Source: www.voip-news.com

Messaging Convergence Gateway from CounterPath Corporation is now available. The carrier-based, core-network messaging server creates a bridge between VoIP and SMS infrastructure used by mobile carriers.

“SMS has surpassed telephone calls to become today’s most widely used means of device-driven communication,” said Donovan Jones, CEO of CounterPath. “Until now, consumers were unable to seamlessly send SMS text messages from their mobile phones to VoIP numbers. CounterPath’s MCG technology changes that by freeing SMS from mobile devices and extending it to consumers using our Bria softphone on their PC. This groundbreaking technology changes the landscape of mobile communications and provides numerous benefits to VoIP service providers and consumers alike, most notably taking consumers one step closer to single-number identity, a key aspect of Unified Communications.”

According to CounterPath:

The MCG is designed for fixed VoIP systems with assigned telephone numbers. Once deployed in the VoIP operator network, the MCG allows two-way text communication between CounterPath’s Bria softphone and mobile devices. Deployed in this manner, there is no need to replicate the mobile operator’s SMS infrastructure, as neither a Messaging Service Centre (SMSC), or Home Location Register (HLR), is required. This greatly reduces capital expenditure and complexity for rolling out SMS services to VoIP users and provides an additional revenue opportunity for service providers.

Additional benefits to service providers include greater adoption and stickiness of their softphone services and ease of implementation. Consumers benefit from significantly more streamlined communications: they no longer have to leave their PCs to send and receive text messages with their mobile contacts and the texting experience is enhanced based on intuitive integration with the CounterPath softphone.

Source: goebel.net

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A commentator to my last post "Why Truphone and Gizmo5 applaud that Nokia turns it’s back on mobile VoIP" doubts my argumentation by asking:

I thought Truphone is based on the built-in SIP client? Then it would seem unlikely that Truphone applauds Nokia dropping the mobile VoIP stack from certain models.

My answer is the following:

Yes, Truphone until now works on top of the built-in SIP client. But the Truphone software develops more and more into a standalone application: with the inclusion of SMS, callthrough where no Wifi is available, presence information and so forth. They aren’t afraid of building their own SIP app since it ties the customer even more to them. Therefore Gigaom wrote:

Truphone isnt waiting around for Nokia to do something. A company spokesman told us: From Truphones perspective Nokia has removed the VoIP client from all the N-Series phones for the planned future. We are putting in a replacement client functionality so that existing customers are not orphaned.

Don’t forget that Truphone has a very high pricing for Wifi calls! Their software is convenient to install, but many other VoIP companies are three times cheaper. That’s why they would be very happy to be your only mobile VoIP provider. Vyke already launched their own client, as you can read here, and Gizmo5’s CEO Michael Robertson officially applauded Nokia’s move in a FierceVoIP article.

The only losers are the cellphone users, since these 3rd party apps are much more difficult to use than the native SIP client. Read this insightful comment, posted at Phoneyboy’s blog:

"Im using VOIP on Nokias phone via my own asterisk server. How can Nokia expect me to develop my own Internet telephony application so that I can continue to use it? There are simply thousands of small users out there for whom this is beyond what they could do. This will leave them out in cold.

And further comment. Any third party application will have hard time to match the comfort of integrated symbian UI, where normal and internet calls are integrated together and one push of a button decides which one to make. Just compare this with Fring whose UI is just terrible."

We tinkerers who use Asterisk, Voxalot, Voipstunt, PBXes and Iptel.org are out of the game for the new Nseries devices. I am afraid that the Nokia E71 is the last cool device for a VoIP aficionado like me. Hopefully the Android devices will have more to give. Phoneboy calls us, who use 10 VoIP providers on our Nokia devices, a "minority". Nevertheless he "understands the frustration". Thank you!

But still I think that he is wrong, or maybe just blue-eyed, when he says: "It sounds like the problem is only limited to these two handsets". The problem affects all Symbian Series 60 3rd generation Feature Pack 2 (S60 3.2)! This means: All new handsets from now on are affected. Nokia’s VoIP isn’t revolutionary disruptive anymore, but has changed to a big boys’ only business.

Source: gigaom.com

Source: goebel.net

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Next week I will be at the Mobile World Congress in Barcelona and I will take my Maxroam SIM card with me. So outgoing calls to German landlines will cost only 0.38 per minute and incoming calls 0.25, instead of 0.58 and 0.28 which a usual German cell phone provider would charge me. Before last year’s regulation these prices where even higher. In some countries I’d still have to pay 2.49 per minute for a local call while roaming. To be reachable in Barcelona I will forward my Berlin office number for free to the UK fixed line number that I have on my Maxroam SIM. I will either use my own ATA for that purpose or a new Maxroam feature.

An outgoing call with Maxroam is a litte bit different from a normal mobile phone call. It doesn’t start directly. Instead you see several status messages running over your phone’s display, like "calling", "requesting" and then again "calling", before you receive an incoming call with no number. That’s Maxroams server calling you. A voice says "connecting, please wait" and contacts the callee. This entire callback system is based on USSD messages. That’s a kind of free short messages in GSM networks, which can be sent only between the user’s handset and the provider. USSD had been invented to let you check the amount of prepaid minutes on your SIM card for free. Nowadays it’s often used as a ‘trigger’ to invoke independent calling services like Maxroam. Think of it like Jajah, but without the need to pay for mobile data usage for the communication with the server.

The Cubic phone, which can also be had from the company for usage with the Maxroam SIM and for VoIP over Wifi, is so packed with software that there is no space left to secure it with a PIN number. Maxroam’s CEO Pat Phelan told me in an interview: "Its very packed on the operating system and we have had to leave room for the two logging on GUI for the hotspots". In the last quarter the company had lots of backend work going on which now result in further improvements, as Pat Phelan told me in an email:

1. Live billing
We now have full live billing for all users on our backend, make a call, hang up and we instantly display it.

2. Add a local number
As of today we can add local FIXED line numbers to your MAXroam sim from 52 countries. (MAXroam only use fixed line numbers unlike other companies which use international mobile or premium UK mobile numbers where the average users have no knowledge of what it costs to dial the SIM and your friends are just subsidizing your roaming). This list is being added to every day. These numbers begin at 1.05 per month and you can pick up, drop as many as you like, minimum commitment is only a month.

3. Free Call forwarding
When you arrive in your destination sometimes you have access to a hotel room or an office number, we will now allow to forward all your MAXroam numbers totally free to fixed line numbers in a list of 48 countries so you are roaming for ZERO COST.

4. SMS only 5c
Once you log into your MAXroam account we will allow all users to send SMS anywhere in the world from the backend for only 5c per message.

But the most interesting announcement he already made in November, when I asked him in an interview for the German magazine ProFirma about Maxroam’s prices compared to other companies like Sunsim, Globalsim or TouristMobile:

Our pricing is only at the beginning, most of these company are just resellers and I dont mean this as an insult, we are building a global brand here, our aim within 1 year is 20c in and out in Europe and under 10c in the USA. Our next sims will be 128k Java with between 6 and 16 IMSI on each sim depending on your travel arrangements, so you arrive in India, we have an Indian IMSI on your SIM and you roam at discounted rates, we are not depending on other peoples roaming agreement and are at present travelling the planet signing independent roaming agreements.

Under 0.10 in and out? Now that would be a great price for USA, where Maxroam still charges 1.10 or more for incoming and outgoing calls. I can’t wait to see these 6 and 16 IMSI SIM cards. That’s like taking 16 cell phones with you, each one with a local card. So you don’t have to pay for incoming calls.

Source: goebel.net

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Last year I was nagging that "Packet8 MobileTalk could be done much better" and I was right. I could learn that now when the Israeli company Mobilemax installed for me on a cellphone the underlying software which powers Packet8’s MobileTalk. It is a real mobile VoIP wonder weapon which I covered in an article for Areamobile.

Internet telephony could be a killer application for mobile phones. But it has it quirks, shows a series of articles I wrote. In most cases you need at least a cell phone which can open mobile websites. It let’s you do a mobile callback with VoIP providers like Jajah or Voxalot. After entering the phone numbers of the caller and the callee on a mobile website, a server rings up both and connects them over the internet. Of course this also costs double, but for international calls it’s still cheaper than the own cell phone provider’s prices. Only Wifi cell phone calls are less expensive. They only cost a few cents per minutes and often they are free. But Wifi isn’t always available.

Mobilemax thus developed a software which automatically sends mobile phone calls the less expensive way over the internet whenever that’s cheaper than normal calls. No need to open mobile websites on the phone or to trigger callbacks by SMS or instant messaging. The software simply sits unobtrusively in the background and automatically determines the way in which the call is connected. The user only needs to enter the number.

Establishing a call with VoIP software from Mobilemax

The program works on about 500 phones with Palm, BlackBerry, Symbian or Windows Mobile operating systems and last month it has been deployed for the first time by the US VoIP provider Packet8. Once a number has a foreign area code, the software starts to act and connects via a landline number to the server by Packet8. The server connects the telephone conversation with the other party over the VoIP network. Calls from the United States to Asia or Europe cost only a 2 to 5 US cents per minute more than the price of a local call. The American mobile operators normally charge up to three dollar minute. German companies like Running Mobile or Cellity offer similar solutions.

But the Israeli software has much more functions which the competitors lack and also Packet8 doesn’t use. The program could also automatically decide to connect a phone call as a callback or over Wifi. Even VoIP calls over 3G will soon be possible, although all German mobile network operators seem to block them now I realized in some self-experiments. Mobilemax’ software is a real miracle weapon for mobile VoIP and the handling is particularly pleasant, because no extra buttons have to be pressed. The software even senses in which country the user is and automatically chooses a local number for callthrough or callback. What a pity that consumers cannot have it. Mobilemax distributes the software only to companies. "We don’t see ourselves providing the underlying service of the application and compete with our customers", said Mobilemax’ Director of Business Development, Perry Nalevka, to me in an interview. The Packet8 customers pay $10 per month only to use it. In addition they get the VoIP telephone minutes billed.

Configuration also allows other VoIP flavours

Other companies want to follow the same business model, Nalevka said, which started as a one-touch-dialing solution for calling card users and roamers who had to use tens of access numbers, PINs and dial flows to make a call without being ripped off by the mobile operators. Six different service providers worldwide and several IPBX and enterprise mobility providers are now testing the software. So far several tens of thousands of licenses purchased.

Other companies who use it:

• Reliance – Indian provider launching in 8 countries soon.
• Gamma Telecom, UK – launched the Crosslink service targeting SMBs and enterprise users
• Tellink, Belgium – callback provider for SMBs and roamers
• Tel3, Miami FL – residential provider
• Boomering – SMB mobility & callback solutions for roamers

Today it supports the following configurations:

• Callthrough with PIN or PINless (CLI based).
• Roaming location based callthrough with multiple access numbers automatically selects the relevant access number according to the user’s location.
• Seamless callback triggered by: DID, USSD, SMS, IP. The application triggers the callback, answers the incoming call and if needed sends the destination number.
• Dial around replacing prefixes in the dialed number (1010).

Further developments:

• Support additional phone models as they are released.
• Add new routes to seamlessly divert calls to: VoWIFI, Vo3G (to SIP or termination).
• Adding in-call Mobility features.

Source: snapvoip.blogspot.com

If you read the previous post on chan_mobile on asterisk, here is the follow up, straight from Asterisk SVN.

Configuring chan_mobile :-
The configuration file for chan_mobile is /etc/asterisk/mobile.conf. It is a normal Asterisk config file consisting of sections and key=value pairs.
See configs/mobile.conf.sample for an example and an explanation of the configuration.

Using chan_mobile :-
chan_mobile.so must be loaded either by loading it using the Asterisk CLI, or by adding it to /etc/asterisk/modules.conf
Search for your bluetooth devices using the CLI command ‘mobile search’. Be patient with this command as it will take 8 – 10 seconds to do the discovery. This requires a free adapter.
Headsets will generally have to be put into ‘pairing’ mode before they will show up here.

This will return something like the following :-

*CLI> mobile search
Address Name Usable Type Port
00:12:56:90:6E:00 LG TU500 Yes Phone 4
00:80:C8:35:52:78 Toaster No Headset 0
00:0B:9E:11:74:A5 Hello II Plus Yes Headset 1
00:0F:86:0E:AE:42 Daves Blackberry Yes Phone 7

Above is a list of all bluetooth devices seen and whether or not they are usable with chan_mobile.
The Address field contains the ‘bd address’ of the device which is like an ethernet mac address.
The Name field is whatever is configured into the device as its name (default name of the device).
The Usable field tells you whether or not the device supports the Bluetooth Handsfree Profile or Headset profile.
The Type field tells you whether the device is usable as a Phone line (FXO) or a headset (FXS)
The Port field is the number to put in the configuration file.
Choose which device(s) you want to use and edit /etc/asterisk/mobile.conf. There is a sample included with the Asterisk-addons source under configs/mobile.conf.sample.
Be sure to configure the right bd address and port number from the search. If you want inbound calls on a device to go to a specific context, add a context= line, otherwise the default will be used. The ‘id’ of the device [bitinbrackets] can be anything you like, just make it unique.
If you are configuring a Headset be sure to include the type=headset line, if left out it defaults to phone.
The CLI command ‘mobile show devices’ can be used at any time to show the status of configured devices, and whether or not the device is capable of sending / receiving SMS via bluetooth.
*CLI> mobile show devices
ID Address Group Adapter Connected State SMS
headset 00:0B:9E:11:AE:C6 0 blue No Init No
LGTU550 00:E0:91:7F:46:44 1 dlink No Init No
*CLI>

As each phone is connected you will see a message on the Asterisk console :-

Loaded chan_mobile.so => (Bluetooth Mobile Device Channel Driver)
— Bluetooth Device blackberry has connected.
— Bluetooth Device dave has connected.

To make outbound calls, add something to you Dialplan like the following :- (modify to suit)

; Calls via LGTU5500
exten => _9X.,1,Dial(Mobile/LGTU550/${EXTEN:1},45)
exten => _9X.,n,Hangup

To use channel groups, add an entry to each phones definition in mobile.conf like group=n
where n is a number.
Then if you do something like Dial(Mobile/g1/123456) Asterisk will dial 123456 on the first connected free phone in group 1.
Phones which do not have a specific ‘group=n’ will be in group 0.
To dial out on a headset, you need to use some other mechanism, because the headset is not likely to have all the needed buttons on it. res_clioriginate is good for this :-

*CLI> originate Mobile/headset extension NNNNN@context

This will call your headset, once you answer, Asterisk will call NNNNN at context context

Dialplan hints :-

chan_mobile supports ‘device status’ so you can do somthing like

exten => 1234,hint,SIP/30&Mobile/dave&Mobile/blackberry

MobileStatus Application :-

chan_mobile also registers an application named MobileStatus. You can use this in your Dialplan to determine the ’state’ of a device.
For example, suppose you wanted to call dave’s extension, but only if he was in the office. You could test to see if his mobile phone was attached to Asterisk, if it is dial his extension, otherwise dial his mobile phone.

exten => 40,1,MobileStatus(dave,DAVECELL)
exten => 40,2,GotoIf($["${DAVECELL}" = "1"]?3:5)
exten => 40,3,Dial(ZAP/g1/0427466412,45,tT)
exten => 40,4,Hangup
exten => 40,5,Dial(SIP/40,45,tT)
exten => 40,6,Hangup

MobileStatus sets the value of the given variable to :-

1 = Disconnected. i.e. Device not in range of Asterisk, or turned off etc etc
2 = Connected and Not on a call. i.e. Free
3 = Connected and on a call. i.e. Busy

SMS Sending / Receiving

If Asterisk has detected your mobile phone is capable of SMS via bluetooth, you will be able to send and receive SMS.

Incoming SMS’s cause Asterisk to create an inbound call to the context you defined in mobile.conf or the default context if you did not define one. The call will start at extension ’sms’. Two channel variables will be available, SMSSRC = the number of the originator of the SMS and SMSTXT which is the text of the SMS.
This is not a voice call, so grab the values of the variables and hang the call up. So, to handle incoming SMS’s, do something like the following in your dialplan

[incoming-mobile]
exten => sms,1,Verbose(Incoming SMS from ${SMSSRC} ${SMSTXT})
exten => sms,n,Hangup()

The above will just print the message on the console.

If you use res_jabber, you could do something like this :-

[incoming-mobile]
exten => sms,1,JabberSend(transport,user@jabber.somewhere.com,SMS from ${SMSRC} ${SMSTXT})
exten => sms,2,Hangup()

To send an SMS, use the application MobileSendSMS like the following :-

exten => 99,1,MobileSendSMS(dave,0427123456,Hello World)

This will send ‘Hello World’ via device ‘dave’ to ‘0427123456′

DTMF Debouncing :-

DTMF detection varies from phone to phone. There is a configuration variable that allows you to tune this to your needs. e.g. in mobile.conf

[LGTU550]
address=00:12:56:90:6E:00
port=4
context=incoming-mobile
dtmfskip=50

change dtmfskip to suit your phone. The default is 200. The larger the number, the more chance of missed DTMF. The smaller the number the more chance of multiple digits being detected.

Debugging :-

Different phone manufacturers have different interpretations of the Bluetooth Hands free Profile Spec. This means that not all phones work the same way, particularly in the connection setup / initialization sequence. I’ve tried to make chan_mobile as general as possible, but it may need modification to support some phone i’ve never tested.

Some phones, most notably Sony Ericsson ‘T’ series, dont quite conform to the Bluetooth HFP spec. chan_mobile will detect these and adapt accordingly. The T-610 and T-630 have been tested and work fine.

If your phone doesnt behave has expected, turn on Asterisk debugging with ‘core set debug 1′.

This will log a bunch of debug messages indicating what the phone is doing, importantly the rfcomm conversation between Asterisk and the phone. This can be used to sort out what your phone is doing and make chan_mobile support it.

Be aware also, that just about all mobile phones behave differently. For example my LG TU500 wont dial unless the phone is a the ‘idle’ screen. i.e. if the phone is showing a ‘menu’ on the display, when you dial via Asterisk, the call will not work. chan_mobile handles this, but there may be other phones that do other things too…

Important: Watch what your mobile phone is doing the first few times. Asterisk wont make random calls but if chan_mobile fails to hangup for some reason and you get a huge bill from your telco, dont blame me (not ME, the developer!)

Feedback, Support, Please can you make Mobile Phone X work… etc :-

as always, bugs should be reported at http://bugs.digium.com

email the man responsible for this mess at david.bowerman at gmail.com or dseeb_ on #asterisk & #asterisk-dev irc.

Source: voip-tech.blogspot.com

Globe7 results to be, by the website My VoIP Provider, the world’s best VoIP provider, according to rates and the service provided.
Globe7 offers a free software downloadable from the official website compatible with PC (Windows 2000/XP/Vista/Linux) and Macintosh that propose features and functionality similar to Skype™, as the possibility to make videocalls, free calls from PC to PC, SMS sending, file transfer, also it’s present a IPTV service to freely watch videos and TV broadcasted in streaming over the Internet.
The rate for one minute of a PC to landline or a cellphone call in the U.S. is 0,01 USD; the rates for all worldwide destinations are available in this page; besides, if you register an account as "Gold Member" you get a free US phone number.

Source: snapvoip.blogspot.com

DÜSSELDORF, Germany, September 11 /PRNewswire/ — sipgate is the first VoIP provider to give customers direct access to its system via a designated, proprietary interface. At the same time, "sipgate API" provides a catalog of functions that – within the scope of the product stipulations – can be integrated into third party products and solutions at the discretion of the user. This allows customers and developers to use virtually any sipgate function for their own applications. Information on the interface and the source codes of several model programs are available at http://www.sipgate.co.uk/api.

In making API available, sipgate is taking a step no other telephony service has ever offered: developers are given the opportunity to customize the service portfolio of the company to their own needs. As a result, VoIP, SMS and fax systems can now be integrated into CRM systems and Office programs, for example. Their integration into commercial software, offering telephony and telephony administration to users directly, is also feasible. Website operators will find callback buttons and phone numbers featuring Click2Dial particularly helpful.

For transport protocol, "sipgate API" uses "XML-RPC," whose specifications are described in detailed documentation. Working with "sipgate API" is free with any sipgate account, regardless of the pricing plan selected. The API solution can be used without a special release.

To facilitate the initial use of the solution, sipgate also discloses the source code of Firefox extension "sipgate FFX." Along with the creation of the connection with Click2Dial, it also demonstrates requests for account status, missed calls, faxes and voice messages received. Moreover, sipgate supports .NET developers with "sipgate API .NET SDK," a convenient library that makes the "sipgate API" services easy to use. Several Perl examples and a KDE panel application, which along with "sipgate FFX" are subject to a GPL 2 license, complete the no-cost service portfolio. A mailing list allows developers to share experiences and tips.

More information and press photos available at http://www.sipgate.co.uk/presse and http://www.sipgate.co.uk/api

Source: goebel.net

The VoIP company Sipgate, one of the biggest in Germany with also significant business in the UK, offers a special service for developers. "Sipgate API" is a new interface to integrate almost every Sipgate function – VoIP, SMS and large administration tools – in own applications. The Sipgate API enables to use central Sipgate functions within your own software or web projects, so that VoIP tinkerers can set up their own mashup services.

In his latest blog post Thomas Howe, the master of mashup, was so kind to explain again what mashup means:

A mashup is an application that uses
1) modern Web integration technologies
2) to take content or services from two independent sources
3) to solve a unique or niche problem.

The first element of mashups are the integration technologies they use. These integration technologies create a web as platform architecture, allowing the mashup developer to integrate his software on top of the world class infrastructures provided by Amazon or AOL, simply, easily and safely. The most common technologies used for mashups include Web services calls, which either come as a SOAP or REST flavors, AJAX, Javascript and Ruby.

The second element of mashups is that they take content or services from more than one independent source. This is where the mashup word comes from. Mashups take things that might not go together, and puts them together in a valuable way. The classic mashup is the Chicago Crime Map, that took data from the Chicago Police Department and plotted it on Google Maps, so that you could see where the burglaries happened.

The "Sipgate API" is provided free of charge and can be used for mashups with every Sipgate account. Up to date the fax function of Sipgate can be used only with the German service.

To make integration easy, Sipgate publishes also the source code of the Firefox extension "Sipgate FFX" as well as several Perl examples and a KDE panel application under a GPL 2 license. Further more .NET developers will find with "sipgate API .NET SDK" a comfortable library to use the "Sipgate API" services easily. Over a mailing list developers can also exchange experiences and tips.

You will find all information about the interface and the exemplar applications including detailed documentations under www.sipgate.co.uk/api.

Source: snapvoip.blogspot.com

When you hear the word MIG and when you are as old as me, the only thing comes to mind is the Cold war era Russian fighter Jets. But Russell Shaw pointed me to a another form of MIG. MIG33 is heading our way to make a jolt in the mobile communication sphere!
After listening to Russell’s advise I also started playing with MIG33 and I think it is a formidable application. According to the website, mig33 is an operator-agnostic global community that makes the most popular Internet applications accessible on any mobile phone. That means consumers can take advantage of low-cost VoIP calls, SMS and IM communications tools, and participate in mobile chat rooms, profiles, and photo sharing, all directly from their mobile phone. I have not tested all the features. The other thing is I use phone for what is originally meant to be, to make calls. (That might change now that Apple has made iPhone affordable and hacks are public!). I have to warn you though, about pricing! MIG33 might still need to polish their pricing. I mostly call Japan and Germany and I already get better rates mobile to mobile. But what matters here is the modus operandi.
But most of the features are available even from a pc. The first thing I tried out was make a call following the instruction on site!;

To make a call back using SMS:

1. Send an SMS to "+447717989963" with **

• Needs to be sent from your mig33 registered mobile number
  

mig33 will connect both numbers together using mig33’s call back technology.MIG33

Source: snapvoip.blogspot.com

Yahoo mail is offering SMS talk or texting with new Mail interface as it emerges from beta phase,
WeSeePeople: Socialize with Yahoo Mail

Source: goebel.net

The Betamax company Voipbuster now offers free SMS from a mobile phone. I have just checked it out and it works great. You just have to install a small Java applet on your cell phone. Voipbuster routes the message as so called "UrlSms" over their servers. The SMS is free, besides of a little 3G data traffic. I have just sent two totally free SMS over Wifi from my Nokia E61. The messages arrive within minutes and seem to be a tiny bit slowlier than normal SMS.

1) How to install:

• On your mobile go to http://gsm.voipbuster.com/ (on some mobiles you need to choose ‘download application’ on some phones you can just go to the webpage).
• download and install the application, start the application (for most mobiles: in the menu, go to applications)
• in "settings" fill in your VoipBuster username / password and your mobile phone number

2) How to send SMS:

• start the application (for most mobiles: in the menu, go to applications)
• go to options –> sms
• you can select the person you want to send a sms from your mobile phone contacts through options –> add from contacts
• type your message and hit SEND!
• your mobile phone will ask you to open a data connection. This uses just a tiny bit of data traffic.
• Voipbuster charges you nothing! Yes –> NOTHING!

Again I wonder how Betamax will refinance this service. But they already surprised me a few times with such offers. Normally they don’t last too long and then become paid services at cheap prices. I guess it’s just an introductory offer.