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.