you are creating a product or a custom solution that requires integrated voice communications, Asterisk is exactly what you need.

Let’s take the product scenario first.  If you want to build a conferencing server that connects to both VoIP and PSTN networks, Asterisk is a great starting point.  Asterisk has all kinds of features that make multi-party conferencing really, really easy.  It also includes native support for every major VoIP and PSTN protocol in use today.  To build a conferencing server out of Asterisk you need to pick out your platform hardware (computer), create an administration interface (probably a web application running on Apache) and possibly an end-user interface.  You’ll probably want to integrate with calendaring systems like Exchange, iCal, Google Calendar, etc.  You probably want to tie in email and possibly IM notifications and reminders.  Given a skilled development team you can probably bang this out in a few months.

Compare that with building from scratch and you can see the power of Asterisk.  You didn’t have to write (or license) a SIP stack.  You didn’t have to write your own DTMF detection algorithm or even wrap a DTMF collection function call for use in your application.  In fact, the actual “telephony programming” probably came down to a few dozen lines of Dialplan script and a bit of SQL to set up the database.  You shaved years off your development and testing path, added value through your snappy web interface and built it all on a free engine.  Nice.

Asterisk fits very nicely into the toolboxes of telephony integrators and data VARs.  If you’ve ever done custom integration work you know how difficult it can be to make systems from different vendors (or different generations) play nicely.  In enterprise scenarios where modern data applications share space in the server room with legacy switching gear, Asterisk can be indispensable.  It acts as a kind of “telephony glue” that ties VoIP to TDM and digital to analog.  It also bolts onto legacy systems as a perfect low-cost adjunct.  Your customer has an Octel voice messaging system that’s on its last legs?  No problem.  Replace it with an Asterisk-based system.  Your biggest client needs a dialer that can call an entire city in an hour?  Sure.  Asterisk can do that.

If you’re already familiar with networks, telephony and scripting, the Asterisk learning curve is fairly easy to overcome.  Read Asterisk: The Future of Telephony by Smith, Madsen and Van Meggelen.  Take a look at the samples and recipes on Asterisk.org.  Take the Asterisk Fast-Start or Asterisk Advanced class for a bit of hands-on training.  You’ll find that building solid solutions with Asterisk is drastically easier than building your own voice engine from scratch using a raw C language API from some proprietary vendor.  (Trust me on this one: I built IVR engines on the Dialogic and Brooktrout APIs before I discovered Asterisk.)

Asterisk is also a terrific way to learn about telephony and communications.  Students, hobbyists and artists have used Asterisk to build some extraordinarily creative applications while at the same time learning about telecommunications.  Some of the most successful developers in the Asterisk ecosystem started out experimenting with the code while in college or even high school.


If you’re technically inclined and want to build a communication product or solution, then Asterisk is for you.  If you’re in need of a great phone system at a great price, check out Switchvox.

Asterisk Administrative Interface                                      Switchvox Administrative Interface



Source:  http://blogs.digium.com/2010/02/19/asterisk-or-switchvox/

Microsoft Unified Communications Open Interoperability Program

2017-07-27T00:01:11+00:00 June 9th, 2010|Uncategorized|

Microsoft Unified Communications Open Interoperability ProgramFind out more about the Microsoft Unified Communications Open Interoperability Program for enterprise telephony infrastructure, including finding qualified SIP-PSTN gateways, IP-PBXs and SIP Trunking Services.OverviewThe qualification program for SIP/PSTN Gateways, IP-PBXs and SIP Trunking Services ensures that customers have seamless experiences with setup, support, and use of qualified telephony infrastructure and services with Microsoft’s unified communications software and Microsoft Office Communications Online (BPOS-Dedicated).

Only products that meet rigorous and extensive testing requirements and conform to the specifications and test plans will receive qualification.

While the specifications are based on industry standards, this program also defines:

  • Specific requirements for interoperability with Office Communications Server & Exchange Server Voice Mail
  • Specific requirements for interoperability with Office Communications Online Services for SIP Trunking Service Providers
  • Testing requirements for qualifying interoperability with Office Communications Server & Exchange Server Voice Mail
  • Installation, set-up and configuration requirements via a Quick Start Guide
  • Release Notes with any known issues
  • Documented support process between Microsoft and the vendor
  • Enterprise-class standards for audio quality, reliability, and scalability

The scope of the qualification is for environments where either Office Communications Server or Exchange Server Voice Mail utilizes a SIP/PSTN Gateway, IP-PBX or SIP Trunking Service for communication with the PSTN. Additionally, qualification is available for Microsoft Office Communications Online Service environments utilizing Office Communications Server with Exchange Online Unified Messaging.

The testing focus of the program is designed to ensure that vendors providing interoperability with Microsoft unified communications solutions do so in a consistent and supportable manner, including SIP and signaling support used with the Mediation Server role of Office Communications Server and the Voice Mail role of Exchange Server.

Direct SIP: Gateways and IP-PBXs Qualified for Office Communications Server

Listed below are gateway and IP-PBX products and necessary firmware combinations that have been independently qualified. It is recommended that you visit the vendor’s Web site for the latest information regarding PSTN/PBX, protocol, capacity, country support and documentation including a Quick Start Guide, release notes and known issues.

— “Qualified” +S —”Qualified with SRTP & TLS”

Vendor Configuration Tested Product Communications Server Version
2007 R2 2007
Aastra IP-PBX MX-ONE V.4.0 +S  
Aculab Basic Hybrid ApplianX Gateway for Office Communications Server 2007, V1.0.0       
Altigen IP-PBX MAXCS,    
AudioCodes Basic Gateway Mediant 1000, 5.60A.013.005 +S  
AudioCodes Basic Gateway Mediant 1000, 5.20A.043    
AudioCodes Basic Gateway Mediant 2000, 5.60A.013.005 +S  
AudioCodes Basic Gateway Mediant 2000, 5.20A.043    
AudioCodes Basic Hybrid Mediant 1000 Hybrid, 5.60A.013.005 +S  
AudioCodes Basic Hybrid Mediant 1000 Hybrid, 5.20A.043    
AudioCodes Basic Hybrid Mediant 2000 Hybrid, 5.60A.013.005 +S  
AudioCodes Basic Hybrid Mediant 2000 Hybrid, 5.20A.043    
Avaya IP-PBX Avaya Aura Session Manager 5.2 with Avaya Aura Communication Manager 5.2.1 SP1    
Avaya Basic Hybrid Secure Router 4134, 10.2.1    
Avaya Basic Hybrid Secure Router 4134, 10.1.0    
Avaya Basic Gateway Secure Router 2330, 10.2.1    
Cisco Basic Gateway**    2851 Integrated Services Router, IOS 12.4(15)T    
Cisco Basic Gateway** 2851 Integrated Services Router, IOS 12.4(24)T    
Cisco Basic Gateway** 3845 Integrated Services Router, IOS 12.4(15)T    
Cisco Basic Gateway** 3845 Integrated Services Router, IOS 12.4(24)T    
Cisco IP-PBX Unified Communication Manager 7.1.3    
Dialogic Basic Gateway DMG2000, 6.0.128 +S  
Dialogic Basic Gateway DMG2000, 5.1.142    
Dialogic Basic Hybrid DMG4000, 1.5.102    
Dialogic Basic Hybrid DMG4000, Dialogic Diva SIPcontrol +S  
Ferrari electronic AG Basic Gateway OfficeMaster Gate, 3.1    
Ferrari electronic AG Basic Gateway OfficeMaster Gate, 3.2 +S  
Ferrari electronic AG Basic Hybrid OfficeMaster Hybrid Gate, 3.2 +S  
Huawei Technologies IP-PBX SoftCo, V100R002    
Innovaphone IP-PBX IP6000, V7.00 ocs-certified 09-7034301    
Media5 Basic Gateway Mediatrix 3000 Series, DGW 2.0    
Mitel IP-PBX 3300,    
NEC Basic Gateway SV70 OCS-GW-A, MG-16SIPC    
NET Basic Gateway VX1200, 4.7v88 +S  
NET Basic Gateway VX1200, 4.4.2.v31    
Nortel IP-PBX CS 1000, 5.50.12 +S  
Nuera Communications    Basic Gateway GX-1K, 5.60A.013.005 +S  
Nuera Communications Basic Gateway GX-1K, 5.20A.043    
Nuera Communications Basic Gateway GX-2K, 5.60A.013.005 +S  
Nuera Communications Basic Gateway GX-2K, 5.20A.043    
Quintum Basic Gateway Tenor DX, P107-06-00-OCSR2-03    
Quintum Basic Gateway Tenor DX, P105-19-10-MS-01    
Quintum Basic Hybrid Tenor Hybrid Gateway 60, P107-06-00-OCSR2-03    
Seltatel IP-PBX SAMoffice 4, 2.8.0    
Tango Networks Basic Gateway Abrazo (qualified with Audiocodes Mediant 2000 5.20A.043), 3.3    
Teldat Basic Gateway Vyda-2M 10.7.55 +S  
VegaStream Basic Gateway Vega400, 8.282S029    

** See partner’s site for known issues and support notes

Other Supported Products

Other supported products are listed by request of the Vendor as the same qualified firmware may be supported across several different products. While these have not been specifically tested, the Vendor does fully support this configuration for the listed Qualification Level. Please contact the Vendor for more information on these products.

Gateways or IP-PBXs Other Supported Products
AudioCodes Mediant 1000 MediaPack 11x, Mediant
Cisco 2851 Integrated Services Router    2800 Series  
Cisco 3851 Integrated Services Router 3800 Series  
Dialogic DMG2000 DMG1000 and DMG2000 Series  
Dialogic DMG4000 Dialogic 4000 Media Gateway Series and Dialogic Diva SIPcontrol  
Mediatrix 3000 Series Mediatrix 4100 Series, Mediatrix 4400 Series  
NET VX1200 VX Series, VX900, VX1800  
Quintum Tenor DX Tenor AS, AF, AX, BX, DX and CMS Series  
Teldat Vyda-2M Vyda Series and Atlas Series  
VegaStream Vega400 Vega50 Europa Vega 5000  

SIP Trunking Services Qualified for Microsoft Office Communications Server 2007 R2

SIP Trunking enables connectivity to the Public Switched Telephony Network (PSTN) directly over SIP. SIP Trunking services are offerings from IP Telephony Service Provider partners that offer PSTN origination, termination and emergency services using the SIP protocol. An enterprise can use SIP Trunking to connect their on-premise voice network implemented by Microsoft Office Communications Server 2007 R2 or to provision PSTN termination capability for Office Communications Online (BPOS-Dedicated).

Listed below are SIP Trunking Services that have been independently qualified to meet the UCOIP requirements along with those services who meet the additive requirements for Office Communications Online (BPOS-Dedicated).

Carrier Service Name Office Communications Online (BPOS-D) Ver 10.1
AT&T AT&T IP Flexible Reach Service  
BT Global Services BT Onevoice  
Global Crossing SIP Trunking Services  
IntelePeer IntelePeer SIP Trunking  
Interoute InterouteOne  
IP Directions OCS Telephony Services  
Jajah JAJAH SIP Trunking  
Orange Business Services SIP Trunking  
Sotel SoTel IP Services  
Sprint Sprint Global MPLS, SIP Trunking     
Swisscom Swisscom VoIP Gate  
Telenor Telenor Samordnet kommunikasjon
(Unified Communication)
ThinkTel OCS Connect  
Verizon Business IP Trunking Services  

Supported IP-PBXs for Microsoft Office Communications Server

The following IP-PBXs are supported by Microsoft but have not gone through the formal OIP qualification process nor was the testing requested by the vendor. Sufficient internal testing has been performed by Microsoft such that specific configurations are supported by Microsoft (where applicable with known limitations). These configurations utilize the commercially available production SIP trunk interface of the IP-PBX vendor but may not be supported by the IP-PBX vendor. In addition, IP-PBX vendor-provided complete documentation for installation and set-up, release notes, or documented support processes may not be available. Wherever possible, Microsoft will endeavor to provide documentation for installation and set-up.

IP-PBX Vendor Tested Product Supported Configuration Software Versions Tested 2007 R2 2007
Avaya Communications Manager SIP Enablement Services Direct SIP 4.0    
Known Limitations:

  • Configuration requires setting “Alternate Route Timer(sec)” value from default of 10 sec to 30 sec. The configuration should show “Alternate Route Timer(sec): 30” in the corresponding SIP signaling group.
  • When an call is ringing to the Office Communicator user, the caller (either on an Avaya station or a PSTN line routed through the PBX) will not get ring back tone. This issue has been resolved by Avaya with the 5.x software releases.
  • Quality of Experience reports will not contain information regarding jitter and packet loss.
  • Comfort noise generation is not supported. As a result, comfort noise is not played on Office Communicator.
  • ISDN Failover is not supported from an OCS perspective. If the Avaya PBX is being used for PSTN connectivity and multiple T1’s are being utilized, an OC client will not retry a call based on a T1 being unavailable. It may be possible to configure the Avaya to not assign outbound calls from OCS to an unavailable T1, but this configuration was not tested.
Cisco Cisco Unified Communications Manager Direct SIP 4.2(3)_SR3a    
Known Limitations:

  • The PRACK message sent by CUCM 4.2(3) is malformed by missing the MAXFORWARDS header.  As a result, this configuration requires PRACK to be disabled.  By default, PRACK is disabled in CUCM 4.2(3)
  • For Office Communications Server 2007, this support requires update package for Communications Server 2007 Mediation Server: August 2008.
  • OCS 2007 may not appropriately normalize the PAI in the 200 OK or UPDATE, resulting in OC displaying a non RFC3966 formatted global number and in failed RNL on OC. When calling from OC 2007 to a Cisco phone number, after the caller gets connected, the name of the person on the Cisco phone may not be shown on Communicator, and instead OC may display the E.164 number (without a “+”) for the person on the Cisco phone. This is resolved in OCS 2007 R2
  • When calling from OC 2007 to a Cisco phone number, where the Cisco extension is disconnected or out of service, the Cisco IP-PBX may not notify OC 2007 in a timely manner. This has been remediated in OCS 2007 R2.
OpenScape Direct SIP 3.1R3    
Known Limitations:

  • Inbound early media/PRACK is not supported on the Siemens PBX. As a result, in some situations the initial audio of a call may be clipped as the call signaling is being set up.
  • Certain Siemens IP phones may not render audio for inbound calls. If this occurs, a phone configuration change to support both symmetric and asymmetric RTP will be needed.
  • For Hold/Un-hold to function properly, OpenScape needs the RTP config parameter Srx/Sip/ZeroIpOnHold set to false.
  • For full SDP versioning support, OpenScape needs the RTP config parameter Srx/Sip/CompareSdpBody set to true.
  • Quality of Experience reports will not contain information regarding jitter and packet loss for PSTN calls coming through the Siemens IP PBX.

Dual Forking Qualified for Microsoft Office Communications Server

Listed below are IP-PBX and firmware combinations that have been independently qualified. Please contact the vendor for more information on these products.

IP-PBX Vendor    Tested Product    Qualification Level    Software Version Tested Other Supported Products    2007
Nortel CS 1000 Dual Forking***Dual Forking
with RCC***
Call Server X2105.00WSignaling Server 5.00.31Multimedia Communications Manager    CS 1000 Series  

*** If you are deploying Dual Forking, you must also install the following updates for Microsoft Office Communications Server 2007:

 Source:  http://technet.microsoft.com/en-us/office/ocs/bb735838.aspx

What is Unified Communications?

2010-05-20T17:54:00+00:00 May 20th, 2010|Uncategorized|

Many people ask what is Unified Communications.  The answer is simple.  It is the ability to streamline communications.  The common way this is done is by integrating all or a few of the following communication tools.

  • Email
  • Calendering
  • Voicemail
  • Instant Messaging
  • Telephone Systems
  • Video and Audio Conferences and Meetings

Microsoft does a good job explaining it in the video below.

Small Business Computer Support in Detroit Metro

2017-07-27T00:01:11+00:00 March 9th, 2010|Uncategorized|

Did you know that Coretek offers computer and network support to small and medium sized business in the the Detroit Michigan metro area?  We do!!!

              Contact us today at (248) 684-9400 for more information.


SMB Computer Support:

Coretek is committed to providing computer and network solutions that address small and medium sized businesses in the Detroit area. Our goal is to help our clients achieve a predictable and cost-effective IT support.  

  • Cost Effective
  • Computer Supportdell_ultra_small_desktop
  • Network Support
  • Server Support

Coretek offers computer and network support on a subscription basis or time and material basis that saves a company time, worry and money.  We will proactively update and stabilize a company’s computing environment, eliminating problems before they occur.  Coretek utilizes processes and tools that over time reduce the number of problems in a network and amount of time it takes to resolve computer issues when they do occur.  This service will make a company and its employees more productive, which is Coretek’s ultimate goal.  Coretek is dedicated to providing its clients with more than just great IT people but innovative computer solutions.  Solutions provided by Coretek include the assessment and design of appropriate technology solutions, technology implementation, ongoing support, and appropriate migration planning and implementation.  Coretek looks forward to building a reputation of excellence with its clients as a provider of computer support services.  We will not waste your employee’s time on the phone instructing them on what to do to fix the issue.  We will either remote control into the machine with the problem and fix it, or we will send someone on-site to repair it quickly.  Our small to medium business team is comprised of a group of highly trained consultants and engineers who understand the needs of smaller businesses and have the experience to provide technology recommendations as well as their implementation and support.


Technology Solutions Offered: 

  • Computer, Printer and Network Support
  • Server Migrations and Upgrades
  • Network Security
  • Technology Assessments and Strategic Planning
  • IP Telephony Systems and Integration (Voice over IP, VoIP)
  • Server and Desktop Virtualization
  • Project Management
  • Microsoft Integrated Solutions (AD, Exchange, System Center, ISA, IIS, SharePoint, Project Server)


       101 Best and Brightes

  Microsoft Gold Partner

VoIP For Business: Stability vs. Savings

2017-07-27T00:01:12+00:00 January 21st, 2010|Uncategorized|


How can VoIP save your business money?Digium Whitepaper

You want to or already have deployed a VoIP (Voice over Internet Protocol) capable phone system for your business, but where are the monthly cost savings, VoIP? You’ve seen some savings by reusing your existing company infrastructure, like network wiring, and you’ve seen a boost in productivity because of all the features that can come with VoIP, and specifically an IP PBX, but do you really need to entrust your voice to the wild west of the Internet to see any real impact on your monthly bill? We’ll explore ways to get the most out of an IP PBX (Internet Protocol Private Branch eXchange) deployment so that your calls are as cheap and as reliable as you are willing to make them. And we’ll look at ways to help you decide how much risk your company can tolerate in the name of slashing phone bills.

Wait, but isn’t VoIP free?

Not exactly, no. If you make a call using VoIP to another user of the same VoIP network, then yes, this call could potentially be free. This is really dependent on what the owner of that network has decided for their policy. If the owner of the network is you, as in the case of multiple IP PBX systems joined together, then yes, those calls are free.

So what are you paying for then?

If you’re not calling another VoIP user, like in the case where a VoIP call is made to a cell phone, somewhere, somehow, that call needs to jump out of the VoIP network and “terminate” into the PSTN (Public Switched Telephone Network). That’s the service you’re paying for when you’re paying for VoIP service (Fig 1).

The main reason that your phone calls are less expensive when using a VoIP provider is because they’re sending your call as far as they can with VoIP, and only sending it as short a distance as they can out on the PSTN. In other words, they’re saving by not sending the call long distance either.

An ITSP (Internet Telephony Service Provider) with many termination points all around the world can have rates well below a traditional carrier for this reason. Take, for example, a call you want to make from Los Angeles to someone’s regular home phone in Paris. If the VoIP carrier you’re using has a termination point in Paris, you’re in luck and the call will travel across the distance just like any other internet traffic (like if you sent an e-mail to someone in Paris), and then when it needs to go from that termination point in a data center out to the PSTN network in Paris, its just a local call, and therefore, cheap!


But all this goes out the window when you consider that most ITSPs are actually just reselling a larger, wholesale carrier’s minutes. So shopping for an ITSP can just come down to shopping for the lowest
rates. But buyer beware! Just like anything else, you tend to get what you pay for. There are definitely bargains to be had, but it’s important to know if the carrier you’re researching is reselling someone else’s minutes or if they actually have their own network. It may be better if they’re reselling a larger carrier’s minutes because that large company has a lot of infrastructure, presence worldwide, and support staff. On the other hand, you will get some frustrating answers from ITSPs that don’t own their own network if they’re experiencing an outage. Basically, there’s not much they can do about it. So if you are going to choose to go with a provider that resells rather than owns their own network, the best bet is to choose a carrier that resells several larger carriers’ minutes, instead of depending on just one.

Big Impact: Routing Calls Wisely

The whole goal here is to explore how routing your business calls through the right channels can impact your bottom line, without forcing you to jump into VoIP “head first” at the outset. VoIP may be cheap, but it’s typically no more reliable than the internet, so balancing with PSTN calls would be the wise deployment, due to the government regulations placed on our old telephone network.

Scenario One: Use VoIP to Just Call Between Offices

So easy, it should actually be difficult to NOT implement an IP PBX this way. From remote employees, to entire remote offices, by deploying IP handsets and IP PBXs at each location, all of your devices can just talk to each other without intervention by an outside agency. That is, they all speak the same language, no translation is necessary, so there’s nothing to pay for in that case but bandwidth. Presuming you’ve moved from a traditional PBX to an IP PBX, and have simply unplugged your old analog lines from the old system and plugged them in to the new system, you could still see some pretty decent savings by deploying this way. You could consider this to be the “safest” way to roll out an IP PBX, but unless your company does a lot of branch to branch calling or has a lot of employees working from home, you could probably aim a little higher.

Scenario Two: Add in VoIP for Select Outbound Calls

By adding in a VoIP provider to the mix, you can pick and choose through the Switchvox interface which calls should be handled by which route. Imagine your phone system, but instead of just having those old analog lines plugged in, you’ve also chosen to sign up for service with a VoIP service provider. In the Switchvox GUI interface, you can actually specify that for 911 calls, the system should send the call over the analog lines, but when you call New York, to use the VoIP service provider’s route. See what we did there? We used VoIP because it’s cheaper to call long distance using VoIP, but we used the analog lines when we didn’t care about cost and just wanted the most reliable call possible.

With Switchvox you can get as specific as you want with defining types of calls (any long distance call, any call to Los Angeles, any call to 858-234-9090) so that you can route them the way that makes sense for your business. And by using several VoIP providers, you can boost the cost savings even more. If you have multiple providers, you can set the Switchvox to use the provider that gives you the best rate for the call, e.g., use VoIPro to call China, but Vongo to call Dallas.

To balance a little reliability back in there, Switchvox can be set up with fail-over, or fallback routes for the calls that you specify. E.g., use Vongo to call Dallas, but if that fails, place the call using the good old PSTN lines. If you have multiple providers, you can get really fancy and stack the routes up as much as you like: use Vongo to call Dallas, but if that fails, use VoIPro, and if that fails too, use the PSTN. These fail-over routes can be put into place wherever it makes sense for your business. Would you like to fail over to the PSTN when your employees are trying to call China? It’s up to you! Do you want to spend$30 on a five minute phone call, or would you rather your employee give it another shot in 5 minutes when the outage has (hopefully) passed?

An even more granular level of control is to define the routes that should be used not just by the number your employee is dialing, but by whom the employee is. Maybe you don’t want your tech support team to call out using the analog lines unless its a 911 call and maybe they shouldn’t be allowed to dial 1-900 numbers at all. Maybe the CEO should be able to use the PSTN fail-over route when she’s trying to call China but the VoIP connection is unavailable. All of these options are open to you with Switchvox.

This deployment scenario is the most common way that Digium’s Switchvox IP PBX systems are deployed in the real world because they offer the most flexibility for balancing cost savings with reliability. If you find that your VoIP providers aren’t as reliable as your business demands, you can ratchet up the use of the PSTN lines for many of your calls. If you find your VoIP lines have never been a problem, you can start scaling up their use and really see the savings add up.

Scenario 3: VoIP Inbound

In the previous scenarios, we’ve been discussing outbound calls but it is possible to use VoIP for inbound calls as well. A phone number (or as many as you need) can be procured from many VoIP service providers. These are often called “DIDs.” It is often cheaper to get a DID than a PSTN phone line (that by nature comes with a phone number) and so it is an attractive option for many businesses trying to squeeze out the most cost savings possible with their new IP PBX. What many businesses fail to consider, however, is that they will often pay for outbound and inbound calls with this new number. With your old analog lines though, you probably didn’t pay for inbound calls. E.g., if John calls Jane using a regular phone line, he pays for the call based on how long he’s on the phone and Jane doesn’t pay a dime. With VoIP, unless you’re signed up for a plan that is a flat fee for both outbound and inbound, you’d pay both ways.

Another point to consider when evaluating moving your numbers to VoIP DIDs is number portability. Unless your VoIP service provider can transfer your numbers, there are generally some costs associated with changing your businesses phone numbers: printing new business cards, informing your clients, updating advertisements or websites, etc.

And the last reason that using VoIP for inbound calls is unusual in a business IP PBX is the reliability factor. If that call can’t reach the IP PBX, it’s a far worse thing for most businesses than if an outbound call fails. Think of it this way, if you’re sitting at your desk and try to call your customer in China and the call route rules don’t fail over to the PSTN so your call simply fails, you’re going to hang up the phone and try again in 5 minutes. If your customer, on the other hand, tries to call you and it doesn’t work, who’s to say they’re ever going to call back again? Ouch. And there’s nothing the IP PBX can do about this to fail-over, because the call isn’t getting to it. It can’t re-route a call that it doesn’t have.

The bottom line is to be cautious when assuming you need to switch all of your numbers to DIDs. A far more common way to deploy VoIP DIDs is to use them as back up numbers in case all of your analog phone lines are full. Making sure you’ve got enough PSTN lines to handle your inbound call volume is important, but if you find that one day your company has been covered in the New York Times and your phone is ringing off the hook, you can at least roll over to your DIDs. And if those happen to be down at that very instant, I’d say you were both having very good luck and very bad luck on the same day. In other words, this is probably a reasonable risk for most businesses.

So how does one implement this scenario? Your PSTN provider probably has an option available that you can add to your plan that will forward inbound calls to another number if all of your phone lines are busy. Just give them your VoIP DID and tell them that’s the number they should forward to. They don’t even have to know it’s a VoIP line, it looks just like a telephone number to them. If you are going to deploy this way, it is a good idea to look for a provider that will allow multiple inbound calls over the same DID, that way you’re pretty much guaranteed not to “ring busy” when your customers call on the busiest day.

Scenario 4: All VoIP, All the Time

Sometimes Switchvox IP PBXs are deployed as an office phone system that strictly use VoIP and they don’t have any analog lines plugged into them. One way that this happens is when a VoIP service provider actually doesn’t route your business’s calls over the Internet, but instead uses a private network. These types of providers can therefore offer data as well as VoIP service and can provide SLAs (Service Level Agreements) that other VoIP providers can not. They are actually in charge of what happens to your calls, rather than trusting them to the Internet. These types of providers can also offer QoS (Quality of Service) which prioritize your voice packets over your data packets, ensuring that your phone calls sound perfect. An often misunderstood aspect of VoIP is that it sounds bad- not true! It actually sounds better because its digital. What can sound bad is the network the call is on. Calls that travel over the Internet can often take on a robot-y sound or be choppy because they’re sharing that “information superhighway” with a lot of other traffic. QoS ensures a clear path from start to finish.

The other, more risky, but cheapest way to do an all VoIP system is to get a regular VoIP service provider account and a DID or two. You will want to make sure you have a plan for dialing 911 (some VoIP providers support this, others do not). This is, of course, both the cheapest (probably) way to deploy and fraught with the most risk (also probably). You may have great luck with this, and you may curse the day you ever tried it. For this reason, it’s recommended that you work up to this gradually, rather than jumping in feet first and having to scale back and make adjustments to cope with trouble.

Failures? Outages? What gives?

With all of this talk about failures and outages, you might be asking yourself just what you’re getting into! As I hope I’ve shown, VoIP can be deployed in such a way as to improve your call’s sound quality, be cheaper and just plain better, but there can be bumps in the road, which is why I’ve outlined these different deployment scenarios. These bumps can be caused by a lot of different things, like outages of your ISP, big Internet backbone style outages (fairly rare, but they happen), and outages at your VoIP service provider. The important thing to remember is that if your IP PBX cannot navigate the route from its location to your service provider, your calls will fail. The regular analog lines that we’ve come to depend on are essentially an old, proven dedicated network that’s regulated by our government to be up with “five nines” of reliability. The Internet is not such a beast, but it’s still pretty darn good.

Balancing Act: The Best of Both Worlds

This straightforward assessment of the situation hopefully gives you the tools to evaluate for yourself how you’d like to deploy VoIP in your network. You can deploy something very simple that almost emulates a traditional PBX to a system with least cost routing implemented with fail-over rules that keep your company’s communication running smoothly. There should be no reason to hesitate when it comes to deploying a next-generation phone system in your business!


To find out more about Digium, Switchvox, or the Open Source Asterisk Project, visit www.digium.com/switchvox or call 1-877-4-CORETEK.

Digium Switchvox Releases New Version 4.5

2017-07-27T00:01:12+00:00 January 21st, 2010|Uncategorized|

Digium Switchvox system  – new 4.5 version features greater handset support

Switchvox 4.5 Release

Switchvox SMB 4.5 Extends the Power of the Web Interface to Phone Handsets

Digium’s Switchvox® SMB 4.5 is a powerful, full-featured and cost-effective VoIP Unified Communications (UC) solution designed for small- to mid-sized businesses. Switchvox SMB 4.5 delivers greater flexibility for your office communications, extending control of mission-critical applications traditionally accessed through the web interface directly to phone handsets.

Based on Digium’s Asterisk, the world’s most popular open source telephony engine, Switchvox SMB is a powerful IP PBX that integrates an easy-to-use web interface with innovative UC features such as fax, chat and video calling.

Call or email us today for an onsite demo!

Phone Feature Packs extend the power of Switchvox to Polycom phone handsets

The new Phone Feature Packs extend the power of the Switchvox to Polycom phone handsets for maximum control of the communications system.  Features that were previously only available through the web interface are now made available through Phone Feature Packs to these phone handset displays. Users can easily access powerful features from their phone handsets, such as call recording, visual voicemail, a searchable company directory and call parking lots. New options for more fundamental phone functions, such as hands free click-to-call dialing, distinctive ringtones for different types of calls, extension failover to a backup Switchvox SMB server, and support for multiple extensions on a single handset, are also available in version 4.5. 

Check out these other powerful Switchvox SMB 4.5 features!

·      User profiles— Caller profile information such as photo, extension, title and location.  This appears on the Switchboard and on the display of Polycom phones with Phone Feature Packs on internal calls.

·      Flexible language support— The Switchvox 4.5 GUI, manual, and inline help are available in English, UK English, Italian, Castilian Spanish, and Latin American Spanish. Sound packs, which include all the audio prompts used in the system, are available for these languages, as well as Australian English, French, and French Canadian. For businesses with international offices, users can customize their language settings to best complement their location and language preference.

·      Comprehensive monitoring—Digium has implemented the Simple Network Management Protocol (SNMP) for Switchvox 4.5, which gives administrators the ability to collect real time data about the status and health of their systems. For a comprehensive description of features, please contact us.