calls's archive
Truphone 3.0 comes to the Apple iPod touch
Truphone today launched Truphone 3.0, a major new upgrade to its mobile VoIP application for the Apple iPod touch. Truphone 3.0 was already available for the Apple iPhone, so this release just brings the latest features to the popular iPhone touch.
IM services currently supported include Skype, MSN Messenger, AIM, Yahoo! Messenger and Google Talk. It also does free calls when in Wi-Fi to other Truphone users as well as free WiFi calls to Skype and Google Talk users. Though I would like to see 3G data support to enable VoIP over 3G. Yes I know Apple blocks VoIP over 3G apps, but if you jailbreak your iPhone, you should be able to make VoIP over 3G calls. (read my tutorial on how to do VoIP over 3G on jailbroken iPhones) Yet, there is no mention whether their truphone app will work over 3G on jailbroken iPhones. Ironic that in 2007 truphone was the first to demonstrate VoIP over WiFi on an Apple iPhone that they jailbreaked.
Of course, you could use Truphone Anywhere for free calls, but that uses the 3G voice channel not 3G data. It leverages a callback system that uses your bucket of cell minutes for ‘relatively’ free calling.
In any case, check out the news.
Tags: 3g, apple, google talk, iphone, ipod touch, skype, wifi
Related tags: apple iphone, jailbroken iphones, truphone, apple, calls, iphone
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GVDialer for Google Voice
GVdialer is an intriguing mobile application for Google Voice, supporting iPhone, BlackBerry, Android, Microsoft, and Symbian. GVdialer enables you to use Google Voice with your mobile phone while enabling some cool features. For instance, you can present your Google number as your Caller ID on outgoing mobile calls, thus keeping your mobile number private. This also gives you a one number identity to share with people.
Using the app installed you can dial directly from your phone’s contacts, speed dial, call log or keypad, and GVdialer will automatically connect the call using Google Voice.
Even cooler you have Google Voice feature access including instant access to Google Voice’s voice mail, Inbox, and GOOG-411.
As seen in the iPhone application, GVdialer lets you configure when GV would be used, i.e. on all calls, international calls, domestic calls, or ask on every call.
It costs $9.99, but definitely worth checking out
Tags: google, google voice, gvdialer, iphone, microsoft, mobile phone
Related tags: google voice, google, voice, gvdialer, mobile, calls
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Bring Out Your Dead! Wait! Skype for Asterisk is not dead!
Digium has an excellent post today titled The Rumors of Our Death discussing Skype for Asterisk (SFA) and the recently launched (beta) SkypeforSIP (SFS). There has been much discussion on the blogosphere, twitter, and elsewhere if SFS means the death of SFA. Some were even seen carting SkypeforAsterisk away into the trashbin of other failed software endeavors, as seen here:
It’s not a pretty sight when people write you off for dead when you’re really not. But wait just a second. Digium’s Steve Sokol explains late today that SFA is not dead. He writes:
With Skype’s recent announcement of Skype For SIP there has been a great deal of pontification on the impending death of the not-yet-released Skype For Asterisk. I’d like to take a moment to explain why Skype For SIP (SFS) does not spell the end for Skype For Asterisk (SFA), and why Skype For Asterisk is still in beta.
First, the key differences between Skype For SIP and Skype For Asterisk:
Steve lists more differences, but I don’t want to steal his thunder. Go read his post. I knew there were key advantages in SFA over SFS and there was so much confusion out there, I was tempted to write a comparative chart, but was too busy. In any event, it’s nice to see Digium clarifying the advantages of Skype for Asterisk. Any questions?
Tags: asterisk, sfa, sfs, skype, skype for asterisk, SkypeforSIP, voip
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Skype For SIP Marries Skype and IP-PBXs
Today, Skype announced it is enabling SIP-based IP-PBX to connect to the Skype network, which will allow low-cost SkypeOut calling, receiving calls from Skype users, and receiving calls from regular PSTN phone lines. Outbound calls from IP-PBX SIP handsets to Skype phones is not part of this news announcement. Skype commented it is too difficult to dial Skype usernames from a desktop handset.
Features:
- Receive and manage inbound calls from the 405 million Skype users worldwide on SIP-enabled PBX systems, connecting the company website to the PBX system using Skype click-to-call buttons
- Place calls via Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX, saving your business money with Skype’s low rates
- Purchase Skype online numbers to receive calls to the corporate PBX from landlines or mobile phones
- Manage Skype calls using your existing hardware and system applications such as call routing, conferencing, phone menus, voicemail and call recording and logging - no additional downloads or training are required
Skype For SIP is perfectly suited to businesses that already have IP-PBXs and want to connect to Skype’s network which offers low-cost calling. Skype for SIP is being launched as a closed beta program, but you can register and try to be part of the beta. Interestingly, Asterisk for Skype, a SIP-based (and IAX) open-source IP-PBX was the first to offer some tight integration with Skype.
If Skype continues to open their proprietary doors (they recently announced they were giving away the SILK codec), and now they’ve finally enabled SIP-to-Skype integration, then we can see a momentous shift from SIP trunking to Skype trunking. Customers will not only choose the lowest cost IP trunk, but also the one with the best quality. Skype is renowned for their high-quality due to their P2P architecture, so companies will look very closely at Skype trunking instead of SIP trunking.
And of course if customers shift to purchasing Skype trunks, the SIP trunks will have to follow suit, which means a win-win for businesses looking for low-cost calling.
Via Skype blog
Tags: ip-pbx, open source, sip, Skype, Skype for SIP, voip
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EZCallerID.com Hosted CNAM for Enhanced Caller-ID on any IP-PBX Launches
EZ Call, Inc. today announced the launch of EZCallerID.com, a new service that provides enhanced Caller ID, also known as CNAM, for VoIP calls. The hosted CNAM service gives you not just the phone number, but the name of the person calling.
Most SIP trunking providers do not provide the caller’s name with Caller ID on inbound calls. EZCallerID.com solves this issue by simply having you route your inbound calls to their server. They insert the caller’s name and send the call back to your IP-PBX.
How’s it work? Simply put, EZCallerID.com connects to the national databases that contain the name associated with each phone number, perform a reverse lookup, insert the CallerID info into the From SIP header and then send the call back to you.
This is similar in concept to my recent article, CNAM (CallerID with Name) on Asterisk using Reverse Phone number lookup, but in this case, no Asterisk PBX is required. It works with any SIP-based IP-PBX. It is a hosted offering, so there is a fee of course, but it’s not expensive. They only charge $0.015 per call (or $1.50 for every 100 calls). The service is available on a pay-as-you-go basis ($10 minimum initial charge), with no recurring charges or minimum monthly commitment.
Head on over to EZCallerID.com if you want to sign-up.
Hat tip to Eric Hernaez for the news tip
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Tags: Asterisk, Caller ID, CallerID, CNAM, EZCallerID.com, IP-PBX, Session Initiation Protocol, SIP, voip
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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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flaphone Enables Free Web-based SIP-to-Skype calls
Today, flaphone (formerly Flashphone) announced that users of their Flash VoIP application can now make a call from flaphone to skype. You simply need to enter sip:skype_username@skype after selecting “none”(global)” for the SIP account. I should mention that flaphone supports multiple SIP credentials, which is a really nice feature. I’ve been testing flaphone for several weeks now and have been meaning to write up their cool Flash-based VoIP application.
In any event, for my first test call I entered sip:tomkeating@skype and pressed the call button. The call was initiated and the call quality was superb!
You can also use this SIP-to-Skype feature for flaphone’s CallMe widgets that you place on your website.
Similarly, Gizmo5 recently launched OpenSky which also enables SIP-to-Skype dialing. However, Gizmo5 calls are free only up to 5 minutes long. For longer calls they are offering a paid service. There is no such restriction that I am aware of with flaphone.
By leveraging Flash, flaphone is cross-platform, has minimal download time, and you can run it from any browser. That and the fast that it supports SIP-to-PSTN calling, SIP URI dialing, and SIP-to-Skype calling, means this is one VoIP app you should check out!
Tags: flaphone, flashphone, gizmo5, sip, skype, voip
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Another Free VoIP Calling Web Application
Goober Networks, recently launched CallingAmerica.com, which offers web-based free VoIP calls to any landline or mobile phone in the U.S. or Canada. The Web-based offering uses Flash for the audio output & microphone input. As for the business model for “free calls” CallingAmerica.com uses advertisements on their website that you must watch before the call is initiated.
I decided to test it for myself to see how well it works. I simply went to their website, entered a phone number, and clicked the FreeCall now button, as seen here:
You’ll be presented with a captcha code which you must enter to prove you are human, as seen by this clipped browser screenshot here:
Then, you’ll see an ad and a short countdown (15s or less) before you can initiate the call as seen by this clipped browser screenshot here:
The countdown was pretty short, so surprisingly it wasn’t annoying. After the countdown, the Flash application confirms your microphone source. Simply by talking into it, it detects the audio signal and then initiates the call. The call quality was pretty good - certainly on par with other web-based VoIP offerings.
I should point out that if you don’t register, the calls are limited up to two minutes in duration each. Pretty useful if travelling and just want to make a quick free call. By registering for free at CallingAmerica.com, users can make an unlimited number of calls for a duration of up to 15 minutes. All in all CallingAmerica.com is worth keeping bookmarked for when you need to make a quick free call.
Tags: CallingAmerica.com, captcha, flash, free calls, goober networks, voip, web
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Nokia runs Skype on Wi-Fi and 3G - Suck on that iPhone!
Skype and Nokia today announced that Skype will be integrated into Nokia devices, starting with the Nokia Nseries. The Nokia N97 flagship phone will be the first to incorporate Skype in the 3rd quarter of 2009.
Skype will be integrated into the address book of the Nokia N97, allowing you to see when Skype contacts are online and perform instant messaging (IM) or VoIP calls.
But here’s the real kicker - the Nokia N97 will be able to use Wi-Fi and 3G to make and receive free Skype-to-Skype voice calls as well as Skype calls to landlines and mobile devices. The Apple iPhone on the other hand, restricts VoIP clients to just Wi-Fi VoIP calls and blocks 3G (data) VoIP calls.
Tags: 3g, apple, iphone, n97, nokia, nseries, skype, voip, wi-fi
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OpenSky enables SIP-to-Skype Calls
Michael Robertson, CEO of Gizmo5 has done it again! He has launched a cool new service called OpenSky http://gizmo5.com/opensky, a service that enables SIP-to-Skype calling. It’s not the first time someone has broken the Great Wall of Skype to enable SIP-to-Skype calling, but it’s always great to have a new method of accessing the popular Skype network using the standard SIP protocol.
Gizmo5’s free Skype gateway called OpenSky is open to all users enabling any SIP-based VoIP software or SIP-based phone system to make calls to Skype users. What they’ve done is to create a SIP alias for every Skype user. So if you want to call a Skype user you simply dial skypeusername@opensky.gizmo5.com from any SIP-capable device.
The service can initiate a call by sending a SMS so any mobile phone can be used to call a Skype user or calling from a web browser using just flash. Users can even have any SIP call forwarded to their Skype address using my.gizmo5.com.
This is currently a free service Gizmo5 for calls up to 5 minutes long. For longer calls they are offering a paid service.
I can see this service being useful for when you want to make an international call and don’t want to be forced to use your PC to make a free Skype-to-Skype call. Instead you can use your SIP desk phone (if it supports URI dialing) or you can use your mobile phone to initiate the call.
Tags: gizmo5, Michael Robertson, opensky, sip, skype, voip
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Analysis of a VoIP Attack
VoIP security is often overlooked by IT administrators as well as VARs and resellers that deploy VoIP in the enterprise. They do so at their own peril, however. One of the main factors behind using VoIP is to save money. Well imagine your IP-PBX has been hacked and you don’t notice anything wrong until you receive the next phone bill with hundreds or thousands of dollars in phone charges. There goes all the savings you anticipated when you decided to install VoIP!
This laissez faire towards defensive security reminds me of Star Trek, where for whatever reason the Enterprise flies through space with danger lurking around every corner but they keep their defensive deflector shields off and often turn them on when it’s too late. The Enterprise has a fusion reactor with nearly limitless power, so why not keep the deflector shields on all the time? Maybe they’re just being “green” and shooting for five nines (99.999%) of efficiency. 
In any event, security in general is often overlooked, whether it’s securing your web server or your email server, or confidential database servers. But in most cases when these particular systems are hacked it’s usually just an inconvenience (defaced web pages, spamming through your email server) with minimal financial impact. Not so when it comes to VoIP. The financial impacts of a hacked SIP server or VoIP gateway could be tremendous. This is especially true for larger organizations which already have hundreds or thousands of calls per month, including international business calls. How does accounting find the fraudulent calls in the phone bills which are 4 inches thick? It’s like finding the proverbial needle in a haystack.
If the hackers are smart, they will limit the amount of traffic they route through a hacked gateway as not to set off any red flags. It could be months or possibly even years before anyone notices anything is amiss. I’m reminded of an old PBX technology called DISA (Dialed In Switch Access) which was one of hackers first tricks to get free calling. DISA was designed to allow employees to remotely call into the PBX and get second dial-tone. With this second dial-tone using touchtones they could logon to ACD queues, monitor agents calls, and of course initiate outbound calls.
In fact, many years ago, TMC was hit with a DISA-like attack on our Comdial PBX resulting in quite a few international calls. If I recall, there was a vulnerability in the Keyvoice voicemail system which allowed someone to make outbound calls. Needless to say, I was able to shut it down pretty quickly.
Part of the attack also involved using a scripted dialer which accessed the voicemail system by automatically sending the # key, then sending a chosen extension (say 100), and then iterating through all the various PINs (0000 - 9999). Since TMC has a toll-free 800 number, the attacker only has to make at most 10,000 calls to find a PIN to a particular extension. Obviously, chances are they’d find the script in much less than 10,000 calls and you get 3 tries before the voicemail hangs up. Once the PIN is found, not only does the attacker have access to the the user’s voicemail, they also have access to any DISA capabilities of the voicemail system. More reason why IP-PBXs today need to have a PIN expiration feature just like Active Directory supports password expiration. No matter how many times IT staff reminds employees to change their passwords/PINs, they just don’t do it unless the system forces them to. I don’t believe any of the Asterisk systems I’ve tested have password expiration - so my open source Asterisk fans, if you’re listening, add it to the code, will ya?
With all this in mind, I was fascinated to read an article by an Austrian company IPCom titled “Analysis of a VoIP Attack”. It’s an excellent read. Let me give you the abstract:
Recently, several IT news websites reported VoIP attacks against home users, containing lots of myths and incorrect statements. Unfortunately, they also give wrong security advices. This article analyzes the attacks and describes the motivations behind. Further, it shows simple workarounds how “insecure” software can be used in a secure way.
Here’s a teaser:
1 The Attack
1.1 Analysis
On 23.09.2008, heise.de reported an attack against VoIP devices of German VoIP users [heise]. This article references a thread in the IP-Phone-Forum [ipphone] in which people report that their VoIP phones started ringing in the middle of the night and displayed incoming calls from the phone number 5199362832664. One of the users presented a log file of a Patton SIP device which captured the suspect INVITE request:
02:12:42 SIP_TR> [GW] < Stack: from 213.130.74.70:3808
INVITE sip:810525551690000@1.2.3.4;transport=udp SIP/2.0
Via: SIP/2.0/UDP 213.130.74.70:3808;branch=100100101101011111101110
00100213.130.74.701.2.3.41863480914;rport
Max-Forwards: 100
From: <sip:5199362832664@1.2.3.4>;tag=21671132663-
4985269162167113266321671132663213.130.74.70
To: <sip:810525551690000@1.2.3.4>
Call-ID: 83764811100011101110010010110101101100111001001011
0101111110111000100213.130.74.701.2.3.41863480914f
df23881052555169000021671132663-
4509759162167113266321671132663213.130.74.70174046 6380
CSeq: 1 INVITE
Contact: <sip:fdf238@213.130.74.70:3808;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 394
Let’s have a look at this SIP message. The funny thing is that absolutely nothing in this SIP message is trustworthy: Probably the SIP message has been received via UDP and the source IP address could be easily spoofed. Further, every data in the SIP message is user generated (in this case by the attacker) and does not necessarily reflect real data. Nevertheless, let us try to analyze the message:
- Source IP address 213.130.74.70 and source port 3808: Although the IP address could be easily spoofed, in this case it may be the real address of the attacker as the IP address is also present in the Via: header (used for sending back responses). Further, if the attacker wants to know the result of the attack, he has to receive the SIP responses meaning that he has to provide his real IP address.
- The Call-ID looks like a random string and contains the source IP address. As the Call-ID is invalid (per RFC 3261 the Call-ID must not contain spaces), it can be assumed that the attacker did not use a fullfledged SIP stack, but some scripts to generate the request.
- The User-Agent header displays “X-Lite” as client. However, if you compare the above request with an INVITE request sent by X-Lite you will find out that the random strings (call-id, tags, branch
Ok, you’ve been thoroughly ‘teased’, now go read the full article (PDF). Good stuff! 
Tags: DISA, hack, security, sip, sip headers, spoofing, star trek, voip
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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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Polycom KIRK DECT SIP Phones
Polycom today announced the launch of its latest KIRK Digital Enhanced Cordless Telecommunications (DECT) wireless products. Polycom has introduced three new products: the KIRK Wireless Server (KWS) 300, the KWS 6000, and the KIRK 5040 handset. which are all SIP-based.
The name KIRK certainly evokes Captain Kirk from Star Trek and most likely intentionally, since Captain Kirk and his crew made the wireless communicator famous 40 years ago. No doubt Polycom has some Trekkies in their engineering or marketing teams. 
Of course, I should mention that Polycom acquired Spectralink, a well known wireless phone manufacturer and Spectralink previously acquired Kirk Telecom, the makers of these DECT wireless phones. So the product name is simply a reflection of their corporate name. So perhaps all this Trekkie analogy business is moot. Or maybe they named their company Kirk Telecom to honor Captain Kirk. Stranger things have happened.
All KIRK solutions are scalable, both in terms of the number of users as well as the coverage areas supported. The latest additions to the KIRK Wireless Server portfolio include:
- The KIRK Wireless Server 300, a SIP-based wireless telephony system, is ideal for smaller sized businesses, by scaling support from one to 12 handsets The KIRK Wireless Server 300 is a single-cell solution that can support up to four simultaneously calls and up to six KIRK repeaters in order to extend the coverage area. Each KIRK repeater increases the coverage area by approximately 50 percent.
- The KWS 6000 is a SIP-based enterprise wireless telephony solution that scales from just a handful up to more than 4,000 users. Up to 256 radio units are supported, which when combined with the KIRK Media Resource, can support more than 1,000 simultaneous calls. Each KIRK base station handles 12 simultaneous calls, and customers can scale up based on their individual needs. Additionally, KIRK repeaters can be added to increase the coverage area by approximately 50 percent.
The KIRK 5040 handset, the newest addition to the KIRK product line, is a lightweight DECT phone that combines an intuitive user interface and wireless headset that can be operated hands-free and wirelessly with a Bluetooth headset. Like the KIRK 5020, the 5040 can quickly be switched to silent mode and will distinguish between external and internal calls by ring tone. The KIRK 5040 handset also features an intuitive user interface and a large color-display offering an experience similar to a mobile phone and with the added benefit of hands-free operation.
Features of the 5040:
- TFT colour display (65.000 colours, 8 lines of text/icons)
- Li-ion battery
- 4 Way navigation key
- 2 Softkeys
- CLIP (40 caller-ID presentations)
- Date and time in display when supported by system
- Internal/external ring pattern
- Volume control
- Telephone book with 250 name entries (4 numbers per name)
- Auto login - roaming between 10 different installations
- Silent mode (mutes all alerts/calls)
- Alerting on silent mode (choice from display flash, vibrator or short ring)
- Call list of incoming/missed/received (last 40 entries)
- Redial function from call list
- Speed dial
- Auto answer with different settings (after 1st ring/when lifted from charger/on headset/loud speaker on)
- 10 different ring signals and adjustable ring volume
- Key lock
- Auto key lock
- Vibrating alert
- Any key answer
- 11 menu languages (UK, FR, DE, ES, IT, NL, CZ, PL, DK, NO, SE)
- Headset connection
- Ring signal in headset
- Adjustable volume in headset
- Answer/end calls via headset button
- Microphone mute
- Speaker on auto-answer
- R-key for transfer and special services
- Adjustable alerting volume (low battery/low coverage/incoming message)
- Adjustable backlight delay (for max. battery conservation)
- Text messaging - max. 72 characters per message (system dependant) 10 user defined messaging templates
- Stores 20 messages
- Speech/stand by time: Up to 15/100 hours
- Weight incl. battery: 110g
- Size (LxWxH): 146×48x19mm
- 2 types of chargers (w/wo USB 2.0 connection)
- Suitable for Bluetooth headsets
Pricing & Availability
The new KIRK solutions are available worldwide through Polycom’s certified reseller partners. The list price for the KWS 300 is U.S. $360. The KWS 6000 list price is U.S. $1,200 and includes a server and one base station, which supports up to 30 users. With the scalable nature of the KWS6000 it can also be set up for more users. The KIRK 5040 handset sells at a list price of U.S. $310. To learn more about Polycom’s KIRK phone solution, head here.
Tags: bluetooth, Captain Kirk, KIRK 5040, KIRK Wireless Server 300, KWS, KWS 6000, Polycom, SIP, 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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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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3CX Free SIP Softphone
3CX has announced the release of a totally free new VoIP softphone called 3CX VoIP Phone that allows users to make and receive calls from their computer using SIP-based VoIP providers or SIP-based IP-PBXs. This free SIP softphone client isn’t the first “free” softphone on the market. That distinction belongs to Counterpath’s popular X-lite SIP softphone. However, I’m glad to see more free SIP softphones on the market.
In fact, I’d like to see free SIP softphone clients for Windows Mobile and the iPhone. Counterpath, had one for the PocketPC but they discontinued it. Of course, you can install fring, which includes a SIP client, and you can IM or call Skype, MSN Messenger, ICQ, and Google Talk. fring is compatible with Symbian 8, 9.1, 9.2, Windows Mobile 5 & 6 and UIQ handsets. GizmoProject and SJPhone are two other options that work on several mobile handset models. I’d also like to see more standardized video softphone clients so you can have videoconference calls from your mobile phone to someone running a softphone client on their PC or Mac.
In any event, one of the key features of 3CX VoIP Phone is its integration with Microsoft Outlook. 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.
Nick Galea, CEO at 3CX said: “3CX VoIP Phone is great for businesses that wish to have an easy to deploy, business-level VoIP soft phone. Because it is free, the usual hassle of administration of client licenses is avoided. The free editions of other VOIP phones do not have key features such as call transfer or the ability to put a call on hold.
Nick put a “green” spin on using softphones over hardphones when he said, “VoIP Phones are an interesting option for businesses - they are easy to administer and environmentally friendly. Hardware phones require additional electricity, administration and desk space.“
3CX VoIP Phone of course somes with a dial pad, buttons for transferring or forwarding calls, put calls on hold, accept, reject or ignore calls, and more. You can also review your history of calls.
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.
Tags: 3CX, 3CX VoIP Phone, counterpath, Nick Galea, SIP, softphone, video, VoIP, X-Lite
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Vonage UK launches new low cost V-Plan calling plans
Vonage today announced two new low cost call plans for their Vonage UK subsidiary. According to Vonage UK, “Following customer research and reacting directly to consumer concern about increasing household costs and spiraling business overheads, Vonage has created two new fixed rate call plans.”
They added, “Vonage subscribers report enormous savings on their monthly bills and comment on the speed and ease of swapping providers as well as installing Vonage. The new call plans fly in the face of increasing utility prices and the new £6.99 plan has been designed for the high percentage of Vonage consumers requesting more cost efficient plans for North America.”
Vonage’s £7.99, £14.99 and £18.99 plans incorporating up to 45 countries remain unchanged. The two new call plans offer Vonage’s lowest ever rates and are called V-Plan UK and V-Plan US.
• £5.99 per month - unlimited calls to the UK (V-Plan UK)
- Premium features such as call waiting, caller ID, call diversion, voicemail, three way calling (normally billed as extras with other providers), are included as standard.
• £6.99 per month - unlimited calls to the UK, United States and Canada (V-Plan US)
- As above plus, for only £1 extra per month, unlimited calls to the US and Canada to include calls to US and Canadian mobile phones.
Here’s a screenshot of the various V-Plan calling options. Click image to see the plans:
Vonage to Vonage calls are free. Also, there are no hidden costs with Vonage - prices are always quoted including VAT.
Vincent Potier, Managing Director of Vonage, said; “We recognise how important it is for customers to keep costs low and as predictable as possible - especially in light of the current economy. Our new plans enable customers to make worry free calls for as long as they want and show our long term commitment to listening and responding to our customers as well as offering the highest level of customer service and value for money”.
Tags: calling plans, United Kingdom, V-Plan, VoIP, Vonage UK
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Plumble Ad Sponsored “Free” Calls
You remember Pudding Media, right? They offered ad sponsored phone calls in exchange for “listening” in on your phone conversations so it could target audio ads. Pudding Media essentially leverages keyword wordspotting using speech recognition. Well, today, I learned about Plumble from Telecom Monthly. The article seems giddy with what it thinks is some new revelation…
Once in a blue moon, a new product comes along with an idea so obvious that you just want to slap your forehead that you didn’t think of it first. For a couple of years, companies like Jajah and Skype have been offering “Free” phone service. But both still collect your credit card number and charge you for calls that you make off their networks. So they aren’t really “Free,” although they can come close if you call mostly other Jajah or Skype users.Plumble, The Free Phone Service
A new beta service called Plumble offers actual free phone calls without collecting your credit card number or even your name.
Well, Pudding Media offers free ad-sponsored calls, so this is nothing new. Of course, I didn’t care for Pudding Media’s eavesdropping ad model. Plumble is limited to U.S. and Canadian calling, which isn’t that expensive anyway. Where’s the international free calling? Heck Jajah has been offering free international calling since 2006. Further, Plumble requires that you dial 818-742-0110. Um, that’s not toll-free - that’s a California area code. So I have to pay to dial long-distance to make a “free” phone call? Not to mention it is currently limited to 5 minute phone calls. Thus, this is only useful for short calls anyway and only for people where 818 is a local call.
Thanks, but no thanks.
Tags: free calls, international calls, Jajah, Plumble, Pudding Media, Skype, voip
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