phone's archive
Apple, Nokia, RIM, Samsung, & others settle on micro-USB phone charger standard
One phone charger for all? There can be only one! According to Reuters, telecom firms have backed standard phone charger in Europe. No, hell hasn’t frozen over. Yes, you will finally be able to use a single charger that works with any future mobile phone you purchase - at least in Europe anyway. That’s one less charger to pack when traveling and one less charger to throw away when you upgrade your mobile phone. The top mobile telephone suppliers, including Nokia, Sony Ericsson, and others have agreed to back an EU-wide standard for phone chargers. New mobile phones will use a standard micro-USB socket to ensure compatibility.
Motorola, LG, NEC, Qualcomm, Research in Motion, Samsung and Texas Instruments have also signed the agreement. Interestingly, Apple has also signed on. You mean Apple which charges licensing rights for their proprietary docking cable is on board too? Well, well… Isn’t Apple fighting Palm over the Pre emulating an Apple iPod to enable iTunes support? Seems against Apple’s nature to support a connection standard that has enabled such a huge third-party cottage industry for Apple’s iPod and iPhone products. For example, Apple licenses their connector to FM transmitters, iPod external speakers, and more.
So one charger for any mobile phone, surely there’s a catch…
Tags: apple, iphone, ipod, itunes, NEC, nokia, palm, qualcomm, RIM, samsung, texas intruments
Related tags: mobile phone, phone charger, standard phone, apple, charger, phone
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sipgate enters U.S VoIP Broadband Market offering free calls
Today, sipgate is launching a new free VoIP broadband phone service called sipgate one. sipgate one is similar to Vonage, Packet8, and other broadband VoIP providers, but with some additional cool features and a fairly unique pricing plan. For instance, you get a free telephone number, no set-up costs and no monthly charges or minimums.
I spoke with sipgate CEO Thilo Salmon to find out more about sipgate one. First, unlike many VoIP broadband providers, sipgate one isn’t tied to any specific SIP hardware (locked ATAs, SIP phones, softphones, etc.), so you can use it with your favorite SIP device or use a SIP-based ATA and connect your favorite analog cordless phone. Want to use a softphone but don’t want to mess with SIP credentials? No problem - sipgate one has a free softphone app that will autoprovision for you. Those that want to use their favorite SIP device will be pleased to see a helpful drop-down list of many popular SIP devices with corresponding screenshots on how to configure the device to work with sipgate one.
What does it cost?
When using a VoIP phone, outbound calls to other sipgate users are completely free, just like Skype. However, inbound calls on the free U.S. phone number are also free, unlike Skype which I know charges a monthly or yearly SkypeIn subscription. I was a bit surprised sipgate was offering the first phone number for free, so I contacted Thilo a second time to confirm. He responded, “The first U.S. number is indeed completely free. So we are inviting everyone to sign up for a free number and as long as one only uses sipgate to receive calls on a SIP phone the service is free in its entirety. We do not even ask for a credit card. That does set us apart from Skype. While they do not charge per incoming minute, Skype charges a yearly (monthly?) fee as you have pointed out.”
When dialing other phone numbers in the U.S. and Canada are just 1.9¢ per minute and calls to toll-free numbers are free. Other rates apply when calling other countries. Thus, sipgate only charges for outgoing calls and E911 ($1.90 monthly), if activated. No other charges or fees are leveraged on a sipgate one subscriber. Thus, to get started you can initially charge your account with a minimum of $10 with automatic crediting if the balance falls below a user-specified amount ($5, $10, $20, $50). Thilo told me that they aim to keep their international rates lower than Skype’s to make them a very compelling option to potential customers. I then asked Thilo who was comparable feature-wise and he responded, “Google Voice. They don’t allow you to connect a SIP phone, but the features are very similar.”
The true power behind sipgate is its Web-based interface which gives you full access to your voicemail messages, recordings, and faxes. I tested a preview version of sipgate one and was pretty impressed with the features and ease of use. My experiences in my mini-review of sipgate follow below…
Thilo told me that they designed the Web-based interface with Google’s Gmail in mind. For instance, you can search, star/unstar a recording, as well as label recordings, which are very similar to Gmail. Additionally, the interface allows users to call someone back with a single click, as well as divert calls to other landline or mobile phones. Starting a call is as simple as clicking the New Call button and then choosing which phone device you wish to use. Below you’ll see I have 3 options for a new call - an arbitary connection (specify any phone number ad hoc), phone of Tom Keating, and Mobile or Landline of Tom Keating.
As seen by the screenshot above, you can bridge a call to any of your phone devices by initiating a call via the web. This is very similar to Jajah, who made the web-based calling method famous. This interface can be used on your web-enabled smartphone - Windows Mobile, Apple iPhone, etc. if the device doesn’t have an embedded SIP stack or 3rd party SIP softphone installed for making outbound calls.
Mobility features are just as powerful. You can have your home, office and mobile phone ring in parallel. If you are using a mobile phone with a SIP stack, (many Nokia series have SIP) you can be in the car, receive a VoIP call, and then simply press *6 on your mobile phone to record the call. When you next gain access to a web browser you can retrieve the call recording. This is perfect for sales people on the go and who need to enter important call details into a CRM system.
Other features included the ability to customize your outgoing message by uploading an mp3 file, divert calls to another number, create an ad-hoc conference with the touch of a button, and the ability to view missed calls with CallerID info. I asked Thilo what they use on the back-end and he told me they use a fork of SIP Express Router, which enables Class 5 type features. Additionally, Thilo stated there is no maximum storage limit for voicemail and faxes.
Outbound faxing is a breeze. You can click some fields on the web interface and add the fax number, from/to info, add a signature, and of course body text. You can also attach a PDF and it will render it. I asked Thilo about Word support and he said that was in the works.
Inbound faxing does require a separate phone number, but there is no fee to receive faxes. There is a nominal $2.90 one-time fee per additional phone number. I asked Thilo about CNG autodetection of fax tones using just a single universal phone number, but he said “We have found that most people don’t really like that.” Personally, I’d rather just have one phone number on my business card, but it’s a minor complaint.
“There is simply no barrier to people disconnecting their old phone lines anymore. Phone and cable companies have long been pushing voice plans in the region of $25 to $40 per month–which end up being as much as $60 or more with extra charges–and that’s just ridiculous,” said Thilo Salmon, CEO of sipgate. “Even with calls to other landlines and mobile phones, most users will spend less than $5 a month using sipgate one. And for those people only receiving incoming calls on their VoIP phone, the service is completely free.”
sipgate is also readying a multi-user edition of the service aimed at small businesses, which will not only replace landlines, but also customer premise phone systems. sipgate is certainly setting a new low-price benchmark with bundled powerful features that should cause VoIP fans to seriously consider them. Free phone number, free toll-free calling, free inbound calling, free inbound faxes, what’s not to love? 
Tags: broadband voip, e911, free calls, iphone, packet8, sipgate, sipgate one, skype, voip, vonage, windows mobile
Related tags: phone number, asked thilo, phone service, mobile phone, based interface, phone
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IPEVO S0-10W Skype Desktop Phone
IPEVO’s S0-10W Skype Desktop Phone is a pretty neat Skype phone, that unlike most Skype phones, was designed for your desk, not to carry around. That includes the IPEVO Wi-Fi Skype phone, I recently reviewed. While the IPEVO Wi-Fi Skype phone was designed to be portable and carry around, Skype phones like these are small which means a very small speakerphone (or none at all), resulting in limited or no hands-free speakerphone capabilities. The IPEVO S0-10W Skype Desktop Phone solves that problem.
Rather than sporting an Ethernet connection, however, it uses a built-in Wi-Fi adapter, so you could carry this around the house / business if you wanted to (still needs power, so would have to plug it back in). The S0-10W offers the normal feel of a desktop phone in usability and appearance, but features a built-in CPU for computer-free Skype communication. The S0-10W features an adjustable 2.4-inch color LCD three user-defined speed dial keys, an LED light indicating incoming and missed calls, as well as new voicemails, and a hands-free speakerphone button with echo cancellation.
It’s too bad this desktop phone doesn’t have an embedded webcam so you could do some Skype videoconferencing. Now that would make this an even sweeter product! 
Add internet surfing to the mix and we’re golden.
“IP calling no doubt offers a financial benefit to consumers, however as with any unfamiliar technology, some may be hesitant to try something new,” said Royce Hong, CEO of IPEVO. “That’s why we designed the S0-10W Desktop Phone. By offering consumers a similar interface to the landline desktop phone most people currently use, we feel confident that families and businesses alike will have the confidence to make the switch for their everyday communications needs.”
The IPEVO S0-10W Skype Desktop Phone is now available at the MSRP of $169.99.
Tags: desktop phone, ipevo, IPEVO S0-10W Skype Desktop Phone, S0-10W, skype, voip, wi-fi
Related tags: desktop phone, skype desktop, skype phone, skype phones, skype, phone
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IPEVO’s S0-10W Skype Desktop Phone is a pretty neat Skype phone, that unlike most Skype phones, was designed for your desk, not to carry around. That includes the IPEVO Wi-Fi Skype phone, I recently reviewed. While the IPEVO Wi-Fi Skype phone was designed to be portable and carry around, Skype phones like these are small which means a very small speakerphone (or none at all), resulting in limited or no hands-free speakerphone capabilities. The IPEVO S0-10W Skype Desktop Phone solves that problem.
Rather than sporting an Ethernet connection, however, it uses a built-in Wi-Fi adapter, so you could carry this around the house / business if you wanted to (still needs power, so would have to plug it back in). The S0-10W offers the normal feel of a desktop phone in usability and appearance, but features a built-in CPU for computer-free Skype communication. The S0-10W features an adjustable 2.4-inch color LCD three user-defined speed dial keys, an LED light indicating incoming and missed calls, as well as new voicemails, and a hands-free speakerphone button with echo cancellation.
It’s too bad this desktop phone doesn’t have an embedded webcam so you could do some Skype videoconferencing. Now that would make this an even sweeter product! Add internet surfing to the mix and we’re golden.
“IP calling no doubt offers a financial benefit to consumers, however as with any unfamiliar technology, some may be hesitant to try something new,” said Royce Hong, CEO of IPEVO. “That’s why we designed the S0-10W Desktop Phone. By offering consumers a similar interface to the landline desktop phone most people currently use, we feel confident that families and businesses alike will have the confidence to make the switch for their everyday communications needs.”
The IPEVO S0-10W Skype Desktop Phone is now available at the MSRP of $169.99.
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IPEVO Wi-Fi Phone for Skype Review
The IPEVO Wi-Fi Phone for Skype launched just last month and IPEVO sent me a review unit to check out, which I’ve been testing for a few weeks.
Installation
Installing the IPEVO Wi-Fi Phone was a snap. Although it only has a numeric keypad for entering in alphanumeric characters for entering in the WEP/WPA/WPA2 key. As you hit a key, it shows you the current key where your cursor is located, but it also shows you a horizontal list of all the characters that key can produce and highlights the current key. This is useful not only to confirm what keys it can produce, but if overshoot the key you want, you can easily see how far away you are to scrolling back around and reproducing that key. Definitely made entering the Wi-Fi key much easier and faster.
After entering in TMC’s Wi-Fi key I proceeded to add my Skype name and password. Similarly, it showed the key as it was being pressed, as well as the horizontal list of keys, making it very easy to enter in my Skype credentials. I should mention that you can add multiple WiFi networks, so you can have a Home profile, Work profile, etc. So when you walk into a WiFi area you have configured it will automatically connect.
The phone then signed into Skype and I saw my list of contacts. From the Contacts screen you can click the center green button or the Options button to access a submenu. From this submenu you can then Call the contact, Send Voicemail (if you paid for this service), View Profile, or go into Advanced where you can rename, remove, block/unblock, or add a contact. If you have a ton of contacts, no worries - holding the up or down key will cause it to autoscroll and it will wrap around to the top/bottom depending on which direction you were scrolling. Of course, it’s worth noting you don’t have to go into the submenu to initiate a call. You can simply highlight the contact and press the green phone icon to initiate a one-click dial.
The IPEVO Skype phone supports both Ad hoc and access point mode for connecting to a wireless network. For security it supports WEP, WPA Personal, and WPA2 Personal. It also supports static IP as well as DHCP (default).
Let’s now do a video demo of the IPEVO Skype WiFi phone which shows the phone in action making a SkypeOut call to a direct dial number which terminates to my Aastra VoIP desk phone.
I like that the phone can display Skype credit, access SkypeIn settings, and Skype voicemail. It also lists the history by all calls, missed calls, incoming calls, outgoing calls. Access to voicemails and contact requests is also available.
It took 71 seconds from off till fully authenticated on the WiFi network and can dial a Skype contact. While other Skype WiFi phones also take over a minute to fully boot-up, I’d like to see all Skype WiFi phones speed up the boot process. I should add that there doesn’t appear to be a speakerphone available, even though the back of the phone sports a speaker, which apparently is only used for the phone’s ringtones.
I was pretty impressed with how lightweight the phone was and the call quality was excellent. I called some people and they said the call quality was just as good as a landline if not better. y only complaint about the phone is the lack of instant message (IM) support, but I haven’t seen other Skype phones support IM either. While typing an IM using just numeric keys (0-9, *, #) is difficult, the phone should at least be able to receive IM messages. But even that isn’t possible. Maybe there is a technical reason while IM wasn’t included or maybe was just a design decision. Still, I’d like to see IM on Skype WiFi phones.
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The standalone Skype phone features a backlit keyboard, 1.8″ LCD screen, and rechargeable battery. The 1.8″ screen supports 128×160 resolution with 65k colors. The phone supports G711 and G729AB codes with a voice sampling rate of 8KHz.
It supports 802.11b/g WiFi standards along with WEP, and WPA/WPA2-PSK
encryption protocols. Talking time is approximately 4hr & standby time is 30hr via its Li-ion 900mAh battery. I tested the standby time, and it was indeed about 30hr.
The IPEVO Wi-Fi Phone for Skype is now available at the MSRP of $129.99 at http://store.ipevo.com/
Tags: IPEVO Wi-Fi Phone for Skype, review, skype, SkypeIn, SkypeOut, voip, wifi
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Dotcom-Monitor announces new SIP Monitoring tool
Today, Dotcom-Monitor announced a new SIP monitoring tool to add to its portfolio of external monitoring services. It’s similar to other web-based Monitoring-as-a-Service (MaaS) services which monitor the uptime of web servers and notify when a problem occurs. In this case, Dotcom-Monitor’s SIP Monitoring service monitors on-premise or hosted IP-PBXs.
How’s it work? Dotcom-Monitor’s SIP monitoring service makes live intermittent SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Basically ,it acts as a SIP end client, placing an actual telephone call to a specified number, and checking the results of that call. The expected result of the call is setup as “Answer”, “No Answer”, “Busy”, or an Error Condition (if there is an unexpected result).
According to their representative, “real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.”
I’m going to talk to then next week to find out more. For now, check out the news release…
Dotcom-Monitor Enhances Unified Suite of Monitoring Services with SIP Monitoring for VoIP Systems
Easy-to-Use, Cost-Effective External Service Monitors and Analyzes SIP Systems or Infrastructure for Uptime and Performance
Minneapolis, Minn. − March 18, 2009 − Dotcom-Monitor, (www.Dotcom-Monitor.com), a leading provider of externally-hosted network monitoring services, today announced the addition of a cost-saving SIP monitoring service to the company’s unified suite of monitoring capabilities. Today’s announcement adds another critical tool to Dotcom-Monitor’s portfolio of external monitoring services, which includes uptime and performance monitoring of websites, web applications, and Internet network infrastructure.
Dotcom-Monitor’s new SIP monitoring service makes live periodic SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Additionally, real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.
“Due to SLA requirements and hybrid VoIP traffic routes, it is important for VoIP monitoring to proactively mimic the end-user’s perspective from external locations, rather than only relying on passive internal network analysis systems,” said Vadim Mazo, founder and chief technical officer of Dotcom-Monitor. “Many organizations’ VoIP monitoring and uptime needs are best addressed by a simple, cost-effective external system, rather than a large, expensive in-house system. Dotcom-Monitor’s SIP monitoring service provides customers a unique, easy-to-use, targeted solution for quickly identifying and pinpointing VoIP connectivity error conditions,” noted Mazo.
The new SIP monitoring service can be configured and managed with little or no IT expertise, which is ideal for the growing number of small and mid-sized businesses (SMBs) with on-premise or hosted IP-PBXs. Its proactive monitoring ensures connectivity errors can be addressed before the errors become downtime problems for customers. Dotcom-Monitor’s SIP monitoring service ensures SMBs can rely on their VoIP systems, Service Providers can monitor their VoIP infrastructure, VoIP Wholesalers can monitor Service Provider connectivity and reliability, and VoIP VARs and managed service providers can count on client uptime and revenue.
“As the VoIP ecosystem continues to grow in scope and complexity the need for simple and affordable SIP monitoring has never been greater,” said Jonathan Fuld, CTO of SIP Print, the only provider of pure, affordable SIP call recording systems for SMBs. “In fact, SMBS and any cost-conscious organization that is dependent on SIP-based communications could benefit by investigating an externally hosted SIP monitoring provider like Dotcom-Monitor.”
Dotcom-Monitor’s SIP Monitoring is available immediately by visiting: www.dotcom-monitor.com
Tags: Dotcom-Monitor, phone, SIP Monitoring, sla, sms, voip
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Microsoft OCS 2010 Will Finally Eliminate the PBX
Well, Microsoft has let the cat out of the bag and leaked word that Microsoft OCS 2010 will “remove the need for PBX equipment within your organization”. I’m certainly not surprised. Let’s flash back to last year where I wrote and article titled Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX. In it I wrote:
“Office Communications Server 2007 R2, debuting just one year after the Microsoft unified communications launch, highlights the pace of innovation that is possible with software,” said Stephen Elop, president of the Microsoft Business Division at Microsoft. “This new release puts Microsoft on a rapid path to deliver voice software that does much more than a network private branch exchange (PBX) and with much less cost.”
Interesting quote, eh? Does this not sound like Microsoft is sounding the death knell for the network PBX (IP-PBX)? This is an interesting turn of events. Microsoft hasn’t been pitching OCS 2007 as an IP-PBX replacement, but rather as something complementary. In fact, I remember talking with Microsoft about this last year (2007) and they went out of their way to explain that OCS 2007 is not an IP-PBX replacement. Also, Microsoft has many IP-PBX partners in the OCS 2007 arena, including Mitel, Nortel, and others. Slip of the tongue? Or is Microsoft going full-out into the IP-PBX arena? Certainly, the fear by many IP-PBX vendors is that one day Microsoft will offer a full-fledged software-based IP-PBX replacement, but I don’t think that day has come yet - even with the new features in OCS 2007 R2.
Now with OCS 2007 R2 fully launched and with added support for direct SIP trunking, the next logical step is a 100% Microsoft UC solution without the need for a PBX/IP-PBX at all. Of course, Microsoft OCS 2007 R2 is still currently very limited in the support it has for SIP IP phones. Most businesses aren’t ready to toss desktop phones for a 100% software-based softphone solution, i.e. Microsoft Communicator. So OCS 2010 will have to support SIP phones from popular SIP phone players such as Aastra, Polycom, and snom. Perhaps Microsoft will borrow or acquire the technology from SmartSIP, which recently launched an add-on for OCS 2007 R2 that enables any SIP phone to work with OCS.
So where did I hear that Microsoft was aiming to eliminate the need for a PBX in OCS? I discovered the information within a document on Microsoft’s website titled ‘Microsoft Unified Communications Business Value Tool’. On Page 24 it states:
You will deploy Office Communications Server 2010, which expands on the communications capabilities delivered in OCS 2007 R2. This release is designed to remove the need for PBX equipment within your organization and replace it with an integrated communications system that dramatically reduces management costs and gives end users innovative tools to communicate and collaborate across geographic boundaries from their office, home or on the road.
Not only do they state they will eliminate the PBX, but they declare the next version name of OCS (OCS 2010), which as far as I know Microsoft hadn’t announced yet. Many UC/VoIP experts predicted that eventually Microsoft would attack the IP-PBX space alone, but one has to wonder if alienating their IP-PBX partners is such a good idea. One of their strongest OCS partners is Nortel, who is experiencing financial difficulties and is probably not in a position to pressure Microsoft to back off. Mitel is another strong partner as well that could be impacted by Microsoft’s decision. Of course, Nortel and Mitel could still go after the SIP-based IP phone space within the OCS arena, but the IP phone market is much more of a commodity with a much lower margin than a full-fledged IP-PBX. Of course, there’s always the high-end media phone market with large margins. For instance, Polycom recently announced their VVX1500 media phone, which created some buzz.
I doubt OCS 2010 will have all the advanced call center functionality you get from Nortel, Avaya, Mitel, etc. After all, this will be Microsoft’s first release that doesn’t rely on the IP-PBX to do the intelligent call routing & handling. They’ll probably have some rudimentary call queues and skills-based routing, but not much else. Don’t expect predictive dialing in OCS 2010, a mainstay of the call center market. Still, a 100% software-based IP-PBX with unified communications capabilities will be a compelling choice for many businesses.
Tags: microsoft, Microsoft OCS, Microsoft OCS 2007 R2, mitel, nortel, sip, SmartSIP, unified communications, voip
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- Microsoft OCS 2007 R2 (next release) to be 64-bit Only - Aug 29, 2008
- Microsoft Drops VoIP from Windows Live Messenger - Jun 26, 2008
- Dell VoIP Products Analysis - Jan 23, 2008
- Microsoft OCS 2007 Review - Oct 16, 2007
- Microsoft and Nortel to co-develop IP-PBX - Jul 18, 2006
- Windows Server 2008 RDS Does VoIP - Mar 11, 2009
- Sametime and Microsoft OCS integration - Nov 10, 2008
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SmartSIP Launches for OCS 2007 R2 Enabling Any SIP Phone & Any SIP Trunking Service Provider
OCS 2007 R2 won’t replace your PBX just yet. However, their latest R2 version adds the ability to do direct SIP trunking, thus bypassing the need for an IP-PBX.
One drawback however is that Microsoft only supports direct SIP trunking with two providers, namely Global Crossing and Sprint. Well that’s pretty lame, considering their are dozens of decent SIP trunking service providers and probably hundreds across the entire world.
Fortunately, Mike Stacy an OCS 2007 guru, over at Evangelyze Communications has some products that enhance OCS 2007 R2 functionality. One such product is SmartSIP which launches tomorrow. According to Mike, the first dot release due next month will add the capability to use standard SIP phones with OCS. Currently, you have limited options namely Tanjay or Snom phones, but with SmartSIP you can use a Polycom IP phone, an Aastra IP phone, or dare I say, a Cisco IP phone connected to OCS 2007 R2. 
With the Cisco SIP firmware load of course.
Tags: aastra, cisco, Evangelyze Communications, ip phone, microsoft, Microsoft OCS 2007 R3, mike stacy, ocs 2007 r2, polycom, service provider, sip, SIP trunking, smartsip, unified communications, voip
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- Hmmm… Buy an IP-PBX at Costco - May 29, 2008
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- Microsoft Response Point Announces SIP Trunking Service Providers - Apr 18, 2008
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ooma Telo vs. magicJack
Rich met with ooma recently to see their latest wares and hear about their current business model. Recently, ooma ditched the ‘P2P voice network’ idea where users actually “share” their home landline with others and instead became a traditional VoIP broadband provider. Apparently, the privacy issues were too much to overcome, since users were concerns about fraudulent activity happening on their home landline by outside ooma users. I had my own reservations about the business model as well, since they claimed it would take 2,000 strategicly placed ooma boxes in all the various local exchanges to get good local call coverage for free P2P calls.
Besides becoming a traditional VoIP broadband provider, ooma is now going to start offering high-end media phones, that according to Rich Tehrani will in the future feature a picture frame, in-house sensors and cameras. As for what they offer today, in early January, ooma launched Telo, which offers unlimited, free VoIP-to-PSTN (U.S.) calls over the Internet along with a DECT 6.0 cordless phone that supports call screening, MP3 ringtones, 12-hour talk time, HD voice, speakerphone, two-line support, mobile transfer, and intercom. It supports up to eight phone numbers and six phones
ooma’s Telo phone system with DECT 6.0 handset.
The Telo phone system is expected to be available in the first half of 2009. The next question you’re probably thinking is “If it’s free unlimited U.S. VoIP-to-PSTN calls, how does ooma make any money?” The answer to that is ooma offers ooma Premier, with advanced features that they hope people will opt & pay for. (See: http://www.ooma.com/company/how_we_make_money.php)
Some of the Premiere features include:
- Instant Second Line allows you to make or take two simultaneous calls from a single phone number
- Blacklists helps you protect your privacy and block telemarketers
- Multiring lets you answer calls from your home phone or cell phone
- Message Screening allows you to listen in as the caller is leaving their message
- Send to Voicemail allows you to transfer a call to your voicemail
- Voicemail Forwarding lets you forward voicemail so that you can listen to it from your favorite email program
- Do Not Disturb allows you to roll your calls into voicemail without ringing your phone
- Personal Numbers allows you to select additional phone numbers in any calling area in the US
The “free” unlimited calling puts them on par with magicJack, but the magicJack is much less expensive (magicJack costs $39.99 1st year, and $19.99/yr in subsequent years). Pricing for Telo has not been announced, but I’m sure it will be much more expensive since the hardware costs so much more. One advantage for Telo is that magicjack requires your PC to be on all the time to make/receive calls over its USB-based dongle. The Telo phone system is a standalone phone that has no such restriction. It’s also a multi-line and multi-handset phone platform, so it’s more suitable to busy households that require multiple lines or phone handsets.
Check out Rich’s post for more on Telo and how the FCC is actually an investor in ooma.
Tags: , dect 6, handset, magicJack, ooma, phone, telo, voip
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- Congrats to Netgear - wins Top 10 Gadgets 2007 Award - Dec 12, 2007
- Ooma Goes Booma? - Jul 19, 2007
- Ooma Free Phone Calls - Jul 19, 2007
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31 Million IP Phones shipped by Mayan 2012 Doomsday, the Media Phone, & Slow Consumer Adoption
According to In-stat, nearly 31 Million Business IP Phones will ship in 2012. That’s if the Mayan 2012 Doomsday Prophecy doesn’t come to pass. You know, the one where the Mayan calendar ends on December 21st, 2012 - the same date as the Winter Solstice and when the Earth will be in galactic alignment with the massive black hole at the center of our galaxy, as well as our sun, resulting in a planetary shift. The date 12-21-12 reads as, A-B-B-A-A-B. Since the Hebrew language is read from right to left, this date would read BA ABBA. The Hebrew translation for BA ABBA is “Father comes” or “Father is coming”. If you believe in that sort of thing. Here’s a History Channel clip about 2012 that might bring out the conspiracy/doomsday nut inside you.
And then there is a Sony Pictures movie titled 2012 coming out this summer, as seen by this trailer:
Anyway, In-stat explains, “Within the business market, corded IP phones remain the standard, and will continue to dominate the enterprise IP phone market through 2012,” says Norm Bogen, In-Stat analyst. “However, WLAN and IP DECT phones continue to grow, especially within some specific vertical and geographical markets.”
Recent research by In-Stat found the following:
- Cisco, Avaya, and Nortel are leading the market for enterprise IP phones.
- Wi-Fi integration in cellular phones is growing rapidly; however, the majority of Wi-Fi/cellular phones are not designed for VoIP.
- Uniden holds top market share for consumer IP corded phones
The In-stat report points out that the IP phone market is “a tale of two markets” with IP phones thriving in business but as for the home consumer, not so much. By 2012, 31 million voice-centric business IP phones will ship but the consumer side will be outpaced by businesses more than 10 to 1. Why the slow consumer adoption of IP phones?
According to In-stat, “the nascent consumer market for voice-centric IP phones is being subjugated by the introduction of IP media phones, such as the Verizon Hub and AT&T HomeManager that support both IP communications, as well as delivery of Internet information and multimedia content.”
I think they’re a little premature in that statement. The Verizon Hub is a cool device, but it literally just came out, so it is not affecting consumers from buying IP phones at home. I think the reason is that consumers are happy with their home cordless phones with built-in answering machines. Some even have the multi-handset cordless phone systems, which allow you to strategically place handsets around the house with the ability to screen callers and remotely check the answering machine without going to the base unit. There just isn’t a good reason to purchase a $150-$300 corded/desktop IP phone for the home.
You could argue that a Wi-Fi phone might be a good option for the consumer. Wi-Fi phones are less expensive than desktop IP phones, they’re mobile, and they can get you cheaper or even free calling (i.e. Skype-to-Skype calls). But Wi-Fi phones have notoriously bad battery life. A better option in my opinion is a 
DECT 6.0 phone device with VoIP capabilities, such as the Philips VOIP841 Skype phone. Although there are other WiFi and DECT phones worth a look. I should point out that Wi-Fi phones have the advantage over DECT of sometimes offering a built-in browser so you can access the web.
Let’s look at how In-stat defines “media phone” and “why the media phone”:
The media phone is a new category of broadband device that combines the power of the PC with the performance of a telephone. The result is an always-on multimedia broadband device that is perfect for accessing online news and weather, viewing videos, and a host of other applications. In-Stat believes that the media phone will complement the PC, TV, and mobile handset, becoming an indispensable 4th screen in the home. Service providers and IP PBX vendors, alike, are introducing media phones because they add value to traditional voice telephones and related services.
Verizon Hub, a sample media phone
I certainly agree that consumers will start to adopt “media phones” in their homes, but only if the media phones are subsidized by the carrier. They’ll be too expensive otherwise. It worked for the cell phone market, so it can work for the home as well. There have been plenty of times I wanted to check the weather or current movie times, but had to boot up my PC in order to look up information. Having a media phone in the living room with instant Internet access is a nice feature to have. I do agree with the In-stat report that businesses will continue to be the main driver behind IP phone sales, but I wouldn’t be surprised to see traditional phone manufacturers such as Uniden developing cordless IP phones for the home market that offer Internet access.
You can download a free copy of In-Stat’s media phone research report: The Media Phone Has Arrived!
Relatedly, the research, “IP Phones Worldwide-On the Desk and Beyond” covers the worldwide market for voice-centric IP phones. It includes:
- IP phone vendor market shares for 2007 and 1H2008, segmented by phone type and consumer versus business
- A 5-year forecast by IP phone type (Corded, WLAN, Cordless DECT, Dual-mode Cellular/WLAN, Consumer, Business)
- Analysis of trends in business and consumer markets
- Profiles of more than two dozen vendors
Tags: 2012, DECT, doomsday, in-stat, ip phones, mayan, media phone, research report, skype, Verizon Hub, voip, wi-fi phone
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- Verizon Hub News - Jan 23, 2009
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- TechCrunch Touchpad - Needs 3G/4G/EVDO, Skype, and VoIP - Jan 19, 2009
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IPEVO Wi-Fi Phone for Skype
The IPEVO Wi-Fi Phone for Skype is a new Wi-Fi phone specifically designed to run Skype without the need for a PC. The standalone Skype phone features a backlit keyboard, 1.8″ LCD screen, and rechargeable battery. The 1.8″ screen supports 128×160 resolution with 65k colors. The phone supports G711 and G729AB codes with a voice sampling rate of 8KHz. No wideband audio support apparently. 
It also sports a headset jack socket for connecting a headset.
It supports 802.11b/g WiFi standards along with WEP, and WPA/WPA2-PSK
encryption protocols. Talking time is approximately 4hr & standby time is 30hr via its Li-ion 900mAh battery.
The IPEVO Wi-Fi Phone for Skype is now available at the MSRP of $129.99, exclusively at http://store.ipevo.com/
Tags: IPEVO WiFi Phone for Skype, skype, voip
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Duper WeenieWideband Codec”>Skype Launches 4.0 with SuperDuper WeenieWideband Codec - Feb 03, 2009 - VoiceGear SkyBridge at ITEXPO - Feb 03, 2009
- Dual Stack SIP and Skype IP Phone Coming - Jan 30, 2009
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- TechCrunch Touchpad - Needs 3G/4G/EVDO, Skype, and VoIP - Jan 19, 2009
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- Apple Large screen iPod Touch in Future Plus Skype & Apple Partnership? - Dec 31, 2008
- Flash Skype Video cards = Flash Skype Client in the Near Future? - Dec 19, 2008
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Plantronics .Audio 1100M USB phone for Office Communicator 2007
Plantronics today unveiled a phone handset optimized for Microsoft Office Communicator 2007. The new wideband .Audio 1100M USB phone handset features a standard, 10-key dial pad combined with call-answer and call-end keys, a visual ring indicator, a speakerphone activation key and ringer volume control. A headset jack is also available on the handset for users who want a hands-free option.
The Plantronics MCD 100 USB speakerphone, also announced today, is an option for those conducting group conference calls via Office Communicator 2007. It provides 360-degree room coverage so more than one person can participate in a PC-based audio conference. To achieve call clarity, the MCD 100 includes acoustic echo cancellation, noise suppression and equalizer algorithms embedded within the product.
The timing of the new product launches is perfect since Microsoft Office Communications Server 2007 R2 launches today at 9:30am PST.
Pricing and Availability
The .Audio 1100M, a USB phone handset, has an MSRP of $79.00. The MCD 100, a USB speakerphone, has an MSRP of $179.00.
Tags: OCS 2007 R2, Office Communicator 2007, plantronics, Plantronics .Audio 1100M, USB phone, voip
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- Office Communications Server 2007 R2 beta released - Sep 02, 2008
- Office Communications Server 2007 R2 Rumors - May 15, 2008
- XML Contest for VoIP Phones - Sep 04, 2007
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D2 Technologies Demos reference design for WiMAX mobile phones at ITEXPO
D2 Technologies is demonstrating a complete reference design for WiMAX mobile phones at ITEXPO. D2’s mCUE communications user interface claims to be the industry’s first turnkey software solution for quickly delivering mobile WiMAX devices to market
D2 Technologies, a market leader in embedded IP communications software platforms, today announced that private demonstrations of the industry’s first complete reference design for the creation of WiMAX mobile phones are available at ITEXPO East 2009 in Miami, Florida from February 2-4, 2009.
Built around the company’s breakthrough mCUE converged communications client, the reference design provides manufacturers with a powerful turnkey software solution for quickly developing and delivering WiMAX handsets and similar portable devices to market. The reference software implementation is running on a dual-mode mobile platform running Linux on a Texas Instruments OMAP-based processor; this version is available immediately, with additional implementations on the Google Android and Microsoft® Windows® CE/Mobile platforms coming in the second quarter of 2009.
ITEXPO attendees as well as those attending the 4G Wireless Evolution Conference collocated with ITEXPO East, can arrange for a first look at the WiMAX mobile phone reference design utilizing D2’s mCUE by contacting the company during the show.
“The industry is pouring tremendous resources into the creation and delivery of a WiMAX infrastructure, and manufacturers will need a fast and cost-effective way to develop mobile devices that take advantage of the greater bandwidth, range and other benefits of this technology,” said Doug Makishima, vice president of marketing and sales at D2 Technologies. “Our mCUE platform offers these companies a quick, easy and affordable path to the delivery of converged, unified WiMAX communications devices.”
The reference design provides all of the components necessary for manufactures to quickly field WiMAX-compatible mobile devices, from the underlying voice engine and communication protocols to the complete user interface. The embedded softDSP VoIP engine includes all of the required CODECs, error correction, PLCs, jitter buffer, DTMF, RTP/SRTP session keys, and other related technology. The complete communication support includes the IMS-SIP protocol and other IMS-, VCC- and 3rd party-related services and protocols required for voice service, IM chat and presence over WiMAX, as well as multi-radio, multi-network seamless voice call handover. The user interface elements, which are customizable for specific OEM and service provider requirements, include a contacts address book with presence and IM chat capabilities for a true unified communications experience.
The reference design showcases D2’s mCUE solution, which pairs an innovative, patent-pending communications user interface with the company’s vPort MP VoIP software platform to address the needs of OEMs and service providers delivering integrated UC and Fixed Mobile Convergence (FMC) functionality. mCUE provides a complete embedded software framework for multi-mode mobile devices for enterprise, carrier and consumer use, such as dual-mode Wi-Fi - and now WiMAX - enabled smartphones.
Mobile devices with mCUE provide users with advanced presence-based and push-to-x control of cellular and VoIP calls, PBX feature activation, IM, email, SMS and other features typically only available on PC-based unified communications soft clients. Its revolutionary user interface, built on top of a multi-identity, multi-session, multi-protocol engine, enables flexible interoperability with multiple communications services such as enterprise IP PBXs and UC systems, and commercial VoIM services like Google Talk, Yahoo!, MSN, AIM, and others. It is highly adaptable to any network deployment scenario for Enterprise UC/FMC services or consumer-oriented converged communications services offered by IMS, pre-IMS or legacy service provider networks.
Tags: converged communications, D2 Technologies, mCUE, phone, WiMAX, wimax mobile phone
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Microsoft Response Point SP2 launches + review
At ITEXPO in Miami, Florida, Microsoft has just launched Response Point SP2, the latest version of their IP-PBX. Microsoft sent me a sneak peek of the SP2 beta code in January to check out. I’ve reviewed the Microsoft Response Point system in the past and have been impressed with how easy it is to setup and install and I like the speech-recognition functionality that is built-in. With SP1 Microsoft added some sorely needed features. SP2 further enhances the feature-set by adding after hours receptionist schedules, one-way paging, two-way intercom, and it now supports out-of-band DTMF (the more accurate method) whenever appropriate in addition to its existing in-band DTMF support (more false positives).
Also new in SP2 is that you can now escape from the voicemail greeting before the recording starts. In SP1, a caller would have to hang up and call back in order to attempt to talk to someone else instead of leaving a message. In SP2, an external caller can press ‘0′ while listening to a voicemail greeting, to be routed back into the system or operator (if enabled). However, once the ‘beep’ plays, indicating that recording has started, pressing ‘0′ has no effect.
My favorite new feature is the one-way intercom paging and two-way intercom, which you initiate by pressing the blue Response Point button and then saying “page <person or group>” or “intercom <person>”. Instead of ringing a fellow co-workers phone, you can intercom them which automatically puts their phone into speakerphone mode with 2-way audio. Similarly, you can page (1-way audio) an individual extension. Even better, you can page an entire group (sales, marketing, entire company) for an important announcement.
You could in theory do a 2-way audio intercom page to the entire company, but having 20-50 IP phones go into speakerphone mode with “open mics” (microphones) will obviously lead to screeching audio feedback and some not-to-happy co-workers. I tested this and it appears that Microsoft was keen to the audio feedback issue since they seem to disable any intercom command to more than one phone.
For instance, when I said “intercom entire company” it swapped the command to “page entire company” instead and paged all the phones I added to the “entire company group”. The big difference of paging vs. intercom is that there is only one open mic versus several so you don’t get the audio feedback. Although I commend Microsoft for disabling corporate-wide two-way intercom calls, I think they should permit up to 3-4 phones to be intercom’ed together. This would allow for quick, impromptu conferences with a small group without having to call each extension individually and conference them together. Of course, these phones would have to be fairly far apart from each other to prevent audio feedback.
Alternatively, a user can perform intercom and page dial without using the RP button. Intercom is done by dialing “4*nnn”, and page dial by “5*nnn”, where nnn is the recipient’s extension.
The Assistant is also improved allowing you to double-click a contact to initiate a call. You can also right-click and click Call. Conveniently, it will automatically place your phone into hands-free speakerphone mode and ring the other phone so you don’t have to touch your phone. One suggestion I have for the Assistant application is the ability to Page a person or a group or intercom an extension. You can however call a “group” which will simultaneously ring a group of phones.
In many cases existing features from SP1 have been updated to fix bugs or improve performance. SP2 adds a new After Hours Receptionist feature. When using the “Receptionist Plan” to answer incoming calls, a customer can schedule times when incoming calls should be routed to the human receptionist. Outside of these scheduled times, calls will be routed to the Automated Receptionist.
Analog phones can be connected to Response Point via FXS gateway devices that support the Response Point discovery and provisioning protocol. One such product is made by Quintum. These analog devices also work with SP1.
One minor usability improvement is that the Call Forwarding prompt is now optional. In some cases, users do not want call forwarding to be announced to the callers. Each user can specify whether or not they want their callers to hear an announcement when forwarding calls.
Another addition in SP2 is the ability to launch a custom URL from an incoming call. This lets developers integrate RP into another application, especially a CRM application for instant customer information lookup. Information appended to the URL will include the extension being called, the user at the extension being called, the time, the Caller ID of the incoming caller, and the name associated with that Caller ID.
Another nice improvement in SP2 is Parked Call Return which will automatically return a call to the extension that parked it after 3 minutes. If there is no answer, the call is directed to the auto-attendant or receptionist. SP1 supports both analog lines, and VoIP service through a broadband connection (i.e. SIP trunk).
SP2 adds two other common phone trunk technologies: Digital trunks (T1, etc); and VoIP trunks delivered via an on-premises gateway device. The addition of digital trunks means Direct Inward Dial (DID) support as well.
VPN & multi-subnet support is perhaps one of the more exciting new features in SP2. In SP1, all Response Point end points need to be on the same subnet. In SP2, phones can be on a different subnet from the Base Unit, allowing a user to specify the IP address of the base unit. This will enable remote agents, telecommuters, etc. to use Response Point phones on their home broadband network. All that you need to do is first provision the phone in the office, then take it home, establish a VPN connection to the office, and plug the phone in. Remote phone capabilities was a sorely missing feature in Response Point that SP2 finally addresses.
Conclusion:
Microsoft Response Point SP2 fills in quite a bit of feature-gap that existed before. Response Point continues to be one of the easiest phone systems to setup and maintain and it doesn’t skimp on features. The only noticeable feature missing might be call queues, but Microsoft insists most SMBs don’t require call queues. All-in-all RP SP2 rounds out the feature-set and I look forward to what Microsoft adds in future releases.
Tags: florida, itexpo, miami, microsoft, response point, sp2
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- Aastralink RP Debut Coincides with Microsoft Response Point SP1 release - Jul 08, 2008
- VoIP & Communications Developers - We’ve heard you… - Mar 28, 2008
- Microsoft Response Point SP1 - Mar 24, 2008
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Dual Stack SIP and Skype IP Phone Coming
Image is of the IPEVO Solo Desktop Skype phone not the forthcoming product. But picture something like this device that supports SIP, Skype, and has a color LCD A source has told me about an interesting new VoIP/Skype product that is coming to market in Q3 2009. It’s your typical desktop IP phone supporting the standard SIP stack, however it has an interesting twist - it also supports the Skype protocol. The product is a dual-stack desktop phone supporting both the SIP and Skype protocols. Picture a Cisco, Polycom, or Aastra SIP-based phone that also supports Skype! Essentially, each desktop IP phone becomes a Skype endpoint with the ability to receive Skype calls to the Skype username (i.e. tomkeating) or the SkypeIn number (i.e. 212-555-1000). My first thought when I heard about this product is why would you want Skype installed on the desktop phone endpoints? Why not just use one of the several Skype appliance or software gateways out there (Actiontec/VoSKY, Callfree, SimplyExchange, SkyStone, Skype for Asterisk), which gives you centralized administration of the Skype accounts and can interface with any SIP-based IP-PBX for Skype-based trunking (outbound & inbound dialing). The manufacturer of this device says their phone is more cost-effective than a Skype appliance since it’s simply a software load on an existing IP telephone. Further, you can slowly add additional SIP/Skype-enabled phones as your business grows or as needed. They’ll also sell a SIP-only IP phone that can later be upgraded to also support Skype via a license key. Additionally, he pointed out that many employees telecommute with a remote IP phone, but also use Skype with a PC headset to stay in touch with business associates and friends. Using their IP phone you don’t need to purchase a headset since you can use the IP phone handset instead to answer or make Skype calls. Also, no need to worry about running Skype or missing a call if you reboot your PC. One major consideration is that Skype is very CPU intensive requiring a hefty processor, which obviously adds to the cost. My source tells me that they have done some extensive benchmarks to ensure the voice quality is good and are in the final stages of deciding between two well-known processor chips. He explained the phones can be programmed via its web interface such that a certain prefix goes through SkypeOut, (i.e. ‘8′) while other calls by default are routed to the SIP-based IP-PBX. I mentioned the difficulty of dialing Skype usernames using just a numeric keypad and he stated they are considering adding a full alphanumeric keypad to the phone, but it would increase manufacturing costs. They are strongly considering a “lite” (standard numeric pad) and a “pro” version (full alphanumeric keyboard). Both phone models will have a color LCD but they haven’t finalized the dimensions yet. He said the phone’s web interface will have some rudimentary call history and a contacts database of your Skype buddies and imported contacts. You will be able to click on a contact hyperlink and initiate a Skype or SIP-based call. He claims the IP phone will be price comparable to Polycom and Aastra IP phones. My only thought now is why hasn’t someone thought of a dual-stack SIP Skype desktop IP phone before? Come to think of it, I’ve thought of this idea many times over the years, but figured only tech-savvy VoIP geeks would want such a desktop phone. But with the cost-saving potential such a phone would bring, I wouldn’t be surprised to see a few dual-stack SIP Skype IP phones come to market in the next year.
Tags: aastra, desktop phone, phone, polycom, sip, skype, SkypeIn, SkypeOut, voip
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Verizon Hub Web browsing VoIP Phone Gadget Coming
Back in March 2007 I wrote about Verizon VoiceWing VoIP FiOS service with a brand new phone called Verizon One, sporting a touch-screen, stylus, Internet access, VoIP, and more. It’s pictured here:
Well, according to Boy Genius Report, it appears that a new product called the Verizon Hub deskphone is launching and is even in Verizon stores with a “do not open until further instructions order”. I looked at the Verizon Hub pictures and it looked identical to the Verizon One. One of my sources told me that the Verizon Hub is the same thing as the Verizon One, but apparently Verizon changed the name to the Verizon Hub deskphone. It’s possible it has additional features. Building in a femtocell into the device would be nice, to help extend and improve the range of your cell phone in poor coverage areas. My source wouldn’t confirm nor deny that it has femtocell capabilities.
As I previously wrote about the Verizon One/Hub, unfortunately, the Web browsing experience is tad limited, since you can only get specific Web content that Verizon has enabled. Some of the approved web content includes categories such as news, weather, and movie listings. But if you try and browse a site not listed, you’re out of luck. I even joked, “perhaps Verizon doesn’t want people going to Yahoo! Yellow Pages and bypassing Verizon’s 411 directory assistance at $1.75 a pop! (not sure exact figure). But I’m sure there plan is to find partners and advertisers for the Verizon One device. Still a neat little gadget even if they do “lock” down the browsing experience.”
The Verizon Hub (formerly Verizon One) will handle regular phone calls, but it was designed for VoIP. The cordless phone, which doesn’t look nearly as “cool” as the rest of the device, will work with Verizon’s upcoming VoiceWing VoIP service for FiOS. It also sports a touch screen with stylus and it will require a two-year contract at $29.99/month as seen by this Verizon Hub search.
Tags: femtocell, FiOS, Verizon, Verizon Hub, Verizon One, voip
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Belkin Desktop Internet Phone for Skype
I was doing some Christmas shopping on Amazon when I was curious what sort of VoIP phones were available on Amazon. I came across the Belkin Desktop Internet Phone for Skype, which I never heard of.
The Belkin Desktop Internet Phone for Skype is a standalone Skype phone with no PC running Skype requirement. All that is required is a wired connection to your LAN with Internet access. Because it’s a standalone Skype phone it’s much more suitable as a landline phone replacement in your house than some of the USB-based Skype phones.
The Belkin Desktop Internet Phone for Skype sports a 1.8-inch color LCD display and a phonebook that supports over 500 sets of phone numbers. Importantly, it supports all key Skype features including Skype-to-Skype calling, SkypeIn, SkypeOut, Skype Voicemail, Skype search directory, Skype contact list, Skype Presence, call waiting, conference call (guest only), call history, and user profiles. (Note: The SkypeIn capability is critical if you want to replace your landline phone.)
Belkin espouses the fact that unlike many other Skype phones that use a WiFi connection and require additional SSID or security-key input, the Belkin Desktop Internet Phone is a plug-and-play device that connects automatically to the Internet & Skype at boot up. While no wireless SSID keys to fuss with is certainly a plus, the fact that this phone is completely “wired” without even a wireless handset, may make this a deal breaker for some. The only time I use a “wired” phone is at work and even then I have a DECT 6.0 wireless headset I can use if need be. The phone does have a speakerphone for a hands-free experience, but that doesn’t make up for no roaming around capabilities.
Still, if you want to have a standalone Skype phone and make this a landline replacement for cheaper calls, it certainly has some nice capabilities. Other features include phone-book dialing, mute, redial, hold, and a call-timer display.
Available on Amazon for $79.99
p.s. You should also check out my Philips VoIP841 review, which sports a DECT cordless phone and is also a standalone Skype phone.
Tags: belkin, Belkin Desktop Internet Phone for Skype, landline, phone, skype, voip
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Grandstream GXV3005 Video Phone
Grandstream just announced the GXV3005 IP video phone is coming this month. The GXV3005 features an additional FXO port and is part of the GXV3000 IP video product line series.
The GXV3005 features the same design and all of the functionality as the GXV3000 model, but it comes with the advanced versatility of an FXO PSTN line port, which enables customers to receive both video telephony service over an IP network and their home land-line. One thing I like about Grandsteam is they follow industry standards, such as SIP, but also H.263 & H.264, so in theory their video phone will work with other H.263/H.264 compliant video phones. We don’t need a videophone war, after all.
The GXV3005 is based on the same feature set as the GXV3000, including a 5.6 inch TFTP adjustable LCD screen, a VGA camera which allows nearly all viewing angles, support for real-time high-quality video (up to 30 frames per second) using H.264/H.263 video standard at bandwidths as low as 32kbps and up to 1Mbps. The additional FXO/PSTN port supports calling over and between two networks (IP and PSTN network) from a single phone and
offers both a layer of security and back up for IP video phone communication by providing the option of a PSTN life line.
“We are excited to announce the availability of GXV3005, which further enhances the versatility and convenience of telephony-over-any-network on top of our popular GXV3000 IP video phone,” said David Li, CEO of Grandstream Networks. “The GXV3005 is the second model in the GXV3000 product family and we will continue to introduce more innovative enhancements to the GXV3000 series of advanced IP video phones in the near future.”
The GXV3005 is priced at $325 and it will be available through Grandstream’s worldwide distribution channels beginning mid December 2008.
Tags: Grandstream, GXV3000, GXV3005, H.263, H.264, video conferencing, video phone, voip
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Ditch Desktop IP phone to Save Money?
With a slowing global economy, one of the cost-cutting measures desired by IT personnel appears to be the desktop phone, including desktop IP phones. That at least according to an OnRelay survey that shows 88% of cost-conscious IT Professionals would ditch the desktop phone. Apparently, the desire for for mobile working makes desktop tools a low priority expense. I should point out that OnRelay offers a hosted PBX solution with mobile phone only extensions, so you might think this survey has a conflict of interest. However, OnRelay’s solutions can work with existing PBXs and desktop phones, so there isn’t necessarily a conflict.
In any event, today OnRelay published results from a survey revealing the personal business tools IT professionals couldn’t be without, and the equipment they would give up as corporate budgets tighten. Their answers show a clear preference for the mobile office, and question corporate spend on technologies that those in IT have already left behind.
OnRelay surveyed 330 IT professionals in UK enterprise. Reflecting today’s tougher economic choices, they were asked to select the four office IT tools they see as essential - out of a list of nine items.
Those surveyed were asked to choose from: the laptop; desktop PC; virtual private network (VPN) access; mobile phone; Blackberry™, desk phone; video conferencing; or their own personal desk.
The top three items chosen by the IT professionals were all mobile office technologies. The laptop came in as the most essential item - making the must-have list of 88% of respondents. The mobile phone came second at 74% of respondents, followed by VPN access (69%).
The least chosen items emphasise the low priority end-users place on fixed desktop equipment. The office desk phone was the tool those questioned were most willing to give up, with only 18% listing the desk phone as essential. This is akin to the 20% who prioritised a desktop PC, and the 22% who chose video conferencing as a must-have.
OnRelay’s survey provides insight into IT professionals’ disengagement with desktop tools. Although the majority of those questioned (87%) currently had an office desk phone, if given the choice, 88% said they would choose the mobile as their one business phone.
The underuse of the desk phone is already apparent from the fact that only 3% of those surveyed said they forwarded their calls to mobile when away from their desk. A full 54% of IT professionals admitted to never forwarding their desk phone calls, whilst only 10% forward their calls when away from the office.
“This survey shows that IT decision makers want wireless, use wireless, and will lead the drive to cut the cord as budgets tighten,” comments OnRelay, CFO Marie Wold. “Costly desk phones will rapidly be phased out of IT budgets to be replaced by mobile-only telephony. The current economic climate is accelerating this shift towards mobile integration in the corporate.”
We do indeed live in a mobile world, but do I see the desktop phone going the way of the dinosaur anytime soon? Not likely. There are just too many advantages to a desktop phone, including a larger LCD, higher quality speakerphone, more feature buttons & speed dials, and more. Still, it was an interesting survey worth sharing.
Tags: blackberry, economy, ip phone, mobile phone, OnRelay, OnRelay Hosted MBX, voip, wireless
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- TalkPlus Voice 2.0 app launches - Oct 30, 2006
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AudioCodes enters IP Phone arena with 300HD Series
AudioCodes has now entered the IP phone arena with a phone that supports wideband codecs (HD) for superior sound quality. Seems a bit odd for a company that makes VoIP PCI and cPCI communication boards and VoIP media gateway modules (PMC form factor) and Analog Media Gateways (2/4/8/24 ports) to be entering the crowded VoIP arena, but enterprise IP phone market is expected to quadruple, from $2.1 billion in 2007 to $8.4 billion in 2001, with some 63 million endpoints being shipped by that time, according the Synergy Research Group. So there is a pretty big pie to go after. Traditional network hardware vendor Adtran has also recently entered the IP phone arena with their Adtran IP700 series (see Adtran IP706 review).
The AudioCodes 300HD Series includes three models: The 310 HD entry level phone with a basic display and interface; the 320HD premium endpoint with a larger screen, and the 350HD executive phone with a color LCD screen. All three models are based on AudioCodes newly announced VoIPerfect software, and include the most commonly used wideband codecs. Of course they are SIP-based so they should work on standards-based IP-PBXs such as Asterisk and these phones are also PoE (Power over Ethernet) compliant.
For more details, check out TMCnet reporter Erik Linask’s article. One interesting take Erik has is when he writes, “First and foremost, it owns the DSPs that enable the higher quality, which means it can provide its HD VOIP-enabled handsets at a price point comparable to other high-end non-HD devices”.
Now I just have to get my hands on on to review. Stay tuned…
Tags: 300HD, 320HD, AudioCodes, Erik Linask, HD, ip phone, PoE, voip, wideband codec
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Nov 04, 2008
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FCC Doing Heavy Lifting
The FCC is holding a meeting on Nov. 4. On the agenda: Inter-Carrier Compensation, Alltel-VZ merger, Clearwire-Sprint merger, and a vote of White Spaces. Lots of heavy lifting on this agenda. Martin wants to give his pals at VZ one more gift before he goes.
The VZ-Alltel merger is big, but the topic that can really rock telecom is the Inter-carrier Comp issue, which has been a stagnant FCC docket for years.
If companies can show high costs, they will continue to benefit from the subsidy program. Martin also wants to eliminate wireless providers’ right to claim government subsidies for offering service in hard-to-reach areas. Martin wants all companies, wireless included, to show they have incurred losses in providing rural service before they can collect the subsidy. Without those changes, Martin worries that the subsidy fund will collapse of its own weight and rates will go up anyhow. [CNN]
It depends want the Compromise looks like — and it will be a large compromise. Democrats want one thing. Republicans another. Cellcos versus Wireline. Rural versus Urban. Inter-Carrier Comp even bleeds into the USF issue. How? Because rural carriers count on both Universal Service Fund subsidies AND rather high call termination charges to keep afloat.
Why now? The ISP inter-carrier comp rule has been in court for six years. Earlier this year, the DC Court ruled that the FCC had to get off the pot:
The court set the deadline for an order from the FCC at November 5, 2008, six months from the date of oral argument, stated it will not grant an extension and warned that if an appropriate order is not timely issued, it will vacate the interim inter-carrier compensation rules.
Consumer groups are against another largess for the monopolies at the expense of the ratepayers.
The head of the Federal Communications Commission wants a massive overhaul of the fees that phone companies pay each other when they connect calls. Supporters say the reforms will help fund improved broadband Internet access for rural America, but consumer advocates question how much the plan will raise people’s phone bills. “This could be potentially a billion-dollar giveaway to phone monopolies, paid for out of consumers’ pocketbooks,” said Chris Murray, an attorney with Consumers Union. [AP]
Intercarrier comp is how the various phone companies pay each other for traffic. VoIP providers and cellular carriers, especially Sprint, would like a fairer shake. The old RBOCs would like the Rural LEC’s to stop getting so much money. (see Free Conference services not getting paid by RBOCs).
The National Telecommunications Cooperative Association, which represents small phone carriers, told FCC officials earlier this month that a new rate of $0.0007 per minute puts many of their members’ livelihoods at risk.
And then there is the White Spaces issue. When broadcasters make the DTV transition in 1Q09, there will be unused spectrum that the Wireless World would like to use for its own bandwidth needs. However, due to bleed over (interference) with cordless microphones and other broadcasting devices, the NAB is opposed. [see dailywireless]
All of this is at one meeting while America votes.
Tags: fcc, inter-carrier compensation, mergers, white spaces
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Tags: fcc, inter-carrier compensation, white spaces, mergers
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Aastra 57i and 57i CT Review
The Aastra 57i is one of my favorite VoIP phones. The 57i and its sister, the 57i CT (cordless telephone adjunct), offers some unique features and is undoubtedly one of the most flexible IP phones you will find. The 57i and 57i CT sport a large 144 x 128 pixel graphical backlit LCD display and 6 dynamic context-sensitive softkeys. Although the resolution isn’t designed for photos, it’s a very large LCD, one of the largest I’ve seen making it very easy to read the number of voicemail messages, the CallerID of an inbound call, and the one touch feature keys you’ve programmed. The 57i is of course SIP-based making it fully interoperable with IP-PBX platforms such as Asterisk. The 57i and 57i CT offer advanced XML capability to access custom applications and support for up to 9 calls simultaneously.
57i CT Web Interface Preferences
The 57i CT is exactly the same as the 57i except it has a built-in wireless transmitter in the base unit and it comes with an integrated WDCT cordless handset with a range of up to 300,000 sq ft. The cordless phone sports 10 previous number redials, a mute button, on hold, Callers List, transfer,4 ringtones, and more. My only complaint is that there isn’t a dedicated transfer button. While on a call using the wireless handset, you have to press the F (Function) button, scroll to Xfer, and then enter the extension number. The mobility the cordless handset gives you is perfectly suited for executives, mobile warehouse personnel, as well as retail staff. Here’s a photo of the 57i CT on my desk along with the cordless handset:
The Aastra 57i has excellent XML browser capabilities. You can for instance install an Aastra developed and freely available XML application that gives you Visual Voicemail on trixbox CE, an Asterisk-based IP-PBX. This allows you to scroll through your messages on the phone, see the CallerID, and pick the one you want to play. Here’s a screenshot of visual voicemail in action:
Another freely available app is meet-me conference with the ability to see attendees along with the ability to kick or mute participants from the LCD. Still another XML app lets you play .mp3 files using the LCD and buttons and it can even read the ID3 tags in the MP3 files to browse by album, artists, or entire song list, as seen here:
One useful business app is a SugarCRM XML app that lets you look up names and numbers in your SugarCRM database simply by typing the name into the LCD using the numeric keypad. After fining the proper record you can instantly dial the contact. 
Other freely available apps include Area code lookup, Ask Google, CNN News, ESPN News, Horoscope, Local Weather, Netflix, Movies, Stock, World Clock, and Yahtzee. Here are some screenshots of many of these free XML applications.
The 57i supports directories, including a shared corporate-wide directory, which is easily created simply by creating a CSV file called directory.csv file and uploading it to the TFTP server. Then you edit the aastra.cfg file on the TFTP server and add:
directory 1: directory.csv
When you next reboot the phone, the corporate directory is automaticly downloaded.
The phone also supports callers logs, a redial with up to the last 100 phone numbers, and 12 programmable keys located next to a large LCD. The 6 top keys are static softkeys with up to 10 programmable functions. The 6 bottom keys are state-based softkeys with up to 20 programmable functions. All top and bottom keys can be configured for specific functions, such as Intercom, Callers List, speed dials, etc. When you program one of the keys, it displays its function within the LCD. For instance, if you look at my phone you’ll see I have China King set as one of my speed dials. One press and I can instantly call my favorite Chinese restaurant!
The 57i phones support shared call and bridged line appearances (SCA & BLA). In fact, Aastra’s line of IP phones do a good job of emulating key system functionality. It fully supports BLF (busy lamp fields) allowing you to see when a person/extension is on the phone. From the web interface you simply pick BLF, enter the extension and the name of the person. When the monitored user is idle, there is a small telephone icon shown with the handset on-hook. When the monitored user is on an active call then a small telephone icon is shown with the handset off-hook.
The BLF also acts a one-click speed dial, allowing you to simply click a BLF and instantly ring that extension. Another BLF capability is the BLF List feature. The BLF List feature on the Aastra IP phones is specifically designed to support the BroadSoft Broadworks Rel 13 Busy Lamp Field feature simply by entering the BLF List URI. The BLF List URI is the name of the BLF list defined on the BroadSoft BroadWorks Busy Lamp field page for your particular user. For example, my480i-blf-list@as.broadworks.com.
57i CT Web Interface Preferences
Some other features of note. First, in addition to shared call and bridged line appearances, typical features such as call forward, call transfer, call waiting, intercom and local 3-way conference are available. Second, the Aastra 57i and 57i CT support stuttered tone letting you know that you have new voicemail waiting. Third, the IP phones support several different languages. You can have the IP Phone UI and the Aastra Web UI display in English, French, Spanish, German, and Italian.
57i CT Web Interface Preferences
It’s also worth mentioning that the Aastra 57i offers two different module extension options (images to right). The 57i supports up to three Aastra 536M modules, each offering 36 keys with LED indicators to create a feature rich attendant console. Or, it will also support up to three of the advanced Aastra 560M modules, each offering 60 keys with a screen based LCD display and LED system.
The 57i Aastra IP telephones have a full-duplex speaker phone with excellent voice quality. I’ve used an Aastra phone’s speaker phone for a long time and have never had any complaints. The phones sport dual auto-sensing switched Ethernet ports along with integrated IEEE 802.3af Power-over-Ethernet.
Configuring Aastra phones are done via configuration files which can then be uploaded to a TFTP server. The syntax of Aastra config files are very straightforward. However, due to the flexibility of Aastra phones and the amount of features, it can get a bit complicated. In fact, Aastra phones are so flexible and configurable, that someone even built an Aastra Configuration Generator, leveraging Microsoft .NET to make it easer.
The stated goal is: “This utility will hopefully take some of the pain out of creating the configuration files for the Aastra range of IP phones.” The website adds, “With it you can create the base config, the buttons (programmable, top softkey and softkey) config, the console button config, the IP phone directory and encrypt the file prior to uploading it. You can even link in with NMap to perform network browsing for Aastra MAC addresses. You should be able to quickly and easily create phone configurations without having a vast knowledge of the options required. I have tried, where possible, to display tooltips for options and restrict the choice of input to what the option expects.”
Conclusion
The Aastra 57i and 57i CT are two of the most flexible VoIP phones you will find on the market today. The large LCD and plethora of configurable buttons make this phone a must have for executives and other enterprise users that require maximum flexibility for increased productivity. My only suggestion to Aastra would be to offer a color LCD version with a higher resolution for displaying Internet graphics pulled via RSS using an XML application. Perhaps the next series, most likely called the Aastra 67i series, will have color? 
I love the Aastra 57i and the Aastra 57i CT and in fact have a 57i CT as my primary desk phone. What more can I say? Two enthusiastic thumbs up!
Price: 57i: $219.95 57i CT: $319.95
57i and 57i CT Resources:
57i
Aastra XML Scripts for trixbox CE (PDF)
57i Installation Guide (PDF)
57i User Guide v2.1 (PDF)
Latest firmware downloads
57i CT
Aastra XML Scripts for trixbox CE (PDF)
57i CT Installation Guide (PDF)
57i CT User Guide v2.1 (PDF)
Latest firmware downloads
Tags: 57i, 57i CT, Aastra, IP phone, phone, review, voip
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Fixed Mobile Convergence (FMC) Leads to Workforce Productivity and Customer Service Improvements
Fixed Mobile Convergence (FMC) enables users to use a single phone handset that connects to both cellular and corporate wireless networks thus enabling users to carry a single phone device to communicate with business associates, family, and friends, no matter where they are. The obvious benefit besides the cost savings of using a single phone device is employee productivity. A new study by OnRelay’s own live deployment data reveals when FMC technology works, user penetration and hardware cost savings follow along with increased customer responsiveness since the employee is able to take inbound business calls anywhere. FMC also enables advanced PBX functionality such as call transfer, call conferencing, on-hold, etc.
Here’s the interesting report info put out by OnRelay:
Tags: Fixed Mobile Convergence, FMC, pbx
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No ♥ for High-end Mobile Apps
Thought I’d share this interesting report from Accenture about mobile phone app utilization that claims higher-end applications remain vastly under-used by U.S. consumers. Funny, I think Apple iPhone users might disagree. The report found that 88 percent of U.S. consumers said they never use their mobile phones or other mobile devices to watch videos. That is high, but expected since most mobile phones can’t even play videos. My main phone, a Windows Mobile 6.1 smart phone doesn’t even natively support Youtube Flash videos using Pocket Internet Explorer. I did install Skyfire though, which adds pretty respectable video performance. I was able to watch Youtube videos on my Windows Mobile with pretty decent video quality, but of course I’m not the ‘typical’ mobile phone user.
The report adds that 84 percent said they never use their mobile phones or mobile devices to send email. Say what? Email is only 4 pts better than video? I would have expected email penetration to be much better, especially with all the Blackberry phones. People want email access on their phone more than any other app, but maybe that’s just my opinion. Further, the report says 79 percent said they never employ them to play games on the go. So more people played games on their phone than sent email? You’ve got to be kidding me. How is that possible? I suppose many phones come with cheezy built-in games, which might skew the results.
Anyway, read the report for yourself…
Continue reading No ♥ for High-end Mobile Apps…
Tags: accenture, Apple, consumers, iphone, mobile applications, mobile apps, mobile phone, skyfire, U.S.
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snom 820 VoIP phone
snom today launched the snom 820, an elegant and attractive VoIP phone. It features a large high-resolution TFT color display (3.5″, 320 x 240 pixels) and can easily display call lists, phone directories, and caller information via the integrated XML browser. I believe this is snom’s first “white” VoIP phone, which gives it a nice clean look. Very Apple-esque. It appears to support “picture / photo” CallerID but I need to confirm. The snom 820 also supports wideband in the handset which captures more than double the spectrum of voice frequencies that by standard phones.
Perhaps the most interesting feature is that the new snom 820 supports WiFi. No need to use a network cable if you don’t want to. In fact, the snom 820 supports Power over plug (A/C adapter), Power over Ethernet (PoE) or WiFi.
Security-wise it features secure VPN, TLS and SRTP. Important business features include three-party conferencing, polyphonic ringtones, and up to twelve different SIP identities.
According to snom, the new look of the snom 820 was designed in cooperation with very renowned German industrial designers. They stated, “this first model of the new product line of snom phones is living up to the highest expectations in functionality and timeless beauty.”
Indeed, that is one beautiful phone. Can I have one?
Tags: ip phone, poe, snom, snom 820, voip, voip phone, wifi
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Differentiation Part II
When I say that word - Differentiation - in a space like VoIP with 1100 providers, what does it mean?
Marketing is about stories. Not stories about your boring tech, but stories about what the consumer can do with your tech. For the most part, the technology works, but why would anyone use it?
Blackberries were the first prolific smartphone for business users. Execs thumbing away at the airport after listening to voicemails. A long comes Voicemail to email and Bingo no more typing, just forward the message with the attached wav file and delegate.
Tell me a story about how tele-workers or virtual offices can be managed remotely with call monitoring, call recording, and ACD stats with a Hosted PBX solution. Now I have a case study and I have something concrete to wrap my head around when talking to prospects. As a telecom sales agent, I need solutions not features. I need to know how any of the 130+ features of Broadsoft can result in productivity or TCO or ROI for a prospect. I need a case study. A story. Something specific. Like how you can monitor remote workers to see if they are making the calls and doing their job.
When you tell me that I can add a second business line to my cell phone for $5 a month, well, now I have a story to tell the small business owner. The light bulb has gone off in my head.
Taking a call on any phone I am standing near, being able to transfer my cell phone call to any landline, now that is a specific message I can carry to the marketplace. It’s a cool tool for contractors who work for different employers. Now you have delivered a real follow-me number.
We are approaching the point when a phone number will call a person, not a location, desktop or home. That’s what Presence is all about, right? Stop playing phone tag and get some work done.
This is differentiation. This is also marketing. Creating a story to spread. Designing a clear, concise message about the benefit of the tech feature. (By the way, this all came about from a call I was having with Eric Thomas, CEO of FreedomVoice and Newber.)
Tags: differentiation, marketing, newber, voip
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Jul 31, 2008
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snom m3 review
The snom m3 SIP wireless (DECT) phone is one of my favorite VoIP phones. I’ve been testing and reviewing it for a few months but haven’t had time to write up the review until now. First, let me point out that the problem with IP-PBXs is they typically give you a desk phone or a softphone with no real mobility options to walk around, which is critical in some vertical markets, such as retail and manufacturing. Even sales professionals want the flexibility to take calls while roaming the office. In the past, I have used analog telephony adapters to connect my cordless phone to my SIP-based IP-PBX, but the cordless phone lacks multiple lines, call transfer, call conference, call waiting, or even a message waiting indication (MWI). Enter the snom m3, a SIP wireless phone that like a home cordless phone which not only gives you mobility while on the phone, but full IP-PBX functionality as well, including call hold, call transfer, message waiting indicator, and more. In fact, while the caller is holding, music-on-hold is available from the IP-PBX, giving the same business professional experience from a desktop phone.
I should mention that there are WiFi SIP phones, but the battery life on these phones isn’t great. snom takes advantage of Digital Enhanced Cordless Telecommunications (DECT), a wireless communication standard which can seamlessly hand off calls as a handset moves between multiple base stations in a large office, but also has superior battery life than WiFi SIP phones. The Lithium Ion battery offers a very good eight hours of talk time and 100 hours of standby. Additionally, DECT devices use the 1.9 GHz band while WiFi uses 2.4Ghz so they don’t interfere with one another. DECT also doesn’t suffer the microwave oven interference that often plagues WiFi access points.
snom m3 Main Menu
The snom m3 supports up to 8 different SIP identities (registrations) allowing you to connect to separate IP-PBXs (or SIP service providers) or the same IP-PBX to support multiple lines. The m3 is 2″ x 5″ and less than an inch thick sporting a nice 1.75″ color LCD (128×128 pixels and 65,536 colors), 2.5mm headset jack, and a speakerphone. The headset jack is a nice feature that I haven’t seen on any cordless DECT phones. The phone also comes with a belt clip so you can easily use the headset for talking while walking. The m3 is surprisingly very lightweight - much lighter than I would have expected. The phone also has volume controls, the basic 12 dialpad keys, five navigation keys, and two function keys. The snom m3 ships with some documentation, but for real technical details, the snom m3 wiki is the place to go.
The m3 communicates with the base station which is connected directly to your network via a standard Ethernet cable. Once connected and booted up, the base station obtains an IP address from the DHCP server. By default (factory setting), snom m3 phones are configured to use HTTP as the transfer protocol for provisioning, but TFTP can also be used. Since I was testing this with an Asterisk-based trixbox system, I changed the gateway to use TFTP. Also, the snom m3 supports Option 66 on the DHCP server to automatically acquire the IP address of the TFTP server. Nice!
The TFTP boot server address can be an IP address, a fully qualified domain name (FQDN), or an URL. I also created a config file (/tftpboot/m3/settings/0004132A10E4.cfg) on the TFTP server for the snom m3 to download. I was able to get access to the firmware, upload the new firmware to /tftpboot/m3/firmware/ and it automatically downloaded the latest firmware. Even better you can have it set to connect directly with snom’s server (http://provisioning.snom.com/m3/firmware/) to download the latest firmware and even set a schedule to automatically grab the latest version.
Features:
- Display: 128 x 128 pixels, 65536 colors, backlit
- Li-Ion battery pack for 20 hours of calls or 100 hours standby
- Range: 50 meters indoors, 100 meters outdoors
- 12 numerical keys, 5 navigation keys, 2 function keys
- Speakerphone on mobile handset
- Polyphonic ringtones
- Automatic registration of handset
- Separate charging cradle for handset
- 8 handsets per base station
- 8 SIP registrations with different servers/registrars
- Up to 3 concurrent calls per base station
- Three-way conference
- Remote setup, password protection
- Open DECT GAP standard
Since the snom m3 supports multiple handsets, this leads to some interesting multi-handset functionality. For instance, the Telephony Settings on the web interface lets you pick which identity (CallerID) each handset will use when making outbound calls. You can also set which handsets will ring on incoming calls for each SIP registration/phone number. Thus, you can have one SIP registration ring your home office m3 handset, another ring your son/daughter’s m3 handset, and another phone number be the shared kitchen m3 phone. In fact, the snom m3 supports three concurrent calls per base station so you can receive 3 simultaneous calls to the handsets.
The snom m3 supports the most common VoIP codecs, including G.711u (PCMU), G.711a (PCMA), G.729ab, and iLBC. G.711 is the standard used by traditional phone systems and it features the best voice quality at the expense of more bandwidth used (80kbs), which isn’t ideal for some DSL connections that only sport 256kbs upstream. Fortunately, the snom m3 supports G.729a which only use 8kbps at a slight loss of voice quality. iLBC (Internet Low Bitrate Codec), although not as widely supported, is designed for narrow band speech and supports two bit rates, 15Kbps (20ms frame rate) and 13.3 Kbps(30ms frame rate), though the m3 only supports 20ms frame rather @15Kbps. iLBC yields slightly better voice quality than G.729a yet also has a higher robustness in dealing with packet loss while using roughly the same amount of bandwidth. It also has a more dynamic range of sound than G.729a. So kudos to snom for including iLBC as a choice.
You can also configure various settings from the phone itself, though it’s more tedious. The VoIP settings is protected by a PIN / password which defaults to 0000. From the phone you can configure the timezone and it even supports NTP time servers for accurate time. Additionally, you can add contacts, however adding contacts via the phone is a bit tedious. I wished the web interface let me add them there and then it would push the contacts down to the multiple handsets.
So how’s the phone’s range? snom claims the phone needs to be within 50 meters indoors or 100 meters outdoors from the base station. I walked around TMC’s offices and didn’t lose a signal. Then I went outside walked about 250 feet and it was crystal clear. Excellent range I have to say. The voice quality of the earpiece was very good and the remote end said I sounded very good during my test calls. I also tested the speakerphone, and although it wasn’t the best voice quality, I didn’t expect a fantastic sounding speakerphone on such a small handset. I should mention that you can also perform intercom calls to either a single m3 handset or you can intercom page all handsets. Useful if you are trying to reach someone and don’t know where they are located.
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All in all, the snom m3 is an excellent wireless VoIP phone with excellent battery life, very good range, and very good features. The multiple simultaneous SIP registrations is a huge plus. I wished the base station supported PoE, but it’s not a big deal for home users since most home users don’t have Power over Ethernet switches. I’ll be interested to compare the snom m3 with the new line of Polycom KIRK wireless DECT SIP phones, but for now the snom m3 is my favorite cordless SIP-based VoIP phone!
Price:
You can buy the snom complete set (with base + handset) <a href=”http://www.amazon.com/SNOM-Technology-snomm3CompleteSet-Snom-Complete/dp/B0013F6IJI%3FSubscriptionId%3D151BWK97V0S8BGYJ8F02%26tag%3Dtechstuff01-20%26linkCode%3Dxm2%26camp%3D2025%26creative%3D165953%26creativeASIN%3DB0013F6IJI” title=”Buy now at amazon.com-only !” onmouseover=”return overlib(’Click for Amazon price:
Snom M3 Complete Set
Buy Now‘, STICKY, TIMEOUT, 6000);” onmouseout=”return nd();”>on Amazon for $172
, and an additional <a href=”http://www.amazon.com/SNOM-Technology-snomm3EnhancementSet-Snom-Enhancement/dp/B0013F6IKC%3FSubscriptionId%3D151BWK97V0S8BGYJ8F02%26tag%3Dtechstuff01-20%26linkCode%3Dxm2%26camp%3D2025%26creative%3D165953%26creativeASIN%3DB0013F6IKC” title=”Buy now at amazon.com-only !” onmouseover=”return overlib(’Click for Amazon price:
Snom M3 Enhancement Set
Buy Now‘, STICKY, TIMEOUT, 6000);” onmouseout=”return nd();”>handset on Amazon for $142.
Tags: DECT, review, SIP, snom m3, VoIP, wifi
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Jul 11, 2008
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Cellular Numbers
Broadband Reports discusses NPD Group data about iPhone conversion. “Nearly half of new AT&T iPhone customers switched from Verizon Wireless, another 24 percent switched from T-Mobile, and 19 percent switched from Sprint. iPhone smartphone share jumped from 11% to 17% of the market with the 3G’s launch.”
T-Mobile can probably keep some folks from switching with their sold out Android G-phone, but what has Sprint got? Actually, they have the Blackbeery Bold, Curve and Palm Centro - the other best selling phones in the summer according to NPD Group.
Other cellular news comes from Gary Kim on his IP Business blog:
More than one fourth of wireless phone customers have replaced their traditional landline connections at home and are now using wireless service exclusively to communicate on a daily basis, according to J.D. Power and Associates. … The study finds that among the 27 percent of current wireless customers who report replacing their traditional landline phone with wireless service, 61 percent have completely disconnected their home landline service.
Tags: cellular
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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
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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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Oct 10, 2006
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Landlines Going the Way of the Buggy Whip …
According to a report released last week by Nielsen, nearly one in five American households will not have a home phone line by the end of the year.
To many, I have a feeling that this is a surprising finding, but it shouldn’t be.
Landline home telephones are quickly becoming an unnecessary expense. What can one do on a landline phone that they can’t do with their cell phone?
It’s a little bit like that essential tool of the horse-drawn carriage age — the buggy whip. It used to be a big business, but don’t think you will find too many of them today.
Now what is going to happen to all of those telephone poles and wires some day when everything goes wireless? It will be good that we won’t be foresting new trees to turn into new poles, but what will become of all of the ones left standing? Will they become a ghost town-like image — although one seeming to have no beginning and no end — of a time gone by?
Good food for thought inspired by and article in Insurance & Technology.
Tags: cell phone, landline, nielsen
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Jan 31, 2008
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Sony’s Bluetooth Watch: Calling Dick Tracy!
Can’t get to the ringing cell phone fast enough ladies because it’s hidden inside your briefcase, purse, bag or pocket?
Well, does Sony have a product for you!
The new Bluetooth MBW-200 range of Bluetooth watches is an expansion of the highly successful Bluetooth MBW-150 range, which enables you to control your phone with your watch.
Missed an important call or SMS because you couldn’t hear or find your phone at the bottom of your handbag? With the MBW-200 this is now a thing of the past. Using innovative Bluetooth technology, the watch displays the caller ID and vibrates as your phone is ringing or when you receive an SMS.
Reject or mute the call with a button on the watch or use the Bluetooth headset or mobile phone to answer the call and start chatting.
Designed in partnership with watch industry leader Fossil and the knowhow of Sony Ericsson, the MBW-200 series comes in three distinct designs; Sparkling Allure, Contemporary Elegance and Evening Classic.
The collection features scratch-resistant mineral crystal glass with an anti-glare coating for optimum readability and glow. The discreet, yet stylish and clear monochrome OLED display is invisible when not in use and clearly visible on a bright day so that you can easily see who is calling or which track is playing.
Features include:
- Reject or mute call through the watch
- Vibrates on new SMS/MMS — and an out of range warning — when 30 feet away from your phone
- Discreet yet clear caller ID — OLED display
- Play, pause, stop or skip a track on your phone’s music player
- Quartz movements for exact time keeping — both digital and analog
- Auto pairing — easier to connect the watch to your phone
- Water resistant up to 3 ATM
- Allergy safe stainless steel
- Mineral crystal scratch resistant face with antiglare coating
Availability in early Q4 2008.
Tags: bluetooth, cell phone, fossil, mbw-200, sony, sony ericsson, wrist watch
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Aug 28, 2008
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Cisco is Jabbering
In 2007, Cisco integrated Jabber components into its conferencing platform. Today, Cisco buys Jabber, “an open-source IM and presence protocol used by Google Talk and Gizmo, for an undisclosed sum”. On our panel at IT Expo, The Role of Apps in VoIP, we talked about Gen Y not liking to talk on the phone. My conclusion is that you will need to incorporate XMPP and XML to enable chat, instant messaging, SMS messaging to IP phone - all to communicate with employees, customers, vendors - without talking on a phone.
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Jun 27, 2008
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Phonevite API released at ITEXPO
At ITEXPO I met with John Nahm, CEO and co-founder of Phonevite, an intriguing VoIP-enabled one-to-many broadcasting service that takes the concept of the popular Evite email-centric invitations service and extends it to the phone world.
Calvin Kim is the other co-founder and he actually founded Dialpad, one of the pioneers of VoIP with their Java-based click-to-call application. Dialpad was later sold to Yahoo. Calvin is now the CTO of Phonevite.
John explained he wants Phonevite to be like Evite but for the phone. Phonevite was incorporated in January of last year and has reached profitability with their premium customers. They offer a free service with the only limitation that you can only blast a maximum of 25 phone numbers/contacts.
John explained that the largest users of his service includes lots of schools, Boy Scouts, soccer / little leagues, emergency rescue teams, and emergency management teams. The most important feature of Phonevite is that it trumps email when it is a time-critical matter and users don’t check email regularly. Or even if they do check email, John explained what if each of the soccer team’s parents are on the way to practice, it starts raining and the coach wants to cancel. They will most likely not be email accessible while driving. Using Phonevite you can blast the entire team’s mobile phone list announcing that practice has been canceled.
John pointed out that SMS is an alternative, but it has a limit to how many people you can SMS and it is difficult to describe situation in the 150 character SMS limit. The premium service offers a tiered bonus structure for tiered VoIP calling rates. Also, schools and non-profits get a discount.
At ITEXPO Phonevite is announcing their API. This will enable websites such as
freeconferencing.com, Evite, socializr, pinger, etc. to directly tie into Phonevite’s service.
Interestingly, the back-end doesn’t use Asterisk, the open-source PBX platform. John told me they developer the back-end VoIP calling interface themselves. They do leverage other open source software though, such as MySQL.
Tags: API, Asterisk, Calvin Kim, ITEXPO, John Nahm, mobile phone, Phonevite, SMS, VoIP
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Newber
FreedomVoice unvieled Newber at CTIA. It is the beta version of the first location-aware business number. Newber Beta is an application that resides in the iPhone as a fully functional second line and uses positioning technology to locate and seamlessly transfer calls to nearby landlines, even during an in-progress call.
Newber Beta delivers an independent number that can be assigned to any phone, sparing the caller the task of dialing multiple numbers for mobile, home, work, etc. Newber Beta also allows a person to take business calls on their private phone without giving out personal information.
A highly anticipated capability of the commercial release is “Contact Finder”. A Newber user will be able to simply tap a name on the contact list and all of that contact’s ‘numbers will be automatically dialed in sequence. Manually dialing one ‘number after another will be a thing of the past.
“Plans are in the works to introduce Newber for other mobile smart phones,” said Eric Thomas, CEO of FreedomVoice. “Newber is making business calls simple again.”
“The Newber application adds a second business line to your iPhone that enables you to redirect incoming calls to any phone using built-in GPS technology. Simply key in a landline phone number at your location, then toggle between taking your business calls on that phone or your iPhone. Newber will automatically detect this phone each time you return to that location, allowing you to change phones with one touch. You can even swap phones in the middle of a live call without interrupting the conversation.” [from the Newber website]
IntoMobile has the above video with a good description as well as a demo of Newber. Newber’s functionality is similar to the Broadsoft Anywhere application.
Disclosure: FreedomeVoice is a consulting client of mine. Congrats! to Eric & Company. And only yesterday I was complaining about all the iPhone news.
Tags: anywhere, freedomvoice, iphone, newber, voip
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Tele-Presence versus Video Conferencing
Andy Abramson writes about how video conferencing from a client company like SightSpeed is better than Cisco’s Tele-Presence. On Sept. 22, Brian Carroll is having a tele-seminar on “Email vs. Phone vs. In-Person Meeting”. Keith Rosen, a respected sales coach, would tell you that, especially in today’s economic climate, nothing beats a face-to-face or a phone call. Keith writes about how Sales 2.0 is diluting inter-personal communications. (If you have ever received an email from someone that using text messaging a lot, you will see what he means).
As one commenter wrote to Andy, Tele-Presence is a richer experience. Well, it should be for equipment that is in excess of $50K per site. Even this price barrier is being removed by companies like WBS Connect leasing out tele-presence rooms for business use.
Another point is that many tele-workers don’t want to shave and dress for video conferencing. There’s no IT guy at the home office to help with the video conference set-up. Messing with the webcam and the software is a pain and when you have so much to do in 8 hours, dialing a conference bridge is easy. Even that isn’t fool-proof as quite a few times stuff has happened to hamper that easy tool.
Maybe the differentiating factor will be personal interaction. When I see couples at restaurants and one or both are on the cell phone, I have to wonder what are they doing. That’s a live person that traveled to meet you. Talk to that person. And the yakking in the car while driving WITH other people in the car. This society is in for a wake-up call and it will come when most of us are too weak to fight back.
You can’t get a human on the phone but ChaCha pays humans to run your Google search while you are mobile. Go figure. Maybe the differentiator to a sale will be the personal touch.
Is travel is mess? Yes. Will virtual conferences replace real ones like IT Expo West next week? No. It’s about the hand-shake; the meals, drinks and ideas shared. It’s about getting a read from someone live. That is not easily replaced. That said, tele-presence as a 3-D meeting experience might work for second or third meetings to go over details or brainstorm or for product demos or training.
Video conferencing like SightSpeed and tokbox can replace phone calls (if we can figure out how to make them on demand like a phone call instead of scheduled). And I need to start replacing some IM and email with the phone. I pay for a landline, a VoIP line, and a cell phone. I need to use it more and the keyboard less.
Tags: cisco, sales, sightspeed, tele-presence, video conferencing
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Cops Befuddled by Vonage 911 Crying baby emergency call
According to the Seminole Chronicle, police were befuddled when they responded to a 911 call with a crying baby on the open phone line, only to discover they were at the wrong house. Apparently, the owner is a Vonage customer and moved without notifying Vonage of the change of address to update the E911 records.
The article explains:
Kelly informed Sanford police of the situation. Upon arrival at the family’s home, officers confronted a rather confused and embarrassed father who admitted his son had been playing with his cell phone.They had moved, the father said, and had not bothered to update their phone carrier, Vonage, with their new address.
What I don’t get is how the baby dialed 911 using a cell phone that is tied to Vonage’s service. Since when did Vonage start offering cell phone service? If 911 was dialed from a cell phone, the tracing of the call should be performed from the wireless carrier not Vonage.
The only thing I can think of is that the father installed some sort of Vonage software on his mobile phone that enables outbound calling through the Vonage service. But if such a piece of software exists, I’m unaware of its existence. The closest thing is Vonage Companion, and that is designed to run on PCs not a mobile phone. I’m more confused over this than the mainstream media was over the Governor Sarah Palin VP pick.
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New 3CX VoIP Phone SIP Softphone
In late July, 3CX launched a new SIP-based VoIP client called 3CX VoIP Phone, with a fully-featured dialpad, and it allows for easy call transfers. It also features history of calls, the ability to put calls on hold, and the ability to accept, reject or ignore calls. Best of all 3CX VoIP phone is completely free and works with most popular IP PBXs and VoIP providers. It even sports multiple SIP profiles support for registering with multiple SIP accounts.
As I wrote back in July, 3CX VoIP Phone features strong Microsoft Outlook integration. Users can launch calls directly from their contacts’ list within Outlook by just right-clicking on the name of the person they wish to call.
Other features of 3CX VoIP Phone
- Supports several SIP profiles
- Shows personal call log/history - ideal for salespeople
- Message Waiting Indication (MWI)
- Supports G.711 (A-Law and u-Law), GSM, iLBC and Speex codecs
- STUN support for NAT/firewall traversal
- Installation provided as MSI for easy deployment
The new 3CX VoIP Phone can be downloaded here: http://www.3cx.com/VOIP/voip-phone.html
Tags: 3CX, 3CX VoIP Phone, G.711, phone, SIP, Speex, VoIP
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