What is new in ShoreTel Contact Center Version 7?

ShoreTel Recently announced the availability of Version 7 of the Enterprise Contact Center. It is not secret that I am a big fan of this product, so I was anxious to see what the new version had to offer. The product has a number of new features and capabilities that are both feature specific and impact the infrastructure of the product. Anyone who has worked with the ShoreTel iPBX for any period of time will become instantly comfortable with the new ECC system administration interface. It is now a browser based portal, very similar to the interface used for the iPBX administration. In fact I would be willing to bet that the new ECC portal will become the standard for the ShoreTel product line and the iPBX Shoreware Director will take on some of the characteristics of the new ECC 7 administration portal. Another area in which the ECC has adopted the exiting iPBX paradigm is in the area of licensing. Early versions of the ECC required the installation of a hardware dongle on the server and each desktop that ran the Supervisor software. Dongles are a pain for everyone and beginning with Version 6, ShoreTel began to migrate away from this requirement by eliminating the dongle requirement for Supervisors. Previous versions of ECC required the installation of the USB Hasp on the server before installing the ECC application. No dongle, not ECC! With ECC 7, not only are the dongles not needed, you get the familiar 45 day grace period to run the ECC application and try all the features before you have to provide the license keys. The server key locks to the MAC of the server, in a fashion similar to the iPBX key. The ECC application also reads the BIOS serial number of the server for added software protection. Other infrastructure changes include the support for Windows 2008 R2 64 Bit Servers. The ECC will support Citrix and WTS clusters and most importantly, roaming profiles are now supported. The system will now allow for 100 concurrent Supervisors and 1000 concurrent Agents, though you may define 2000 agents in the database configuration. The application support 400 DNIS reports and historical data can now be kept for 24 months allowing for year over year trend analysis. An exciting new feature is the addition of Personal Queues. I am sure we have all had the experience of working with an “agent” on a particular customer service issue, maybe given a “home work “ assignment only to call back and have to start over with a new agent. The concept of a “Personal Queue” makes it possible for inbound callers to reach specific agents, if you desire that option. In this way, after completing the “home work” assignment, you can call back in and queue for the agent that originally handled your call in the first place. Agents can move high priority calls to their personal queue with a simple mouse click. Historically, if you wanted this option you had to configure a group and service for each agent that required a personal queue. With Version 7, this process has been streamlined with the creation go a new entity that defines how the caller should be handled while in the personal queue. A very powerful option and very useful in direct selling environments. The familiar graphical scripting tools has not changed and scripts are generated using the established procedure. The Diagnostic console has been upgraded and is more usable for trouble shooting at the System Administrator level. I am particularly excited about the creation of a Lab SKU, something I would like to see ShoreTel do with the entire product line. The Lab SKU makes it possible for you to purchase and run a small scale Contact Center along side your production environment. In this way you can create new scripts, strategies, call flows and of course, test new upgrades before putting them into your production environment!

FastVoip now offers Free Calls to 13 Countries including India

Yeah, India is now independent again. FastVoip announced their best ever deal to make India calls totally free along with other 12 countries where you can make free calls. Lets find out about the deal. FastVoip, a leading VOIP company from Europe has just announced their biggest ever promotion to date. If you thought betamax was the only VOIP Provider to offer free calls per month to various destinations for a flat fee per month, then fastvoip has proved you wrong. Betamax as you all know is a unreliable VOIP Provider which can go block your accounts without notice and freeze your balance. FastVoip has now emerged as a strong competitor after this latest announcement to offer free calls to over 13 countries including India Landine. You can make calls to following countries using FastVoip. Belgium Netherlands Canada (+mobile) Poland China Sweden France Thailand Germany United Kingdom India United States (+mobile) Italy Please note the only calls to landline phones are free except for USA and Canada where calls to mobile phones are free too. In order to use the Free Calls promotion, you need to make a minimum deposit of 5 Euro by Credit card or $7.50 by Ukash. You will need to recharge your account every month for this amount in order to keep the free call service active. There is one more solid opportunity for all of you to make some money from fastvoip. I will discuss it later this week. For now, Download FastVoip and start making those free calls.

Call India for 0 cents per minute

Surprised or shocked? Yes, its true. You can now make calls to India for 0 cents per minute and if you miss this offer, you will surely regret later. IndiaLD, one of the largest India focused VOIP Provider has launched a smashing offer that will make you drool. With IndiaLD Quarter Call Offer, you can call any Indian Mobile for only 25 cents per call. Don't get confused, its per call and not per minute. Each call will last for an hour i.e 60 minutes which makes the effective calling rate per minute as $0.004 (less than 1 cent). This is a perfect offer for anyone looking to call India and talk at length. Don't we all love talking back home for literally hours? Even for people who actually hang up within 1 minute, there will be no charge for that short call, yes it will be free. IndiaLD currently offering $5, $10, $20 & $30 call credits to be bought for this offer. The quarter call offer is only available till end of April 2011. However, IndiaLD is seriously considering to extend the offer further into the summer holidays. You can call from the USA, Australia, Canada, United Kingdom, Bahrain, India, HK, Ireland, Japan, NZ, Spain Norway, Netherlands and Singapore using IndiaLD Access numbers. To be honest, I find this offer "the best ever" till date. The calling rates are unmatched and unheard of. Don't think and procrastinate, Join the party till it lasts and Sign up now at IndiaLD

Cypress Claims Hosted VoIP Enhances Business

Entrust your communications to someone offsite eliminates the need for expensive solutions to combat disaster situations. Hosted VoIP solutions can provide an added dimension of flexibility in times of disaster. It’s scalable; resilient; and can handle a “wide range of applications.” Cypress Communications has aimed its new product at the hosted VoIP solutions market. The complete end-to-end hosted VoIP and unified communications solution, called C4 IP, earns its name from its design to “help organizations connect, communicate, collaborate and continue even during a disaster or catastrophe.” Cypress Communications is the operator of a national private network designed for VoIP. It claims that its network has been “engineered for 99.999 percent availability and local site survivability with no single point of failure” due to its design to automatically track down any problems before they can cause service interruptions and repair them, with redundancy built in for “ensure maximum availability of voice communications.” Cypress distributes its hosted VoIP over carrier-grade Nortel CS2000 switching and Nortel MCS5200 multimedia. The CS2000, a central-office-grade switching platform with mirrored hardware components, is located in Dallas. Both halves operate with separate power supplies to reduce interruptions in the event of power failure or any failure of hardware or software.  The MCS5200 handles C4 IP’s multimedia components. Collaboration, soft phone functionality, presence, chat, video calling and Web conferencing all fall under its domain, headquartered in Atlanta, and it also provides a backup to the backup for the CS2000. Resiliency is created by located the C4 IP platforms in hardened carrier-grade collection centers with dual power feeds, overlapping generators and uninterrupted power supplies. These measures ensure that C4 IP customers won’t be without communications during a disaster. Scalability comes from an architecture able to handle up to 180,000 lines, 200,000 trunk channels and 2 million busy hour call attempts. Additional disaster recovery components include redundant voicemail servers with failover capacity, and a feature that allows users to copy voicemail over to email. C4 IP is fully compatible with Microsoft Outlook and offers features including “click-to-call,” instant messaging, unified inbox and other unified communications applications, video/audio/Web conferencing, find me/follow me, virtual number and related remote/mobile features.

VoIP Supply Named to Fast Track 50

For the third consecutive year, VoIP Supply earned a place in the Fast Track 50, a list of the 50 fastest growing companies in Western New York. Buffalo’s Business First newspaper bestowed the honor based on VoIP Supply’s growth in two areas: Total sales and number of U.S. employees. “The success of VoIP Supply has paralleled on the dedication, talent, and hard work of our tremendous staff. I cannot thank them enough, and this award belongs to each of them as well,” said Benjamin P. Sayers, president and CEO of VoIP Supply. Sayers was recently elected to the infoTech Niagara board, a trade association for western New York’s information technology industry. Sayers received the award June 17 in Buffalo. Sayers joins the board’s 20-member panel, which features influential names in New York’s technology industry, including those from the recruitment, academic, health, and other various industries. “It is a great honor to be named to such a forward-thinking group as InfoTech Niagara. They have great plans not only for professional and economic development, but for community outreach to students and the less fortunate as well,” Sayers said.

Citrix Incorporates VoIP With Conferencing Software

Citrix Online will release new versions of GoToMeeting and GoToWebinar software that now incorporate VoIP capability. Both software suites feature support for Mac and telephone audio, and one-click web conferencing, designed for small and medium businesses. “Our goal at Citrix Online has always been to develop the simplest and easiest-to-use Web solutions, which is extremely important to our SMB customers,” said Elizabeth Cholawsky, vice president, products and services for Citrix Online, in an article on MarketWatch.com. “With these newest releases of GoToMeeting and GoToWebinar, every feature is based on extensive user feedback, from full Macintosh access to this new total audio solution. Used effectively, these technologies can change not only the operating style but also the business economics of companies everywhere.” Integrating VoIP into the software will streamline the conferencing process. Often, conference calls include users of both VoIP and traditional phone services, and patching both types of users together was problematical. “The new VoIP capability gives us the opportunity to significantly scale up the business,” said John Lane, founder and owner of TaxSaleLists.com LLC. “Having both VoIP and phone connectivity options will encourage more attendees at our weekly marketing seminars, and now we can handle up to a thousand people. With this new product, we are free to ramp up our promotional efforts and really make an impact on the bottom line.” GoToMeeting was designed for smaller conferences, and GoToWebinar can support audiences of up to1,000. Both programs allow users to incorporate Keynote, PowerPoint, text documents, pictures and other material into presentations, training or product demonstrations. Video is not directly supported by Citrix products, but the company asserts many do use video in GoToMeeting or GoToWebinar presentations. Both programs include tools to monitor audience and gather feedback as well as determine who is participating and who is not. “With collaboration products, ease of use is critical, especially for SMBs,” said Mark Levitt, vice president for collaborative computing at research firm IDC. “Citrix Online’s one-click access to VoIP and traditional phone connections for both PC and Mac users is designed to give growing businesses one easy-to-use interface for a rich combination of popular tools that are easy to manage and afford.” Making the software Mac-compatible was a move the company hopes will be a popular one, citing a sharp increase in Mac use among customers.

C2Call - New browser-based Java VoIP app

C2Call (C2Call) lets you make web-based P2P VoIP phone calls direct from your browser whether you're using Linux, Windows, or an Apple Mac. The software is a Java widget that is loaded temporary from the C2Call web page into your browser each time you use C2Call. Using the C2Call Java widget I spoke with Martin Feuerhahn, Director of Actai Networks Pte Ltd Taiwan Branch. The voice quality was excellent and the latency wasn't too bad either considering I'm in Connecticut and he's in Taiwan! In Internet Explorer when you first load the widget you'll see that the application has a digital signature that has been validated by a trusted source: Actai Networks GmbH, the company mentioned in the above screenshot is the parent company of www.c2call.com. In Firefox the app just ran without showing the digital signature verification message. The client is just 1.1MB in size so it's fairly small. The beauty of C2Call is that you can send a call invitation to anybody you want to talk to simply by sending an email invite directly from the www.c2call.com applet. The recipient can click on the link and can instantly talk to you. This is a really great viral feature that is especially good for non-technical people. For instance, you can invite your grandma to talk simply by emailing her. Unlike Skype which would require her to install the software, register a username, configure Skype, etc., grandma can just click a link and speak to you almost instantly. Here's a sample email invite being sent out to my boss and industry-leading VoIP pundit, Rich Tehrani. The widget is SIP-based and importantly it uses the Speex codec, which gives the software its very good voice quality. So you're probably wondering how a Java-based VoIP app fares versus a Flash-based VoIP app, such as the Flashphone app I've written about. Well, for one if you decide to use an Adobe Flash-based solution, you have to use the Nellymoser codec, which is not a free codec. Second, Adobe Flash requires the use of a media server, which Martin claims is not as scalable as their true P2P (peer-to-peer) VoIP solution. Another feature of note is that C2Call can traverse about 90% of firewalls using standard SIP technology. The official launch isn't for another couple weeks, but if anyone wants an invite give me a shout (Contact link above).

D2 Technologies Releases Mobile Handset Solution Powered by Google Android

D2 Technologies today released their mobile handset solution powered by Google Android. D2's mCUE mobile convergence software solution combines a communications user interface with the company’s vPort MP VoIP software platform and is targeted towards OEMs and service providers to help deliver integrated Fixed Mobile Convergence (FMC) and Unified Communications (UC) functionality. mCUE provides a complete embedded software framework for multi-mode mobile handsets for enterprise and consumer use, such as dual-mode cellular plus Wi-Fi phones. Its completely Java-based user interface framework for Linux can be ported to other GUI platforms. “mCUE revolutionizes mobile communications by tying together the best aspects of PC-based communications, such as VoIP, instant messaging and presence support, with the roaming benefits of mobile cellular and connection speed and quality of in-building wireless,” said Doug Makishima, vice president of marketing at D2 Technologies. “It is a complete turnkey solution for multi-mode mobile communication devices.” mCUE is interoperable with enterprise IP-PBXs and unified communications systems. Looking at the GUI and the feature-specs, this definitely seems like a pretty cool product that I need to get my hands on and test. Check out the news today. LAS VEGAS, CTIA Wireless 2008 (Meeting Room 355) — April 1, 2008 — D2 Technologies, the market leader in embedded software platforms that power IP communications, today announced that its mCUE™ converged communications client for mobile devices and handsets now supports Google’s Android, one of the industry’s first open-source mobile platforms. D2’s mCUE product line, coupled with Google’s Android software development kit, now enables OEMs to rapidly develop handsets powered by Android that offer a premium, graphic- and media-rich user experience. The mCUE solution delivers device interoperability with enterprise IP PBXs, Unified Communications (UC) systems and service provider networks, integrating all session types including voice, IM, SMS and email messaging through a converged presence-based communications user interface (CUI). D2 is unveiling the integration solution at CTIA Wireless 2008, being held at the Las Vegas (NV) Convention Center April 1-3 (Meeting Room 355). “This offering will give developers a complete turnkey solution for developing multi-mode mobile communication devices based on Android,” said Doug Makishima, vice president of marketing and sales at D2 Technologies. “There is considerable market anticipation around Google’s Android platform, and our goal is to make it as fast, easy and affordable as possible for OEMs to deliver converged, unified communications devices to meet that demand.” D2’s mCUE pairs an innovative, patent pending communications user interface with the company’s vPort MP VoIP software platform to address the needs of OEMs and service providers delivering integrated Fixed Mobile Convergence (FMC) and UC functionality. mCUE provides a complete embedded software framework for multi-mode mobile handsets for enterprise and consumer use, such as dual-mode cellular plus Wi-Fi phones. Mobile devices with mCUE provide users with advanced presence-based and push-to-x control of cellular and VoIP calls, PBX extension feature activation, IM, email, SMS and other features typically only available on PC-based unified communications soft clients. Its revolutionary user interface, built on top of a multi-identity, multi-session, multi-protocol engine, enables users to simultaneously utilize multiple different communications services such as enterprise IP PBXs and UC systems, and commercial VoIM services like Google Talk, Yahoo!, MSN, AIM, and others. Developed by the Open Handset Alliance, the Android platform is a Linux-based software stack for mobile devices that includes an operating system, middleware and key applications.

Yahoo Messenger for Vista Adds VoIP

If you recall in my December Yahoo! Messenger for Vista post, this version which was pretty much completely rewritten didn't have VoIP yet -- unlike the XP version which does have VoIP capabilities. Well good news my friends! I learned via Josh Jacobson, Senior Product Manager, Yahoo! Messenger for Vista on his blog that PC-to-PC VoIP (free calling) is now available as well as PC-to-PSTN (Yahoo MessengerOut?) It also features "Phone In" number so you can receive calls. This latest beta of Yahoo Messenger for Vista will have cool voice visualizations for the product that leverage the WPF 3D integration. (see image right) SMS will also be part of this next release, allowing users to send text messages from Yahoo! Messenger for Vista to a friend’s mobile phone for free. The plan is to add Yahoo! Mail alerts to this next version as well. I'd like to see Yahoo! Messenger for Vista be able to make VoIP calls to Microsoft Live Messenger. They're both SIP-based after all and they already interoperate with IM. But then again, Yahoo! and Microsoft have been butting heads lately with Microsoft trying to acquire Yahoo and Yahoo playing "hard to get" wanting to be courted for more cash. Anyway, you can download and try out the preview version available at http://messenger.yahoo.com/vista Also, read more about the new features on the Yahoo! Messenger blog. http://www.ymessengerblog.com/blog/2008/03/06/in-the-mix-with-yahoo-messenger-for-vista/

SippySkype SIP-to-Skype Gateway

Last month I wrote about my strong disagreements with a guest blog post on Skype Journal that was titled "A SIP/Skype Gateway Is NOT In The Forecast". I disagreed with the premise that a SIP/Skype gateway isn't in the forecast. Well, Skype Inc. may not care about offering a SIP-to-Skype gateway, but that doesn't mean VoIP enthusiasts are going to sit around and not build their own SIP-to-Skype gateways! I've blogged on a few occasions where I discussed the desire for Skype users to have SIP connectivity and various home-brewed solutions. Well, we can add another SIP-to-Skype gateway solution to the mix. I just discovered SippySkype today, which is an open source & free SIP-to-Skype gateway. Check out the features: SippySkype is Java software that allows you to make and receive Skype calls from your SIP/VoIP adapter. Basically a Skype/SIP Bridge/Gateway/Proxy Call Skype Users using speed dial or use Skype out. SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks - 1.1 or higher Skype to SIP authentication/denial mappings via incoming Skype User ID - 1.1 or higher Support RFC2833 touchtone decoding (DTMF) - 2.0beta or higher Could be used as an endpoint with Asterisk Auto play pre-recorded file(s) to SIP callers - 2.0beta or higher Incoming SIP Pin number authentication and dialing - 2.0beta or higher Open Source - You can modify/fix it if you like. It's free System Requirements: Skype Client Working Java 1.6.0 or better runtime mjsip/mjua 1.6 http://www.mjsip.org/ - Use those included with SippySkype as some bugs have been fixed. Skype4Java 1.0 https://developer.skype.com/wiki/Java_API - Unmodified SIP/VOIP adapter such as a spa-3102 to make and receive Skype calls or register with a provider or Asterisk. Should work where Skype4Java works (windows/linux/osx). (I'm using it on Windows XP)

PBXtra 4.0 Released

Fonality today announced the release of PBXtra 4.0. New features in PBXtra include FindMe with Boomerang Mobile Integration, a feature that uses presence detection to automatically find employees on their mobile devices, allowing them to answer the call or bounce it to another extension. Other new features include tighter integration with mobile phones and web browsers, and enhanced support of branch offices. One really cool feature that is part of the Boomerang Mobile feature is that you can dial *1 to record the mobile phone call and have it automatically stored on the PBXtra server. Another cool feature that Chris Lyman CEO of Fonality gave me a sneak preview a few weeks ago was FONcall, a new PBXtra plug-in for the Firefox web browser. It turns any phone number on any site into a link. When you click the link, PBXtra will automatically take an Aastra and Polycom off the hook and dial out to the number hands-free. I pointed out to Chris that there are similar plug-ins out there including Skype's browser plugin, and I added that Skype's plugin often brings your computer browser to its knees. Chris said he was well aware of that fact and they spent countless hours developing their plugin to make sure it wasn't a CPU hog. Also, no support for Internet Explorer yet - possibly never since Chris stated it was much harder to develop plugins for IE. (though I should point out that Skype's plugin works on IE) Greg Galitzine has more on this news and was the first to post the story about this new release.

Talkonaut 4.0 native Symbian S60 edition released

The GTalk2VoIP dev team just released their VoIP+chat application for Symbian S60 phones that runs as a native (.sis file) application. I should stress that several Nokia smart phones come with a SIP stack, however they are limited to WiFi use only and won't work over your cellular network. You can thank the carriers for that one. Talkonaut on the other hand is not bound to WiFi only, allowing you to make calls over GPRS, EDGE, 3G or WiFi. Talkonaut has essentially developed their own VoIP SIP stack that allows Talkonaut users to make voice calls over most data connections your mobile phone might have, such as GPRS, EDGE, 3G and WiFi. The application also sports the ability to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients. I believe you can even chat with MSN Messenger, Yahoo, AIM, and ICQ users. Here's a list of the new features in the 4th release according to the release: - Talkonaut 4.0 was entirely rewritten in C/C++ and now runs on Symbian S60 3rd edition based Nokia smart-phones. Moving from Java to native platform allowed to reduce memory usage, improved speed and to add some functionalities that were not previously available for Jave applications, like VoIP, access to file local system (improved file transfer) and interaction with other applications (Web Browser for opening URLs right from chat window). - Using a set of narrow-band Speex codecs and relying on Jingle Audio extension to XMPP protocol (same as implemented in Google Talk), Talkonaut now brings to the world a very powerful combination of IM chat, Presence and VoIP calls made over data connection. - Talkonaut 4.0 allows to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients, as well as to other Talkonaut fellows. - Talkonaut 4.0 allows to receive free calls from SIP phones (or any other VoIP networks), from Google Talk, MSN, Yahoo, AIM and ICQ users. - Talkonaut 4.0 allows to make cheap VoIP calls to any mobile or landline phone number in the world. - Talkonaut 4.0 allows to define any number of SIP accounts and use them to make free or cheap calls over third-party VoIP/SIP carriers. A flexible Dialing Plan feature is helpful for choosing routes to destinations between different carriers. - Talkonaut 4.0 is fully equipped with all the features previously available in J2ME version of Talkonaut 3.0, inherits the same graphical user interface and menu structure. Talkonaut 4.0 runs on the following sets of Symbian S60 based Nokia smart-phones: o 3rd Edition: 3250, 5500, 5700, 6110, 6120, 6121, 6290, E50, E51, E60, E61, E61i, E62, E65, E70, E90, N71, N73, N75, N76, N77, N80, N81, N81 8GB, N82, N91, N92, N93, N93i, N95, N95 8GB Talkonaut 4.0 can be downloaded over-the-air using mobile web browser from http://get.talkonaut.com/, or via PC from http://www.talkonaut.com/download.shtml

Junction Networks Adds Inbound Bridge to onSIP Hosted PBX

Junction Networks, provider of the hosted onSIP PBX service, has launched Inbound Bridge, an accessory service that ties in third-party providers of international and domestic DID numbers. Inbound Bridge saves money for Junction Networks’ business customers by allowing them to find the best per-minute price for inbound VoIP calling minutes in their chosen geographic regions. Essentially this bridging application lets hosting customers use third-party providers of DID numbers for best LD price in desired footprint. According to Junction Networks, "Inbound Bridge solves a vexing problem for companies that want local numbers and 800 numbers in foreign countries or specific domestic regions—numbers typically rented from other VoIP network providers. Many of these providers simply deliver calls from one end point to the other; they don’t implement the key function of the SIP VoIP signaling protocol that allows callers to navigate phone menus, transfer calls, put callers on hold and trigger other events. Without this function (specified in the IETF’s RFC 3515), inbound callers are disconnected when they “press 1 for sales,” or try to transfer to extension numbers for specific people. Or at the least the DTMF touch tones are not recognized. Junction Networks' Inbound Bridge performs the missing SIP function, maintaining the two-way SIP signaling needed for interactivity after a call is already established. It can therefore pass the entered touch-tone digits to the auto attendant and other PBX applications. It can integrate onSIP with any network vendor having an open SIP implementation (permitting communication with other SIP networks). “This is a perfect opportunity if you’re a company that wants a toll-free service from Europe or Asia, or you have some other reason to use a third-party phone number provider,” said Rob Wolpov, president, Junction Networks. “But you do want to use our hosted solution, because it works very well, you have company extensions in one or more sites, and you like the price of that.” ($39.95 a month for core voice applications and an unlimited number of SIP extensions, with free calling between them.) “You can get your toll-free origination from any provider that offers SIP delivery of inbound calls, and just have it go to us through the Inbound Bridge. We charge $1.95 a month per DID, and half a cent a minute to cover our cost of the Bridge. If you can get your toll-free for one or one and a half cents a minute, you wind up paying two cents a minute for toll-free, as opposed to paying almost four cents a minute to Junction Networks. “That’s completely OK with us,” Wolpov stressed. “Go somewhere else to get your minutes and then come to us for the auto attendant, the ACD queue, the voice mail, all those sorts of applications.” To date, Junction Networks has tested the Inbound Bridge with international DID provider Voxbone. Other third-party DID providers are to be tested and added in the coming months. “We’re happy to supply onSIP customers with our core value—locally dialed phone presence in more than 40 countries around the world,” said Rodrigue Ullens, Voxbone CEO. “With our intercontinental voice-only backbone, Junction Networks’ Inbound Bridge and onSIP platform, these enterprises can get the best value in DID numbers and international VoIP transport, plus all the convenience and flexibility of hosted IP PBX.”

Hosted trixbox IP-PBX

Just learned via TMCs' VoIP Forums that Click4PBX just announced hosted trixbox plans starting at $109/month (1-10 extensions) and going up to a maximum of $279/month (1-300 extensions). It's unclear from the website if that monthly fee is per extension or the total cost for each particular plan. I believe it is the total cost since the alternative (per extension) would be cost prohibitive. Not to mention the price would go down per extension as you scale up where as the plans listed increase in price as you increase the number of extensions. It just seems so cheap to only pay $109/month for up to 10 extensions. A more typically example for the SMB is 50 extensions and that's still only $139/month. Actually, all Click4PBX is doing is hosting the trixbox software. Well, they are hosting the PC hardware as well and providing some failover redundancy I suppose. So there is some value in that if you are an SMB that doesn't want to maintain hardware & do periodic trixbox software upgrades. In any event, with this solution all you need is DID numbers and SIP trunking termination services from Click4PBX's list of certified VoIP providers. Your IP phones connect directly with the SIP service provider, while Click4PBX hosts the trixbox management interface, which leverages FreePBX, a web front-end to several Asterisk-based solutions. I'm very high on SIP trunking as you know, so it'll be interesting to see if more hosted trixbox or hosted Asterisk-based solutions come to market. We already have thousands of website hosting companies - could a plethora of hosted PBX providers be far behind? I should add that Click4PBX also hosts VICIDIAL, a set of programs that are designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound call center suite. It includes predictive dialing capabilities, inbound, Outbound and Blended call handling and more.

SimplyExchange Skype PBX Gateway

Wisecom Technologies Ltd is now selling a Skype Certified gateway called SimplyExchange, which hooks up to your corporate PBX using four analog FXS ports. This is similar to the Actiontec VoSKY Skype gateway, though SimplyExchange is a much smaller unit than the 1U VoSKY gateway appliances. In any event, you simply connect 4 SimplyExchange analog FXS ports to 4 PBX FXO trunk ports, which will "Skype enable" your PBX. Alternatively you can connect to 4 telephones (or some to trunk ports and some to telephones) Next, you configure your PBX by setting a hunting group number for the four trunk ports to a specific number, such as 8. This hunting group will be act as Skype lines allowing a corporate user to simply press '8' to get access the Skype network, as seen by these 2 diagrams: Then you hook up the SimpleExchange gateway to a PC (via USB) running Skype and install their software. After configuring four Skype IDs you'll see this in your System Tray - 4 SimplyExchangeExt icons, 4 secondary Skype account icons, 1 primary Skype account icon, and 1 SimplyExchangeCenter icon. Pretty cool seeing 4 green Skype icons in the System Tray! Technically, you'll need 5 Skype IDs, since one of them is used as your "primary" Skype account which runs on your PC and communicates with these 4 secondary accounts. You just need to tell this primary account to other Skype users. When somebody calls this primary account, the incoming call will be distributed automatically to one free account of these four secondary accounts. You can obviously configure a SkypeIn phone number as well for inbound PSTN calling. What's really cool is that the installation installs a local PostgreSQL database and a local web server on your PC so you can do some cool web-based management. The web interface keeps a call history, lets you view your contacts and has other capabilities. You can even import your Skype contacts using the web interface as seen here: Features Integration of Skype into existing PBX system Supports 4 lines Skype call in and call out at the same time Unique enterprise Skype account as hunt group with automatic distribution for all incoming Skype calls Up to 16 digits Skype contact speed-dials Centralizes all Skype activities to a single server - good for enterprise management Web interface for contacts management and history/log check Uses only one Windows account to handle 4 Skype accounts Supports 4 inbound and/or outbound calls concurrently Automatically finds a free line/port for the incoming or outgoing Skype call Works with the existing PBX to provide other telephony features such as Call Transfer, Call Pickup,Call Hold, Call Waiting, Conference Call... etc. (PBX dependent)

Jajah liberates VoIP users from PC

The rivalry between Cisco and Microsoft has once again intensified, as Microsoft launched its Unified Communications portfolio to take a significant share of the VoIP market pie. Cisco has already admitted that Microsoft�s entry into the VoIP sector has already slowed down its sales cycle with customers are waiting longer to make a comparison between the portfolios of the two companies before proceeding to buy. As for Microsoft's Unified Communications initiative, the company promises to treat voice as just another data type that can be routed around an office, much like an email. The new products include Microsoft Office Communications Server2007, Microsoft Office Communicator 2007, Microsoft Office Live Meeting, a service pack update of Microsoft Exchange Server 2007 and a Round Table, Ethernet-connected, 360-degree videoconferencing VoIP phone that will cost $3,000. Microsoft's new initiative is to eliminate the traditional PBX systems that run office environments. As for Cisco, its CTO of unified communications has said: There's a couple of extra conversation while we work through what does Microsoft actually have, what does Cisco have and what are the various ways they can work together.

What is VoIP

Voice over Internet Protocol (VoIP) is the family of technologies that allow IP networks to be used for voice applications, such as telephony, voice instant messaging, and teleconferencing. VoIP entails solutions at almost every layer of an IP network–from specialized voice applications (like Skype) all the way down to low-level quality measures that keep those applications running smoothly.

Unless you’ve been sleeping under a very big rock for the last year, you’ve certainly heard the phrase “Voice over IP” uttered. Perhaps you’ve seen those hilarious Vonage commercials that feature painful and embarrassing accidents caught on tape, promising to let you dump your local phone company in order save big on your phone bill. You may also have seen the Cisco telephones that are curiously inserted in prime-time shows like 24.

What is all the hubbub about, anyway? Why, VoIP, of course! VoIP, the fabulous secret ingredient in Vonage, Skype, Cisco CallManager, and a host of other revolutionary technology products you may have already encountered on TV, in the news, or in person. But what makes these products so revolutionary? What is it about VoIP that is such a big deal?

Voice over Internet Protocol is a family of technologies that enable voice communications using IP networks like the internet. Inventive developers and entrepreneurs have created an industry around VoIP technology in its many forms: desktop applications, telephone services, and corporate phone systems. VoIP is a core technology that drives everything from voice-chat software loaded on a desktop PC to Mac full-blown IP-based telecommunications networks in large corporations. To the Wall Street speculator, VoIP is a single technology investment with many revenue streams. To the enterprise network engineer, it’s a way to simplify the corporate network and improve the telephony experience for users of the network. To the home user, it’s a really cool way to save money on the old phone bill.

What is Internet Telephony?

The question “What is Internet Telephony?” has answers that prove that the face of communications as we know it is changing. Today, the internet is not only a pathway for data transmissions. It is now also possible to make telephone calls over the internet, a technology which makes paying long-distance charges optional instead of mandatory.

So exactly what is internet telephony? Also called Voice over Internet Protocol (VOIP), it is the technology that gives the ability to take an analog voice signal and translate it to a digital signal. That digital signal is then sent from one computer across the internet to the receiving computer, where it is reconverted to an analog signal that the receiver hears as the caller’s voice.

Or, if only the originating caller has VOIP, they can still make a call using VOIP. The provider will route the call across the internet to the ‘node’ (local phone switch) nearest the receiver’s location. Usually the caller has to pay a small fee for this service. By using the internet instead of a long-distance company’s telephone network, the caller avoids paying any long-distance fees.

There are, however, variations to the answer of the question about what is internet telephony. Depending on what the customer wants or needs, their level of comfort with technology and their other telephony needs, they have a choice as to what kind of internet telephony service they select.

So when we consider the question of what is internet telephony, we must start with the basic VOIP technology and then look at the packages in which it is offered to the customer.

There are VOIP providers who provide both the internet (usually, broadband internet) service, local phone service with all the calling features the local phone company offers, plus the software and hardware needed to make calls. They usually charge equipment fees and monthly service fees.

Then there are those who provide only the VOIP service over your existing broadband connection, and equipment is optional. But if you have a PC with speakers, a sound card, a microphone and an internet connection, you can even purchase or download software that will allow you to set up your own VOIP calls, free of any monthly or long-distance charges.

88% of business users back VoIP technology

Nearly nine out of ten corporate users of voice over internet protocol (VoIP) applications are happy with the technology but some businesses are missing out on the benefits it can offer, according to a new report.

Research carried out by Data Integration in conjunction with Extreme Networks found that VoIP telephony systems have around 60 per cent market penetration among big businesses, with 88 per cent of users giving the technology positive feedback, reports vnu.net.

Another 27.2 per cent of businesses intend to implement VoIP communication systems within the next three years, the survey found. However, less than a third of those questioned were currently running IP video over their networks.

According to Extreme Networks’ regional director for the UK and Ireland, this low uptake reflects a “lack of understanding” about the potential of video communication for businesses.

“This [uptake] could be down to the false start caused by early and expensive video solutions, or the lack of understanding of its future benefits,” he said. Benefits of VoIP and video conferencing include reduced environmental impact from less travel and lower telecommunications costs.

The survey questioned UK firms with over 500 employees.

Top VoIP Phone System Features

Before you can make the decision about what you need from a phone system and what features should go on your checklist, you need to be sure you know all about the surrounding parameters. You need to know about the network you are connecting to, the size and expandability of the system you want and what the mission critical business application you need from a phone may be. This includes subtle decisions like how much disruption to operations and training your organization can tolerate in the transition.

There are three basic kinds of phone system. The first is what is called a keyless system or a KSU-less system. These are very lightweight phone systems that are for organizations with fewer than ten employees that also have very basic phone requirements. The advantage of these systems is that they are very inexpensive and very easy to set up and run. They are portable and can be moved easily if you relocate. Typically you get very little support and maintenance from your service provider in a keyless system. In addition, features like preventing users from picking up a line in the middle of another call are often impossible to implement.

Next largest is what is called a key system. These use a key system unit (KSU) that is a central control unit that provides features and functions for your phone system that aren’t available on ordinary phones such as extension management and locking users off from lines that are in use. These are essentially closed, ‘black-box’ units that provide 80% of the functions of a basic PBX system for a lower cost. They are typically used in businesses with from 5 to 40 or 50 employees that have only basic business telephony needs.

Finally are PBX systems – these are the most common kind of business phone system and they are the most flexible. With PBX systems that are designed fro as few as 5 users and as many as several thousand users in multiple locations. You can choose to have the PBX system on your business premises or you can have it hosted at a service provider who then performs upgrades and maintenance for you.

Then you have to look at the features of the phone system – these are different than the features of a particular phone itself. They are the features that provide the added business value which is the reason you turn to a phone system in the first place. It may be that you business needs clear, easy-to-set up conference calling, or call tracking that can be tied in to a client billing system, or the ability to tie employees’ cell phones into the overall phone system, or any of over a hundred other complex features that can be important in business.

Beyond the basic decision of what the phone system should do is also the decision about how to connect the phone system and if you have the required infrastructure or networking in place to support the phone system you want. You may find that you have to budget a network upgrade or a broadband supplier upgrade. And that may change your budget priorities.

But it is easy to forget the basics when you start looking at advanced features. Before you let yourself get seduced by ‘find-me-follow-me’ and click-to-call, you might want to check that your basic business-critical functions are covered. For example, there are many VoIP phone systems that have advanced features but do not offer paging or intercom features.

The other non-negotiable issue is call quality. Fortunately there are a few basic issues to make sure of in terms of call quality and if they are dealt with then VoIP will often offer better call quality than any older phone technology. The primary issue is bandwidth. You need about 128kbps of bandwidth per active call line in your phone system. You also want to evaluate your internal network to look at priority of traffic and latency to make sure that packets get routed internally fast enough to maintain call quality. A good VoIP service provider can help with analysis and guidelines to make sure your system can keep up and some service providers will offer real-time monitoring to make sure your phone system can keep up.

The last issue is around hosted versus on-premise systems. This comes down mostly to a business decision. Do you need to maintain control and core access to management and features of your system in house? Or do you want to install and forget and not think about your system? In the latter case you should look seriously at hosted solutions and in the former at on-premise solutions.



Key Questions to ask yourself before deciding on a phone system.

What is the size of my business in terms of number of employees using the phone now?
And in two years?

What kind of internal computer and network systems do I have? Can they handle VoIP?
Do I need to install something new or perform and upgrade?

What are the core features I need from a phone system – no matter what? The features that I cannot leave out for the good of my business?

The answers to these questions need to be your foremost guide. In terms of size you need to be able to figure out how many extensions (actual phones) you need AND you need to know how many outside lines you need. Typically this is about a third of the number of phones but if your business involves a lot of calling for sales or service reasons, you may need more. In terms of your network you do NOT want to install a phone system and then discover that to make it work effectively you need to spend as much again on upgrading your network. And in terms of features you do not want to compromise on the features you need just to get something fancy which may or may not fit into your day-to-day operations.