sipXecs Enterprise SIP PBX

sipXecs is an Enterprise SIP PBX that comes complete with voice mail and auto-attendant. It can also be used as a high performance enterprise toll-bypass SIP router. sipXecs combines all common calling features, XML-based SIP call routing, Web-based configuration, and integrated management and configuration of the PBX and attached phones and gateways. It is a modular server-based solution that does not require any additional hardware, as it interoperates with any SIP compliant gateway, phone, or application.

Digium Open Source Telephony

About Digium
Digium®, Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Switchvox, Digium's Unified Communications solution to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of telephony hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom communications solutions. At Digium, we're changing the way businesses communicate.

A Peek at Cloud Telephony: SIPfoundry’s sipXecs

A recent article posted in The Technoverse Blog (in Telecom Patchboard)

My curiosity got the better of me.  While I’m completely content to use turn-key cloud telephony–OnSIP, in my case—the lure of DIY telecom is sometimes too enticing to resist.

This led me to SIPfoundry’s sipXecs, an open-source PBX that many are using instead of an on-premises metal-based solution.

SIPfoundry has grand goals for open VoIP solutions. They are an independent non-profit that hopes to promote “free and unencumbered” telephony. Which is another way of saying their sipXecs PBX software is 100% standards based. So if enough companies, small and large, install sipXecs on their servers, we can all communicate via SIP over the Internet and not pay a dime in per minute charges.

I thought I’d experiment with sipXecs to see what all the shouting was about.

I am not part of a corporate structure with spare LINUX servers on tap.  I did what a lot of business that need on-the-fly access to  data center servers are trying: I grabbed a virtual server from Amazon’s Elastic Computing Cloud or EC2.

Did I also mention I’m not really an IT person? Certainly sipXecs requires a very tech savvy person to install and maintain. I knew about EC2 from a previous writing assignment, and I was after all a former UNIX developer.

Configuring an auto-attendant with sipXecs

Just enough background to get me  into trouble.

The first speedbump I ran into was learning enough about EC2 to grab an appropriate virtual instances—in my case I was looking for Red Hat’s Centos version 5 OS—from their  data center in the clouds. There are references at the end to explain how to access EC2: the key tool being Amazon Web Services or AWS console.

EC2 and AWS are not terribly difficult to comprehend and work with, but there are subtleties with private and public IP addresses and DNS—some of which is still stumping me.

My goal was modest: just to bring sipXecs up and experiment with its browser-based interface. If I could get a SIP endpoint connected, I would consider that just gravy.

I forged ahead with my foundry.

The documentation on their Wiki explains how to install the software—you’ll need to learn a little about the yum software installation utility.  So … once the sipXec is installed, you then  configure this soft PBX using their sipxecs-setup command.

After a little trial-and-error, I got sipXecs on-line and then scooted into the browser interface.

From what I can tell, this thing looks like a real PBX:ACD, auto-attendant, conferencing, hunt groups, intercom (automatic answer), along with support for lots of SIP phones (Cisco, Avaya, Linksys, Polycom, Audiocodes, …) and gateway integrations to the TDM world.

Overall, I am impressed. Quibbles: the responsiveness of the Amazon virtual OS instance I’m using is sluggish, but I didn’t pay for anything very powerful.

Yes, did try to get my X-Lite softphone to connect;unfortunately that involves a level of DNS prowess that I don’t possess at this point. I hope to have a resolution soon enough and should have another post on how sipXecs plays with endpoints.

I was on do-it-yourself-mission in this post but that shouldn’t take away from the fact that sipXecs is a serious product for large companies.  For the enterprise,  a new player, eZuce, has recent stepped in to provide corporate-level support.

Even with pay-for-support model, I believe that sipXecs is very competitive proposition versus on-premise hardware. Check out the eZuce site for my information. read more....

Sneak Peek: Upcoming openACD release

As many of you know we have been working diligently on a new contact center ACD solution called openACD to replace the currently existing ACD application in openUC / sipXecs.  We are making rapid progress and it is time to give you a sneak peek into what is coming.  There is a lot of excitement about openACD as it holds the promise of becoming a highly flexible, scalable and resilient contact center solution.  openACD is one of these next generation contact center solutions that is media agnostic and allows queuing lots of things, including basic calls, voicemail messages, email and at some point instant messages (IM) or even FAX.   It is skills based and offers priority queuing implementing rules that look at different criteria for incoming messages.

Here are some of the highlights:

- Contact center ACD solution with skills based routing, priority and unified queuing, openUC / sipXecs integration for Web administration, supervisor interface for managing agents and call flow, and detailed CDR recording.

- Agents now have the flexibility to transfer calls, using an agent Web UI, in a variety of ways; something that was not possible with our old ACD. Agents can transfer a call back into queue, or into another queue, or to another agent, or they can do an attended warm transfer to an external number.  All these transfers can be tracked for historic reporting.

- Callers can press star '*' while in queue and leave a voicemail instead of staying in queue.  The voicemail message then stays in queue and is presented to an agent.  The agent can then decide to either call the caller back or respond in a different way if the caller's contact information is available.  The voicemail is treated with lower priority and put at the end of the queue by default, but that could be configurable at some point so as to maintain position in the queue.

- openACD can queue email and such incoming email messages are presented to agents over a Web UI.  As with calls there is a wrap-up time that applies to responding to an email.

- Supervisors have a lot of flexibility and different ways to intervene, including transfer calls to a particular agent, send them to voicemail, listen in on a call including whisper and barge.  If emails are in queue supervisors can view the email including mark it as spam.  Call recording is supported, although we will have to improve on recording granularity a bit.  The supervisors can also move agents between profiles to balance load across clients/departments during a call volume spike. There's also support for the supervisor to record an outage message to be played as part of the IVR.

- Agents can be remote either as remote workers connected to the system over SIP, or using an external phone connected to the PSTN.  Care needs to be taken so that calls do not ring out to voicemail systems outside the call center environment.

One of the most exciting aspects of openACD is the possibility to achieve significant scale and redundancy.  Some more work will be required in this area, but here is the basic concept:  Several openACD nodes can be setup as part of an openUC / sipXecs system, all centrally managed and part of the cluster.  Queues can span more than one node and accept calls from any node and deliver to an agent on any node.  Queues are global to the cluster by default but can also be split and then merged again.  This is useful to establish redundancy in case connectivity is lost between nodes or to scale a queue to include several servers.  When a queue is split it continues to operate with its own version of the queue containing only the calls present on that node.  The queue can then be merged again later, combining the calls.

Early performance and scalability tests are very promising with a range of different results.  We will publish results once we are a bit further along in testing.  If you have results to share let us know.

openACD is available in our developer builds and you can participate in the testing effort.  A first official release will be included in the openUC / sipXecs 4.6 release.  If you are a company with a need for a cost-effective contact center and would like to work with us, now is the time to influence the direction we are taking with this. 

And finally, I'd like to thank everyone involved, in particular the openACD team Andrew and Micah, the KGB team without whom this would not have been possible, as well as the team here at eZuce.   We are very excited about this new capability added to openUC / sipXecs and we are looking forward to sharing more details soon.

SIPfoundry and the University Community Announce Unified Communications Initiative

NEWBURYPORT, MA--(Marketwire - September 29, 2010) - Today SIPfoundry launches its EDU initiative, a comprehensive program designed to bring lower cost and modern communications to universities, colleges and school districts. As a proven replacement for legacy IP-PBX systems and as an open alternative to an all Microsoft infrastructure, the SIPfoundry sipXecs solution brings a robust all software solution with a sophisticated user experience to the higher education community.

Colorado State University (CSU) and Cedarville University are the first full members in this joint effort to transition from costly, limiting, legacy IP-PBXs and build 21st century communications that provide a rich, connected experience for faculty, staff, students and alumni. As investing members, CSU and Cedarville represent the higher education community to drive the future direction and define the requirements for an open, interoperable communication solution.

News Facts

The SIPfoundry EDU program is designed to inspire innovation for the open source UC solution of the future based upon the sipXecs open source code, the largest and most comprehensive open source effort to build unified communications using the Session Initiation Protocol (SIP).SIPfoundry embraces a 'For EDU - By EDU' approach that ensures higher education institutions save significant money and it gives them direct influence over the future direction and roadmap of the open source development.The SIPfoundry website has been updated to make it easy for members to collaborate, share best practices and leverage open source applications.In addition to investing in SIPfoundry, Patrick Burns, VP for Information Technology and Dean of Libraries at CSU has joined the SIPfoundry governing board of directors. You Tube: Video Testimonial from Patrick Burns at Colorado State University

Supporting Quote
"Our campus selected sipXecs because we wanted to invest in a product that allowed us to save money while facilitating collaboration with our peers," said David Rotman, Associate VP for Technology CIO, Cedarville College. "We are confident that this program will enhance our ability to communicate and is well suited to meet the learning, research and academic needs of our university and others. We are especially pleased to be involved in a project with significant user-community involvement."

Supporting Resources
SIPfoundry Homepage
SIPfoundry and eZuce EDU Program
SIPfoundry Twitter Page 
SIPfoundry Facebook Page

About SIPfoundry 
SIPfoundry, the leading collaborative open source community dedicated to unified communications and founded in 2004 by Dr. Martin Steinmann and Jerry Stabile as a not-for-profit organization, promotes and advances Session Initiation protocol (SIP)-related open source projects. eZuce Inc., established in 2010 by the creators of SIPfoundry, is the commercial entity that delivers open enterprise communications solutions and support based on the standards established by the SIPfoundry community.

BroadVoice Internet Phone Service

BroadVoice™ Internet phone service allows residential and business customers to use their cable modem, DSL modem, or other BroadBand Internet connection to make and receive Voice over IP (VoIP) phone calls using an ordinary touchtone telephone. Bring Your Own Device™ (BYOD™) plans allow customers to connect their own SIP devices, including IP phones, softphones, and Asterisk PBXs. BroadVoice utilizes our SecureSIP™ technology to ensure accurate connectivity throughout the user experience. SmartSIP™ technology by BroadVoice is used to optimize the routing of network voice traffic, provides the best possible quality voice transmission for each customer's phone device, and automatically configures BroadVoice Authorized BYOD™ devices

Vocalocity Review

The VoIP provider Vocalocity say that it markets only to small businesses, and that almost all of its customers are companies with 50 employees or fewer. As a result, Vocalocity claims to understand the small business market and offer products that work best for small businesses. The company claims in its website that using a Vocalocity system can reduce your costs by 50-80% over a traditional on-premise PBX system.

How does it work? By delivering all of your telephone services over the internet instead of the copper lines of the phones company. All you actually need are some phones and a broadband internet connection. Vocalocity owns all the heavy-duty equipment like routers and switches and takes care of maintaining and upgrading th software.

You just pick up your phone like you always have and make your calls. 

Vocalocity’s most popular plan costs just $39.95 per month per person for unlimited inbound and outbound calls, and includes a whole laundry list of features including: caller ID, call waiting, forwarding and holding, 911, 411, conferencing, do not disturb, voicemail and detailed call logging, to name a few.

And because it all runs on the internet, you can access all your phone features from and computer and phone with an internet connection. So your employees in remote locations or even with cell phones can access all the same features. Plus, you get an internet portal; to help you add extensions, change rules for night lines and forwarding, and to manage every other aspect of your account.

There is no limit to the number of callers you can have, save for bandwidth considerations. Figure that you will need about 64 kbps of bandwidth per simultaneous call. Check with your internet service provider to see how much bandwidth you have available in your connection. If you will be making a large number of simultaneous calls, you might need to increase your bandwidth.

Vocalocity also offers a lower-cost plan. The metered plan is just $14.95 per month plus 3 cents per minute for calls. Yo might use this in a conference room or a visitor lobby, where call are mostly internal.

With all the above plans, international calls are extra, with prices based on the country you are calling and the phone system within that country. Prices run from about a nickel (France to as much as $1 (Belarus) per minute.

Other add-ons, which you would purchase a-la-carte, include additional local numbers, toll-free numbers, virtual local numbers ( a local number in a city where you are not located, but perhaps your customers are), paperless faxing call groups and more.

In addition to traditional customer service, the company offers webinars, white papers, case stories and other resources such as FAQs.

Vocalocity is a privately held company headquartered in Atlanta, Georgia. They can be reached at 1-877-862-2562 or at vocalocity.com. Customer service can be reached Monday through Friday from 9 am to 9 pm (EST) at 866-499-9474.

Mitel PBX Phone Systems

Mitel’s offerings to the voice communications market seem oriented to the large user and the service provider with smaller customers, rather than the small- or medium-sized business markets. By providing both equipment and negotiated rates with U.S. signal carriers, Mitel offers to be your communications partner.

Hardware / Infrastructure – The company’s flagship product line – the model 3300 IP Communications Platform – is scalable from 10 to 65,000 users – -that’s right – 65,000 users. It is a full-function PBX system with embedded capabilities including unified messaging and auto-attendants. It is designed to run in most Local Area Network (LAN) and Wide Area Network (WAN) environments, and supports IP phones, wireless phones and audio conferencing gear.

Other “infrastructure” offerings include set-ups for up to 52, 250, 600 and 1,000 users. A further offering enables the 3300 series to run on Sun Microsystems hardware.

Software – the Communications Director program — is designed to run on Mitel’s proprietary hardware or on industry-standard servers from third-party suppliers. The company also offers a Multi-Instance version in which multiple copies can be run on a server or servers to offer the same functionality to a large user or – in the case of a service provider – multiple small and medium users.

Management and Reporting – Mitel offers programs aimed at simplifying and improving the management, reporting and troubleshooting of Mitel networks. Enterprise manager includes a suite of tolls for configuring and controlling even enterprise-wide systems from a single portal, reducing costs and streamlining administration.

Another program offers remote management functions, enabling outside engineers to diagnose and troubleshoot system problems even from a faraway location, reducing the need – and expense – of on-site visits.

Phones and Accessories — Mitel offers dozens of of telephones and accessories to go with its systems. Ranging from basic IP-enabled desktop phones to high-end multi-line speakerphones and special-use units such as the model 5560 IPT which designed to withstand the rigors of the stock-exchange trading floor. Accessories and peripherals include everything from headsets and stands to interfaces for analog phones, programmable key modules softphones (a phone run on a PC) and digital handsets.

 

Applications – Mitel’s applications suite offers a broad range of programs designed to add increased functionality to the company’s systems. Applications include Business Dashboard for telephone-system reporting and management, Customer Service Manager, Unified Messaging to integrates voice with other communications, audio and web conferencing, teleworkers and a mobile program for unified communications.

Wireless — Another area in which Mitel has a broad range of offerings is in wireless products. It offers phones, gateways, wireless IP phones, phones designed to work in the ubiquitous 802.11b wi-fi environment (including PDA-based softphones).

Finally, Mitel offers a variety of services to complement it’s hardware and software offerings. Choices include everything from systems design and engineering services to hosted solution to completely outsourced management of your phone system.

Mitel has even negotiated deals with a number of ISPs around the country for reduced cost signal carriage, and may be able to offer competitive rates for phone, Internet, data transport and other messaging services.

Mitel sells through a network of reseller partners. Given the large numbers of both resellers and possible combinations of software, hardware and services, it is impossible to say anything meaningful here about pricing.

According to Techcrunch.com, Mitel, a privately-held company with annual sales of almost $750 million and more than 2,000 employees, is in the midst of an IPO filing in Canada worth an estimated USD$230 million.

The company is headquartered in Kanata, Ontario, Canada and can be reached there at 613-592-2122. U.S. operations are based in Chandler, Arizona, and can be reached at 480-961-9000. Customer Service is open from 8 am to 6 pm EST and can be reached at 1-800-267-6244.

TelaSIP SIP Overview

Telasip is an ITSP – Internet Telephone Service Provider – that offers Session Initiation Protocol (SIP)-based Internet telephone service to residential and business customers. With an SIP plan, the provider replaces the phone company’s twisted-copper wires with the Internet, where it is much cheaper to send data.

The company offers three residential plans:

VoIP Plus – $14.95 per month for unlimited incoming plus 500 minutes to the US and CanadaVoIP Premium – $24.95 per month for unlimited incoming plus 1500 minutes o the US48 and CanadaVoIP World – $34.95 per month for unlimited incoming plus 1500 minutes to the US48, Canada and selected international locations.

All residential programs include the ability to have 2 concurrent calls, and include call waiting, caller ID, three-way calling, voicemail, anonymous call rejection and caller-ID block plus emergency calling services. Virtual numbers and enhanced 911 are available as options for additional fees, as is international calling, with rates depending on the country called (a full list is available on the company website).

There are various ways you can purchase business VoIP services from Telasip. While the company website is very short on details (they are planing on re-designing it this year), a customer service representative was very helpful in filing in the blanks. Available plans include:

A by-the-minute plan (1.1¢/minute for incoming and 2.5¢/minute for outgoing) with a minimum of $40.
A basic 3,000-outgoing-minutes plan with unlimited incoming calls, voice-mail to e-mail and fax-to-e-mail features for $44.95 per month. Additional minutes cost 2.5¢ each.
Hosted flat-rate plans with 12 or 25 extensions for $249.94 and $399.95 per month respectively
Enterprise plans beginning at 24 ports with up to 60,000 minutes starting at $699.95 per month

Business plans include such features as call waiting and forwarding and caller ID.

The company also offers per minute, per port and carrier-level wholesale pricing on both inbound and outbound calling services.

As described in the company website, the company is as expanding its network to improve service levels and call quality. It has plans top build new facilities in Salt Lake City, Atlanta and Boston tp supplement the eight gateways it currently operates.

Telasip, which also does business as Epraxys, Inc., is a privately-held company based in Montgomery Village, Maryland. They can be reached at 240-396-1450.

Vitelity SIP Trunking

When you use SIP trunking, you no longer make your phone calls through the twisted-pair copper-wires of the phone company.  You connect your PBX to the Internet.  Your Internet Telephony Service Provider, in this case a company called Vitelity, takes calls made on your office PBX and makes them compatible with VoIP technologies, routes them as far as they can using the Internet, and then, if necessary, back onto the POTS (Plain Old Telephone System, also known as the PSTN, or Public Switched Telephone Network) if needed, to complete the call.

People generally switch to VoIP and SIP trunking because it can save them money on their phone service versus running calls over the regular phone system.  Many companies report as much as a 70% drop in costs. There will be some up front costs because you will need specialized hardware and software to use VoIP, but you should be able to recoup your costs in short order.

Vitelity Link Service is a no-commitment, pay-as-you-go service for small and medium businesses that starts at just $35 per month for the “dial-tone”. A local phone number in most areas of the U.S., with unlimited incoming calls from the “lower 48”, costs $7.95 per month with a $1 activation fee and outgoing call terminations cost 1.44¢ per minute for calls terminating anywhere in the U.S.  Custom toll-free and V-fax numbers are also available at an extra charge. Local and toll-free numbers can also be ported over.

With phone company service, you calculated how many concurrent calls you wanted to be able to support, and got that many incoming lines.  With SIP trunking, the limiting factor isn’t lines, it is bandwidth.  An average call uses around 65kbps (65,000 bits per second). SIP trunks carry VoIP signals along with your Internet traffic, so it is a good idea to work with your ITSP to determine how many calls you can support with your available bandwidth.

Vitelity supports SIP and IAX protocols and supports several compression/decompression (codec) protocols.  In larger markets, Vitelity can bring in to your office a T-1 line (1.544 megabits/second) or enough to carry 23 average phone calls at once.

A variety of features are available through Vitelity, all on an a la carte pricing model, including:

Enhanced 411 (99¢ per call)Outbound caller ID name ($10 per number)Call forwarding (2.8¢ – 3.5¢ per minute)Enhanced 911 ($1.49/number)Virtual PRI – If you need large numbers of inbound lines (pay by the line, not the minute)Voicemail – up to 1000 mailboxesCustomer portal – An Internet-based portal to help manager your account and calls, produce usage reports, add users and more

In a review of several chat forums and review sites dedicated to this market, Vitelity earned very good marks for customer service and support.  With few exceptions, the company was praised for answering customer questions quickly and completely.  Even on those sites dedicated to complaints, Vitelity fared well. Customers felt the company gave excellent service and delivered on what it promised.

Vitelity is part of U.S. National Telecom, a Miami, Florida company traded over the counter under the symbol USNL.  Vitelity, based in Denver, Colorado, can be reached at 1-720-257-5400 or 888-89VITEL (888-898-4835).  Customer support is available by phone from 9 am to 5:30 pm MST.

Digium SwitchVox Systems

Imagine when a phone call comes in, you can look at your computer screen and the Switchvox “switchboard”  program knows who is calling, shows you their name, number and a map of where they are calling from and even brings up an automatic “Google”  page about the caller.  If you have SalesForce, you ‘ll also get that program’s file about the caller plus any comments left on the system after the last time that person called – all before you even pick up your phone.   

You can then decide to answer, transfer to another extension, send it to voicemail or record the call – or finally pick up the phone and answer!  Add to that the fact that you can do so from any IP enabled phone on your network – no matter where it is located – and you begin to see the design philosophy of the company’s premier proprietary product, the Switchvox. 

Come for the hardware lineup……stay for the awesome dance song

The Switchvox phone system comes in three sizes:

Concurrent recordings/conference call users5 recording5 conference callers10 recording15 conference callers20 recording30 conference callersRAID controller, mirrored drive, redundant power supply & optional cold spare failover

Unlike most suppliers, Digium does offer a free trial version of Switchvox.  It is downloadable from the company website and it supports up to 15 users.  But be careful – only download to a computer you don’t use for something else because the software you download will wipe your hard drive.

Support Plans

Silver – $55/user first year then $11/ userActivation plus unlimited e-mail supportGold -  $77/user first year then $17/userActivation plus unlimited e-mail and phone support (during normal business hours of 6 am to 6 pm PST)Platinum – $110/user first year then $28/userActivation plus unlimited e-mail and phone support (during normal business hours of 6 am to 6 pm PST) plus 5 after hours incidents

Beyond the above basic components, options of all sorts are available on an a la carte basis, including:

Analog phone linesEquipment maintenance and warranty plansTelephones – Mostly Polycom Soundpoint phones (from $174 to $549)Conference phonesT-1 lines

Switchvox components and systems are available direct from the manufacturer or through a variety of third-party resellers. 

The company is also the creator of “Asterisk” software, an open source program that can turn an ordinary computer into a communications portal or server.  In combination with Linux and other open-source third-party programs you can assemble a complete communications system from open-sources (non-licensed) programs. 

Both Asterisk and Switchvox are products of Digium, Inc., a privately-held company located in Huntsville, Alabama.  Digium can be contacted at 256-428-6000 or 1-877-DIGIUM1.  Customer support is available from 7 am to 8 pm M-F at 1-258-428-6000.  The company website is www. Digium.com

AireSpring Data/SIP Review

AireSpring offers a broad range of telecommunications services for customers ranging from small businesses with just a few lines all the way up to full enterprise coverage.  Product offerings extend to not just local and long distance voice calls, but also include high-volume data applications: 

High-volume local and long distanceVoIP / SIP (local and long-distance)High-bandwidth data (up to 10 Mbps)Conference CallingMultiple-location and enterprise networks 

If you have a standard PBX requiring a TDM handoff, AireSpring’s offers Dynamic X,  a T-1 line which brings together voice and data over the same line.  Instead of being forced to allocate fixed percentages of your bandwidth to either voice or data,  Dynamic X, as it name suggests, dynamically allocates your bandwidth to voice or data (across up to 16 IP addresses), giving voice calls priority when needed. 

If your PBX is IP-enabled, AireSpring offers traditional SIP trunking.  Get rid of your copper phone lines and PRI (T-1 service) completely and go exclusively over the Internet for your phone service.  Because calls go out over the Internet, AireSpring can use least-cost routing to cut your phone bills.  Trunks start at just $8 each per month.   

As with all SIP-based systems, available bandwidth will be the primary factor determining how many simultaneous calls your phone system will be able to handle.  Make sure you have enough to handle your calls plus your anticipated data needs.  

AireSpring also offers virtual DIDs (local phone numbers) as a way to reduce your use of expensive 800 numbers.  Let’s say you are based in New York, but you get a lot of 800 number calls from Chicago.  Drop the 800 number and replace it with a virtual DID in Chicago for 50¢ per month (including usage).  It works like a local number for your customers but costs you much less. 

SIP trunking is entirely Internet based, so all the features available in your headquarters are available to your entire network. You will no longer be location-dependent for your phone service. Callers won’t see a difference between an extension in your office and an at-home worker or someone calling from on the road.  And all calls between and among network users will be free. 

If you have an existing data or voice network running across multiple carriers, AireSpring’s offers a Virtual Private Network (VPN) offering Multiple Protocol Label Switching (MPLS) a technology which enables packets to be switched between carriers regardless of the transmission protocols involved.  Run all your  traffic -voice, video and data – over a single network. 

Pricing for the company’s varying product offerings is based on a number of factors, including: 

The number of concurrent calls you want your system to be able to handleThe combination you want of local, long distance, data and teleconferencingThe types of equipment you have and its locations. 

 AireSpring has negotiated deals with Tier-1 carriers and currently covers 95% of the United States.

According to the company website, the network is designed to accommodate large-volume and high-capacity users.  Partner companies include such well-known names as Verizon, Global Crossing and Qwest. 

AireSpring, is a privately-held company located in Van Nuys, California, and can be reached at 1-800-825-1055.  For support, call 888-389-2899 or visit www.airespring.com.

VoicePulse SIP Overview

VoicePulse is a supplier of VoIP Telephony services aimed mostly at the residential (SOHO) and small business market, with the ability to serve larger customers and resellers as needed.  VoicePulse services use SIP (Session Initiation Protocol) technology that routes calls over the Internet instead of traditional phone company copper lines.

VoicePulse is primarily a supplier of services.  Although it does provide routers for certain service plans, VoicePulse customers generally supply their own PBX hardware and phones.  VoicePulse’s phone service includes a comprehensive range of features, included three-way calling, caller-ID with name, call waiting with caller-ID name, call return (*69) and call transfer. 

Other features enable you to block selected calls (such as telemarketers or calls without caller-ID) and to filter which calls get through and which ones won’t.  Also included are additional features such as speed-dialing and distinctive rings.

There are five basic service plans listed on the company’s website: 

Outgoing Long Distance (50 US States) call minutesAs low as 1.5¢ / Minute (Lower 48 states only)Outgoing Local and Regional US Calls*Cost/Min. Varies (see website)Cost/Min. Varies (see website)

*See website for breakdown of what comprises local and regional calls 

** Customer is responsible for providing SIP-compliant equipment.  SIP trunking typically comes with enough bandwidth for four simultaneous calls.  Additional simultaneous call c\apacity costs $20 per call.

As with most suppliers of VoIP, there is an entire laundry list of services and features available at an extra cost.  With VoicePulse, that list of extras includes phone numbers, activations, conference-calling capabilities (free with Business Unlimited only), virtual numbers (local numbers in another area code) and more.  Check the website carefully to be sure what will cost extra and what is included.

VoicePulse also offers custom “solutions” for such applications as large offices (20+ users), call centers, colleges and universities, and a wide variety of wholesalers / resellers of VoIP services.

VoicePulse is a privately-held company headquartered in North Brunswick, New Jersey.  They can be reached at 732-339-5100.  Customer service can be reached at the same number on weekdays between 9 am and 5 pm EST.

BroadVox SIP Overview

Broadvox is a supplier of low-cost SIP telephone service, which saves you money by routing calls over the Internet instead of over the Public Switched Telephone Network (PSTN), also known as the phone company. By using its own, proprietary network instead of the phone company’s twisted-copper wires, you get much higher quality calls at a lower price.

You can gradually migrate your phone service from old-style Time Division Multiplexing (TDM) technology to the newer digital Session Initiation Protocol (SIP) used in Broadvox’s Voice over Internet Protocol (VoIP) systems. In this new method, your phone call gets converted into data packets, and it travels the Internet the same way as a YouTube video or a spreadsheet.

Broadvox’s basic product is called Go!Anywhere, and it allows unlimited local and long distance

calling, with what the company says will be a roughly 70% savings over the traditional telephone company. You can use your existing equipment or get an Internet-Enabled PBX system. The basic service costs $35 per concurrent call session, with a minimum of 3 sessions required. Prices drop for all services if you contract for more than 1 year of service. Incoming phone service is available for an additional fee, in increments of 5,000 minutes, starting at a little over 3 cents per minute. Or for 3.5 cents per minute a la carte.

The company also offers two variations of the program, one called Go!Local, which features unlimited local calling with the option of adding long-distance. Go!Domestic is a plan for call or contact centers that use T-1 lines. It eliminates the local component of incoming calls, reducing the cost per minute.

To all three of the above programs, the program offers the ability to add extras, such as enhanced local numbers, with which you can customize the outgoing caller ID information, virtual numbers, directory listings and local number portability.

Finally, Broadvox offers wholesale call origination and termination services for resellers.

The company has an extensive group of partner companies with which it works to provide the best possible level of service. Hardware partners include such well-known names as Alcatel / Lucent and Cisco. It also works with more than 35 vendors of IP phones, PBXs and other equipment, to be sure that their equipment is compatible with its networks.

Broadvox is a privately-held company headquartered in Dallas, Texas with it Network Operations Center in Cleveland, Ohio. It can be reached in Dallas at 213-646-8000 and in Cleveland at 216-373-4600.

Shoretel PBX Phone Systems

Shoretel offers VoIP hardware and systems designed for companies with multiple locations.  With built-in “N+1 redundancy”, Shoretel systems are designed to keep working even if you experience a failure in your network.   

The company offers three main system components – switches, phones and software – designed to work together to create a phone system oriented toward companies with geographically-dispersed operations – sales offices or manufacturing plants, for example – that need seamless phone communications. 

The Switch Is The Thing 

At the heart of Shoretel systems are its voice switches.  Available in capacities from 30 to 120 phones (analog trunking), 30-90 phones (BRI trunking) and 220 phones (digital – PRI – trunking) the switches are scalable, enabling systems as large as 10,000 users.  The overall architecture, according to the company, incorporates n+1 redundancy (every connection has at least one alternate) to keep your phone system up and running, even in the event of a network failure.   

Each switch operates as an independent call processor, even when separated from its servers.  As you mix and match switches, all functions remain integrated, including voicemail and auto-attendants.  Shoretel call this a “distributed architecture”. 

The Phones Are Another Thing 

Shoretel offers phones ranging from basic to full featured. 

At the high end is the model IP 565g, which feature eight buttons, six lines, a color display, full-duplex speakerphone and bluetooth for use with a wireless headset.  The IP 565g also includes integrated 10/100/1000 Ethernet and full VPN connectivity, enabling it to be used outside of company firewalls with a high degree of security.   

Shoretel offers additional phones with fewer features at lower prices.  At the low end of its offerings the IP 110 is a single-line phone suitable for a waiting room or lobby. 

The Software is The Third Thing 

Shoretel also offers a number of software programs to enhance the usefulness of the phone systems it sells.  ShoreWare Call Manager, for example, is a computer-based program which enables users to switch and move seamlessly among video, voice and IM. Other programs include:

Contact center is a program for call centers than manages call routing, enabling you to run a contact center from multiple locations as easily as ou could a centralized call center. Converged Conferencing simplifies the use of audio conferencing, desktop and application sharing, instant messaging, virtual meeting rooms, on-line presentations, and multi-media recording by providing a unified interface.Other available programs include time tracking and billing, SalesForce Integration and more.

Shoretel products are available through a network of channel partners (dealers), so it is difficult to say what prices will be.  We did find a large purchasing consortium on the Internet, so to give you a sense of costs, here are a few excerpts (your prices may vary):

ShoreGear 90V  Analog Switch with 90 phones & voicemail boxes  $3,140.75 each

IP 565 Phone   Top of the line IP phone (silver)         509.15 each

Operator Call Manager Operator Call Manager Software Program        505.75 each 

Shoretel offers e-mail and telephone support, and makes available to its customers a variety of learning tools through “Shoretel University” including FAQs, training and reference guides and other downloadable materials. 

Shoretel is traded on the NASDAQ under the symbol SHOR.  The company, located in Sunnyvale, California, can be reached at 800-425-9385 or 1-408-331-3300 and is on the web at shoretel.com.  Customer support can be reached at 800-742-2348.

Ingate SIP Firewalls Overview

As much time as we spend worying about data security, how many of us give a lot of thought to the security of our voice communications?  If you are a lawyer, accountant or scientist, for example, you probably talk about private and confidential matters on the phone without really worrying about someone eavesdropping on your conversation. 

But once you switch to the digitally-driven SIP environment of VoIP, the possibility that your conversation can be intercepted and overheard multiplies dramatically.  Hackers using packet sniffers and such well-known tools as VOMIT (we’re not kidding – google it) can intercept and turn your private conversations into .wav files they can listen to on any PC. 

Ingate supplies products that can reduce the chances someone will overhear what you say in conversations on an IP phone.   

First and foremost they make SIP firewalls.  Many personal and corporate firewalls –  especially many of the larger corporate legacy systems don’t recognize SIP (the protocol under which VoIP runs) so those calls don’t get the protection of a firewall.  Ingate makes firewalls in 5 models, ranging in capccity from 10 to 50 users.  All firewalls come complete with all the needed software to replace your existing firewall, including full NAT (network address translation) and PAT (port address translation) and TLS (transport layer security) support for encrypted instant messaging.   

If you have a substantial investment in your firewall and don’t want to replace it, Ingate also makes the aptly-named Siparator that connects to our existing firewall. SIParators offer all the functionality of the Ingate firewall.  The Siparator examines every packet of information coming into or going out of your network and blocks those that don’t belong. 

Employees who routinely work and place calls away from your office have unique security concerns. If you now have, or want to add, employees who will work outside of the system protected by your firewall, Ingate offers a software-based VPN (Virtual Private Network) program. Using strong encryption (AES, 3DES or IDSec encryption algorithms) this module enables you to establish a secure link over the internet and between networks without having to use a dedicated line.  It can communicate with any VPN client using IPSec (Internet Protocol Security) or IKE (Internet Key Exchange).  VPN is included with all firewalls and SIParators. 

Ingate is a great company with tons of great insight and information.  We highly suggest their services. 

Finally, for even more security, Ingate offers its Enhanced Security Module.  This software add-on supports both the IDS (Intrusion Detection System) and IPS (Intusion Protection System) and encrypts all real-time communications.  It supports both TLS (Transport Layer Security) and offers SRTP (Support For Realtime Protocol), and includes encryption, authentication and replay protection. 

Keep in mind that the more layers of security you use, the greater the chance that you will introduce noticable lag into your communications.  The manufacturer can best advise you as to how to protect the security of your voice calls without slowing them down. 

Other Ingate products include a QoS (Quality of Speech) module, which enables you to monitor calls and assess how to improve the quality of transmission, and a Remote SIP Connectivity module, which allopws you to extend SIP features and performance to remote locations, even when they are behind non-SIP-aware firewalls. 

Ingate is owned by Ingate Systems AB of Stockholm, Sweden.  They can be reached at +46-(0)8-600 77 50 or at ingate.com.  In the U.S., the company is located in Hollis, New Hampshire and can be reached at 603-883-6568.  Customer service can be accessed via e-mail at support  (@) ingate.com.  Because they are located in Sweden, the company’s customer support hours are from 2 am to 11 am EST, and they can be reached at +46-13-21 08 52.