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Digium Launches Support Services for Asterisk
Some big news from Digium. Rich Tehrani met with them yesterday to get the inside scoop. Rich takes copious notes on his iPhone, which he sent off to me to try and write up this news. Alas, I’ve been pretty busy myself, but I wanted to share Rich’s notes below, since there are some good “nuggets” in there.
For instance, from Rich’s notes I see that Switchvox 4.0 is on the verge of shipping. But the really big news is that Asterisk has announced the general availability of technical support subscriptions for open source Asterisk. Before if you wanted support from Digium, you had to purchase Asterisk Business Edition. Well, no longer. Now, all of you Asterisk fans out there that try Asterisk and get stuck can now contact Digium and get some support. No more relying on the Asterisk community to answer you questions. Not that asking the Asterisk community is a bad thing, but if you phone system is down, you can’t wait hours for someone to respond to an online posting. This could be a huge revenue-generating opportunity for Digium, which can now monetize the open source version of Asterisk with support subscriptions. I’m surprised they didn’t offer it sooner. Maybe they were afraid it would upset channel partners?
Rich’s notes:
- Open Source Subscriptions
- 2 smb subs
- And 2 enterprise class
- Incident based
- Problem: up to today needed community support or consultant with hourly rate
- Now annual sub - 3 year 10% discount
- Can call Digium based
- Level one - support local hours 12 hours - starting at your 8:00 - 7:00
- For 5 days a week
- Buys sub online
- Available in a month through the channel
- Get a key, name contact and get details when you call
- Get incident/case handled
- Can open via we or phone
- Find a bug - gets entered in bug tracker
- Gets handled like any biz edition type of bug
- Not really SLA like a commercial licensed product
- Biz edition - now only available as OEM or commercially licensed product
- They want people to buy the open source - engineering opens up 1.4 and 1.6 - first time Digium provides support for open source asterisk
- Up till now consultants, etc
- Open source - people buying business edition for support reasons
- Now getting open source subs
- Can now support enterprise class apps
- In the past - anyone who built a large network - 2 levels of enterprise class support
- 24×7 - server based
- Unlimited users
- Up to 3 names contacts
- First foray into enterprise from server side
- Up to 24×7 support
- Switchvox 4.0 on the verge of shipping
The new Asterisk support services enable companies to leverage the power of open source Asterisk with the confidence that their system will be supported by the very founders of the Asterisk movement. According to the news release, “The support subscriptions provide technical support, hardware replacements and substantial discounts on training programs to enable users to take full advantage of the power of the Asterisk platform.”
“Digium’s new subscription services give Asterisk users the best of both worlds–they can download and use Asterisk free of charge, as always, and now they can also call on Digium for technical support when needed,” said Spencer. “We think the combo of free and open, with support, is going to appeal to many of our most technical users. The Asterisk community has long been a source of great expertise through online forums, and now we’re supplementing that with the ability to call us, 24×7, for access to our Asterisk experts.”
Danny Windham, CEO of Digium, said: “As Asterisk gains traction within large businesses, demand for professional support is on the rise. Our deep knowledge of open source Asterisk and total commitment to its development makes us ideally suited to offer these new services. Companies that purchase subscriptions will receive support from the most knowledgeable group of Asterisk experts in the industry. We see this offering as a substantial step forward for Asterisk in the enterprise and a valuable service for companies of all sizes.”
Asterisk support subscriptions are bundles of services sold on an annual basis. They include technical and engineering support, consultative services, advance hardware replacement, and discounts on Asterisk training and conference passes.
Asterisk support subscriptions are available immediately from the Digium webstore at http://store.digium.com and will be available through Digium channel partners in Q2. SMB pricing begins at U.S. $595 per year for support during the subscriber’s business hours (8:00 a.m.-5:00 p.m., Monday through Friday); 24×7 support for an SMB begins at U.S. $1,995 per year. Enterprise subscriptions, including 24×7 support, begin at U.S. $3,995 per year. Pricing includes a defined number of servers supported and cases opened per year.
You can read the official news announcement here.
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Tags: technical support, subscriptions, asterisk, open source, digium, ip-pbx, linux, switchvox 4.0, voip, mark spencer, danny windham
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Skype For SIP Marries Skype and IP-PBXs
Today, Skype announced it is enabling SIP-based IP-PBX to connect to the Skype network, which will allow low-cost SkypeOut calling, receiving calls from Skype users, and receiving calls from regular PSTN phone lines. Outbound calls from IP-PBX SIP handsets to Skype phones is not part of this news announcement. Skype commented it is too difficult to dial Skype usernames from a desktop handset.
Features:
- Receive and manage inbound calls from the 405 million Skype users worldwide on SIP-enabled PBX systems, connecting the company website to the PBX system using Skype click-to-call buttons
- Place calls via Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX, saving your business money with Skype’s low rates
- Purchase Skype online numbers to receive calls to the corporate PBX from landlines or mobile phones
- Manage Skype calls using your existing hardware and system applications such as call routing, conferencing, phone menus, voicemail and call recording and logging - no additional downloads or training are required
Skype For SIP is perfectly suited to businesses that already have IP-PBXs and want to connect to Skype’s network which offers low-cost calling. Skype for SIP is being launched as a closed beta program, but you can register and try to be part of the beta. Interestingly, Asterisk for Skype, a SIP-based (and IAX) open-source IP-PBX was the first to offer some tight integration with Skype.
If Skype continues to open their proprietary doors (they recently announced they were giving away the SILK codec), and now they’ve finally enabled SIP-to-Skype integration, then we can see a momentous shift from SIP trunking to Skype trunking. Customers will not only choose the lowest cost IP trunk, but also the one with the best quality. Skype is renowned for their high-quality due to their P2P architecture, so companies will look very closely at Skype trunking instead of SIP trunking.
And of course if customers shift to purchasing Skype trunks, the SIP trunks will have to follow suit, which means a win-win for businesses looking for low-cost calling.
Via Skype blog
Tags: ip-pbx, open source, sip, Skype, Skype for SIP, voip
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Polycom VVX 1500 Media Phone Game Changer?
Today, Polycom has launched the Polycom VVX 1500 touch-screen business media phone, a new VoIP phone that combines IP telephony with business-class video and the ability to integrate with business applications. Recently, Verizon make a big splash with their consumer-class Verizon Hub, a multimedia phone that combines VoIP, Internet access, color screen, video streaming, and more. One could easily make the case that the Polycom VVX 1500 is the “business-class” version of the consumer-oriented Verizon Hub phone.
Although there are many similar features and both could be classified as “media phones“, the Verizon Hub does not do video conferencing, since it does not have an embedded camera. The Polycom VVX 1500 on the other hand does have a video camera embedded and is therefore more suited to video conferencing, which is more prevalent in the business world any way.
The Polycom VVX 1500 combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone along with Polycom HD Voice (wideband telephony) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content. It also includes a color touch-screen interface making this a very unique business IP phone.
So is a business-class media phone with a color touch-screen, web browsing, and video conferencing capabilities a game changer in the VoIP space? Well, the VVX 1500 has a list price of U.S. $1,099, so this is not an IP phone for everyone’s desk in a corporate office. A decent IP phone for the every day worker can be had for $150-$300 which is much less expensive. However, for business executives, CEOs, VPs, and other high-level management, the VVX 1500 is a very attractive IP phone. Often times if a VP or CEO has to have a high-quality video conference, they have to reserve a high-quality video conferencing system located in a particular boardroom. With the VVX 1500 they can stay at their desk and have their meeting. Further, impromptu video conferencing with co-workers sporting a VVX 1500 on their desk can be had allowing for quick collaborative meetings.
In-Stat is very high on the prospects for business-class media phones. According to Keith Nissen, principal analyst at In-Stat, “We anticipate that within five years, nearly 10 million business media phones will be shipped worldwide, generating more than U.S. $3 billion in annual revenue. They are a key to the future of the IP PBX business.” He added, “With its rich heritage in voice and visual communications and content sharing, Polycom is well positioned to be a leader in this new world of communications. The company’s VVX 1500 is the first business media phone that enables customers to work more efficiently and effectively than ever before by tying together voice and visual communication with critical business processes.”
Polycom VVX 1500 Touch-Screen
“There is growing demand from our service providers and customers to help them configure video within our BroadWorks call control platform,” said Mike Tessler, CEO of BroadSoft. “We have a long history of teaming with Polycom to deliver high quality hosted VoIP solutions, and the VVX 1500 is especially compelling because it goes far beyond the functionality of a traditional video phone by combining rich telephony, business-class video and an applications platform that is all deeply integrated with the BroadWorks platform, and it is extremely easy-to-use.”
The VVX 1500 was also specifically designed for lower power consumption, using power over Ethernet (PoE) using IEEE 802.3af, and requiring less than half the power of similar competing products such as traditional video phones. The device’s cool smart-motion technology enables the screen to go into power-save mode when no one is in the office.
The VVX 1500 features an open API and microbrowser that enable third-party application developers to integrate VVX 1500 with business applications such as unified communications, customer relationship management (CRM), and appointment management systems. The always-on, touch-screen user interface of the VVX 1500 includes a menu screen on which developers can place icons for users to locate and start their applications.
Polycom VVX 1500 Profile View
The VVX 1500 comes bundled with several applications including the Polycom Productivity Suite, which enables users to initiate and control audio conference calls right from the device’s screen as well as record calls locally using a flash drive in the phone’s USB port. The VVX 1500 also features a free Web service called My Info Portal through which customers can select to receive content such as local weather reports and other personalized information on the screen when the device is not in a voice or video call.
Interoperability is not a problem since the VVX 1500 uses the same Session Initiation Protocol (SIP) software as incorporated in Polycom’s SoundPoint IP and SoundStation IP desktop and conference phone product lines to communicate with SIP based IP-PBXs and hosted SIP servers. The product is in the process of being SIP video-certified by Polycom’s ecosystem of more than 30 VIP and VoIP Field Verified call control partners, including BroadSoft, Deltapath, NEC Sphere, Objectworld, and Zultys.
“Our customers consistently seek better leverage of their communication systems to improve productivity and reduce costs. They also expect Polycom to continuously deliver innovative, intuitive products to market,” said Sunil Bhalla, senior vice president and general manager of Voice Communications Solutions at Polycom. “Our leadership and legacy in both voice and video communications enables us to develop a truly unique device. The VVX 1500 is the business media phone to combine a superior business-grade VoIP telephone that features our renowned HD Voice with one-touch video and access to key enterprise applications. We’re delighted propel collaborative communications to the next level with this ground-breaking device.”
The Polycom VVX 1500 will be available this month through Polycom’s channel partner network at a list price of U.S. $1,099. To learn more about the Polycom VVX 1500, visit www.polycom.com/vvx1500.
Tags: broadsoft, ip phone, ip-pbx, keith nissen, mike tessler, polycom, sip, SoundPoint IP, SoundStation IP, sunil bhalla, video, video conferencing, voice, voip, VVX 1500
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SHSU Switches Back to Cisco CallManager from Asterisk
In 2006, I came across a Network World article, which espoused the fact that Sam Houston State University (SHSU) had switched from the Cisco CallManager IP-PBX to open source Asterisk. I wrote about this news since 6,000 students and faculty were moved off Cisco to the open source Asterisk IP-PBX, which was great news for the open source Asterisk community. This deployment demonstrated that Asterisk could scale and put to rest one of the main complaints against Asterisk.
Well, 3 years have passed, and according to this thread written by Jason Fuermann, who is responsible for SHSU’s IP phone system, SHSU has switched back to Cisco from Asterisk. Say what?
Continue reading SHSU Switches Back to Cisco CallManager from Asterisk…
Tags: asterisk, CCVP, cisco, cisco call manager, Cisco Certified Voice Professional, digium, ip-pbx, Jason Fuermann, tco, voip
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- Signate, an Asterisk provider, bites the dust? - Dec 08, 2006
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- Another IP-PBX company bites the dust? - Aug 08, 2006
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EZCallerID.com Hosted CNAM for Enhanced Caller-ID on any IP-PBX Launches
EZ Call, Inc. today announced the launch of EZCallerID.com, a new service that provides enhanced Caller ID, also known as CNAM, for VoIP calls. The hosted CNAM service gives you not just the phone number, but the name of the person calling.
Most SIP trunking providers do not provide the caller’s name with Caller ID on inbound calls. EZCallerID.com solves this issue by simply having you route your inbound calls to their server. They insert the caller’s name and send the call back to your IP-PBX.
How’s it work? Simply put, EZCallerID.com connects to the national databases that contain the name associated with each phone number, perform a reverse lookup, insert the CallerID info into the From SIP header and then send the call back to you.
This is similar in concept to my recent article, CNAM (CallerID with Name) on Asterisk using Reverse Phone number lookup, but in this case, no Asterisk PBX is required. It works with any SIP-based IP-PBX. It is a hosted offering, so there is a fee of course, but it’s not expensive. They only charge $0.015 per call (or $1.50 for every 100 calls). The service is available on a pay-as-you-go basis ($10 minimum initial charge), with no recurring charges or minimum monthly commitment.
Head on over to EZCallerID.com if you want to sign-up.
Hat tip to Eric Hernaez for the news tip
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Tags: Asterisk, Caller ID, CallerID, CNAM, EZCallerID.com, IP-PBX, Session Initiation Protocol, SIP, voip
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ContactQ Enhances Asterisk’s Call Center Functionality
ContactQ is a new call center application server created by Braxtel Communications designed to run on Asterisk that brings advanced call center functionality to the Asterisk platform. Their aim is to handle any sort of contact method and put it into their advanced multi-media queue. For instance, they plan on queuing video calls, text messages, web callbacks, and of course regular calls. ContactQ is a fully featured multi-media skills based routing ACD. In my meeting with Braxtel at ITEXPO, Lee McCabe, Director of sales said video call queuing brings up some interesting possibilities, such as the ability to play corporate video promotions while the call is on hold. The concept is intriguing and it takes traditional music-on-hold to the next level with video-on-hold.
Lee mentioned that ContactQ is currently being developed as both a commercial product and also as an open source GPL project. It supports industry standards such as SIP, Voice XML(VXML), AJAX (Web 2.0), XMLRPC and designed for inter-working with VoIP softswitches like Asterisk, FreeSWITCH, PingTel and Cisco.
I asked Lee if they leverage FreePBX at all for the front-end GUI or if they use their own and Lee stated they use their own front-end interface for configuration as well as monitoring of call center queues and call center statistics. Importantly, ContactQ sports the ability for call center supervisors to listen in on agents using DTMF/touch tones on their phone.
Features include a powerful IVR with drag and drop programming tool and historical reporting delivered via the web browser. It features powerful dialing capabilities critical to call centers, including Outbound Preview, Progressive and Predictive dialing modes.
Other features include:
- Fully featured ACD supporting
- Multiple queue modes
- Call pull back on no answer
- Overflow to voicemail
- Queued voicemails
- Time / Day routing rules
- Agent unavailable types
- Skills based routing (9999 skill levels)
- Telephony based agent logon
- Web browser based system configuration
- Multi Language support (US English only in Version 1)
- LDAP integration
- Multiple partition configuration
- Role based login
- Supervisor Monitor / Listen-in
- SNMP support
- Real-time Supervisor Dashboard web application providing
- Agent and queue performance statistics
- Agent and queue drill down statistics
- Real-time Agent Dashboard
- Agent performance statistics web application
Lee said their software always uses the latest version of Asterisk with nightly builds available. Installation is via a bootable .iso image which will automatically format and install ContactQ in just minutes. I tried to get some screenshots of the admin from their website, but the website seems a bit of a work in progress. But the feature-set seems pretty powerful and I hope to check it out soon.
Tags: asterisk, call center, ContactQ, ip-pbx, voip
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2008 World Series Wager: Oranges for Cheesesteaks?
I was sent a release about a wager between two hosted VoIP companies over which team will win the World Series - either the Philadelphia Phillies or the Tampa Bay Rays. The bet involves a dirty dozen cheesestakes by Alteva against Telovations’ two bushels of oranges. Ok, I’ll bite (pun intended) and blog this since it involves VoIP and our national past time. 
Check out the wager:
2008 World Series: Oranges for Cheesesteaks?
The Philadelphia Phillies and Tampa Bay Rays will soon battle it out on the field in the 2008 World Series. In each respective state, thousands of people are already placing bets on who should win this year’s title. In particular, two hosted phone system providers are actually betting oranges for cheesesteaks in a friendly, VoIP competitor match up. Alteva is apparently putting a dozen cheesestakes on the line against Telovations’ two bushels of oranges.
Representing the Phillies: Hailing from Philadelphia is Alteva, leading provider of Hosted VoIP solutions in the Northeast. Alteva provides businesses with a telephone system that requires minimal installation and configuration while eliminating monthly phone system maintenance charges and reducing overall telephony costs. With a sophisticated open standards infrastructure, Alteva enables businesses to easily integrate existing business applications and add a steady stream of new Alteva or third-party “mashup” applications.
Representing the Rays: Based in Tampa, Telovations provides growing businesses in the Southeast with a hosted, managed, IP communications solution, including telephone equipment, telephone service and a high-speed T-1 connection to the Telovations private, secure and fully managed Cisco Powered RealTime Network(SM).
“Just like major league sports, business is all about competition. These guys on the field are fierce competitors, but off the field many of them are good friends,” said William Bumbernick, President and CEO of Alteva. “Our industry is no different. While we compete on opportunities, we also have a common goal to educate businesses about how Hosted VoIP or IP PBX solutions are superior to the traditional customer premise based PBX model. Most importantly, it is a great opportunity to show our support for our Phillies, especially since our offices are located in the heart of Philadelphia,” added William Bumbernick.
“Hosted IP communication solutions have surpassed premise based systems and there’s no reason for businesses to purchase, install and maintain systems that will be paid off by the time they’re obsolete,” said Telovations President and CEO Rick Schonbrun. “Plus, Tampa is the best team in baseball and it’s hard to find a good cheesesteak down here.”
About Alteva, LLC
Alteva, North America’s largest provider of Enterprise Hosted VoIP, has become the poster child for showcasing the high quality and reliability of Hosted VoIP solutions. Alteva provides businesses with a telephone system that requires minimal installation and configuration while eliminating monthly phone system maintenance charges and reducing overall telephony costs. With a sophisticated open standards infrastructure, Alteva enables businesses to easily integrate existing business applications and add a steady stream of exciting new Alteva or third-party “mashup” applications. Rather than building their processes around their phone system, Alteva’s customers build their phone systems around their ideal processes. Alteva provides its “Communication as a Service” solution to growth-oriented companies in 48 states and 9 countries. For additional information about Alteva and its solutions, please visit www.altevatel.com or call1-877-258-3821.
About Telovations, Inc.
Based in Tampa, Fla., Telovations provides growing businesses in the Southeast with a hosted, managed, IP communications solution, including telephone equipment, telephone service and a high-speed T-1 connection to the Telovations private, secure and fully managed Cisco Powered RealTime Network(SM). Telovations’ Innovate services also include over 20 innovative applications, professional support and systems management for one low monthly price. For additional information about Telovations and the Innovate managed communications solution, please visit www.telovations.com or call 1-877-We-Innovate (1-877-934-6668).
Tags: 2008 World Series, baseball, cheesesteaks, hosted VoIP, ip pbx, oranges, Philadelphia Phillies, Tampa Bay Rays, voip
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Adtran IP 706 Review
Adtran recently launched their IP 700 series of IP phones in late April. Adtran sent TMC Labs the IP 706 model, which supports up to 6 lines, but the 700 series also includes the IP 712 which is identical feature-wise but supports up to 12 lines. Each line can be configured to register with unique SIP proxy/registrar servers. This allows a different line for every line key on the phone. A line is called a multiple call appearance (MCA) type if it will be assigned to one or more line keys on the same phone. It is called a shared call appearance (SCA) type if the line is shared across multiple phones. This is not to be confused with SLA (Shared Line Appearance) which maps PSTN lines to buttons on all the phones. Of course you need to assign two lines with the same SIP credentials to two different lines (MCA) for full call handling functionality.
Like most if not all IP phones these days, the IP 706 supports 802.3af Power over Ethernet (PoE) as well as TFTP booting of firmware and configuration from a TFTP Server. The Adtran phone will connect to your TFTP Server (option 66 on DHCP server) and look for a file called adtran_[MAC address of Adran phone].txt. So for instance, for the IP 706 phone I tested, it looked for adtran_00a0c831593c.txt on the TFTP Server when the phone was booted.
The configuration files are pretty easy to figure out and sample files are available. For instance, one of the first things you’ll want to do to configure any IP phone is to setup the dialplan. I was able to easily figure out how to setup the syntax for the Adtran dialplan, as seen here:
# DialPlanExternal is for realm GE line types and DialPlanPBX is for realm GP line types
DialPlanExternal |911|2-9]xxxxxx+T3|2-9]xx[2-9]xxxxxx|[0-1][2-9]xx[2-9]xxxxxx|011xxx+T3|xx+#
DialPlanPBX |911|9911|1-8]xxx|9[2-9]xxxxxx+T3|9[2-9]xx[2-9]xxxxxx|9[0-1][2-9]xx[2-9]xxxxxx|9011xxx+T3|*2-9]0123456789*]+T3|*1xx|#xx+#|xx+#|**xxxx
The web admin was pretty intuitive and can be used instead of a config file on a TFTP server. Here’s a screenshot of a the web interface:
Want to specify a corporate directory? No problem. Just export a comma separated file containing your corporate directory, upload it to the TFTP server and then add this line to the Adtran config file:
SystemPhonebook adtran_phonebook.csv
I exported my Outlook Contacts to a CSV file, including first name, last name, company name, title, email, street, street2, street3, city, state, ZIP, country, mobile phone, home phone, FAX, with column/field headings in the first row. The IP 706 will read the first row to automatically map the contact data into the system phonebook. Once imported, you can scroll through the system directory using the 4-way navigation button. Holding the up/down arrow doesn’t cause it to auto-repeat. Fortunately, you can press the left or right arrow to page up/down through the contacts. The 4-way navigation button also acts a shortcut buttons. When on the home screen you can press one of the four directions to access Incoming calls, Missed Calls, Placed calls, and the Personal address book. The detailed contact details is pretty cool, especially since most phones only store name and/or company and the phone number.
Defining buttons is pretty easy. Here’s some examples from my config file:
Button.1.Label Line 1
Button.1.Type line
Button.1.Line 0
Button.2.Label Line 2
Button.2.Type line
Button.2.Line 0
Button.4.Label x149 Tom
Button.4.Type speed
Button.4.Number 149
Button.5.Label DND
Button.5.Type DND
Button.6.Label vm
Button.6.Type speed
Button.6.Number 8555
Although the Adtran IP700 series was probably designed initially to work with the Adtran NetVanta 7060 and 7100, the Adtran IP700 series are SIP-based so the phones work with any SIP-based IP-PBX. I was able to register the phones on the Asterisk-based trixbox platform very easily. Once registered, I was able to make calls to Aastra and Polycom IP phones. The voice quality on both ends seemed very good. Usually the sound quality when using a handset is not an issue for any IP phone - it’s when you try and use the speakerphone that sound quality issues arise. You need good echo cancellation to make sure the remote speaker’s audio isn’t fed back into the speakerphone. Polycom is renowned for their superior sounding voice quality in speakerphones, however, I was pretty impressed with the sound quality on the Adtran IP 706 when in speakerphone mode. The speakerphone volume when set to maximum is extremely loud and without any distortion. I doubt even in the largest of conference rooms that the loudest volume setting be required, but it’s good to know it has the capability.
Overall, I like the button feel. not too hard, not too soft. Navigating the menus and options was very intuitive, though there is no key auto-repeat, which would be handy when scrolling quickly through the built-in directory book. Though, as I previously stated, you can use the left or right arrow to page up/down. The LCD was excellent - it’s very bright and uses icons to indicate various features. For instance, a bell indicates your phone will ring, while an ‘X’ through the bell indicates DND mode. Similarly, a phone icon displays next to each line with or without an ‘X’ depending on if the line was registered with the SIP registrar or not. A U-turn arrow indicates a line is being forwarded. An envelope displays at the top of the phone if you have voicemail, along with the number of new messages. The phone has a slightly slow boot-up time taking 83s to fully boot. Comparatively, an Aastra 57iCT took 53s and a Polycom IP650 took 65s. Not a big deal, since you don’t typically reboot your IP phone.
The Adtran IP phone supports busy lamp fields (BLF) using the Broadsoft method not the Sylantro method. This may be important if you are deploying Asterisk, since Asterisk only supports the Sylantro method. Personally, I have no need for BLF on our Asterisk-based IP-PBX, and no one in our office uses BLF, but certainly receptionists might find BLF useful. Other than the BLF feature, all other features worked on the trixbox system I was testing it with.
I was able to make outbound hands-free auto-answer intercom calls from the IP 706 to an Aastra phone. First I had to define the star code (*74) for initiating hands-free intercom calls. From the IP 706 I simply pressed the HFAAI (hands-free auto answer intercom) button on the LCD display under the More menu and dialed an extension which will immediately cause the remote phone to ring off-hook into hands-free speakerphone mode. You can also setup a speed dial for HFAAI so you don’t have to go into the More submenu - a two step process.
Although outbound HFAAI calls from the IP 706 work, I wasn’t able to get the Adtran phone to receive hands-free intercom calls from an Aastra phone. For instance, I made a from x149 Aastra phone to the IP 706, and although the IP 706 LCD displayed “Intercom - 149″ it rang normally and did not go off-hook into speakerphone. I have to lift the receiver or press the speakerphone button to answer the call. I contacted Adtran technical support and they were quickly able to determine the issue. The phone responds to “alert-autoanswer” or “autoanswer” in the SIP header, so it’s possible to tweak Asterisk to get it to work.
For speed dials, the Adtran IP phone supports 100 Personal and 300 System entries, no matter how many fields are in each record. You can even enter in pauses for speed dials with a “P” for a 2 second pause, useful for dialing through auto-attendants to an extension (i.e. 98005551234PP100).
In addition, you can export Outlook Contacts into a CSV file and put the CSV file on the TFTP server, which will be the global (not personal) system phonebook. You can also import a .CSV file directly to the phone via the phone’s Web interface for your own personal phonebook and speed dials. The personal contact directory can be imported from the personal web GUI. You log into http://x.x.x.x/admin for the admin GUI, but just log into http://x.x.x.x for the user GUI. It allows for the upload (append or replace), and backup of the personal directory. The format is the same as the System Directory csv file.
Users can even enable call forwarding from the phone’s web configuration. This is useful for when the IP-PBX doesn’t support call forwarding. It even supports forwarding to an outside number.
From the phone itself you can test the audio of the handset speaker and the phone speakerphone. You can set the input to the handset microphone and have the output directed to the handset speaker or the speakerphone. Further you can test the button LEDs by turning them all on and you can test the LCD on the phone. Adtran claims that the IP700 series draws less than 6.49 watts of power under normal operating conditions. I was going to test it with my Kill a Watt electric meter, but I seemed to have misplaced it.
One nicety is you can modify the splash screen simply by downloading a 216×336 pixel 16-bit bitmap file to the parameter IconPixmap. This might be useful for OEMs or even IP-PBX vendors that want to do branding.
On inbound calls, the blue Messages light flashes, which is the button used to check your voicemail. You can’t press the flashing Messages button to answer the call on speakerphone mode. I would prefer that it flash the speakerphone button instead. The reason is that when I first hooked it up and called it for the first time, I instinctively pressed the Messages button since it was flashing and I wanted to answer it via speakerphone mode. A minor complaint for sure.
Another test I performed was redirecting an inbound call to voicemail. You have a couple options. First, you can simply click ‘Ignore’ on the LCD and that will simply mute the ringing, but the caller has to wait until the ring duration setting has been met before going to voicemail. The proper way is to press the ‘Vmail’ icon on the LCD which will redirect the caller to the voicemail system. When I first attempted this, it sent the caller into the voicemail logon asking the caller for their extension and password. After perusing through the Admin Guide, it seemed like I had the voicemail settings correct. But then I realized I needed to do a call transfer direct to voicemail (*86 code) to the phone’s extension (135). So I needed the *86 code. I simply needed these two lines in the Adtran config file:
MessagesCallback 8555 # For 1-button access to check voicemail
Reg.0.Voicemail *86135 # For redirecting callers to voicemail.
The phones include an adjustable desk stand or can be wall mounted. An integrated headset jack with electronic hook-switch eliminates the need for a mechanical handset lifter. The electronic hook switch is compatible with GN Netcom and Plantronics headsets.
Features:
- Adaptive jitter buffers and packet loss concealment algorithms
- Six programmable buttons
- Large backlit display, with 6 rows by 35 characters (IP 706), 9 rows by 35 characters (IP 712)
- Message waiting indicator
- Four-way navigation
- 802.3af Power over Ethernet (PoE)
- Integrated headset jack
- Distinctive ring tones by number
- Multiple call appearances
- Three-way conferencing
- Busy Lamp Field (BLF)
- Shared Line Appearance (SLA)
- Hands-free auto-answer intercom
- Distinctive incoming call treatment/call waiting
- Visual ringing alert/message waiting indicator
- Voice activity detection and comfort noise fill
- Full-duplex speaker phone
- Three-way conferencing
- G.711u, G.711a, G.729A (Annex
|
Pricing: The Adtran IP 706 is $249 and the Adtran IP 712 is $299.
Conclusion
I like the aesthetics of the IP 706. It’s a nice clean design with a bright LCD and it has a very intuitive navigation menu on the phone. Similarly, the web interface was easy enough to navigate and figure out. The adaptive jitter buffers and packet loss concealment algorithms are a nice addition to ensure voice quality. A way of importing personal contacts into the phone itself via the web interface would be nice, but I do like that the Adtran speed dials support pauses - not all IP phones do, which makes them less useful when dialing auto-attendants with extensions. Overall, I was pretty pleased with the Adtran IP 706’s style, performance, and features. Customers have yet another choice when choosing a SIP-based IP phone. Watch out Aastra, Grandstream, Linksys, Polycom, and Snom - there’s a new IP phone in town!
Tags: adtran, asterisk, IP 700, IP 706, IP 712, IP phone, IP-PBX, SIP, trixbox, voip
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Jazinga
Jazinga launched its entry into the SMB PBX space after winning the Best of Show Award at Internet Telephony Conference & EXPO. Jazinga’s box is about the size of a D-Link router, but is more that a wireless access point and QOS router. It is a full fledged, SIP-capable IP-PBX that can use IP Phones or Plain old RJ11 phones. (You know those ugly ones on your desk now).
One big selling point is the easy configuration, which comes from a consumer focus that means you don’t need an IT gal or a PBX guy to set it up or manage it. Jazinga claims that the DIY set-up time is about 10 minutes after you plug your IP or PSTN phones in.
It’s a space-saver too. Router and wireless access point rolled into the PBX. (The router even prioritizes voice traffic). Other features include an auto-attendant, voice mail, conferencing, call forwarding, on-hold music - all for 20 or less users. The Jazinga system is available directly from the company and its channel partners for a $1,095.
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Gizmo5 SIP Trunks available in trixbox CE
Gizmo5 SIP trunks have always been available in trixbox CE, but it was a manual process. The Gizmo5 team has built a module to be part of the trixbox package manager that allows you to purchase your trunks, see your account balance, purchase more minutes, and automatically setup your inbound and outbound routes. The module is now available via the trixbox package manager and will be built into all upcoming ISO builds. 
Additionally, the calling service for trixbox CE is pre-configured to use the Gizmo5 calling network and includes a new UI for easy administration. Also included is a Tech Check system that confirms basic setup of a trixbox CE system and notifies users when new Gizmo modules are available. Finally, the new offering also includes pay-as-you-go and Gizmo5 has also joined Fonality’s FACE program (Fonality Authorized Certified Ecosystem) as a Gold partner to ensure its products are optimized and compatible with the trixbox CE platform.
Tags: gizmo5, IP-PBX, SIP trunking, SIP trunks, trixbox CE, voip
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Skype for Asterisk Launches
Skype and Digium have hooked up to bring Skype to Asterisk called Skype For Asterisk. Skype For Asterisk launched minutes ago enables Asterisk users to get access to Skype features coupled with the capabilities of Asterisk. For example, the beta version of Skype For Asterisk will allow customers to make, receive and transfer Skype calls from within Asterisk systems using their existing hardware; enable inbound calling solutions like free click-to-call from company websites or virtual offices; and manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.
Hey, I guess I was right in my (Astricon) prognostications earlier today about it having to do with Skype.
The Skype For Asterisk Beta program begins today. Asterisk users, system administrators and developers are invited to apply to participate at http://www.astricon.net/skype
I’m trying to figure out how you transfer a call to a Skype username (i.e. tkeating) using a traditional (Asterisk) IP phone with no keyboard - just a numeric keypad. Of course, maybe the transfer feature is only to other Asterisk extensions or outside phone numbers and you can’t initiate calls to Skype usernames. Of course, I’m guessing that you can map inbound Skype calls to usernames to specific Asterisk IP phone extensions.
Update (1pm): Some other thoughts…
Will Skype for Asterisk work exclusively on Digium’s flavors of Asterisk (AsteriskNOW, Switchvox, etc.) or will it also work on trixbox CE, PBX in a Flash, etc? Is the Skype channel driver licensed by Digium or is it a free driver, which can then be used on other Asterisk distros. Since Asterisk offers a free version of their open source solution, I’m going to have to assume the Skype channel driver will also be free.
Update (1:20pm): Some info from TMCnet reporters at Astricon
- Majority of questions were about access to code. Mark says their will be some limited access.
- Caller ID - they say it can work.
- Number portability - Oberg says that is a ‘local issue’ and not built in to this beta.
- No pricing announced.
- Commercial license model, Not open source.
More on this news after the jump…
Continue reading Skype for Asterisk Launches …
Tags: Asterisk, Digium, ip-pbx, Skype, voip
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Digium Major Announcement - what can it be?
Today, Digium, creator and primary developer of Asterisk, the leading open source telephony platform will be making a major announcement at Astricon later today. Digium hinted to me that a major announcement would be made at Astricon when I visited their Huntsville, Alabama headquarters in August.
I tried to find out what the news will be, but alas Digium couldn’t tell me. So I thought it would be fun to prognosticate what this deal could be.
1) Digium’s Switchvox will be distributed by Dell, which currently carries another Asterisk competitor, Fonality.
2) Digium will be acquired by Adtran, an avid supporter of Digium in the past.
3) HP seeing that competitor Dell is offering IP-PBXs (i.e. Fonality and Nortel) will partner with Digium to offer Digium’s line of IP-PBXs and telephony hardware
4) Now that Adtran, a financial supporter of Digium, offers their own line of IP phones, including the IP706 and IP712, perhaps Digium will offer a bundled IP-PBX package that includes Switchvox and a few Adtran IP phones.
5) Digium will announce this whole open source thing is nonsense, there’s no money in it, and they’re announcing that they are making Asterisk closed source effective immediately. Ok, maybe in some alternate universe!
6) CDW, a major distributor, will be carrying and distributing Digium’s products.
Ok, so what’s your guess? Costco? And remember the clock is ticking. The announcement could be made at any moment.
Tags: Adtran, alabama, Astricon, Digium, fonality, Huntsville, ip-pbx, IP706, IP712, voip
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Jazinga IP-PBX Launches
Today Jazinga, launched it’s IP telephone system solution with a simplified installation and configuration process takes designed to take about 15 minutes to set up.
I’ve had a Jazinga unit in TMC Labs for several weeks now, but just haven’t had the chance to hook it up and test it. But is on my to-do list. It’s a pretty inexpensive IP-PBX available directly from Jazinga Inc. and its channel partners for a MSRP of $1,095 USD.
In the meantime, thought I’d share the news after the jump…
Continue reading Jazinga IP-PBX Launches…
Tags: IP-PBX, Jazinga, phone system, voip, www.jazinga.com
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HUD 3.0 for trixbox CE and any other Asterisk IP-PBX!
I spoke with Kerry Garrison last week at ITEXPO and he gave me a news scoop that Fonality would soon offer the HUD 3.0 Unified Communications client for the open source trixbox CE Asterisk-based platform. trixbox CE is one of the most popular Asterisk distributions. I recently commented in my trixbox Pro review (paid version),”The feature-rich HUD Pro client is certainly a competitive advantage Fonality has over many other Asterisk-based solutions.” As such, offering HUD 3.0 for trixbox CE is a major move by Fonality.
In fact, Kerry told me that once HUD has been ported to trixbox CE, the new HUD will work not only on trixbox CE but should also work other Asterisk flavors, i.e. Switchvox, PBX in a Flash, Voiceroute, etc. Fonality plans to license the HUD 3.0 client, however, pricing has not been set. While Fonality loses the “exclusivity” of their feature-rich HUD client, they will certainly gain revenue from other Asterisk distros that license HUD.
The new HUD 3.0 will provide trixbox CE users with presence management and detection in a single interface for all types of office communications, including SMS, instant message, landline calling, mobile calling, chat, voicemail, email, conferencing, recording, and barging.
HUD 3.0 permissions in the Group Manager:
“Open source rarely lacks in features, but often lacks in ease-of-use and polish. Our intention with this announcement is to bring the polish of the HUD 3.0 unified communications platform, which is in use by more than 100,000 paid users, to the trixbox community. This should allow them, now more than ever, to compete with the high-prices of the big-iron oligopoly,” said Chris Lyman, CEO of Fonality.
HUD 3.0 for trixbox CE delivers a unified communications dashboard that shows availability for onsite and remote employees, eliminating wasted time created by busy signals, voice mails and phone tag. Users can easily drag and drop calls onto a colleague’s desk or mobile phone, quickly convert IM chats to voice calls, and even personalize calls with photo caller ID.
One caveat with the new HUD 3.0 client that some may object to is that it uses a hybrid-hosted model for the sign-in process. It uses Fonality’s hybrid-hosted data centers to authenticate the sign-on process to determine if you have a valid license. I asked Kerry what happens if the Internet is down and he said you can’t sign-in to HUD. However if Internet goes down after you sign-in, HUD still works.
The new capabilities of multi-lingual HUD 3.0 include:
• Integration with Google Talk from any desktop, BlackBerry, or iPhone. Users will now be able to chat and exchange presence with any of Google’s more than five million Google Talk users.
• Deep and instantaneous Web 2.0 integration with all applications such as CRM, Google, ticketing, billing, and financial systems. Web pages can either actively launch upon call activity or send silent (background http) notifications to web-based applications.
• Mobile presence displays mobile users to the rest of the company when they connect to the trixbox CE phone system. Mobile presence also supports Busy-Ring Back™ so as not to disturb a fellow employee who is on a mobile device.
• Personalized photo caller-ID displays photos of all inbound and outbound callers, and displays caller photos when listening to voicemail or joining a conference bridge.
• SMS one-way text messaging is now included, in addition to the already supported one-click call, mobile call, voicemail, email, and chat.
• Visual conferencing for the easy creation and recording of conferences. Drag users in, kick, and mute all with a single click. Each conference bridge participant has a face photo displayed, which makes it easier to talk to them, see them, and communicate with them using HUD 3.0 built-in instant messaging.
• Visual voice mail allows users to receive voicemail directly on their PC or Mac desktop. A MWI (message waiting indicator) provides notification when a new voicemail has arrived and, with a few simple clicks of the mouse, messages can be played, fast-forwarded, rewound, or saved to disk. Also, voicemail contacts can be easily added to Outlook, and users can click to return calls, or click to chat.
• HUD Queues take trixbox CE and HUD 3.0 into larger call centers without the typical associated costs. HUD 3.0 now supports big call center features such as real-time queue displays, up-to-the-second queue stats including ASA, dragging-and-dropping any holding queue call to an agent, call center alarms, abandon alerts, broadcast agent messaging, virtual wall boards, and much more.
• Seven languages at launch including English, Spanish, French, Japanese, Russian, and simplified and traditional Chinese.
An open beta program will begin in a couple of weeks. HUD 3.0 for trixbox CE will be available in October.
Tags: fonality, HUD 3.0, IP-PBX, itexpo, Kerry Garrison, trixbox CE, VoIP
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Microsoft Response Point Adds T1 Support and SIP Trunking Service Providers
Today at ITEXPO, Microsoft will announce a few more SIP trunking service providers that come pre-configured in the Response Point unit making it virtually plug-and-play when selecting a SIP trunking service provider. At ITEXPO Microsoft will announce they are adding NGT, Cbeyond, and Junction Networks. In addition they are also adding support for broadband VoIP service provider Packet8, as well as Bandwidth.com, a provider of Internet and managed services, including SIP trunking.
At ITEXPO Microsoft will also announce that they will be supporting the ClearOne IP conference phone, what Microsoft told TMCNet they consider one of the best quality IP conference phones out there.
Additionally, Sangoma will announce a board/gateway in a PC that supports Microsoft Response Point along with Response Points network auto-discovery method. Perhaps the biggest news is that Microsoft is announcing T1 support through a pertnership with Quintum and a special Response Point compatible T1 gateway. It also supports Response Points auto-discovery, so it’s very easy to add a T1 trunk line to the system. This marks the first time Response Point supports digital T1 trunk lines. With analog trunk lines already supported, the addition of T1 support should open up more opportunities to sell this product in the SMB space.
Tags: 8×8, Bandwidth.com, ClearOne, IP-PBX, Microsoft, Packet8, Quintum, Response Point, Sangoma, SIP, T1, voip
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Microsoft Response Point News
I just met with Microsoft to talk about what’s new in their Response Point IP-PBX. They demo’ed the Aastra AastraLink RP phone system which works with Aastra IP phones with the Response Point “magic” blue button embedded onto the phones. The magic button allows you to use speech-recognition to perform call transfers, directory lookups, and more. I met with Richard Sprague, Senior Director, Microsoft Response Point and Xuedong Huang, General Manager, Microsoft Response Point to find out what’s new and to see a demo of RP SP1.
Microsoft’s Richard Sprague & Xuedong Huang (holding a cordless Aastra Response Point phone)
Microsoft gave TMC’s Michael Dinan and myself a demo of the Response Point system, which now features a fully-functional click-to-dial software application, and presence integration. They mentioned they recently added the ability to say “free411″ and have it dial the free directory assistance service. Microsoft will be announcing some interesting Response Point news tomorrow at ITEXPO, so stay tuned. I’m also told Xuedong Huang will give an interesting keynote with some insights in the future direction Microsoft will be taking in (unified) communications. He said he will showing a video and it’s a one-time only video and won’t be available anywhere for viewing. So I’m hoping to capture the keynote session using my liveblogging mobile phone!
Microsoft’s demo setup at ITEXPO
During our meeting, Richard Sprague mentioned he read my blog about live video blogging from ITEXPO using my Windows Mobile phone. So I gave him a quick demo and recorded the video. In the video you will see the table Microsoft setup at ITEXPO in a meeting room to demo the platform to the media and analysts attending the show. The video shows some Aastra phones and the AastraLink RP system, which I should mention features PoE (Power over Ethernet) ports for easy plug-and-play installs. Check out the video below, which even features a special hidden dial string that gets you into some uber secret auto-attendant featuring Bill Gates himself as the recorded voice prompts. Alas, my camera didn’t pick up the dial string. 
Too bad Jerry Seinfeld doesn’t make a special voice-over appearance.
Tags: Aastra, Aastralink RP, Bill Gates, IP-PBX, itexpo, jerry seinfeld, Microsoft, Response Point, Richard Sprague, voip, Xuedong Huang
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PIKA WARP Appliance for Asterisk Review
There is no doubt that open source Asterisk has taken IP telephony by storm resulting in many vendors offering solutions based on Asterisk. Because the “free” Asterisk software is open source, it has helped drive down the cost of installing an IP-PBX. Only one major cost factor remains - the hardware. If you install Asterisk on a traditional PC, you have several hardware components - the motherboard, the CPU, the memory, the hard drive, the power supply, CD/DVD drive, etc. Some of these hardware components aren’t necessarily required to operate a fully-functional IP-PBX or could be replaced with inexpensive alternatives. For instance, instead of a hard drive, why not use Flash memory? It’s cheaper, more reliable, is more easily upgraded, and can be easily swapped after a failure. It also uses less electricity than a hard drive resulting in a “greener” Asterisk solution. When you consider how tight profit margins are when offering an IP-PBX to the SMB market, shaving off a few dollars in hardware costs can be a huge pricing competitive edge. For these reasons, PIKA Technologies offers an embedded Asterisk appliance called the WARP Appliance targeting the SMB market with a cost-effective telephony development platform. PIKA pointed out that WARP is not exclusively tied to Asterisk explaining, “Our customers have developed call logging system, IVRs, predictive dialers and 50% of them use Asterisk, 50% don’t.”
The PIKA WARP Appliance isn’t a turn-key Asterisk IP-PBX, but instead is a development platform that enables resellers and VARs configure Asterisk 1.4.x to their liking, and then offer a customized version of Asterisk through their distribution channel. In fact, PIKA sells what they call the “PIKA WARP Appliance for Asterisk Developers Kit”, which includes a PIKA WARP Appliance for Asterisk, one 4 port FXO (trunk) module, one 4 port FXS (station) module, one SD Memory Card (1Gb), one Serial Cable (programming), a network cable, and Getting Started Guide. The PIKA WARP Appliance for Asterisk Developers Kit is discounted to $550 (limit 1 per company) to encourage developers. The normal non-discounted list price is $725. The main concept behind the WARP Appliance is to offer resellers and VARs an inexpensive Asterisk hardware platform that they can OEM and offer under their own brand name. I should also mention that the WARP Appliance now also works with FreeSWITCH, so developers can also choose to embed FreeSWITCH instead of Asterisk. In fact, any telephony application such as IVRs, call logging, predictive dialer etc. built using PIKA’s telephony APIs can be integrated onto the appliance, whether it is a proprietary application or based on an open source platform.
Back Panel and the cover taken off to show the inner guts of the WARP Appliance
Importantly, the Kit comes with 4 analog phone ports and 4 analog trunk lines, or essentially a 4×4 “development” phone system, which is perfect for many SOHO and SMBs. In fact, considering many IP Phones are >$200 and analog phones can be had for $20, one should not underestimate how many small businesses would like to dip their feet into VoIP but aren’t ready to commit to expensive IP phones. The PIKA WARP Appliance allows them to get a fully-featured Asterisk IP-PBX while offering up to 4 analog phone stations and 4 analog trunk lines. In fact, resellers can even offer 8 analog stations by swapping out one FXO card and instead including two FXS cards - all modules can be mixed and matched in any combination, including BRI in future For inbound and outbound calling the reseller can offer 100% SIP for the trunking side, which has the added benefit of lower per minute charges compared to traditional PSTN dialing. The configuration of the appliance is modular and can include up to 9 ports of a combination of FXO/FXS/BRI plus VoIP stations and trunks.
Top View looking inside the WARP Appliance
While there are many DIYers (Do It Yourself) out there that have built their own home-brewed embedded Linux Asterisk appliances, PIKA has spent considerable resources on choosing reliable embedded hardware and performing quality assurance (QA) testing. When building your own appliance, DIYers have to be concerned with EOL (end of life) on components such as motherboard, memory etc and have to deal with software installation issues and integration with the hardware (ie. drivers)
WARP comes pre-loaded with the 2.6x Linux Kernel (stripped down PIKA version) and includes SSH (Dropbear), Asterisk and Asterisk GUI (1.4.x), database (SQlite3), Httpd (webserver), PHP5, NTP, DHCP, TFTP server & client, as well as VLAN and DNS. As previously mentioned, you have the ability to add any software package that your application requires.
I got to test drive the PIKA WARP Appliance in the lab and was pretty impressed how easy it was to load firmware, add packages, and build a fully-functional copy of Asterisk. The unit includes a RAM disk, full root access, 256MB of RAM, and 256MB of Flash for loading the Linux kernel. Additionally, you can add an SD memory card for additional memory storage, useful for storing voicemail. The processor is powered by a AMCC Power PC 440EP, which operated at 533MHz. The outside of the unit features a 2 x 20 backlit LCD display, with API-controlled front-panel scroll button. You can even control the LED with simple shell commands.
Make the LED red:
echo 1 > /sys/class/leds/warp-red/brightness
echo 0 > /sys/class/leds/warp-green/brightness
Make the LED green:
~ #> echo 0 > /sys/class/leds/warp-red/brightness
~ #> echo 1 > /sys/class/leds/warp-green/brightness
Make the LED orange:
~ #> echo 1 > /sys/class/leds/warp-red/brightness
~ #> echo 1 > /sys/class/leds/warp-green/brightness
To turn the LED off you just echo 0 to both.
I learned this tip on David Clarke’s blog/community. David is the Business Development Manager at PIKA Technologies and he started the blog of a place where developers can find 3rd party add-ons such as various Asterisk GUIs and WARP tips. It is relatively new but content is growing daily. You can check it out here: www.pikawarp.org
The back of the unit includes Music-on-hold audio in, paging system audio out, an SD slot, a single Ethernet port and one USB port. I’m told PIKA is working on a dual-Ethernet port WARP Appliance in the near future. This would allow the appliance to add NAT firewall capabilities. Importantly, the unit includes a power failure switchover emergency PSTN port. In the event of a power failure, you can still make an outbound call, i.e. 911.
The appliance can run software from flash memory or via a network file system (NFS) located on your development computer. According to PIKA, “Initially, you will use NFS to execute the software( kernel and ramdisk). NFS will be the primary method for running software on the appliance during development. It is faster to boot using NFS, updates to files can be done without taking the time to write new images into flash and, depending on the file type being modified, without rebooting.” There are 3 methods available to load software onto the Appliance.
a) svn checkout of PADS
b) tarball of PADS
c) pre-built images file for the appliance
Building the software is very straightforward using PADS (Pika Application Development Suite) to compile the various packages and then transferring it to the WARP Appliance. You can also compile directly on the WARP Appliance itself using gcc. (See: http://pikawarp.org/?p=53) If using PADS, your development computer requires the following Linux packages in order to use PADS:
• A serial client (e.g. minicom on Linux or HyperTerminal on Windows)
• TFTP (Trival File Transfer Protocol) Server
• NFS (Network File System) Server
• WGET
• Subversion (SVN)
• AUTOCONF
• AUTOMAKE
• LIBTOOL
• NCURSES
• SSH client
• GCC 4.x or greater
On your development Linux PC you go to the location of your unpacked source or SVN checkout of PADS and simply type:
#make menuconfig
This command displays the package selection menu. This will include default menu selections, but you can easily add/remove packages from the Appliance. Next you select ‘Exit, choose ‘Yes’ when asked if you want to save your configuration and then enter the command:
#make
This will build the software with the packages you chose. When the build is complete, you will have an NFS mount point at <Your PADS path>/build_warp/root.
The software image for the kernel (cuImage.warp) is created during the previous step. To create software images for the ramdisk and the persistent file system, you simply enter the command:
#make image
The following compressed images will be located in <Your PADS path>/images:
• cuImage.warp (kernel)
• uRamdisk (ramdisk)
• image.jffs2 (persistent filesystem)
The next step is loading the images into the appliance. There’s a few ways of doing it, including entering a special bootloader mode called U-Boot and using a serial cable and software like HyperTerminal. But a much easier method is doing it across the network using TFTP or SCP to transfer and load the software onto the appliance. To actually write software to flash you use warploader. warploader is a PIKA’s tool that allows you to write software into flash memory while the appliance is running. The tool provides a single step to replace software eliminating the need to enter the special U-Boot prompt and a serial connection to load new software.
After transferring the image to the appliance, you just type this command to load the software into Flash:
#warploader -p <partition name> filename
For instance:
#warploader -p kernel /root/cuImage.warp (kernel)
#warploader -p root /root/uRamdisk (RAM Disk)
Flash memory has a limited number of write-erase cycles. A utility is provided to track the writes to the NAND flash and can be used to monitor excessive or rapidly increasing amounts of data written to flash which may indicate a problem with an application.
To view the amount of data written, enter the following at the Linux prompt on the appliance:
cat /proc/driver/ndfc
or on my version:
cat /proc/driver/pikasd
Unfortunately, the number of writes is reset to zero after a reboot, but still a useful utility.
Two additional partitions called persisent1 and persistent2 are provided in flash memory for user-defined purposes. This space can be used for additional persistent data or for files that will not fit into the ramdisk image. I should point out that when the system is booted, the ramdisk is read from flash or NFS into memory and therefore, the size of the ramdisk is an important consideration for system performance. The maximum size of the ramdisk, using the current settings is 64 Megabytes, out of the total 256M of RAM. PIKA claims that this size is sufficient for a load that includes all of the packages currently made available by PIKA in PADS, with the exception of GDB (GNU Project Debugger).
I hooked up some analog trunk lines using a Teltone analog simulator as well as a few analog phones. I also registered a Polycom IP650 and an Aastra 57i IP phone. I was able to make extension-to-extension calls, outbound calls through the Teltone simulator, and inbound calls to the auto-attendant. In my testing of the PIKA WARP Appliance, it handles fax just fine. It doesn’t currently support T.38 real-time fax over IP because T.38 is very processor intensive, but PIKA told me T.38 support is in the works. PIKA includes some special built-in extensions to speed development and testing. For instance, I was able to dial 500 and make an IAX VoIP call to Digium’s corporate auto-attendant (misery.digium.com) with no firewall configuration. I’m always impressed how IAX is able to traverse NAT firewalls without messing with the firewall.
Here’s a list of the built-in testing extensions:
Extension Purpose/Destination
2222 - Connects to the audio in port to listen to the audio sent from an external device such as an MP3 player.
2233 - Connects the handset microphone to the audio out port on the appliance, used for paging.
2244 - Begins playing pre-recorded prompts to the audio out port on the appliance. After dialing, if you hang up, the prompts will continue to play.
2255 - Stops the pre-recorded prompts started by dialing extension 2244.
4001 to 4005 - These extensions call FXS lines 1 to 5, respectively. If the FXS module is not present, the call will be routed to voice mail.
4006 to 4010 - These extensions call the sample SIP Agents defined in sip.conf. If the SIP agent associated with the extension is not registered, the call will be routed to voice mail.
4060 - PIKA FAX receive test. Connect a FAX machine to one of the FXS ports, dial this extension and the
appliance will receive the FAX. A tiff file will be stored in /tmp/warpfax.
4061 - PIKA FAX transmit test. Connect a FAX machine to one of the FXS ports, dial this extension and the appliance will send a test FAX (the PIKA logo) to your FAX machine.
9<number> - Calls out on an available FXO extension. If no FXO extensions are available, congestion will be received.
500 - IAX test call to Digium’s auto-attendant.
Features/Specs:
- Operating system — Denx ELDK, with a 2.6.19.2 Linux kernel
- AMCC Power PC 440EP Embedded 533 MHz Processor 1200 mips
- Supports floating point and MMU (memory management unit)
- Internal flash 256 MB NAND(OS + apps) plus 4 MB NOR memory (uboot)
- 256MB RAM
- External removable 1 GB SD flash memory (no hard drive improves reliability) for additional voice mail prompts / storage
- back -up of configuration files and custom settings (facilitates unit replacement)
- Reset function remotely controlled
- Maximum IP ports 75
- Maximum FXS ports 9
- Built-in FXS ports 1
- Maximum FXO ports 8
- Maximum BRI ports 4 / channels 8 (future)
- Simultaneous calls 32
- Dynamic thermal management (fan)
- Power failure transfer
- Music on Hold input
- Paging system output
- Echo cancellation
Connectivity
- WAN/LAN ports 1
- RS-232 interface
- USB ports 1
Display
- Size 2×20 character
- Backlight
- ScrollButton
- API
Physical
- Brandable
- Desk mountable
- Wall mountable
- 9.25″ W x 6.65″ D x 2.18″H
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Conclusion
Comparisons will no doubt be made with Digium’s Asterisk Appliance 50 (AA50), so I thought it might be useful to offer my own comparative analysis. I haven’t tested the Digium Asterisk Appliance, so I can’t compare the development environments between the two. Though, I am under the impression that Digium not longer supports an open development environment. Looking strictly at feature-specs, I see that the PIKA WARP Appliance does have some key advantages, including built-in Music-on-Hold, paging, LCD display, 5 more FXS ports, and higher scalability (75 vs. 50). The Digium Asterisk Appliance does however have 4 LAN ports to the WARP’s single WAN/LAN port and the Digium Asterisk Appliance has an additional WAN port which currently the PIKA WARP Appliance does not have. The PIKA WARP Appliance offers 256MB of RAM to Digium’s 64MB of RAM and WARP offers 256MB of Flash memory to Digium’s 8MB of Flash.
Feature-specs aside, perhaps the WARP’s greatest advantage is that is flexible and customizable while the AA50 supports Asterisk only. I asked PIKA why the AA50 isn’t conducive to 3rd party applications and development and PIKA told me, “With such a small amount of memory and a more complex development environment, allowing 3rd party apps is not realistic to the typical Asterisk developer and Digium likely found it too difficult to support. PIKA has made the development process easy with PADS so it is a more viable option for Asterisk developers.”
The PIKA WARP Appliance for Asterisk is a compelling platform for developers, resellers, and VARs looking for a low cost, reliable, feature-rich Asterisk appliance to offer to the SMB market. The complete customizability and its ability to support analog trunks, analog phones, as well as IP phones and IP trunks makes it a great solution for small businesses that don’t have voice T1/PRI lines. Further, unlike Asterisk on a traditional PC, the WARP Appliance comes pre-installed with Music on Hold (MOH) and Paging built-in, as well as power failure transfer (PFT). Another key advantage is that it is modular allowing you’re the choice of up to 9 ports of a combination of FXO/FXS/BRI ports. Further, the WARP Appliance can handle up to 75 IP phones and 32 simultaneous calls, which is quite impressive for this very small and surprisingly light device. I should point out that many new small businesses are started each day and these “green fields” are looking for a cost-effective and feature-rich phone system. The PIKA Warp Appliance fits the small business market segment quite nicely both from a price and feature perspective.
Further, medium-sized businesses that have outgrown their current key system or PBX could be enticed to switch to the PIKA WARP Appliance even if their current phone system lease isn’t up yet. The reason is super low-cost of the WARP Appliance. Of course, resellers, VARs, and developers will no doubt package together their own applications and offer a profit premium over the $725 list price. Still, I’d expect the PIKA Appliance to allow developers to offer a full-fledged Asterisk IP-PBX with strong analog support for around $1000, which is a very competitive price. All-in-all, I really liked the PIKA WARP Appliance and I think developers will too.
Tags: AA50, Asterisk, Digium Asterisk Appliance 50, ip-pbx, PIKA Technologies, PIKA WARP Appliance, VoIP, WARP Appliance
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Aug 26, 2008 - Digium AsteriskNOW News and more… -
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May 16, 2008
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Fonality’s trixbox Pro Unified Agent Edition integrates with Salesforce.com
Fonality’s trixbox Unified Agent Edition (UAE) can automatically match all inbound and outbound calls with the corresponding record in salesforce.com’s AppExchange, and call data is captured and logged eliminating manual entry. This is a big step for Fonality in taking their Asterisk-based IP-PBX from simply an enterprise phone system to a “true” call center platform that can compete with Avaya, Nortel, Interactive Intelligence, and other major call center platforms.
Fonality, today announced the collaboration of the trixbox Pro Unified Agent Edition and salesforce.com’s AppExchange.
Fonality’s trixbox Pro, Unified Agent Edition (UAE) is immediately available for test drive and deployment from Fonality Australia. It’s unclear whether trixbox UAE is available in the U.S. or elsewhere as well. I’m looking into it. Relatedly, go check out my in-depth trixbox Pro 2.0 Call Center Edition (CCE) review I wrote yesterday.
In the meantime, Fonality UAE features:
- Performance Management:
- Automatic Call History provides a detailed history of who employees are calling and the duration of each call.
- Integrated Call Recording allows any call to be recorded from the PBX system and attached to the corresponding record in Salesforce.com.
- Lead Management Reporting provides reports on how many calls were required for conversion, lead activity, etc.
- Agent Activity Reporting ranks agents based on activity and productivity.
- Outbound Call Reporting shows which reports are busiest on the phone, how many calls have been made to leads and customers.
Agent Productivity:
- Click to Call allows any number is Salesforce.com to be automatically dialed with the click of a mouse.
- Improved Screen Pops address the problem of multiple matching records. A screen is automatically launched for immediate note taking, and then can be attached to the proper record after the call.
- Deal Size Alerts pop-up on a rep’s desktop and display the name, company and size of the opportunity before the phone starts ringing.
- Account Ownership Routing automatically directs incoming calls go to the account owners
Tags: Asterisk, Avaya, call center, Fonality, Interactive Intelligence, IP-PBX, Nortel, salesforce.com, trixbox Pro UAE, trixbox Pro Unified Agent Edition, VoIP
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May 22, 2007
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trixbox Pro 2.0 review
Fonality is one of the premiere providers of Asterisk-based IP-PBX solutions. Fonality offers three products: PBXtra, trixbox CE (community edition), and trixbox Pro (commercial/reseller edition). trixbox Pro. which is their commercial edition runs on Fonality’s “hardened” PBXtra technology, which Fonality claims has 5 thousand installations and over 325 million calls to date. trixbox Pro is a hybrid-hosted solution, which means you get 24/7 monitoring, phone mobility with no NAT traversal issues, and automatic software updates.
trixbox appliance
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trixbox Pro comes in three editions. The trixbox Pro family starts with Standard Edition (SE), which is free.The remaining two editions, Enterprise Edition (EE) and Call Center Edition (CCE), are available at a low monthly cost or for a lifetime fee. TMC Labs checked out Fonality’s flagship product, trixbox Pro Call Center Edition (CCE) which has all the features of trixbox Pro SE and trixbox Pro EE, plus additional call center functionality. trixbox Pro Call Center Edition scored very well in my ratings. It’s lowest rating was still a very good 4 stars for ‘Performance’. The reason for this not achieving 4.5 or a 5 star rating was that their hosted web interface can be occasionally slow.
trixbox Pro CCE is based on Asterisk and includes an easy web-based user interface, web-based voice mail, exportable reporting, click-to-call, mouse-driven operator panel, Outlook integration, real-time resource graphs, system alerts, auto-card configuring, seamless VoIP trunking, and more. trixbox Pro CCE is designed for companies with 2 - 200 agents and includes ACD and IVR capabilities with unlimited queues, skills-based routing, real-time queue statistics, graphical reports, and web-based recording access.
One of the most powerful features included in trixbox Pro CCE is HUD Pro, a communications software application which features enhanced presence, on-the-fly recording, call barge, call monitor, CRM integration, and one-touch agent login. Each extension has up to 6 icons at the bottom. Depending on your permissions, you can click on one of these icons to call their voicemail, record, barge, email, call their alternate number (mobile phone), or chat.
Here’s a screenshot of HUD during an internal extension call.
You’ll notice there are different colors to indicate presence and color coding of calls.
Green - Inbound/outbound call
Orange - Queue call
Purple - Intraoffice extension call
Grey - Unregistered
HUD also features drag-and-drop call control, which not only lets you drag calls to someone’s extension, but if that person is out of the office, you can drag the call to the mobile phone icon which will transfer the call to their mobile phone. One neat feature is that HUD can launch a Web browser to a custom URL when your extension rings. This can be used to look up inbound callers in your Web-based CRM software or even direct the search query to Google, AnyWho reverse number lookup, etc. HUD Pro also features secure chat for intraoffice instant messaging.
For users that use Outlook Contacts, there is a TAPI plugin that lets you simply right-click on a Contact and initiate a call. They have also extended TAPI so that you can call directly from your Inbox or any other Outlook email folder. What’s even cooler is that trixbox Pro takes care of dialing the “9″ and it automatically takes your phone off-hook (speakerphone mode) so you don’t have to even touch the phone. The off-hook speakerphone mode actually works throughout HUD. So you can also double-click an internal extension from within HUD and your phone will go off-hook automatically. Similarly, you can highlight a phone number anywhere on your computer, and then “drag” that number into HUD and HUD will dial the number. I should mention that currently trixbox Pro supports this off-hook feature on Polycom and Aastra phones.
trixbox Pro has some other interesting tricks up its sleeve. For instance, it has a Firefox plugin called FONcall which automatically highlights a phone number on a web-page. You simply click it and it initiates the call - once again automatically taking your phone off-hook.
Similar to an Outlook toast popup on an incoming email, HUD also displays a toast alert in the lower right of your screen on an inbound or outbound call. This allows you to direct callers to voicemail, record, or other functions without having to open the main HUD interface. The feature-rich HUD Pro client is certainly a competitive advantage Fonality has over many other Asterisk-based solutions. Though, Fonality does offer a free version HUD Lite which has a slimmed down feature-set and which works on most Asterisk flavors. HUD Lite for instance, doesn’t have on-the-fly recording, recording of others, log in & out of queues, call barging, call monitoring, and some other features. Thus, the powerful functionality in HUD Pro is a compelling reason for prospective buyers evaluating various flavors of Asterisk.
trixbox Pro supports your traditional telephony features such as auto-attendant, IVR, and voicemail. Similarly, from your desktop phone you get your traditional features such as call parking, call transfer (after a flash hook), and call conferencing. All of these work as expected. trixbox Pro also supports Ring-All (Blast Group) and similarly the ability to intercom page an extension or a group.
For trunk support, trixbox Pro supports analog, T1/E1, and now BRI. The user-friendly web-based administrator supports plug-and-play detection of your telephony hardware, which is typically Sangoma hardware in trixbox Pro. I liked the ease at which I was able to detect and configure the T1/E1 card and the two analog cards in the machine.
Faxing is also supported on trixbox Pro. While faxing on Asterisk-based platforms often gets a bad rap, (due to timing/clock syncing issues) Sangoma has recently built some very good analog hardware to solve this problem. Actually, the developed a simple bridging cable that connects from the T1/E1 card to the analog hardware to keep the timing in sync. Presto, bango! - reliable faxing on an Asterisk-based platform!
trixbox Pro has extensive BLF support, although only for Aastra phones. You can easily drag-and-drop users into your BLF area on any Aastra model with BLF support. Fonality also added automatic detection and support for the Aastra 536M and 560M sidecars to extend the number of BLF keys available on your Aastra phone.
Another key feature is the built-in conference bridges. The 5 built-in conference bridges each support an unlimited number of internal participants and as many external participants as you have inbound phone lines.
Mobility features are very strong in trixbox Pro. Each user can logon to their own personal web control panel and specify rules for how/when/where they are contacted as part of their FindMe feature (see screenshot below). FindMe supports presence detection (via HUD) to know when you have walked away from your desk and thereby ring your cell phone.Further, it features a “white list” to only allow specific people to access “findme”, as well as a VIP list (spouse, important contacts) that can reach you regardless of the schedule or your HUD presence status. Very useful feature to help stay in touch with your important contacts while respecting the times you don’t wish to be contacted.
Part of FindMe, the Boomerang feature allows you to send a call that has been forwarded to your cell phone right back to any extension on your PBX. Simply press some touch-tones on your cell and the call can be redirected to your assistant or back to your own desk. You can also record calls on your mobile using Boomerang - a powerful feature. Call screening is included and one of my favorite features. The caller is prompted for their name and once again you have the choice to accept or reject the call.
Telecommuters/Home workers Support
Telecommuting support in trixbox Pro is very good. trixbox Pro’s hybrid-hosted approach means no more pesky VoIP over NAT firewall issues. I was able to take an Aastra phone I auto-provisioned in the office home with me, connect to my home broadband connection, and simply change the DNS setting on the phone to the external DNS entry of the trixbox server and voilà, I was able to make & receive calls to my extension. No need to poke any holes in the corporate firewall or my home firewall. Nice and simple.
Features:
- Outlook Integration
- Voicemail
- Voicemail-to-Email
- Hot Desk
- Music-on-Hold
- Scheduler
- Night Mode New!
- Custom CTI (AGI)
- Analog & IP Phones
- Call Forwarding
- Name Directory
- DIDs
- Unlimited VoIP Accounts
- PSTN Fallback
- Branch Office Support
- Web-based Control Panel
- Powerful Reporting
- Hands Free Auto Phone Provisioning
- FAX Support
- BLF Support
- BRI Detection
- E1 Support
- Live Backup Server
- Multiple Deployment Management
- Conference Bridges
- Routing by DIDs
- Paging / Zone Paging
- Intercom / Zone Intercom
- Voicemail Groups
- Advanced Call Forwarding
- Call Return
- Call Out
- Custom Caller IDs
- SMS/Pager Voicemail Notify
- Alerts & Notifications
- Trunks Status Pages
- Real-Time System Graphs
- FindMe
- Boomerang Mobile Integration
- Call Screening
- Music-On-Hold (Unlimited)
- Historical System Graphs
- Unlimited Call Queues
- Full Featured A.C.D.
- Skills-Based Routing
- Graphical Queue Reports
- Barge Report
- Agent Call Recording
- Agent Variable Log-off
- Agents on Cell Phones
- Agents Shared across Sites
- Real-Time Queue Stats
HUD features:
- Operator Panel (w/ BLF)
- Call Parking Area
- Drag & Drop Call Control
- Color-Coded Call Status
- Drag & Drop to Voicemail
- Extension Sorting
- Enterprise Instant Messaging
- Outlook Integration
- Presence Management
- Click-to-Call Mobile Phones
- Click-to-Email
- Desktop Alerts
- Interactive Desktop Alerts
- Group & User Permissions
- Extension Grouping
- Extension Search
- Extension SearchQuickMenu
- Shortcuts (Hotkeys)
- On-the-Fly Recording
- Queue Status
- Agent Login/Logout
- Call Barging (active)
- Call Monitoring (passive)
- Web Access to Recordings
- Advanced CRM Integration
Some important new features in trixbox Pro 2.0 worth highlighting:
First, calendar-based scheduling has been added allowing you to have your call menu do something specific, such as special holiday greetings. trixbox Pro has had very good auto-provisioning already, but they’ve improved it in 2.0. I was able to connect several Aastra and Polycom phones on our network and trixbox Pro auto-detected them and auto-assigned them an extension in sequential order. Hands free phone auto-provisioning is supported on all supported Aastra models and Polycom models with firmware 2.2 and above.
Resellers will especially like the single-screen management for all of their customers. From one Admin web interface you can switch between installations with two clicks of the mouse in the lower-right corner of the Admin Panel.
Room for Improvement:
The call recordings (screenshot of interface below) within the web-based interface should have a memo text field to allow users to add recording details, such as caller’s name, topic discussed, etc. If you keep a lot of recordings, this will make it easier to reference them in the future.
I’d like to see standard-based videoconferencing support in HUD Pro. If Counterpath can offer a slick videoconferencing app (eyeBeam softphone) based on SIP and other industry standards, then surely so can Fonality. In addition to video, perhaps collaboration capabilities (WebEx, Microsoft Live Meeting) would be a nice addition. This would negate the need for a separate collaboration platform, the associated licensing fees, and duplication of the same employee information which must also be maintained due to employee turnover.
One last suggestion would be to email the call recordings automatically, similar to the voicemail-to-email feature. You can of course access the recordings via the web and download & save to your local PC, but an automatic email option would be nice.
Conclusion
I was very impressed with the ease-of-use of the admin interface, and I especially liked the strong mobility features. Users will like the web-based visual voicemail and voicemail-to-email features. The web-based GUI is one of the best you’ll see on any Asterisk-based platform and it makes extensive use of AJAX and tool-tips. Lastly, HUD’s motley of features, including desktop call control, presence, and on-the-fly recording make trixbox Pro a compelling choice when deciding which IP-PBX to purchase.
Perhaps the only caveat with trixbox Pro as compared with other Asterisk-based solutions is the trixbox pricing. Many Asterisk-based solutions are extremely inexpensive, some under $1000. trixbox Pro does offer a relatively inexpensive monthly option. For instance, for trixbox Pro Call Center Edition (CCE) that I tested, it’s $11.39/month per extension for 26-50 extensions. However, the lifetime fee is $159.99/month per extension which works out to be $14,259.50 for a 50 extension IP-PBX. That’s not ‘terribly’ expensive, but it is more than some competing Asterisk-based solutions, though still much less expensive than many Nortel, Avaya, or Cisco systems.
I’m sure Fonality would argue their main competitors are not other Asterisk solutions but the “Big 3″ (Avaya, Nortel, Cisco). Further, in my past discussions with Fonality, they’ve told me that they believe they add a lot of value to the “core” Asterisk that gives it a much stronger feature-set than other Asterisk-based systems. In testing trixbox Pro I would agree that it has many features I have not seen in many other Asterisk-based systems, including HUD Pro, call screening, call recording (some others do have this), strong call center functionality (queues), easy-to-use web-based admin, etc.
I recently saw a demo of Digium’s latest version of Switchvox down at Digium’s headquarters and did notice there are features in Switchvox that trixbox Pro doesn’t have, like Web 2.0 features. I plan on doing a review on Switchvox in the near future. It’ll be interesting to compare the two. Overall, I think trixbox Pro is one of the most feature-rich IP-PBXs I’ve reviewed and I would not hesitate to recommend it to companies looking for an easy-to-use and easy to maintain IP-PBX.
Tags: Asterisk, fonality, HUD, IP-PBX, trixbox CE, trixbox Pro, trixbox Pro Call Center Edition, trixbox Pro CCE, trixbox Pro Enterprise Edition, VoIP
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Fonality Lands $12M Financing Round from Draper Fisher Jurvetson (DFJ)
Fonality, a provider of IP-PBX systems based on Asterisk, has just secured a $12 million financing round led by Draper Fisher Jurvetson (DFJ) Growth Fund with participation from existing investor Intel Capital. Draper Fisher Jurvetson is a well known venture capital firm backing more than 600 companies including industry-changing companies such as Hotmail, Overture, Four11, Baidu, and fellow VoIP company, Skype.
I spoke with Fonality CFO Dan Rosenthal who explained that Fonality has had 16 successive quarters of growth, and that the main goal of this funding was to grow the distribution channel and accelerate growth. As part of the financing deal, DFJ will become part of Fonality’s board. Dan said, “We’re growing at a pretty good clip. Growth takes cash. We’ve got relationships with big partners such as Dell, PCMall coming down the line. In order to really accelerate our growth we want to make sure we’ve got the bulk and the balance sheet to support that growth.”
Dan also mentioned they are focusing on OEM distribution. He pointed to Dell’s deal earlier this year with Fonality and stated that other major co-branded and OEM branded products in lots of different places were forthcoming. PCMall being another example that Dan mentioned.
According to Fonality, they will use this additional capital to continue expanding its share of the $7B domestic telephony market and take a position in the $25B global telephony market, with a special focus on affordability and mobility for businesses with 5 - 500 people per location.
“Fonality’s model has always been to free people from the prison of their cubicles, and to do so affordably,” said Fonality CEO Chris Lyman. “But, with the rising price of gas and the increased adoption of high quality broadband, our vision of ‘your office is the world’ is nearly upon us. This investment by DFJ and Intel Capital, two top-tier investment firms, gives us the freedom to further accelerate our growth and acquisition strategy.”
Randy Glein, managing director of DFJ Growth Fund, has joined Fonality’s board of directors. “Fonality is the right solution at the right time,” said Glein. “The company has developed a powerful, affordable communications platform for small and medium-sized businesses, allowing Fonality-enabled businesses to compete more effectively in the global economy. Fonality combines the benefits of open source development, rich proprietary features, and professional support to help its 5,000 customers take advantage of the latest communications innovations very affordably. Fonality supports workers wherever they are and lets them communicate inside and outside their organizations with easy-to-use software available everywhere, on any device.”
Earlier this year, the company launched a relationship with Dell to sell Fonality products online at Dell.com and via Dell telephone sales. Fonality currently has 140 employees, offices in three countries, and operates four global data centers.
Tags: Chris Lyman, Dan Rosenthal, DFJ, Draper Fisher Jurvetson, Fonality, funding, IP-PBX, Skype, venture capital, voip
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Digium Headquarters Tour
During my tour of Digium’s headquarters in Huntsville, Alabama I snapped quite a few photos that I thought I’d share. I’ll just share a few here, but if you want to see them all, go check out the online photo album I created specifically for Digium:
http://blog.tmcnet.com/blog/tom-keating/photos/digium/
Be sure to check my other photo albums as well:
http://blog.tmcnet.com/blog/tom-keating/photos/
The new Digium headquarters is very impressive. I couldn’t help but be in awe of what Mark Spencer created. Obviously, there is money to be made in open source, and seeing the fruits of Mark’s and his team’s labor in this state-of-the-art building was proof of that. I can’t help but reminisce back to my November 2001 column titled “In Search Of A Linux-Based PBX” where I listed several “young” Linux-based PBX solutions. One such solution I discovered was Asterisk. Check out what I wrote back in November 2001 to see how far Asterisk has come:
As my search continued, I stumbled upon a company called Asterisk. According to their Web site, Asterisk is an open source PBX and general telephony toolkit that runs on the Linux operating system. Asterisk provides a set of APIs that essentially make it a type of middleware between Internet and telephony channels like VoIP, voice over frame relay, etc., and telephony and Internet applications like voice mail, phone directories, call parking, and so forth. Asterisk supports a flexible and extensible channel API, allowing any number of real hardware or software interfaces. It is purported to support ISDN, PRI, T1, and POTS through an Adtran Atlas. The Quicknet Internet PhoneJack and Quicknet Internet LineJack are both supported. Plans to support Lucent-based Winmodems also in the works. This product is not a turnkey out-of-the-box Linux-based PBX, and its still under development, but it does have several telephony functions working, such as call bridging, call transfer, call parking, and rudimentary voice mail. Check them out.
There was no Digium telephony cards back then. It ran on Adtran hardware and some Quicknet PhoneJack cards. Who knew that Asterisk would explode and become the dominant Linux-based IP-PBX solution?
Well, there is no better proof on how far Asterisk and Digium has come than to see their brand new building. Check out these photos:
Front side of the building
Digium front lobby - Nice cafe to get your morning caffeine fix (coffee) before you head off to do some Asterisk coding.
This is the Asterisk logo in the center of Digium’s headquarters on the 1st floor taken from the 2nd or 3rd floor above. Doesn’t this remind you of the Star Trek Enterprise’s warp core? (Seen below from a side view. Couldn’t find a top-view image which would be more similar)
I count 6 fans and at least 4 heat sinks. Now that is some serious coolage! This might be the AA350 I snapped but not sure.
This is Rotary Dial Phone with an Ethernet Jack making it the world’s only rotary IP phone!
It was in one of the Digium offices I passed by.
Digium has many Collaboration Rooms. This is just one of them. Note the (almost) floor-to-ceiling whiteboard for plenty of collaboration space.
Just a shot of the hallway with the cabling system overhead. Hmm, I only see like 7 cables. What sort of magical networking is Digium doing that they only need 7 cables in a company of 150 people?
Digium has several Collaboration Rooms. Note the orange chairs to match the orange Asterisk logo.
Don’t zoom in on the whiteboard. It contains top secret Digium information on how they plan on taking over the open source telephony world. Oh wait, they’re already doing that.
Can’t call yourself a software company without a foosball gaming table! Though I did notice Digium’s vending machines only sold Dr. Pepper and not Diet Coke. Sacrilege I say!
Digium’s training class room. Each student gets their own PC to install & configure Asterisk.
Digium and TMC having dinner
Left-to-right clockwise: Steve Sokol, John Todd, Russell Bryant, Dave Rodriguez (TMC), Greg Galitzine (TMC), Jane Brooks, Tom Keating (TMC), Bill Miller
There are several other pictures in the Digium photo album. Go check them out.
Tags: Alabama, Asterisk, Bill Miller, Dave Rodriguez, Digium, Greg Galitzine, Huntsville, IP-PBX, Jane Brooks, John Todd, linux, Mark Spencer, Russell Bryant, Steve Sokol, Tom Keating, voip
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The IP-PBX Energy Wars…
So today I get a new report from the Tolly Group stating that the ShoreTel Unified Communications system is significantly more energy efficient than the Cisco Unified Communications Manager. ShoreTel apparently topped Cisco in using less energy to drive VoIP communications in specific large, medium and small enterprise-class scenarios.
This energy usage comparison reminded me of Nortel’s The Nortel Tax Relief Plan which aims to “stop paying the ‘Cisco Energy Tax’ and save up to 40%”. Tony Rybczynski who works for Nortel and writes a TMCNet blog called The HyperConnected Enterprise sparked some controversy with some of his blog posts promoting the fact that Nortel is more efficient than Cisco. He even cites one customer that put a stop order on a $2 million dollar Cisco order once they did the energy efficiency calculations.
Is that what it’s come down to? Instead of feature-to-feature comparisons where going to have to start comparing energy consumption? I’m not against the idea, I just find it kind of humorous that everyone is jumping on the enviro-green bandwagon.
I hate to wonder if an IT manager, CTO, etc. might purchase a more efficient IP-PBX over a less efficient one that has many more features? Well, certainly in San Francisco and other uber-green areas that might be the case. Green trumps everything when you’re a greenie - not that there’s anything wrong with that.
I should point out that the IT Manager or CTO often not held accountable to what the electricity costs are. Many businesses see their electricity bills just as one of the costs of running their businesses. Other than instructing their users to turn off their radios, monitors, and computers at night, most businesses don’t delve into purchasing energy efficient computer or phone equipment. That is changing due to high energy costs - albeit slowly.
What’s missing from this ShoreTel energy comparison report is a comparison with Nortel, Avaya, and other IP-PBX players. Just who is the “king” of energy efficiency? Inquiring minds want to know.
So what are your thoughts on the IP-PBX Energy Wars? Do you care about efficiency or are features for important to you? Post a comment.
Lastly, the press release is included after the jump for your perusal…
Continue reading The IP-PBX Energy Wars……
Tags: Avaya, Cisco, Cisco Unified Communications Manager, efficiency, electricity, energy, IP-PBX, Nortel, ShoreTel, Tony Rybczynski, VoIP
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Aastralink RP Deput Coincides with Microsoft Response Point SP1 release
As I wrote back in March, the Microsoft Response Point Service Pack 1 (SP1) release would coincide with Aastra’s Response Point phone system debut, which is interesting when you consider Aastra got into the IP-PBX game with a Microsoft IP-PBX competitor - the AastraLink Pro 160 appliance, an Asterisk IP-PBX derivative. Well, today marks the launch of Microsoft Response Point SP1 and the Aastra Response Point system aptly named the AastraLink RP.
Before I go into the AastraLink RP, let me first mention that the Microsoft Response Point Service Pack 1 is available as a free download to current Response Point users. Secondly, the announcement is being made at the Microsoft Worldwide Partner Conference 2008. I’m told, “a very special senior Microsoft executive will make an appearance to help announce.” I would have guessed it will be Bill Gates’s but he retired recently on June 27th. Perhaps a special final appearance in a Microsoft capacity?
SP1 is a significant feature update to Microsoft Response Point designed for small businesses with one to 50 employees. Current Response Point customers and partners in the U.S. and Canada can download SP1 for free at http://www.microsoft.com/responsepoint. SP1 includes improved performance, SIP trunking, a call history log, the ability to push out new firmware to the phones automatically, and more. As for the SIP trunking, Junction Networks is the newest VoIP provider supported on Response Point SP1. New Global Telecom Inc. (NGT) and Cbeyond are two previously announced SIP trunking providers that work on Response Point.
Response Point features an easy-to-use admin, voice-enabled user interface, advanced call routing, built-in voice mail, automated receptionist and contact integration with Microsoft Office Outlook. In my pre-SP1 review of Response Point, I opined about the Asssitant’s lack of click-to-dial when I wrote, “Unfortunately, you can’t click-to-dial anyone in your corporate directory to initiate a call, transfer, or conference.” Well, Microsoft added click-to-call functionality for any contact using the Assistant software. The Assistant also now has “presence” so you know when co-workers are on the phone.
I also complained, “Currently, the Music On Hold (MOH) is statically defined in the firmware by the OEM manufacturer and cannot be changed. It would be nice if the MOH could be customizable to support .wav or .mp3 files.” They took that suggestion to heart as well since and SP1 adds the ability to select music for parked calls and hold time. I also wished Response Point had call screening of caller’s name, call screening of voicemail being left live, and the ability to use RP phones remotely for telecommuters. Alas none of these 3 feature ideas made it into SP1, however, I was told by Microsoft that remote phone support is on the roadmap.
I should point out that using the Response Point button on the phone employees can press one button, wait for the chime, and then simply by issuing verbal commands to access anyone in the company directory, including anyone imported via their Outlook contacts. Press the RP button say “Dial John Smith” and it instantly dials. It’s a great usability feature, especially when combined with the Aastra cordless handset.
The AastraLink RP comes with a cordless phone option (part of the 6757i CT RP model) which Microsoft and Aastra told me includes the “Response Point” button with speech-rec functionality. Aastra is one of three Response Point hardware manufacturers and the AastraLink RP base unit comes pre-loaded with SP1 and is available to customers in the U.S. and Canada starting today.
As part of this announcement, Aastra will offer a starter package that includes a base unit, external analog telephony adapter and three phones (including one cordless model) for a suggested list price of $2,400. Additional desktop and cordless phones are available ranging in price from $139 to $399 MSRP for the cordless model; prices are estimated retail prices. D-Link Corp. and Quanta Computer Inc. also will begin to roll out SP1 on their Voice Center IP Phone System and Syspine systems, respectively.
The AastraLink RP phone system is comprised of the RP 500 Base Unit which hosts the Response Point system software, the RP 540 Gateway with 4 analog telephone ports, and a choice of three Aastra Response Point IP phone models. The gateway and phone devices all feature auto-discovery and auto-configuration, making installation a snap.
Three different enterprise-grade phones have been developed for the AastraLink RP system; the entry level 6751i RP, the full featured 6753i RP, and the advanced 6757i CT RP which comes with a cordless handset. Offering larger display screens, superior audio, programmable keys, full duplex speakerphones, and POE support, AastraLink RP terminals deliver enterprise level features and functionality to the SMB market. In addition, the 6753i RP and 6757i CT RP models have headset jacks and support up to three expansion modules.
The All in all, Microsoft seems to be making great strides in the feature-set for their Response Point IP-PBX targetting the SMB, including adding some of my suggested features from my 2007 review. Also, the AastraLink RP phone system seems like a good IP-PBX solution for the SMB at an attractive price-point. I might just have to get one to review and check out.
Tags: 6751i RP, 6753i RP, 6757i CT RP, Aastralink RP, appliance, ip-pbx, microsoft, Response Point, Response Point Service Pack 1, sp1, voip
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Sangoma Acquires Paraxip Technologies
Sangoma has acquired Paraxip Technologies giving Sangama a broader communications play than just offering telephony boards for Asterisk-based solutions. Sangoma is positioning the news as a new foothold in the Windows Unified Communications and IP Contact Center markets.
This kind of reminds me of Aastra which first started offering IP-phones used in Asterisk, trixbox, FreePBX, etc. but then launched the Aastra Aastralink Pro 160 Appliance, making Aastra a ‘competitor’ to IP-PBX platforms such as Digium’s Asterisk, trixbox, etc. Similarly, by Sangoma acquiring Paraxip Technologies, they get their entire NetBorder product line, which enables IP telephony applications such as IP-PBXs, IVRs, call center routing, outbound dialers, and more. Obviously, the IP-PBX and call center functionality is a direct shot across the bow of some of their IP-PBX partners that use Sangoma hardware. Very interesting…
They have a conference call later today that I’m going to try and join and ask this very question. For now, check out the news release after the jump…
Continue reading Sangoma Acquires Paraxip Technologies…
Tags: Aastra, AastraLink Pro 160, asterisk, freepbx, ip-pbx, paraxip technologies, sangoma, voip
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