It is that time of year when we like to reflect on what has happened during the year and look forward to what will be in 2009. Here in the UK the ‘Credit Crunch’ has dominated the headlines and had an effect on business in all areas and for sure the effect will be felt in 2009 and passibly beyond.
Ignoring the Credit Crunch for now, then 2008 has seen VoIP widely adopted and accepted by businesses and we all need to try and continue this in 2009, where clear savings can be demonstrated then businesses will be looking for for such technology.
Early indications have shown that GSM gateways will be more readily adopted as for the vast majority of businesses really big savings on mobile costs can be demonstrated and remember to talk to us about our ranges from Topex and Teles.
Happy New Year to all our readers and here is to a successful 2009 despite it all 
We are seeing some rumours appear that Sylantro has been or is about to be acquired by Broadsoft, reports are appearing over on the FierceVoIP site here and we have also seen a blog from an alleged ex Sylantro employee here and both of these make for some interesting reading but of course until Broadsoft make a formal statement we cannot be sure of anything.
So what will all this mean - there are suggestions Sylantro are in trouble and that Broadsoft will acquire the technology - who knows. We may see one of the main VoIP platform providers dissapear to be swallowed by the competition but only time will tell.
We know are industry loves to create some terms and that Phishing is now a widespread threat to the online community. With various products, solutions and services available out there claiming to minimize the phishing threat, it is wise to take a step back and look at the phishing threat in itself and identify its various components. This is done through studying various phishing attacks, the processes involved, and then comparing them to other previous attacks. Vishing is the same as Phishing but done via VoIP
Today, phishing attacks and its many variants are in the forefront of the news. Due to this publicity, organizations that offer services online, especially the financial institutions, are looking into various controls to mitigate the risks of phishing attacks.
Have a look at the full article over on Top Tech News - Spam & Hackers - Phishing: Don’t Be the Catch of the Day!.
We have seen a press announcement that Voxbone has announced iNum, a one global telephone number that enables anyone to establish a local presence anywhere in the world. Voxbone is making use of the +883 “country code,” newly created by the International Telecommunication Union (ITU), to create a global country code and associated banks of direct inward dial (DID) numbers.
”iNum is a new kind of phone number for a new kind of world–a world with a new geography that’s about local presence and global relationships, not about distance or national borders,” said Rodrigue Ullens, CEO and co-founder of Voxbone. “We believe the new geography is defined by the markets, customers and vendors that businesses need to connect with most. We need ‘local’ communication with these people whether calls originate on public-switched or VoIP networks, whether they are truly local or ‘virtually’ local.”
“We received our first iNum allocation and began testing ‘+883′ numbers in remote networks,” he said. “This month we begin a preview with several service providers, Jajah, Mobivox, Gizmo5, Ribbit, iotum, Rebtel, Voipbuster, Voxeo and Voipuser.” To create a number with “local” PSTN access around the world, Voxbone worked with the ITU to establish the +883 country code, in the same way that +44 is the country code that refers internationally to the United Kingdom and +1 to the United States.
“Voxbone provides iNum numbers for free to carriers and service providers,” Ullens said. As the number wholesaler, Voxbone will receive calls to these numbers and deliver them to the appropriate service provider. In this way, it connects incoming calls from PSTN callers as well as calls between different VoIP islands. Some iNum participating carriers will resell iNum numbers to their customers so they can receive calls; other participants will merely route (terminate) calls to the new numbers. Most participating carriers will even deliver calls to iNum numbers free of charge to their users, effectively providing free connectivity between their network and remote networks.
“Customers and prospects who see an advertised iNum telephone number that costs little or nothing to dial will be just as inclined to call an international business as they would any business physically located nearby or in-country,” said Ullens. “We see iNum encouraging more people to call globally, enabling business subscribers to have wider interactions with callers from new markets,” he added. “For carriers, iNum will increase call minutes and generate additional revenue through innovative global services.”
This is an interesting developement in the IP telephony market and could mean users selecting a call to particular numbers as this could be a free or low cost call - we will watch this one with interest
For more information see iNum or for the press relaease go to Voxbone.
In the past we mentioned that Fonality caused a few waves in the Opensource Community by developing their own version of FreePBX to work more closely with their Trixbox product, it seems that FreePBX has now entered the hands of bandwidth.com by hiring the project’s main developer as its Open Source Community Developer and is said to be committing significant resources and effort to expand the scope of the project.
An informal announcement of bandwidth.com’s commitment to FreePBX came through the main developer Philippe Lindheimer’s blog at www.freepbx.org on 14th November in a post called “A Bright Future for FreePBX.” Lindheimer said he had “joined forces” with bandwidth.com as its Open Source Community Director and indicated both he and bandwidth.com would work to expand the scope of FreePBX and to assure it remains “open and strong.”
Lindheimer cited bandwidth.com’s efforts in purchasing the FreePBX trademark and its efforts with FreeSwitch as areas where the company has been helpful to the open source community. Since bandwidth.com sells VoIP and data services not software then there appears to be no conflict of interest.
This is the latest coup for FreePBX; the web based GUI provides preprogrammed functionality and ease of use on top of Asterisk, including features such as follow me, ring groups with calls confirmation, music on hold, conferencing, and paging and intercom functionality for many SIP phones. Digium incorporated FreePBX into its compilation of the AsteriskNow 1.5 turnkey release in October.
See the blog entry here.
We see that British Telecom has released one of it’s regular VoIP surveys, which they say showed increased adoption rates and planned VoIP expenditures over the next two years. Half of the respondents to the survey planned to increase VoIP expenditures during next year.
Demonstrating the Return on Investment of moving from a PSTN system to VoIP services was listed by the highest percentage (27%) of respondents as the main hurdle for a move to VoIP. Network reliability, voice quality and security were the three most important considerations for managers considering the migration to VoIP products; these three concerns were also the top concerns for the BT surveys conducted in 2005 and 2007. Around 25% of respondents stated that they were planning on switching to an IP-based network in the next year.
Read the survey in full here
We saw a great article over at processor.com that descibes problems and pitfalls to watch out for if you are implementing VoIP or Unified Communications - the key points listed are:
- Converged Network Woes
- Ageing Networks will not fly
- Business Issues Bog Down Process
- Beware the UC Promise
- Network Threats Plague Voice Systems
We have discussed many of these points in the past but it is useful to see the warnings laid out in a single article - the key points to watch for are:
Voice data is far more sensitive than other network data, in turn requiring specific planning and troubleshooting to avoid latency, jitter, and other quality related problems inherent in VoIP and UC technologies.
Because IP based voice systems reside on the same architecture that can suffer traditional network based attacks, it’s imperative to ensure that these systems are included in the overall enterprise security policy.
See the article Here
The IridiaBlog runs on Wordpress and we are big fans of it but one of the (small) limitations for us is that we like to have a nice graphical theme which tends not to play well for our readers who are using mobile devices. We have found a great plugin called MobilePress that renders our blog in a perfect fashion for mobile devices, the following screenshot is our blog using MobilePress as viewed on a Nokia N95 - pretty cool !!!
MobilePress on Nokia N95
To get a copy visit the MobilePress guys here.
Our regular readers will know that one of our pet subjects that we watch with interest is the different ways Mobile carriers attempt to stop VoIP over WiFi being used on handsets connected to their networks. Obviously they see it as a big threat to their revenue and in many cases handsets are provided ‘free’ as part of a contract.
We see in Germany T-Mobile has taken issue with a VoIP application developed for the iPhone. Sipgate, has developed a SIP client for iPhones (first version) and this has upset T-Mobile who has been able to persuade a German court to ban the use of Sipgate on the iPhone in Germany.
Overall this has to be a very small issue as there are many handsets VoIP capable and many applications available for them.
We will keep our eye on this issue as this one is going to run and run….
Skype have been trying to push hard to penetrate the business market and especially the medium to large enterprise type market, we see that Digium and Skype now plan to work together to Skype enable Asterisk.
The Skype for Asterisk connector will enable a presence on the Skype network while getting all the functionality of a PBX. A general Skype login name could be routed into an Asterisk system for call processing and delivery to the next available person in a queue, a single Skype ID could be used for sales, technical support or other types of customer service queues. So in essence a Skype connection will be treated in the same way as a regular trunk or CO line.
While the two companies have had various discussions over the years on how they could work together, a serious effort came to a head at a June meeting at Digium’s Headquarters in Huntsville USA, since then Digium’s software engineers have worked to produce the solution, making it more robust and suitable for a formal beta, but as yet there is no firm date for release. The software is slated to run with all versions of Asterisk including trixbox.
We will watch this one with interest to see if it elevates Skype to be a serious tool for many businesses.
VoIP | admin | 
31 December, 2008 10:49 pm | 
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